From 1d533de998e2887f23c8cf6c39d5db55f8d202af Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 22 Oct 2011 22:48:27 +0800 Subject: ASoC: wm8400: Fix setting Fout clock divider for FLL Control 4 What we want here is to clear the WM8400_FLL_OUTDIV_MASK bits then OR with factors.outdiv. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index dc13be2a09c5..f29bc26ba416 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1059,7 +1059,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); - reg &= WM8400_FLL_OUTDIV_MASK; + reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); -- cgit v1.2.1 From 753ddf52153b60be924109df3bebab0cd60b3297 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 24 Oct 2011 11:31:12 +0800 Subject: ASoC: wm8996: Avoid a redundant i2c_get_clientdata call in wm8996_i2c_remove Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 645c980d6b80..32324c9ddc37 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3144,7 +3144,7 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); - kfree(i2c_get_clientdata(client)); + kfree(wm8996); return 0; } -- cgit v1.2.1 From 49fa4d9b5aeafb985abe8cb8cdf6432690c49ad3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 24 Oct 2011 11:32:41 +0800 Subject: ASoC: wm8940: Fix setting PLL Output clock division ratio According to the datasheet: The PLL Output clock division ratio is controlled by BIT[5:4] of WM8940_GPIO register(08h). Current code read/write the WM8940_ADDCNTRL(07h) register which is wrong. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a4abfdfb217b..3cc3bce61316 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -627,8 +627,8 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 5)); break; case WM8940_OPCLKDIV: - reg = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFCF; - ret = snd_soc_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + reg = snd_soc_read(codec, WM8940_GPIO) & 0xFFCF; + ret = snd_soc_write(codec, WM8940_GPIO, reg | (div << 4)); break; } return ret; -- cgit v1.2.1 From bdb527e9ae038d76917a999108176c5f5be5e35e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 24 Oct 2011 11:33:55 +0800 Subject: ASoC: wm8940: Fix a typo for the mask of setting WM8940_BCLKDIV The registers are 16 bits, thus remove an extra F for the mask. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 3cc3bce61316..fec3892b234c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -619,7 +619,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8940_BCLKDIV: - reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3; + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFEF3; ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2)); break; case WM8940_MCLKDIV: -- cgit v1.2.1 From 9c173d15f99ef182ac4b27e3e03779026d8e6cf1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 26 Oct 2011 22:13:17 +0800 Subject: ASoC: tlv320aic3x: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7a49390bc30d..a77f6ea47198 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -833,7 +833,6 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 d, pll_d = 1; - u8 reg; int clk; /* select data word length */ @@ -869,14 +868,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); /* disable PLL if it is bypassed */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0); } else { snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); /* enable PLL when it is used */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } /* Route Left DAC to left channel input and @@ -1155,7 +1153,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - u8 reg; switch (level) { case SND_SOC_BIAS_ON: @@ -1164,9 +1161,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } break; case SND_SOC_BIAS_STANDBY: @@ -1175,9 +1171,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, 0); } break; case SND_SOC_BIAS_OFF: @@ -1294,7 +1289,6 @@ static int aic3x_resume(struct snd_soc_codec *codec) static int aic3x_init(struct snd_soc_codec *codec) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int reg; snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); @@ -1316,20 +1310,13 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ - reg = snd_soc_read(codec, LLOPM_CTRL); - snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, RLOPM_CTRL); - snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLOUT_CTRL); - snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPROUT_CTRL); - snd_soc_write(codec, HPROUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLCOM_CTRL); - snd_soc_write(codec, HPLCOM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPRCOM_CTRL); - snd_soc_write(codec, HPRCOM_CTRL, reg | UNMUTE); + snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPRCOM_CTRL, UNMUTE, UNMUTE); /* ADC default volume and unmute */ snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); -- cgit v1.2.1 From e50fad4f029c36ed85a71fe7413684cfd3c7d78c Mon Sep 17 00:00:00 2001 From: "ramesh.babu@linux.intel.com" Date: Thu, 27 Oct 2011 12:12:33 +0530 Subject: ASoC: Allow machines to ignore pmdown_time per-link With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: Ramesh Babu K V Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337353fa..52a7259f6184 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,7 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time)) { + if (unlikely(codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- cgit v1.2.1 From 9ce316236b572b437a9a96234a8cc9664927c0c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:49:11 +0200 Subject: ASoC: Convert wm8995 MICBIASes to supply widgets Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 78eeb21e6696..4d109b1ad124 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -688,8 +688,10 @@ static const struct snd_soc_dapm_widget wm8995_dapm_widgets[] = { SND_SOC_DAPM_MIXER("IN1R PGA", SND_SOC_NOPM, 0, 0, &in1r_pga, 1), - SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8995_POWER_MANAGEMENT_1, 8, 0), - SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8995_POWER_MANAGEMENT_1, 9, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8995_POWER_MANAGEMENT_1, 8, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8995_POWER_MANAGEMENT_1, 9, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8995_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8995_AIF2_CLOCKING_1, 0, 0, NULL, 0), -- cgit v1.2.1 From b6406a80278a09d19c31717e68312dbd59dd51fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:48:39 +0200 Subject: ASoC: Convert wm8991 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index c9ab3ba9bced..1d46d59c82a3 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -770,8 +770,8 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { NULL, 0), /* MICBIAS */ - SND_SOC_DAPM_MICBIAS("MICBIAS", WM8991_POWER_MANAGEMENT_1, - WM8991_MICBIAS_ENA_BIT, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", WM8991_POWER_MANAGEMENT_1, + WM8991_MICBIAS_ENA_BIT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LON"), SND_SOC_DAPM_OUTPUT("LOP"), -- cgit v1.2.1 From e1fc3f21c22023b0bb6859c896f1bca979f5cfcc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:48:09 +0200 Subject: ASoC: Convert wm8990 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index d29a9622964c..d4cbec6372db 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -776,8 +776,8 @@ SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0, NULL, 0), /* MICBIAS */ -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1, - WM8990_MICBIAS_ENA_BIT, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8990_POWER_MANAGEMENT_1, + WM8990_MICBIAS_ENA_BIT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LON"), SND_SOC_DAPM_OUTPUT("LOP"), -- cgit v1.2.1 From be48f20d8fe2e4c5998f38dd71c79f97c6fced4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:47:53 +0200 Subject: ASoC: Convert wm8988 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 2e9eba717d1a..514189d1923e 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -267,7 +267,7 @@ static const struct snd_kcontrol_new wm8988_monomux_controls = SOC_DAPM_ENUM("Route", monomux); static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { - SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", WM8988_PWR1, 1, 0, NULL, 0), SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, &wm8988_diffmux_controls), -- cgit v1.2.1 From 812f8a3524b9d369a170428acec79c57786d4670 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:47:40 +0200 Subject: ASoC: Convert wm8985 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index bae510acdec8..36c4ee08e159 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -411,7 +411,8 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right Speaker Out", WM8985_POWER_MANAGEMENT_3, 6, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias", WM8985_POWER_MANAGEMENT_1, 4, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", WM8985_POWER_MANAGEMENT_1, 4, 0, + NULL, 0), SND_SOC_DAPM_INPUT("LIN"), SND_SOC_DAPM_INPUT("LIP"), -- cgit v1.2.1 From 605b151ae3e025e69f89db46e878c782fdc6489b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:47:24 +0200 Subject: ASoC: Convert wm8983 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 93ee28439be5..58e067b5a6a3 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -481,7 +481,8 @@ static const struct snd_soc_dapm_widget wm8983_dapm_widgets[] = { SND_SOC_DAPM_PGA("OUT4 Out", WM8983_POWER_MANAGEMENT_3, 8, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias", WM8983_POWER_MANAGEMENT_1, 4, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", WM8983_POWER_MANAGEMENT_1, 4, 0, + NULL, 0), SND_SOC_DAPM_INPUT("LIN"), SND_SOC_DAPM_INPUT("LIP"), -- cgit v1.2.1 From 48dd231b0bb3a51f5c13e5b53f4e8d798f8d828e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:47:09 +0200 Subject: ASoC: Convert wm8974 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 9352f1e088d2..7bd35b8fdcd2 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -226,7 +226,7 @@ SND_SOC_DAPM_MIXER("Input PGA", WM8974_POWER2, 2, 0, wm8974_inpga, SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, wm8974_boost_mixer, ARRAY_SIZE(wm8974_boost_mixer)), -SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias", WM8974_POWER1, 4, 0, NULL, 0), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), -- cgit v1.2.1 From 20abf088792b2ae5e1c16159aef7d742722f967c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:46:32 +0200 Subject: ASoC: Convert wm8961 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 9568c8a49f96..7f2df7ba27f6 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -531,7 +531,7 @@ SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8961_PWR_MGMT_1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8961_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux), SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux), -- cgit v1.2.1 From 187774cbe3b67a3ea644cfbf9b57e7695ab37558 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:46:17 +0200 Subject: ASoC: Convert wm8960 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 2df253c18568..6e22f9b3d967 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -265,7 +265,7 @@ SND_SOC_DAPM_INPUT("RINPUT2"), SND_SOC_DAPM_INPUT("LINPUT3"), SND_SOC_DAPM_INPUT("RINPUT3"), -SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICB", WM8960_POWER1, 1, 0, NULL, 0), SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), -- cgit v1.2.1 From dcd658c56b5d40d010a1540aa475fe49260a4c91 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:46:01 +0200 Subject: ASoC: Convert wm8904 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9fc8f4c0a9a9..bf325bb12d77 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1196,7 +1196,7 @@ SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lin_mux), SND_SOC_DAPM_MUX("Left Capture Inverting Mux", SND_SOC_NOPM, 0, 0, -- cgit v1.2.1 From 8a709d92c7e0f2015e12b45af506ac64f4c28dda Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:45:42 +0200 Subject: ASoC: Convert wm8900 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3d0dc1591ecc..17a12c2df8da 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -513,7 +513,7 @@ SND_SOC_DAPM_MIXER("Right Input Mixer", WM8900_REG_POWER2, 4, 0, wm8900_rinmix_controls, ARRAY_SIZE(wm8900_rinmix_controls)), -SND_SOC_DAPM_MICBIAS("Mic Bias", WM8900_REG_POWER1, 4, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias", WM8900_REG_POWER1, 4, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8900_REG_POWER2, 1, 0), SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8900_REG_POWER2, 0, 0), -- cgit v1.2.1 From 3ff51c859f086036710b375eb70a84f2efda97f9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:44:59 +0200 Subject: ASoC: Convert wm8400 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index f29bc26ba416..585def1ffca6 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -766,8 +766,8 @@ SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0, NULL, 0), /* MICBIAS */ -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1, - WM8400_MIC1BIAS_ENA_SHIFT, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8400_POWER_MANAGEMENT_1, + WM8400_MIC1BIAS_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LON"), SND_SOC_DAPM_OUTPUT("LOP"), -- cgit v1.2.1 From 2a761cde31fddfe5e22f29bc5e241d597204e095 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Nov 2011 15:19:23 +0000 Subject: ASoC: Start WM8962 FLL if SYSCLK is enabled Since we have code to automatically manage the start and stop of the FLL based on the SYSCLK widget if SYSCLK is already enabled and the FLL is configured then we need to start it up. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f60dfa16545e..74ed8831990e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3394,6 +3394,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned long timeout; int ret; int fll1 = snd_soc_read(codec, WM8962_FLL_CONTROL_1) & WM8962_FLL_ENA; + int sysclk = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_ENA; /* Any change? */ if (source == wm8962->fll_src && Fref == wm8962->fll_fref && @@ -3454,6 +3455,9 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, try_wait_for_completion(&wm8962->fll_lock); + if (sysclk) + fll1 |= WM8962_FLL_ENA; + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); -- cgit v1.2.1 From db0e55438c39c5afa6b7674f5cef86b200bd89ba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Nov 2011 15:59:03 +0000 Subject: ASoC: Enable SYSCLK last when enabling WM8962 mic detection Ensure everything is set up before we start detecting. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 74ed8831990e..430bf535d546 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3661,6 +3661,9 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) snd_soc_jack_report(wm8962->jack, 0, SND_JACK_MICROPHONE | SND_JACK_BTN_0); + if (jack) + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + return 0; } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -- cgit v1.2.1 From a5ef9884088de4ed87ee9490923f277e805b38b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Nov 2011 16:00:15 +0000 Subject: ASoC: WM8962 accessory detection requires MICBIAS Force MICBIAS on as well as SYSCLK as the WM8962 accessory detection can't function without both. No point in making machine drivers manually enable it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 430bf535d546..b9c64a826ff6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3661,8 +3661,10 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) snd_soc_jack_report(wm8962->jack, 0, SND_JACK_MICROPHONE | SND_JACK_BTN_0); - if (jack) + if (jack) { snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS"); + } return 0; } -- cgit v1.2.1 From 00ae3b8691e6486895d92de05d7d1d3a70bb5077 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Nov 2011 16:02:01 +0000 Subject: ASoC: Disable MICBIAS and SYSCLK when stopping WM8962 accessory detection They aren't needed any more. If machines need them for other purposes then further changes will be required. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9c64a826ff6..cf7df9e61ef3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3664,6 +3664,9 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) if (jack) { snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS"); + } else { + snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS"); } return 0; -- cgit v1.2.1 From 8aafc43556cadc24dd9cf8563c66ab68c8d05748 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Nov 2011 13:05:17 +0000 Subject: ASoC: Sort LM4857 with the CODECs in the Makefile Having a separate list for amps is a little confusing now the official driver model for them is the same as for other CODECs so let's sort them into the CODEC list, but only do this for those that are actual CODEC drivers so it's easier to remember which ones need updating. Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842e357b..d7a5ff30a3a5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o +snd-soc-lm4857-objs := lm4857.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o @@ -91,7 +92,6 @@ snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-jz4740-codec-objs := jz4740.o # Amp -snd-soc-lm4857-objs := lm4857.o snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o @@ -122,6 +122,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o @@ -190,7 +191,6 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp -obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o -- cgit v1.2.1 From 98dd932b6a02b3eea2d6c671b48b2d5d7deb5afe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Nov 2011 13:28:23 +0000 Subject: ASoC: Fix sort of jz4740 in Makefile Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7a5ff30a3a5..a7c415dc22fe 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-max98088-objs := max98088.o @@ -89,7 +90,6 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o -snd-soc-jz4740-codec-objs := jz4740.o # Amp snd-soc-max9877-objs := max9877.o -- cgit v1.2.1 From 5b6247abc90a94a38c7e7314191871792645aa6a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 12:11:26 +0200 Subject: ASoC: Remove needless unlikely() There's no point in adding unlikely() annotations outside of hot paths and on systems using these features the annotation will always be wrong (as opposed to being something that only comes up once in a while) so the annotation may even be harmful. Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 52a7259f6184..49aa71e0d7e6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,8 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time)) { + if (codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- cgit v1.2.1 From a04e0c868058b6df7cb5b9a92b469ff72288bbc7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Nov 2011 22:13:36 +0000 Subject: ASoC: Only enable thermal shutdown when required on WM9081 The WM9081 thermal shutdown is only effective when the speaker output is enabled so disable it when that is not in use for a small current saving. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a02c28c..7563a91c9ed3 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -737,6 +737,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), }; @@ -759,6 +760,7 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "CLK_SYS" }, { "Speaker", NULL, "Speaker PGA" }, + { "Speaker", NULL, "TSENSE" }, { "SPKN", NULL, "Speaker" }, { "SPKP", NULL, "Speaker" }, -- cgit v1.2.1 From 94b88e647c795b2ba5add6d43dc7a454c6d02356 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Nov 2011 17:48:28 +0000 Subject: ASoC: Manage thermal shutdown for WM8962 Disable the thermal shutdown circuits for headphone and speaker when the relevant outputs are not enabled in order to save current in idle modes. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index cf7df9e61ef3..c9ba826ccb36 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2679,6 +2679,8 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2834,6 +2836,9 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "HPOUTL", NULL, "HPOUT" }, { "HPOUTR", NULL, "HPOUT" }, + + { "HPOUTL", NULL, "TEMP_HP" }, + { "HPOUTR", NULL, "TEMP_HP" }, }; static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { @@ -2850,6 +2855,7 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker Output", NULL, "Speaker PGA" }, { "Speaker Output", NULL, "SYSCLK" }, { "Speaker Output", NULL, "TOCLK" }, + { "Speaker Output", NULL, "TEMP_SPK" }, { "SPKOUT", NULL, "Speaker Output" }, }; @@ -2878,10 +2884,12 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { { "SPKOUTL Output", NULL, "SPKOUTL PGA" }, { "SPKOUTL Output", NULL, "SYSCLK" }, { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, { "SPKOUTR Output", NULL, "SPKOUTR PGA" }, { "SPKOUTR Output", NULL, "SYSCLK" }, { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, { "SPKOUTL", NULL, "SPKOUTL Output" }, { "SPKOUTR", NULL, "SPKOUTR Output" }, -- cgit v1.2.1 From 03431972ac16bbfcbfb831bb37c419f8f71bf16d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Nov 2011 17:11:54 +0000 Subject: ASoC: Disable thermal shutdown when not using speakers in wm_hubs The thermal shutdown support in wm_hubs devices is tied to the speaker drivers (which are the only high power subsystems within the device). Ensure minimal current usage when the thermal shutdown support is not required by disabling the circuit when the speaker drivers are powered down. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4ea2cd..f98170e3eb89 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -654,6 +654,7 @@ SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), +SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, @@ -789,10 +790,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKL Driver", NULL, "TSHUT" }, { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "TSHUT" }, { "SPKOUTLP", NULL, "SPKL Driver" }, { "SPKOUTLN", NULL, "SPKL Driver" }, -- cgit v1.2.1 From 8918b843aff3236de6301b1137434d3f0bc0a0f5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 31 Oct 2011 22:11:53 -0700 Subject: ASoC: fsi: fixup compile warning This patch fixup below warning ${linux}/sound/soc/sh/fsi.c:442:3:\ warning: passing argument 1 of '__fsi_reg_read' makes pointer\ from integer without a cast ${linux}/sound/soc/sh/fsi.c:517:3: \ warning: passing argument 1 of '__fsi_reg_write' makes pointer\ from integer without a cast ${linux}/sound/soc/sh/fsi.c:663:3: \ warning: passing argument 1 of '__fsi_reg_mask_set' makes pointer\ from integer without a cast Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3d7016e128f9..e620cb17cd2c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -235,13 +235,13 @@ static void __fsi_reg_mask_set(u32 __iomem *reg, u32 mask, u32 data) } #define fsi_reg_write(p, r, d)\ - __fsi_reg_write((u32)(p->base + REG_##r), d) + __fsi_reg_write((p->base + REG_##r), d) #define fsi_reg_read(p, r)\ - __fsi_reg_read((u32)(p->base + REG_##r)) + __fsi_reg_read((p->base + REG_##r)) #define fsi_reg_mask_set(p, r, m, d)\ - __fsi_reg_mask_set((u32)(p->base + REG_##r), m, d) + __fsi_reg_mask_set((p->base + REG_##r), m, d) #define fsi_master_read(p, r) _fsi_master_read(p, MST_##r) #define fsi_core_read(p, r) _fsi_master_read(p, p->core->r) -- cgit v1.2.1 From 202113912ba117b5c5f36e45529921b4cca4be6a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 6 Nov 2011 22:04:53 -0800 Subject: ASoC: ak4642: ak4642 was tested ak4642 was tested by ms7724se board Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 12c1bdef6732..b854eb0e6ad1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -18,7 +18,7 @@ * This is very simple driver. * It can use headphone output / stereo input only * - * AK4642 is not tested. + * AK4642 is tested. * AK4643 is tested. */ -- cgit v1.2.1 From 65ff03f4624d12ad6c19a01a0af7385eda09e4a6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 6 Nov 2011 22:05:25 -0800 Subject: ASoC: fsi: add valid data position control support FSI2 can control valid data position, like package in front/back or stream mode (16bit x 2). But current fsi driver is assuming it was in-back. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e620cb17cd2c..99ed61024166 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -32,7 +32,9 @@ #define REG_DIDT 0x0020 #define REG_DODT 0x0024 #define REG_MUTE_ST 0x0028 +#define REG_OUT_DMAC 0x002C #define REG_OUT_SEL 0x0030 +#define REG_IN_DMAC 0x0038 /* master register */ #define MST_CLK_RST 0x0210 @@ -886,6 +888,8 @@ static int fsi_hw_startup(struct fsi_priv *fsi, int is_play, struct device *dev) { + struct fsi_master *master = fsi_get_master(fsi); + int fsi_ver = master->core->ver; u32 flags = fsi_get_info_flags(fsi); u32 data = 0; @@ -920,6 +924,17 @@ static int fsi_hw_startup(struct fsi_priv *fsi, fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); } + /* + * FIXME + * + * FSI driver assumed that data package is in-back. + * FSI2 chip can select it. + */ + if (fsi_ver >= 2) { + fsi_reg_write(fsi, OUT_DMAC, (1 << 4)); + fsi_reg_write(fsi, IN_DMAC, (1 << 4)); + } + /* irq clear */ fsi_irq_disable(fsi, is_play); fsi_irq_clear_status(fsi); -- cgit v1.2.1 From e94de1e864d2d205e4e503b0f083c07f288b45fe Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 1 Nov 2011 15:17:57 +0800 Subject: ASoC: Avoid a redundant read in cs42l51_pdn_event snd_soc_update_bits already does read-modify-write, no need to read the register before calling snd_soc_update_bits. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8c3c8205d19e..00718b5e747b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -175,21 +175,18 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - unsigned long value; - - value = snd_soc_read(w->codec, CS42L51_POWER_CTL1); - value &= ~CS42L51_POWER_CTL1_PDN; - switch (event) { case SND_SOC_DAPM_PRE_PMD: - value |= CS42L51_POWER_CTL1_PDN; + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, + CS42L51_POWER_CTL1_PDN); break; default: case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, 0); break; } - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, - CS42L51_POWER_CTL1_PDN, value); return 0; } -- cgit v1.2.1 From 79172746827d0579900fa382733f5769d32952eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Sep 2011 16:15:58 +0100 Subject: ASoC: Convert WM8996 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 699 ++++++++++++++++++++++++++-------------------- 1 file changed, 391 insertions(+), 308 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 32324c9ddc37..5671fd398e8a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -114,297 +115,365 @@ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -static const u16 wm8996_reg[WM8996_MAX_REGISTER] = { - [WM8996_SOFTWARE_RESET] = 0x8996, - [WM8996_POWER_MANAGEMENT_7] = 0x10, - [WM8996_DAC1_HPOUT1_VOLUME] = 0x88, - [WM8996_DAC2_HPOUT2_VOLUME] = 0x88, - [WM8996_DAC1_LEFT_VOLUME] = 0x2c0, - [WM8996_DAC1_RIGHT_VOLUME] = 0x2c0, - [WM8996_DAC2_LEFT_VOLUME] = 0x2c0, - [WM8996_DAC2_RIGHT_VOLUME] = 0x2c0, - [WM8996_OUTPUT1_LEFT_VOLUME] = 0x80, - [WM8996_OUTPUT1_RIGHT_VOLUME] = 0x80, - [WM8996_OUTPUT2_LEFT_VOLUME] = 0x80, - [WM8996_OUTPUT2_RIGHT_VOLUME] = 0x80, - [WM8996_MICBIAS_1] = 0x39, - [WM8996_MICBIAS_2] = 0x39, - [WM8996_LDO_1] = 0x3, - [WM8996_LDO_2] = 0x13, - [WM8996_ACCESSORY_DETECT_MODE_1] = 0x4, - [WM8996_HEADPHONE_DETECT_1] = 0x20, - [WM8996_MIC_DETECT_1] = 0x7600, - [WM8996_MIC_DETECT_2] = 0xbf, - [WM8996_CHARGE_PUMP_1] = 0x1f25, - [WM8996_CHARGE_PUMP_2] = 0xab19, - [WM8996_DC_SERVO_5] = 0x2a2a, - [WM8996_CONTROL_INTERFACE_1] = 0x8004, - [WM8996_CLOCKING_1] = 0x10, - [WM8996_AIF_RATE] = 0x83, - [WM8996_FLL_CONTROL_4] = 0x5dc0, - [WM8996_FLL_CONTROL_5] = 0xc84, - [WM8996_FLL_EFS_2] = 0x2, - [WM8996_AIF1_TX_LRCLK_1] = 0x80, - [WM8996_AIF1_TX_LRCLK_2] = 0x8, - [WM8996_AIF1_RX_LRCLK_1] = 0x80, - [WM8996_AIF1TX_DATA_CONFIGURATION_1] = 0x1818, - [WM8996_AIF1RX_DATA_CONFIGURATION] = 0x1818, - [WM8996_AIF1TX_TEST] = 0x7, - [WM8996_AIF2_TX_LRCLK_1] = 0x80, - [WM8996_AIF2_TX_LRCLK_2] = 0x8, - [WM8996_AIF2_RX_LRCLK_1] = 0x80, - [WM8996_AIF2TX_DATA_CONFIGURATION_1] = 0x1818, - [WM8996_AIF2RX_DATA_CONFIGURATION] = 0x1818, - [WM8996_AIF2TX_TEST] = 0x1, - [WM8996_DSP1_TX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP1_TX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP1_RX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP1_RX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP1_TX_FILTERS] = 0x2000, - [WM8996_DSP1_RX_FILTERS_1] = 0x200, - [WM8996_DSP1_RX_FILTERS_2] = 0x10, - [WM8996_DSP1_DRC_1] = 0x98, - [WM8996_DSP1_DRC_2] = 0x845, - [WM8996_DSP1_RX_EQ_GAINS_1] = 0x6318, - [WM8996_DSP1_RX_EQ_GAINS_2] = 0x6300, - [WM8996_DSP1_RX_EQ_BAND_1_A] = 0xfca, - [WM8996_DSP1_RX_EQ_BAND_1_B] = 0x400, - [WM8996_DSP1_RX_EQ_BAND_1_PG] = 0xd8, - [WM8996_DSP1_RX_EQ_BAND_2_A] = 0x1eb5, - [WM8996_DSP1_RX_EQ_BAND_2_B] = 0xf145, - [WM8996_DSP1_RX_EQ_BAND_2_C] = 0xb75, - [WM8996_DSP1_RX_EQ_BAND_2_PG] = 0x1c5, - [WM8996_DSP1_RX_EQ_BAND_3_A] = 0x1c58, - [WM8996_DSP1_RX_EQ_BAND_3_B] = 0xf373, - [WM8996_DSP1_RX_EQ_BAND_3_C] = 0xa54, - [WM8996_DSP1_RX_EQ_BAND_3_PG] = 0x558, - [WM8996_DSP1_RX_EQ_BAND_4_A] = 0x168e, - [WM8996_DSP1_RX_EQ_BAND_4_B] = 0xf829, - [WM8996_DSP1_RX_EQ_BAND_4_C] = 0x7ad, - [WM8996_DSP1_RX_EQ_BAND_4_PG] = 0x1103, - [WM8996_DSP1_RX_EQ_BAND_5_A] = 0x564, - [WM8996_DSP1_RX_EQ_BAND_5_B] = 0x559, - [WM8996_DSP1_RX_EQ_BAND_5_PG] = 0x4000, - [WM8996_DSP2_TX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP2_TX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP2_RX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP2_RX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP2_TX_FILTERS] = 0x2000, - [WM8996_DSP2_RX_FILTERS_1] = 0x200, - [WM8996_DSP2_RX_FILTERS_2] = 0x10, - [WM8996_DSP2_DRC_1] = 0x98, - [WM8996_DSP2_DRC_2] = 0x845, - [WM8996_DSP2_RX_EQ_GAINS_1] = 0x6318, - [WM8996_DSP2_RX_EQ_GAINS_2] = 0x6300, - [WM8996_DSP2_RX_EQ_BAND_1_A] = 0xfca, - [WM8996_DSP2_RX_EQ_BAND_1_B] = 0x400, - [WM8996_DSP2_RX_EQ_BAND_1_PG] = 0xd8, - [WM8996_DSP2_RX_EQ_BAND_2_A] = 0x1eb5, - [WM8996_DSP2_RX_EQ_BAND_2_B] = 0xf145, - [WM8996_DSP2_RX_EQ_BAND_2_C] = 0xb75, - [WM8996_DSP2_RX_EQ_BAND_2_PG] = 0x1c5, - [WM8996_DSP2_RX_EQ_BAND_3_A] = 0x1c58, - [WM8996_DSP2_RX_EQ_BAND_3_B] = 0xf373, - [WM8996_DSP2_RX_EQ_BAND_3_C] = 0xa54, - [WM8996_DSP2_RX_EQ_BAND_3_PG] = 0x558, - [WM8996_DSP2_RX_EQ_BAND_4_A] = 0x168e, - [WM8996_DSP2_RX_EQ_BAND_4_B] = 0xf829, - [WM8996_DSP2_RX_EQ_BAND_4_C] = 0x7ad, - [WM8996_DSP2_RX_EQ_BAND_4_PG] = 0x1103, - [WM8996_DSP2_RX_EQ_BAND_5_A] = 0x564, - [WM8996_DSP2_RX_EQ_BAND_5_B] = 0x559, - [WM8996_DSP2_RX_EQ_BAND_5_PG] = 0x4000, - [WM8996_OVERSAMPLING] = 0xd, - [WM8996_SIDETONE] = 0x1040, - [WM8996_GPIO_1] = 0xa101, - [WM8996_GPIO_2] = 0xa101, - [WM8996_GPIO_3] = 0xa101, - [WM8996_GPIO_4] = 0xa101, - [WM8996_GPIO_5] = 0xa101, - [WM8996_PULL_CONTROL_2] = 0x140, - [WM8996_INTERRUPT_STATUS_1_MASK] = 0x1f, - [WM8996_INTERRUPT_STATUS_2_MASK] = 0x1ecf, - [WM8996_RIGHT_PDM_SPEAKER] = 0x1, - [WM8996_PDM_SPEAKER_MUTE_SEQUENCE] = 0x69, - [WM8996_PDM_SPEAKER_VOLUME] = 0x66, - [WM8996_WRITE_SEQUENCER_0] = 0x1, - [WM8996_WRITE_SEQUENCER_1] = 0x1, - [WM8996_WRITE_SEQUENCER_3] = 0x6, - [WM8996_WRITE_SEQUENCER_4] = 0x40, - [WM8996_WRITE_SEQUENCER_5] = 0x1, - [WM8996_WRITE_SEQUENCER_6] = 0xf, - [WM8996_WRITE_SEQUENCER_7] = 0x6, - [WM8996_WRITE_SEQUENCER_8] = 0x1, - [WM8996_WRITE_SEQUENCER_9] = 0x3, - [WM8996_WRITE_SEQUENCER_10] = 0x104, - [WM8996_WRITE_SEQUENCER_12] = 0x60, - [WM8996_WRITE_SEQUENCER_13] = 0x11, - [WM8996_WRITE_SEQUENCER_14] = 0x401, - [WM8996_WRITE_SEQUENCER_16] = 0x50, - [WM8996_WRITE_SEQUENCER_17] = 0x3, - [WM8996_WRITE_SEQUENCER_18] = 0x100, - [WM8996_WRITE_SEQUENCER_20] = 0x51, - [WM8996_WRITE_SEQUENCER_21] = 0x3, - [WM8996_WRITE_SEQUENCER_22] = 0x104, - [WM8996_WRITE_SEQUENCER_23] = 0xa, - [WM8996_WRITE_SEQUENCER_24] = 0x60, - [WM8996_WRITE_SEQUENCER_25] = 0x3b, - [WM8996_WRITE_SEQUENCER_26] = 0x502, - [WM8996_WRITE_SEQUENCER_27] = 0x100, - [WM8996_WRITE_SEQUENCER_28] = 0x2fff, - [WM8996_WRITE_SEQUENCER_32] = 0x2fff, - [WM8996_WRITE_SEQUENCER_36] = 0x2fff, - [WM8996_WRITE_SEQUENCER_40] = 0x2fff, - [WM8996_WRITE_SEQUENCER_44] = 0x2fff, - [WM8996_WRITE_SEQUENCER_48] = 0x2fff, - [WM8996_WRITE_SEQUENCER_52] = 0x2fff, - [WM8996_WRITE_SEQUENCER_56] = 0x2fff, - [WM8996_WRITE_SEQUENCER_60] = 0x2fff, - [WM8996_WRITE_SEQUENCER_64] = 0x1, - [WM8996_WRITE_SEQUENCER_65] = 0x1, - [WM8996_WRITE_SEQUENCER_67] = 0x6, - [WM8996_WRITE_SEQUENCER_68] = 0x40, - [WM8996_WRITE_SEQUENCER_69] = 0x1, - [WM8996_WRITE_SEQUENCER_70] = 0xf, - [WM8996_WRITE_SEQUENCER_71] = 0x6, - [WM8996_WRITE_SEQUENCER_72] = 0x1, - [WM8996_WRITE_SEQUENCER_73] = 0x3, - [WM8996_WRITE_SEQUENCER_74] = 0x104, - [WM8996_WRITE_SEQUENCER_76] = 0x60, - [WM8996_WRITE_SEQUENCER_77] = 0x11, - [WM8996_WRITE_SEQUENCER_78] = 0x401, - [WM8996_WRITE_SEQUENCER_80] = 0x50, - [WM8996_WRITE_SEQUENCER_81] = 0x3, - [WM8996_WRITE_SEQUENCER_82] = 0x100, - [WM8996_WRITE_SEQUENCER_84] = 0x60, - [WM8996_WRITE_SEQUENCER_85] = 0x3b, - [WM8996_WRITE_SEQUENCER_86] = 0x502, - [WM8996_WRITE_SEQUENCER_87] = 0x100, - [WM8996_WRITE_SEQUENCER_88] = 0x2fff, - [WM8996_WRITE_SEQUENCER_92] = 0x2fff, - [WM8996_WRITE_SEQUENCER_96] = 0x2fff, - [WM8996_WRITE_SEQUENCER_100] = 0x2fff, - [WM8996_WRITE_SEQUENCER_104] = 0x2fff, - [WM8996_WRITE_SEQUENCER_108] = 0x2fff, - [WM8996_WRITE_SEQUENCER_112] = 0x2fff, - [WM8996_WRITE_SEQUENCER_116] = 0x2fff, - [WM8996_WRITE_SEQUENCER_120] = 0x2fff, - [WM8996_WRITE_SEQUENCER_124] = 0x2fff, - [WM8996_WRITE_SEQUENCER_128] = 0x1, - [WM8996_WRITE_SEQUENCER_129] = 0x1, - [WM8996_WRITE_SEQUENCER_131] = 0x6, - [WM8996_WRITE_SEQUENCER_132] = 0x40, - [WM8996_WRITE_SEQUENCER_133] = 0x1, - [WM8996_WRITE_SEQUENCER_134] = 0xf, - [WM8996_WRITE_SEQUENCER_135] = 0x6, - [WM8996_WRITE_SEQUENCER_136] = 0x1, - [WM8996_WRITE_SEQUENCER_137] = 0x3, - [WM8996_WRITE_SEQUENCER_138] = 0x106, - [WM8996_WRITE_SEQUENCER_140] = 0x61, - [WM8996_WRITE_SEQUENCER_141] = 0x11, - [WM8996_WRITE_SEQUENCER_142] = 0x401, - [WM8996_WRITE_SEQUENCER_144] = 0x50, - [WM8996_WRITE_SEQUENCER_145] = 0x3, - [WM8996_WRITE_SEQUENCER_146] = 0x102, - [WM8996_WRITE_SEQUENCER_148] = 0x51, - [WM8996_WRITE_SEQUENCER_149] = 0x3, - [WM8996_WRITE_SEQUENCER_150] = 0x106, - [WM8996_WRITE_SEQUENCER_151] = 0xa, - [WM8996_WRITE_SEQUENCER_152] = 0x61, - [WM8996_WRITE_SEQUENCER_153] = 0x3b, - [WM8996_WRITE_SEQUENCER_154] = 0x502, - [WM8996_WRITE_SEQUENCER_155] = 0x100, - [WM8996_WRITE_SEQUENCER_156] = 0x2fff, - [WM8996_WRITE_SEQUENCER_160] = 0x2fff, - [WM8996_WRITE_SEQUENCER_164] = 0x2fff, - [WM8996_WRITE_SEQUENCER_168] = 0x2fff, - [WM8996_WRITE_SEQUENCER_172] = 0x2fff, - [WM8996_WRITE_SEQUENCER_176] = 0x2fff, - [WM8996_WRITE_SEQUENCER_180] = 0x2fff, - [WM8996_WRITE_SEQUENCER_184] = 0x2fff, - [WM8996_WRITE_SEQUENCER_188] = 0x2fff, - [WM8996_WRITE_SEQUENCER_192] = 0x1, - [WM8996_WRITE_SEQUENCER_193] = 0x1, - [WM8996_WRITE_SEQUENCER_195] = 0x6, - [WM8996_WRITE_SEQUENCER_196] = 0x40, - [WM8996_WRITE_SEQUENCER_197] = 0x1, - [WM8996_WRITE_SEQUENCER_198] = 0xf, - [WM8996_WRITE_SEQUENCER_199] = 0x6, - [WM8996_WRITE_SEQUENCER_200] = 0x1, - [WM8996_WRITE_SEQUENCER_201] = 0x3, - [WM8996_WRITE_SEQUENCER_202] = 0x106, - [WM8996_WRITE_SEQUENCER_204] = 0x61, - [WM8996_WRITE_SEQUENCER_205] = 0x11, - [WM8996_WRITE_SEQUENCER_206] = 0x401, - [WM8996_WRITE_SEQUENCER_208] = 0x50, - [WM8996_WRITE_SEQUENCER_209] = 0x3, - [WM8996_WRITE_SEQUENCER_210] = 0x102, - [WM8996_WRITE_SEQUENCER_212] = 0x61, - [WM8996_WRITE_SEQUENCER_213] = 0x3b, - [WM8996_WRITE_SEQUENCER_214] = 0x502, - [WM8996_WRITE_SEQUENCER_215] = 0x100, - [WM8996_WRITE_SEQUENCER_216] = 0x2fff, - [WM8996_WRITE_SEQUENCER_220] = 0x2fff, - [WM8996_WRITE_SEQUENCER_224] = 0x2fff, - [WM8996_WRITE_SEQUENCER_228] = 0x2fff, - [WM8996_WRITE_SEQUENCER_232] = 0x2fff, - [WM8996_WRITE_SEQUENCER_236] = 0x2fff, - [WM8996_WRITE_SEQUENCER_240] = 0x2fff, - [WM8996_WRITE_SEQUENCER_244] = 0x2fff, - [WM8996_WRITE_SEQUENCER_248] = 0x2fff, - [WM8996_WRITE_SEQUENCER_252] = 0x2fff, - [WM8996_WRITE_SEQUENCER_256] = 0x60, - [WM8996_WRITE_SEQUENCER_258] = 0x601, - [WM8996_WRITE_SEQUENCER_260] = 0x50, - [WM8996_WRITE_SEQUENCER_262] = 0x100, - [WM8996_WRITE_SEQUENCER_264] = 0x1, - [WM8996_WRITE_SEQUENCER_266] = 0x104, - [WM8996_WRITE_SEQUENCER_267] = 0x100, - [WM8996_WRITE_SEQUENCER_268] = 0x2fff, - [WM8996_WRITE_SEQUENCER_272] = 0x2fff, - [WM8996_WRITE_SEQUENCER_276] = 0x2fff, - [WM8996_WRITE_SEQUENCER_280] = 0x2fff, - [WM8996_WRITE_SEQUENCER_284] = 0x2fff, - [WM8996_WRITE_SEQUENCER_288] = 0x2fff, - [WM8996_WRITE_SEQUENCER_292] = 0x2fff, - [WM8996_WRITE_SEQUENCER_296] = 0x2fff, - [WM8996_WRITE_SEQUENCER_300] = 0x2fff, - [WM8996_WRITE_SEQUENCER_304] = 0x2fff, - [WM8996_WRITE_SEQUENCER_308] = 0x2fff, - [WM8996_WRITE_SEQUENCER_312] = 0x2fff, - [WM8996_WRITE_SEQUENCER_316] = 0x2fff, - [WM8996_WRITE_SEQUENCER_320] = 0x61, - [WM8996_WRITE_SEQUENCER_322] = 0x601, - [WM8996_WRITE_SEQUENCER_324] = 0x50, - [WM8996_WRITE_SEQUENCER_326] = 0x102, - [WM8996_WRITE_SEQUENCER_328] = 0x1, - [WM8996_WRITE_SEQUENCER_330] = 0x106, - [WM8996_WRITE_SEQUENCER_331] = 0x100, - [WM8996_WRITE_SEQUENCER_332] = 0x2fff, - [WM8996_WRITE_SEQUENCER_336] = 0x2fff, - [WM8996_WRITE_SEQUENCER_340] = 0x2fff, - [WM8996_WRITE_SEQUENCER_344] = 0x2fff, - [WM8996_WRITE_SEQUENCER_348] = 0x2fff, - [WM8996_WRITE_SEQUENCER_352] = 0x2fff, - [WM8996_WRITE_SEQUENCER_356] = 0x2fff, - [WM8996_WRITE_SEQUENCER_360] = 0x2fff, - [WM8996_WRITE_SEQUENCER_364] = 0x2fff, - [WM8996_WRITE_SEQUENCER_368] = 0x2fff, - [WM8996_WRITE_SEQUENCER_372] = 0x2fff, - [WM8996_WRITE_SEQUENCER_376] = 0x2fff, - [WM8996_WRITE_SEQUENCER_380] = 0x2fff, - [WM8996_WRITE_SEQUENCER_384] = 0x60, - [WM8996_WRITE_SEQUENCER_386] = 0x601, - [WM8996_WRITE_SEQUENCER_388] = 0x61, - [WM8996_WRITE_SEQUENCER_390] = 0x601, - [WM8996_WRITE_SEQUENCER_392] = 0x50, - [WM8996_WRITE_SEQUENCER_394] = 0x300, - [WM8996_WRITE_SEQUENCER_396] = 0x1, - [WM8996_WRITE_SEQUENCER_398] = 0x304, - [WM8996_WRITE_SEQUENCER_400] = 0x40, - [WM8996_WRITE_SEQUENCER_402] = 0xf, - [WM8996_WRITE_SEQUENCER_404] = 0x1, - [WM8996_WRITE_SEQUENCER_407] = 0x100, +static struct reg_default wm8996_reg[] = { + { WM8996_SOFTWARE_RESET, 0x8996 }, + { WM8996_POWER_MANAGEMENT_1, 0x0 }, + { WM8996_POWER_MANAGEMENT_2, 0x0 }, + { WM8996_POWER_MANAGEMENT_3, 0x0 }, + { WM8996_POWER_MANAGEMENT_4, 0x0 }, + { WM8996_POWER_MANAGEMENT_5, 0x0 }, + { WM8996_POWER_MANAGEMENT_6, 0x0 }, + { WM8996_POWER_MANAGEMENT_7, 0x10 }, + { WM8996_POWER_MANAGEMENT_8, 0x0 }, + { WM8996_LEFT_LINE_INPUT_VOLUME, 0x0 }, + { WM8996_RIGHT_LINE_INPUT_VOLUME, 0x0 }, + { WM8996_LINE_INPUT_CONTROL, 0x0 }, + { WM8996_DAC1_HPOUT1_VOLUME, 0x88 }, + { WM8996_DAC2_HPOUT2_VOLUME, 0x88 }, + { WM8996_DAC1_LEFT_VOLUME, 0x2c0 }, + { WM8996_DAC1_RIGHT_VOLUME, 0x2c0 }, + { WM8996_DAC2_LEFT_VOLUME, 0x2c0 }, + { WM8996_DAC2_RIGHT_VOLUME, 0x2c0 }, + { WM8996_OUTPUT1_LEFT_VOLUME, 0x80 }, + { WM8996_OUTPUT1_RIGHT_VOLUME, 0x80 }, + { WM8996_OUTPUT2_LEFT_VOLUME, 0x80 }, + { WM8996_OUTPUT2_RIGHT_VOLUME, 0x80 }, + { WM8996_MICBIAS_1, 0x39 }, + { WM8996_MICBIAS_2, 0x39 }, + { WM8996_LDO_1, 0x3 }, + { WM8996_LDO_2, 0x13 }, + { WM8996_ACCESSORY_DETECT_MODE_1, 0x4 }, + { WM8996_ACCESSORY_DETECT_MODE_2, 0x0 }, + { WM8996_HEADPHONE_DETECT_1, 0x20 }, + { WM8996_HEADPHONE_DETECT_2, 0x0 }, + { WM8996_MIC_DETECT_1, 0x7600 }, + { WM8996_MIC_DETECT_2, 0xbf }, + { WM8996_CHARGE_PUMP_1, 0x1f25 }, + { WM8996_CHARGE_PUMP_2, 0xab19 }, + { WM8996_DC_SERVO_1, 0x0 }, + { WM8996_DC_SERVO_2, 0x0 }, + { WM8996_DC_SERVO_3, 0x0 }, + { WM8996_DC_SERVO_5, 0x2a2a }, + { WM8996_DC_SERVO_6, 0x0 }, + { WM8996_DC_SERVO_7, 0x0 }, + { WM8996_ANALOGUE_HP_1, 0x0 }, + { WM8996_ANALOGUE_HP_2, 0x0 }, + { WM8996_CONTROL_INTERFACE_1, 0x8004 }, + { WM8996_WRITE_SEQUENCER_CTRL_1, 0x0 }, + { WM8996_WRITE_SEQUENCER_CTRL_2, 0x0 }, + { WM8996_AIF_CLOCKING_1, 0x0 }, + { WM8996_AIF_CLOCKING_2, 0x0 }, + { WM8996_CLOCKING_1, 0x10 }, + { WM8996_CLOCKING_2, 0x0 }, + { WM8996_AIF_RATE, 0x83 }, + { WM8996_FLL_CONTROL_1, 0x0 }, + { WM8996_FLL_CONTROL_2, 0x0 }, + { WM8996_FLL_CONTROL_3, 0x0 }, + { WM8996_FLL_CONTROL_4, 0x5dc0 }, + { WM8996_FLL_CONTROL_5, 0xc84 }, + { WM8996_FLL_EFS_1, 0x0 }, + { WM8996_FLL_EFS_2, 0x2 }, + { WM8996_AIF1_CONTROL, 0x0 }, + { WM8996_AIF1_BCLK, 0x0 }, + { WM8996_AIF1_TX_LRCLK_1, 0x80 }, + { WM8996_AIF1_TX_LRCLK_2, 0x8 }, + { WM8996_AIF1_RX_LRCLK_1, 0x80 }, + { WM8996_AIF1_RX_LRCLK_2, 0x0 }, + { WM8996_AIF1TX_DATA_CONFIGURATION_1, 0x1818 }, + { WM8996_AIF1TX_DATA_CONFIGURATION_2, 0 }, + { WM8996_AIF1RX_DATA_CONFIGURATION, 0x1818 }, + { WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_MONO_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_TEST, 0x7 }, + { WM8996_AIF2_CONTROL, 0x0 }, + { WM8996_AIF2_BCLK, 0x0 }, + { WM8996_AIF2_TX_LRCLK_1, 0x80 }, + { WM8996_AIF2_TX_LRCLK_2, 0x8 }, + { WM8996_AIF2_RX_LRCLK_1, 0x80 }, + { WM8996_AIF2_RX_LRCLK_2, 0x0 }, + { WM8996_AIF2TX_DATA_CONFIGURATION_1, 0x1818 }, + { WM8996_AIF2RX_DATA_CONFIGURATION, 0x1818 }, + { WM8996_AIF2RX_DATA_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_MONO_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_TEST, 0x1 }, + { WM8996_DSP1_TX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP1_TX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP1_RX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP1_RX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP1_TX_FILTERS, 0x2000 }, + { WM8996_DSP1_RX_FILTERS_1, 0x200 }, + { WM8996_DSP1_RX_FILTERS_2, 0x10 }, + { WM8996_DSP1_DRC_1, 0x98 }, + { WM8996_DSP1_DRC_2, 0x845 }, + { WM8996_DSP1_RX_EQ_GAINS_1, 0x6318 }, + { WM8996_DSP1_RX_EQ_GAINS_2, 0x6300 }, + { WM8996_DSP1_RX_EQ_BAND_1_A, 0xfca }, + { WM8996_DSP1_RX_EQ_BAND_1_B, 0x400 }, + { WM8996_DSP1_RX_EQ_BAND_1_PG, 0xd8 }, + { WM8996_DSP1_RX_EQ_BAND_2_A, 0x1eb5 }, + { WM8996_DSP1_RX_EQ_BAND_2_B, 0xf145 }, + { WM8996_DSP1_RX_EQ_BAND_2_C, 0xb75 }, + { WM8996_DSP1_RX_EQ_BAND_2_PG, 0x1c5 }, + { WM8996_DSP1_RX_EQ_BAND_3_A, 0x1c58 }, + { WM8996_DSP1_RX_EQ_BAND_3_B, 0xf373 }, + { WM8996_DSP1_RX_EQ_BAND_3_C, 0xa54 }, + { WM8996_DSP1_RX_EQ_BAND_3_PG, 0x558 }, + { WM8996_DSP1_RX_EQ_BAND_4_A, 0x168e }, + { WM8996_DSP1_RX_EQ_BAND_4_B, 0xf829 }, + { WM8996_DSP1_RX_EQ_BAND_4_C, 0x7ad }, + { WM8996_DSP1_RX_EQ_BAND_4_PG, 0x1103 }, + { WM8996_DSP1_RX_EQ_BAND_5_A, 0x564 }, + { WM8996_DSP1_RX_EQ_BAND_5_B, 0x559 }, + { WM8996_DSP1_RX_EQ_BAND_5_PG, 0x4000 }, + { WM8996_DSP2_TX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP2_TX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP2_RX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP2_RX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP2_TX_FILTERS, 0x2000 }, + { WM8996_DSP2_RX_FILTERS_1, 0x200 }, + { WM8996_DSP2_RX_FILTERS_2, 0x10 }, + { WM8996_DSP2_DRC_1, 0x98 }, + { WM8996_DSP2_DRC_2, 0x845 }, + { WM8996_DSP2_RX_EQ_GAINS_1, 0x6318 }, + { WM8996_DSP2_RX_EQ_GAINS_2, 0x6300 }, + { WM8996_DSP2_RX_EQ_BAND_1_A, 0xfca }, + { WM8996_DSP2_RX_EQ_BAND_1_B, 0x400 }, + { WM8996_DSP2_RX_EQ_BAND_1_PG, 0xd8 }, + { WM8996_DSP2_RX_EQ_BAND_2_A, 0x1eb5 }, + { WM8996_DSP2_RX_EQ_BAND_2_B, 0xf145 }, + { WM8996_DSP2_RX_EQ_BAND_2_C, 0xb75 }, + { WM8996_DSP2_RX_EQ_BAND_2_PG, 0x1c5 }, + { WM8996_DSP2_RX_EQ_BAND_3_A, 0x1c58 }, + { WM8996_DSP2_RX_EQ_BAND_3_B, 0xf373 }, + { WM8996_DSP2_RX_EQ_BAND_3_C, 0xa54 }, + { WM8996_DSP2_RX_EQ_BAND_3_PG, 0x558 }, + { WM8996_DSP2_RX_EQ_BAND_4_A, 0x168e }, + { WM8996_DSP2_RX_EQ_BAND_4_B, 0xf829 }, + { WM8996_DSP2_RX_EQ_BAND_4_C, 0x7ad }, + { WM8996_DSP2_RX_EQ_BAND_4_PG, 0x1103 }, + { WM8996_DSP2_RX_EQ_BAND_5_A, 0x564 }, + { WM8996_DSP2_RX_EQ_BAND_5_B, 0x559 }, + { WM8996_DSP2_RX_EQ_BAND_5_PG, 0x4000 }, + { WM8996_DAC1_MIXER_VOLUMES, 0x0 }, + { WM8996_DAC1_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC1_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC2_MIXER_VOLUMES, 0x0 }, + { WM8996_DAC2_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC2_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP1_TX_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP1_TX_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP2_TX_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP2_TX_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP_TX_MIXER_SELECT, 0x0 }, + { WM8996_DAC_SOFTMUTE, 0x0 }, + { WM8996_OVERSAMPLING, 0xd }, + { WM8996_SIDETONE, 0x1040 }, + { WM8996_GPIO_1, 0xa101 }, + { WM8996_GPIO_2, 0xa101 }, + { WM8996_GPIO_3, 0xa101 }, + { WM8996_GPIO_4, 0xa101 }, + { WM8996_GPIO_5, 0xa101 }, + { WM8996_PULL_CONTROL_1, 0x0 }, + { WM8996_PULL_CONTROL_2, 0x140 }, + { WM8996_INTERRUPT_STATUS_1_MASK, 0x1f }, + { WM8996_INTERRUPT_STATUS_2_MASK, 0x1ecf }, + { WM8996_LEFT_PDM_SPEAKER, 0x0 }, + { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, + { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, + { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, + { WM8996_WRITE_SEQUENCER_0, 0x1 }, + { WM8996_WRITE_SEQUENCER_1, 0x1 }, + { WM8996_WRITE_SEQUENCER_3, 0x6 }, + { WM8996_WRITE_SEQUENCER_4, 0x40 }, + { WM8996_WRITE_SEQUENCER_5, 0x1 }, + { WM8996_WRITE_SEQUENCER_6, 0xf }, + { WM8996_WRITE_SEQUENCER_7, 0x6 }, + { WM8996_WRITE_SEQUENCER_8, 0x1 }, + { WM8996_WRITE_SEQUENCER_9, 0x3 }, + { WM8996_WRITE_SEQUENCER_10, 0x104 }, + { WM8996_WRITE_SEQUENCER_12, 0x60 }, + { WM8996_WRITE_SEQUENCER_13, 0x11 }, + { WM8996_WRITE_SEQUENCER_14, 0x401 }, + { WM8996_WRITE_SEQUENCER_16, 0x50 }, + { WM8996_WRITE_SEQUENCER_17, 0x3 }, + { WM8996_WRITE_SEQUENCER_18, 0x100 }, + { WM8996_WRITE_SEQUENCER_20, 0x51 }, + { WM8996_WRITE_SEQUENCER_21, 0x3 }, + { WM8996_WRITE_SEQUENCER_22, 0x104 }, + { WM8996_WRITE_SEQUENCER_23, 0xa }, + { WM8996_WRITE_SEQUENCER_24, 0x60 }, + { WM8996_WRITE_SEQUENCER_25, 0x3b }, + { WM8996_WRITE_SEQUENCER_26, 0x502 }, + { WM8996_WRITE_SEQUENCER_27, 0x100 }, + { WM8996_WRITE_SEQUENCER_28, 0x2fff }, + { WM8996_WRITE_SEQUENCER_32, 0x2fff }, + { WM8996_WRITE_SEQUENCER_36, 0x2fff }, + { WM8996_WRITE_SEQUENCER_40, 0x2fff }, + { WM8996_WRITE_SEQUENCER_44, 0x2fff }, + { WM8996_WRITE_SEQUENCER_48, 0x2fff }, + { WM8996_WRITE_SEQUENCER_52, 0x2fff }, + { WM8996_WRITE_SEQUENCER_56, 0x2fff }, + { WM8996_WRITE_SEQUENCER_60, 0x2fff }, + { WM8996_WRITE_SEQUENCER_64, 0x1 }, + { WM8996_WRITE_SEQUENCER_65, 0x1 }, + { WM8996_WRITE_SEQUENCER_67, 0x6 }, + { WM8996_WRITE_SEQUENCER_68, 0x40 }, + { WM8996_WRITE_SEQUENCER_69, 0x1 }, + { WM8996_WRITE_SEQUENCER_70, 0xf }, + { WM8996_WRITE_SEQUENCER_71, 0x6 }, + { WM8996_WRITE_SEQUENCER_72, 0x1 }, + { WM8996_WRITE_SEQUENCER_73, 0x3 }, + { WM8996_WRITE_SEQUENCER_74, 0x104 }, + { WM8996_WRITE_SEQUENCER_76, 0x60 }, + { WM8996_WRITE_SEQUENCER_77, 0x11 }, + { WM8996_WRITE_SEQUENCER_78, 0x401 }, + { WM8996_WRITE_SEQUENCER_80, 0x50 }, + { WM8996_WRITE_SEQUENCER_81, 0x3 }, + { WM8996_WRITE_SEQUENCER_82, 0x100 }, + { WM8996_WRITE_SEQUENCER_84, 0x60 }, + { WM8996_WRITE_SEQUENCER_85, 0x3b }, + { WM8996_WRITE_SEQUENCER_86, 0x502 }, + { WM8996_WRITE_SEQUENCER_87, 0x100 }, + { WM8996_WRITE_SEQUENCER_88, 0x2fff }, + { WM8996_WRITE_SEQUENCER_92, 0x2fff }, + { WM8996_WRITE_SEQUENCER_96, 0x2fff }, + { WM8996_WRITE_SEQUENCER_100, 0x2fff }, + { WM8996_WRITE_SEQUENCER_104, 0x2fff }, + { WM8996_WRITE_SEQUENCER_108, 0x2fff }, + { WM8996_WRITE_SEQUENCER_112, 0x2fff }, + { WM8996_WRITE_SEQUENCER_116, 0x2fff }, + { WM8996_WRITE_SEQUENCER_120, 0x2fff }, + { WM8996_WRITE_SEQUENCER_124, 0x2fff }, + { WM8996_WRITE_SEQUENCER_128, 0x1 }, + { WM8996_WRITE_SEQUENCER_129, 0x1 }, + { WM8996_WRITE_SEQUENCER_131, 0x6 }, + { WM8996_WRITE_SEQUENCER_132, 0x40 }, + { WM8996_WRITE_SEQUENCER_133, 0x1 }, + { WM8996_WRITE_SEQUENCER_134, 0xf }, + { WM8996_WRITE_SEQUENCER_135, 0x6 }, + { WM8996_WRITE_SEQUENCER_136, 0x1 }, + { WM8996_WRITE_SEQUENCER_137, 0x3 }, + { WM8996_WRITE_SEQUENCER_138, 0x106 }, + { WM8996_WRITE_SEQUENCER_140, 0x61 }, + { WM8996_WRITE_SEQUENCER_141, 0x11 }, + { WM8996_WRITE_SEQUENCER_142, 0x401 }, + { WM8996_WRITE_SEQUENCER_144, 0x50 }, + { WM8996_WRITE_SEQUENCER_145, 0x3 }, + { WM8996_WRITE_SEQUENCER_146, 0x102 }, + { WM8996_WRITE_SEQUENCER_148, 0x51 }, + { WM8996_WRITE_SEQUENCER_149, 0x3 }, + { WM8996_WRITE_SEQUENCER_150, 0x106 }, + { WM8996_WRITE_SEQUENCER_151, 0xa }, + { WM8996_WRITE_SEQUENCER_152, 0x61 }, + { WM8996_WRITE_SEQUENCER_153, 0x3b }, + { WM8996_WRITE_SEQUENCER_154, 0x502 }, + { WM8996_WRITE_SEQUENCER_155, 0x100 }, + { WM8996_WRITE_SEQUENCER_156, 0x2fff }, + { WM8996_WRITE_SEQUENCER_160, 0x2fff }, + { WM8996_WRITE_SEQUENCER_164, 0x2fff }, + { WM8996_WRITE_SEQUENCER_168, 0x2fff }, + { WM8996_WRITE_SEQUENCER_172, 0x2fff }, + { WM8996_WRITE_SEQUENCER_176, 0x2fff }, + { WM8996_WRITE_SEQUENCER_180, 0x2fff }, + { WM8996_WRITE_SEQUENCER_184, 0x2fff }, + { WM8996_WRITE_SEQUENCER_188, 0x2fff }, + { WM8996_WRITE_SEQUENCER_192, 0x1 }, + { WM8996_WRITE_SEQUENCER_193, 0x1 }, + { WM8996_WRITE_SEQUENCER_195, 0x6 }, + { WM8996_WRITE_SEQUENCER_196, 0x40 }, + { WM8996_WRITE_SEQUENCER_197, 0x1 }, + { WM8996_WRITE_SEQUENCER_198, 0xf }, + { WM8996_WRITE_SEQUENCER_199, 0x6 }, + { WM8996_WRITE_SEQUENCER_200, 0x1 }, + { WM8996_WRITE_SEQUENCER_201, 0x3 }, + { WM8996_WRITE_SEQUENCER_202, 0x106 }, + { WM8996_WRITE_SEQUENCER_204, 0x61 }, + { WM8996_WRITE_SEQUENCER_205, 0x11 }, + { WM8996_WRITE_SEQUENCER_206, 0x401 }, + { WM8996_WRITE_SEQUENCER_208, 0x50 }, + { WM8996_WRITE_SEQUENCER_209, 0x3 }, + { WM8996_WRITE_SEQUENCER_210, 0x102 }, + { WM8996_WRITE_SEQUENCER_212, 0x61 }, + { WM8996_WRITE_SEQUENCER_213, 0x3b }, + { WM8996_WRITE_SEQUENCER_214, 0x502 }, + { WM8996_WRITE_SEQUENCER_215, 0x100 }, + { WM8996_WRITE_SEQUENCER_216, 0x2fff }, + { WM8996_WRITE_SEQUENCER_220, 0x2fff }, + { WM8996_WRITE_SEQUENCER_224, 0x2fff }, + { WM8996_WRITE_SEQUENCER_228, 0x2fff }, + { WM8996_WRITE_SEQUENCER_232, 0x2fff }, + { WM8996_WRITE_SEQUENCER_236, 0x2fff }, + { WM8996_WRITE_SEQUENCER_240, 0x2fff }, + { WM8996_WRITE_SEQUENCER_244, 0x2fff }, + { WM8996_WRITE_SEQUENCER_248, 0x2fff }, + { WM8996_WRITE_SEQUENCER_252, 0x2fff }, + { WM8996_WRITE_SEQUENCER_256, 0x60 }, + { WM8996_WRITE_SEQUENCER_258, 0x601 }, + { WM8996_WRITE_SEQUENCER_260, 0x50 }, + { WM8996_WRITE_SEQUENCER_262, 0x100 }, + { WM8996_WRITE_SEQUENCER_264, 0x1 }, + { WM8996_WRITE_SEQUENCER_266, 0x104 }, + { WM8996_WRITE_SEQUENCER_267, 0x100 }, + { WM8996_WRITE_SEQUENCER_268, 0x2fff }, + { WM8996_WRITE_SEQUENCER_272, 0x2fff }, + { WM8996_WRITE_SEQUENCER_276, 0x2fff }, + { WM8996_WRITE_SEQUENCER_280, 0x2fff }, + { WM8996_WRITE_SEQUENCER_284, 0x2fff }, + { WM8996_WRITE_SEQUENCER_288, 0x2fff }, + { WM8996_WRITE_SEQUENCER_292, 0x2fff }, + { WM8996_WRITE_SEQUENCER_296, 0x2fff }, + { WM8996_WRITE_SEQUENCER_300, 0x2fff }, + { WM8996_WRITE_SEQUENCER_304, 0x2fff }, + { WM8996_WRITE_SEQUENCER_308, 0x2fff }, + { WM8996_WRITE_SEQUENCER_312, 0x2fff }, + { WM8996_WRITE_SEQUENCER_316, 0x2fff }, + { WM8996_WRITE_SEQUENCER_320, 0x61 }, + { WM8996_WRITE_SEQUENCER_322, 0x601 }, + { WM8996_WRITE_SEQUENCER_324, 0x50 }, + { WM8996_WRITE_SEQUENCER_326, 0x102 }, + { WM8996_WRITE_SEQUENCER_328, 0x1 }, + { WM8996_WRITE_SEQUENCER_330, 0x106 }, + { WM8996_WRITE_SEQUENCER_331, 0x100 }, + { WM8996_WRITE_SEQUENCER_332, 0x2fff }, + { WM8996_WRITE_SEQUENCER_336, 0x2fff }, + { WM8996_WRITE_SEQUENCER_340, 0x2fff }, + { WM8996_WRITE_SEQUENCER_344, 0x2fff }, + { WM8996_WRITE_SEQUENCER_348, 0x2fff }, + { WM8996_WRITE_SEQUENCER_352, 0x2fff }, + { WM8996_WRITE_SEQUENCER_356, 0x2fff }, + { WM8996_WRITE_SEQUENCER_360, 0x2fff }, + { WM8996_WRITE_SEQUENCER_364, 0x2fff }, + { WM8996_WRITE_SEQUENCER_368, 0x2fff }, + { WM8996_WRITE_SEQUENCER_372, 0x2fff }, + { WM8996_WRITE_SEQUENCER_376, 0x2fff }, + { WM8996_WRITE_SEQUENCER_380, 0x2fff }, + { WM8996_WRITE_SEQUENCER_384, 0x60 }, + { WM8996_WRITE_SEQUENCER_386, 0x601 }, + { WM8996_WRITE_SEQUENCER_388, 0x61 }, + { WM8996_WRITE_SEQUENCER_390, 0x601 }, + { WM8996_WRITE_SEQUENCER_392, 0x50 }, + { WM8996_WRITE_SEQUENCER_394, 0x300 }, + { WM8996_WRITE_SEQUENCER_396, 0x1 }, + { WM8996_WRITE_SEQUENCER_398, 0x304 }, + { WM8996_WRITE_SEQUENCER_400, 0x40 }, + { WM8996_WRITE_SEQUENCER_402, 0xf }, + { WM8996_WRITE_SEQUENCER_404, 0x1 }, + { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); @@ -1413,8 +1482,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "SPKDAT", NULL, "SPKR PGA" }, }; -static int wm8996_readable_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8996_readable_register(struct device *dev, unsigned int reg) { /* Due to the sparseness of the register map the compiler * output from an explicit switch statement ends up being much @@ -1621,8 +1689,7 @@ static int wm8996_readable_register(struct snd_soc_codec *codec, } } -static int wm8996_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8996_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8996_SOFTWARE_RESET: @@ -1723,13 +1790,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - codec->cache_only = false; - snd_soc_cache_sync(codec); + regcache_cache_only(codec->control_data, false); + regcache_sync(codec->control_data); } break; case SND_SOC_BIAS_OFF: - codec->cache_only = true; + regcache_cache_only(codec->control_data, true); if (wm8996->pdata.ldo_ena >= 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), @@ -2692,6 +2759,18 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) "Failed to add ReTune Mobile controls: %d\n", ret); } +static const struct regmap_config wm8996_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM8996_MAX_REGISTER, + .reg_defaults = wm8996_reg, + .num_reg_defaults = ARRAY_SIZE(wm8996_reg), + .volatile_reg = wm8996_volatile_register, + .readable_reg = wm8996_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + static int wm8996_probe(struct snd_soc_codec *codec) { int ret; @@ -2707,10 +2786,17 @@ static int wm8996_probe(struct snd_soc_codec *codec) dapm->idle_bias_off = true; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = regmap_init_i2c(i2c, &wm8996_regmap); + if (IS_ERR(codec->control_data)) { + ret = PTR_ERR(codec->control_data); + dev_err(codec->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; + goto err_regmap; } for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) @@ -2720,7 +2806,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; + goto err_regmap; } wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; @@ -2788,7 +2874,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - codec->cache_only = true; + regcache_cache_only(codec->control_data, true); /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, @@ -2996,6 +3082,8 @@ err_cpvdd: regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_regmap: + regmap_exit(codec->control_data); err: return ret; } @@ -3019,6 +3107,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) &wm8996->disable_nb[i]); regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + regmap_exit(codec->control_data); return 0; } @@ -3028,12 +3117,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, .seq_notifier = wm8996_seq_notifier, - .reg_cache_size = WM8996_MAX_REGISTER + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8996_reg, - .volatile_register = wm8996_volatile_register, - .readable_register = wm8996_readable_register, - .compress_type = SND_SOC_RBTREE_COMPRESSION, .controls = wm8996_snd_controls, .num_controls = ARRAY_SIZE(wm8996_snd_controls), .dapm_widgets = wm8996_dapm_widgets, -- cgit v1.2.1 From ee5f387226d13535f41bda0e8a2cf3843fc4c080 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Sep 2011 19:51:07 +0100 Subject: ASoC: Move most WM8996 resource acquisition to I2C probe Now that the WM8996 driver is using the regmap API for register I/O we no longer need the ASoC card to be active in order to interact with the chip. In order to be more idiomatic for Linux move most of the existing probe() function out into the I2C probe() function prior to registration with ASoC. The IRQ and GPIO init will be moved separately as these are slightly more involved. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 183 ++++++++++++++++++++++++---------------------- 1 file changed, 96 insertions(+), 87 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 5671fd398e8a..cb9709ad66fd 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -50,6 +50,7 @@ static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = { }; struct wm8996_priv { + struct regmap *regmap; struct snd_soc_codec *codec; int ldo1ena; @@ -106,7 +107,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8996->codec->cache_sync = 1; \ + regcache_cache_only(wm8996->regmap, true); \ } \ return 0; \ } @@ -1713,9 +1714,15 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) } } -static int wm8996_reset(struct snd_soc_codec *codec) +static int wm8996_reset(struct wm8996_priv *wm8996) { - return snd_soc_write(codec, WM8996_SOFTWARE_RESET, 0x8915); + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + return 0; + } else { + return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, + 0x8915); + } } static const int bclk_divs[] = { @@ -2786,40 +2793,18 @@ static int wm8996_probe(struct snd_soc_codec *codec) dapm->idle_bias_off = true; - codec->control_data = regmap_init_i2c(i2c, &wm8996_regmap); - if (IS_ERR(codec->control_data)) { - ret = PTR_ERR(codec->control_data); - dev_err(codec->dev, "regmap_init() failed: %d\n", ret); - goto err; - } + codec->control_data = wm8996->regmap; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_regmap; - } - - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - wm8996->supplies[i].supply = wm8996_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8996->supplies), - wm8996->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err_regmap; + goto err; } wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); - if (IS_ERR(wm8996->cpvdd)) { - ret = PTR_ERR(wm8996->cpvdd); - dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); - goto err_get; - } - /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { ret = regulator_register_notifier(wm8996->supplies[i].consumer, @@ -2831,49 +2816,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), - wm8996->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_cpvdd; - } - - if (wm8996->pdata.ldo_ena >= 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); - msleep(5); - } - - ret = snd_soc_read(codec, WM8996_SOFTWARE_RESET); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register: %d\n", ret); - goto err_enable; - } - if (ret != 0x8915) { - dev_err(codec->dev, "Device is not a WM8996, ID %x\n", ret); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8996_CHIP_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_enable; - } - - dev_info(codec->dev, "revision %c\n", - (ret & WM8996_CHIP_REV_MASK) + 'A'); - - if (wm8996->pdata.ldo_ena >= 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - } else { - ret = wm8996_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - } - regcache_cache_only(codec->control_data, true); /* Apply platform data settings */ @@ -3032,8 +2974,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) WM8996_AIF2TX_LRCLK_MODE, WM8996_AIF2TX_LRCLK_MODE); - regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - wm8996_init_gpio(codec); if (i2c->irq) { @@ -3073,17 +3013,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) return 0; -err_enable: - if (wm8996->pdata.ldo_ena >= 0) - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - - regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -err_cpvdd: - regulator_put(wm8996->cpvdd); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -err_regmap: - regmap_exit(codec->control_data); err: return ret; } @@ -3107,7 +3036,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) &wm8996->disable_nb[i]); regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - regmap_exit(codec->control_data); return 0; } @@ -3181,7 +3109,8 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8996_priv *wm8996; - int ret; + int ret, i; + unsigned int reg; wm8996 = kzalloc(sizeof(struct wm8996_priv), GFP_KERNEL); if (wm8996 == NULL) @@ -3203,14 +3132,89 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, } } + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) + wm8996->supplies[i].supply = wm8996_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8996->supplies), + wm8996->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err_gpio; + } + + wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm8996->cpvdd)) { + ret = PTR_ERR(wm8996->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), + wm8996->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + goto err_cpvdd; + } + + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); + msleep(5); + } + + wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap); + if (IS_ERR(wm8996->regmap)) { + ret = PTR_ERR(wm8996->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + goto err_enable; + } + + ret = regmap_read(wm8996->regmap, WM8996_SOFTWARE_RESET, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err_regmap; + } + if (reg != 0x8915) { + dev_err(&i2c->dev, "Device is not a WM8996, ID %x\n", ret); + ret = -EINVAL; + goto err_regmap; + } + + ret = regmap_read(wm8996->regmap, WM8996_CHIP_REVISION, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read device revision: %d\n", + ret); + goto err_regmap; + } + + dev_info(&i2c->dev, "revision %c\n", + (reg & WM8996_CHIP_REV_MASK) + 'A'); + + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + + ret = wm8996_reset(wm8996); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_regmap; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8996, wm8996_dai, ARRAY_SIZE(wm8996_dai)); if (ret < 0) - goto err_gpio; + goto err_regmap; return ret; +err_regmap: + regmap_exit(wm8996->regmap); +err_enable: + if (wm8996->pdata.ldo_ena > 0) + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_cpvdd: + regulator_put(wm8996->cpvdd); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err_gpio: if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); @@ -3225,8 +3229,13 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) struct wm8996_priv *wm8996 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - if (wm8996->pdata.ldo_ena > 0) + regulator_put(wm8996->cpvdd); + regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + regmap_exit(wm8996->regmap); + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); + } kfree(wm8996); return 0; } -- cgit v1.2.1 From b2d1e23373fde66d5532ffdfd0f1e650174b83f6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Sep 2011 23:04:06 +0100 Subject: ASoC: Convert WM8996 gpiolib to regmap Actually pretty straightforward. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 56 +++++++++++++++++++++++------------------------ 1 file changed, 27 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index cb9709ad66fd..fd5bb1ad6912 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -50,6 +50,7 @@ static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = { }; struct wm8996_priv { + struct device *dev; struct regmap *regmap; struct snd_soc_codec *codec; @@ -2325,48 +2326,45 @@ static inline struct wm8996_priv *gpio_to_wm8996(struct gpio_chip *chip) static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); - struct snd_soc_codec *codec = wm8996->codec; - snd_soc_update_bits(codec, WM8996_GPIO_1 + offset, - WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT); + regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, + WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT); } static int wm8996_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); - struct snd_soc_codec *codec = wm8996->codec; int val; val = (1 << WM8996_GP1_FN_SHIFT) | (!!value << WM8996_GP1_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8996_GPIO_1 + offset, - WM8996_GP1_FN_MASK | WM8996_GP1_DIR | - WM8996_GP1_LVL, val); + return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, + WM8996_GP1_FN_MASK | WM8996_GP1_DIR | + WM8996_GP1_LVL, val); } static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); - struct snd_soc_codec *codec = wm8996->codec; + unsigned int reg; int ret; - ret = snd_soc_read(codec, WM8996_GPIO_1 + offset); + ret = regmap_read(wm8996->regmap, WM8996_GPIO_1 + offset, ®); if (ret < 0) return ret; - return (ret & WM8996_GP1_LVL) != 0; + return (reg & WM8996_GP1_LVL) != 0; } static int wm8996_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); - struct snd_soc_codec *codec = wm8996->codec; - return snd_soc_update_bits(codec, WM8996_GPIO_1 + offset, - WM8996_GP1_FN_MASK | WM8996_GP1_DIR, - (1 << WM8996_GP1_FN_SHIFT) | - (1 << WM8996_GP1_DIR_SHIFT)); + return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, + WM8996_GP1_FN_MASK | WM8996_GP1_DIR, + (1 << WM8996_GP1_FN_SHIFT) | + (1 << WM8996_GP1_DIR_SHIFT)); } static struct gpio_chip wm8996_template_chip = { @@ -2379,14 +2377,13 @@ static struct gpio_chip wm8996_template_chip = { .can_sleep = 1, }; -static void wm8996_init_gpio(struct snd_soc_codec *codec) +static void wm8996_init_gpio(struct wm8996_priv *wm8996) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret; wm8996->gpio_chip = wm8996_template_chip; wm8996->gpio_chip.ngpio = 5; - wm8996->gpio_chip.dev = codec->dev; + wm8996->gpio_chip.dev = wm8996->dev; if (wm8996->pdata.gpio_base) wm8996->gpio_chip.base = wm8996->pdata.gpio_base; @@ -2395,24 +2392,23 @@ static void wm8996_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm8996->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm8996_free_gpio(struct snd_soc_codec *codec) +static void wm8996_free_gpio(struct wm8996_priv *wm8996) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret; ret = gpiochip_remove(&wm8996->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm8996_init_gpio(struct snd_soc_codec *codec) +static void wm8996_init_gpio(struct wm8996_priv *wm8996) { } -static void wm8996_free_gpio(struct snd_soc_codec *codec) +static void wm8996_free_gpio(struct wm8996_priv *wm8996) { } #endif @@ -2974,8 +2970,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) WM8996_AIF2TX_LRCLK_MODE, WM8996_AIF2TX_LRCLK_MODE); - wm8996_init_gpio(codec); - if (i2c->irq) { if (wm8996->pdata.irq_flags) irq_flags = wm8996->pdata.irq_flags; @@ -3029,8 +3023,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); - wm8996_free_gpio(codec); - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); @@ -3117,6 +3109,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8996); + wm8996->dev = &i2c->dev; if (dev_get_platdata(&i2c->dev)) memcpy(&wm8996->pdata, dev_get_platdata(&i2c->dev), @@ -3197,14 +3190,18 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_regmap; } + wm8996_init_gpio(wm8996); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8996, wm8996_dai, ARRAY_SIZE(wm8996_dai)); if (ret < 0) - goto err_regmap; + goto err_gpiolib; return ret; +err_gpiolib: + wm8996_free_gpio(wm8996); err_regmap: regmap_exit(wm8996->regmap); err_enable: @@ -3229,6 +3226,7 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) struct wm8996_priv *wm8996 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + wm8996_free_gpio(wm8996); regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); regmap_exit(wm8996->regmap); -- cgit v1.2.1 From 7b16f5601295d0dfd0d48753b9253d41957587fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Nov 2011 19:32:25 +0000 Subject: ASoC: Convert WM8962 to direct regmap usage This initial conversion just moves the register init, regulator acquisition and device verification out to the I2C probe(). Movement of other parts of the driver like the GPIO and beep generation code will follow. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1593 +++++++++++++++++++++++---------------------- 1 file changed, 802 insertions(+), 791 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3fc9d2f74735..6d82b35a70d0 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -50,6 +51,7 @@ static const char *wm8962_supply_names[WM8962_NUM_SUPPLIES] = { /* codec private data */ struct wm8962_priv { + struct regmap *regmap; struct snd_soc_codec *codec; int sysclk; @@ -95,7 +97,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \ struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8962->codec->cache_sync = 1; \ + regcache_cache_only(wm8962->regmap, true); \ } \ return 0; \ } @@ -109,691 +111,691 @@ WM8962_REGULATOR_EVENT(5) WM8962_REGULATOR_EVENT(6) WM8962_REGULATOR_EVENT(7) -static const u16 wm8962_reg[WM8962_MAX_REGISTER + 1] = { - [0] = 0x009F, /* R0 - Left Input volume */ - [1] = 0x049F, /* R1 - Right Input volume */ - [2] = 0x0000, /* R2 - HPOUTL volume */ - [3] = 0x0000, /* R3 - HPOUTR volume */ - [4] = 0x0020, /* R4 - Clocking1 */ - [5] = 0x0018, /* R5 - ADC & DAC Control 1 */ - [6] = 0x2008, /* R6 - ADC & DAC Control 2 */ - [7] = 0x000A, /* R7 - Audio Interface 0 */ - [8] = 0x01E4, /* R8 - Clocking2 */ - [9] = 0x0300, /* R9 - Audio Interface 1 */ - [10] = 0x00C0, /* R10 - Left DAC volume */ - [11] = 0x00C0, /* R11 - Right DAC volume */ - - [14] = 0x0040, /* R14 - Audio Interface 2 */ - [15] = 0x6243, /* R15 - Software Reset */ - - [17] = 0x007B, /* R17 - ALC1 */ - [18] = 0x0000, /* R18 - ALC2 */ - [19] = 0x1C32, /* R19 - ALC3 */ - [20] = 0x3200, /* R20 - Noise Gate */ - [21] = 0x00C0, /* R21 - Left ADC volume */ - [22] = 0x00C0, /* R22 - Right ADC volume */ - [23] = 0x0160, /* R23 - Additional control(1) */ - [24] = 0x0000, /* R24 - Additional control(2) */ - [25] = 0x0000, /* R25 - Pwr Mgmt (1) */ - [26] = 0x0000, /* R26 - Pwr Mgmt (2) */ - [27] = 0x0010, /* R27 - Additional Control (3) */ - [28] = 0x0000, /* R28 - Anti-pop */ - - [30] = 0x005E, /* R30 - Clocking 3 */ - [31] = 0x0000, /* R31 - Input mixer control (1) */ - [32] = 0x0145, /* R32 - Left input mixer volume */ - [33] = 0x0145, /* R33 - Right input mixer volume */ - [34] = 0x0009, /* R34 - Input mixer control (2) */ - [35] = 0x0003, /* R35 - Input bias control */ - [37] = 0x0008, /* R37 - Left input PGA control */ - [38] = 0x0008, /* R38 - Right input PGA control */ - - [40] = 0x0000, /* R40 - SPKOUTL volume */ - [41] = 0x0000, /* R41 - SPKOUTR volume */ - - [47] = 0x0000, /* R47 - Thermal Shutdown Status */ - [48] = 0x8027, /* R48 - Additional Control (4) */ - [49] = 0x0010, /* R49 - Class D Control 1 */ - - [51] = 0x0003, /* R51 - Class D Control 2 */ - - [56] = 0x0506, /* R56 - Clocking 4 */ - [57] = 0x0000, /* R57 - DAC DSP Mixing (1) */ - [58] = 0x0000, /* R58 - DAC DSP Mixing (2) */ - - [60] = 0x0300, /* R60 - DC Servo 0 */ - [61] = 0x0300, /* R61 - DC Servo 1 */ - - [64] = 0x0810, /* R64 - DC Servo 4 */ - - [66] = 0x0000, /* R66 - DC Servo 6 */ - - [68] = 0x001B, /* R68 - Analogue PGA Bias */ - [69] = 0x0000, /* R69 - Analogue HP 0 */ - - [71] = 0x01FB, /* R71 - Analogue HP 2 */ - [72] = 0x0000, /* R72 - Charge Pump 1 */ - - [82] = 0x0004, /* R82 - Charge Pump B */ - - [87] = 0x0000, /* R87 - Write Sequencer Control 1 */ - - [90] = 0x0000, /* R90 - Write Sequencer Control 2 */ - - [93] = 0x0000, /* R93 - Write Sequencer Control 3 */ - [94] = 0x0000, /* R94 - Control Interface */ - - [99] = 0x0000, /* R99 - Mixer Enables */ - [100] = 0x0000, /* R100 - Headphone Mixer (1) */ - [101] = 0x0000, /* R101 - Headphone Mixer (2) */ - [102] = 0x013F, /* R102 - Headphone Mixer (3) */ - [103] = 0x013F, /* R103 - Headphone Mixer (4) */ - - [105] = 0x0000, /* R105 - Speaker Mixer (1) */ - [106] = 0x0000, /* R106 - Speaker Mixer (2) */ - [107] = 0x013F, /* R107 - Speaker Mixer (3) */ - [108] = 0x013F, /* R108 - Speaker Mixer (4) */ - [109] = 0x0003, /* R109 - Speaker Mixer (5) */ - [110] = 0x0002, /* R110 - Beep Generator (1) */ - - [115] = 0x0006, /* R115 - Oscillator Trim (3) */ - [116] = 0x0026, /* R116 - Oscillator Trim (4) */ - - [119] = 0x0000, /* R119 - Oscillator Trim (7) */ - - [124] = 0x0011, /* R124 - Analogue Clocking1 */ - [125] = 0x004B, /* R125 - Analogue Clocking2 */ - [126] = 0x000D, /* R126 - Analogue Clocking3 */ - [127] = 0x0000, /* R127 - PLL Software Reset */ - - [129] = 0x0000, /* R129 - PLL2 */ - - [131] = 0x0000, /* R131 - PLL 4 */ - - [136] = 0x0067, /* R136 - PLL 9 */ - [137] = 0x001C, /* R137 - PLL 10 */ - [138] = 0x0071, /* R138 - PLL 11 */ - [139] = 0x00C7, /* R139 - PLL 12 */ - [140] = 0x0067, /* R140 - PLL 13 */ - [141] = 0x0048, /* R141 - PLL 14 */ - [142] = 0x0022, /* R142 - PLL 15 */ - [143] = 0x0097, /* R143 - PLL 16 */ - - [155] = 0x000C, /* R155 - FLL Control (1) */ - [156] = 0x0039, /* R156 - FLL Control (2) */ - [157] = 0x0180, /* R157 - FLL Control (3) */ - - [159] = 0x0032, /* R159 - FLL Control (5) */ - [160] = 0x0018, /* R160 - FLL Control (6) */ - [161] = 0x007D, /* R161 - FLL Control (7) */ - [162] = 0x0008, /* R162 - FLL Control (8) */ - - [252] = 0x0005, /* R252 - General test 1 */ - - [256] = 0x0000, /* R256 - DF1 */ - [257] = 0x0000, /* R257 - DF2 */ - [258] = 0x0000, /* R258 - DF3 */ - [259] = 0x0000, /* R259 - DF4 */ - [260] = 0x0000, /* R260 - DF5 */ - [261] = 0x0000, /* R261 - DF6 */ - [262] = 0x0000, /* R262 - DF7 */ - - [264] = 0x0000, /* R264 - LHPF1 */ - [265] = 0x0000, /* R265 - LHPF2 */ - - [268] = 0x0000, /* R268 - THREED1 */ - [269] = 0x0000, /* R269 - THREED2 */ - [270] = 0x0000, /* R270 - THREED3 */ - [271] = 0x0000, /* R271 - THREED4 */ - - [276] = 0x000C, /* R276 - DRC 1 */ - [277] = 0x0925, /* R277 - DRC 2 */ - [278] = 0x0000, /* R278 - DRC 3 */ - [279] = 0x0000, /* R279 - DRC 4 */ - [280] = 0x0000, /* R280 - DRC 5 */ - - [285] = 0x0000, /* R285 - Tloopback */ - - [335] = 0x0004, /* R335 - EQ1 */ - [336] = 0x6318, /* R336 - EQ2 */ - [337] = 0x6300, /* R337 - EQ3 */ - [338] = 0x0FCA, /* R338 - EQ4 */ - [339] = 0x0400, /* R339 - EQ5 */ - [340] = 0x00D8, /* R340 - EQ6 */ - [341] = 0x1EB5, /* R341 - EQ7 */ - [342] = 0xF145, /* R342 - EQ8 */ - [343] = 0x0B75, /* R343 - EQ9 */ - [344] = 0x01C5, /* R344 - EQ10 */ - [345] = 0x1C58, /* R345 - EQ11 */ - [346] = 0xF373, /* R346 - EQ12 */ - [347] = 0x0A54, /* R347 - EQ13 */ - [348] = 0x0558, /* R348 - EQ14 */ - [349] = 0x168E, /* R349 - EQ15 */ - [350] = 0xF829, /* R350 - EQ16 */ - [351] = 0x07AD, /* R351 - EQ17 */ - [352] = 0x1103, /* R352 - EQ18 */ - [353] = 0x0564, /* R353 - EQ19 */ - [354] = 0x0559, /* R354 - EQ20 */ - [355] = 0x4000, /* R355 - EQ21 */ - [356] = 0x6318, /* R356 - EQ22 */ - [357] = 0x6300, /* R357 - EQ23 */ - [358] = 0x0FCA, /* R358 - EQ24 */ - [359] = 0x0400, /* R359 - EQ25 */ - [360] = 0x00D8, /* R360 - EQ26 */ - [361] = 0x1EB5, /* R361 - EQ27 */ - [362] = 0xF145, /* R362 - EQ28 */ - [363] = 0x0B75, /* R363 - EQ29 */ - [364] = 0x01C5, /* R364 - EQ30 */ - [365] = 0x1C58, /* R365 - EQ31 */ - [366] = 0xF373, /* R366 - EQ32 */ - [367] = 0x0A54, /* R367 - EQ33 */ - [368] = 0x0558, /* R368 - EQ34 */ - [369] = 0x168E, /* R369 - EQ35 */ - [370] = 0xF829, /* R370 - EQ36 */ - [371] = 0x07AD, /* R371 - EQ37 */ - [372] = 0x1103, /* R372 - EQ38 */ - [373] = 0x0564, /* R373 - EQ39 */ - [374] = 0x0559, /* R374 - EQ40 */ - [375] = 0x4000, /* R375 - EQ41 */ - - [513] = 0x0000, /* R513 - GPIO 2 */ - [514] = 0x0000, /* R514 - GPIO 3 */ - - [516] = 0x8100, /* R516 - GPIO 5 */ - [517] = 0x8100, /* R517 - GPIO 6 */ - - [560] = 0x0000, /* R560 - Interrupt Status 1 */ - [561] = 0x0000, /* R561 - Interrupt Status 2 */ - - [568] = 0x0030, /* R568 - Interrupt Status 1 Mask */ - [569] = 0xFFED, /* R569 - Interrupt Status 2 Mask */ - - [576] = 0x0000, /* R576 - Interrupt Control */ - - [584] = 0x002D, /* R584 - IRQ Debounce */ - - [586] = 0x0000, /* R586 - MICINT Source Pol */ - - [768] = 0x1C00, /* R768 - DSP2 Power Management */ - - [1037] = 0x0000, /* R1037 - DSP2_ExecControl */ - - [8192] = 0x0000, /* R8192 - DSP2 Instruction RAM 0 */ - - [9216] = 0x0030, /* R9216 - DSP2 Address RAM 2 */ - [9217] = 0x0000, /* R9217 - DSP2 Address RAM 1 */ - [9218] = 0x0000, /* R9218 - DSP2 Address RAM 0 */ - - [12288] = 0x0000, /* R12288 - DSP2 Data1 RAM 1 */ - [12289] = 0x0000, /* R12289 - DSP2 Data1 RAM 0 */ - - [13312] = 0x0000, /* R13312 - DSP2 Data2 RAM 1 */ - [13313] = 0x0000, /* R13313 - DSP2 Data2 RAM 0 */ - - [14336] = 0x0000, /* R14336 - DSP2 Data3 RAM 1 */ - [14337] = 0x0000, /* R14337 - DSP2 Data3 RAM 0 */ - - [15360] = 0x000A, /* R15360 - DSP2 Coeff RAM 0 */ - - [16384] = 0x0000, /* R16384 - RETUNEADC_SHARED_COEFF_1 */ - [16385] = 0x0000, /* R16385 - RETUNEADC_SHARED_COEFF_0 */ - [16386] = 0x0000, /* R16386 - RETUNEDAC_SHARED_COEFF_1 */ - [16387] = 0x0000, /* R16387 - RETUNEDAC_SHARED_COEFF_0 */ - [16388] = 0x0000, /* R16388 - SOUNDSTAGE_ENABLES_1 */ - [16389] = 0x0000, /* R16389 - SOUNDSTAGE_ENABLES_0 */ - - [16896] = 0x0002, /* R16896 - HDBASS_AI_1 */ - [16897] = 0xBD12, /* R16897 - HDBASS_AI_0 */ - [16898] = 0x007C, /* R16898 - HDBASS_AR_1 */ - [16899] = 0x586C, /* R16899 - HDBASS_AR_0 */ - [16900] = 0x0053, /* R16900 - HDBASS_B_1 */ - [16901] = 0x8121, /* R16901 - HDBASS_B_0 */ - [16902] = 0x003F, /* R16902 - HDBASS_K_1 */ - [16903] = 0x8BD8, /* R16903 - HDBASS_K_0 */ - [16904] = 0x0032, /* R16904 - HDBASS_N1_1 */ - [16905] = 0xF52D, /* R16905 - HDBASS_N1_0 */ - [16906] = 0x0065, /* R16906 - HDBASS_N2_1 */ - [16907] = 0xAC8C, /* R16907 - HDBASS_N2_0 */ - [16908] = 0x006B, /* R16908 - HDBASS_N3_1 */ - [16909] = 0xE087, /* R16909 - HDBASS_N3_0 */ - [16910] = 0x0072, /* R16910 - HDBASS_N4_1 */ - [16911] = 0x1483, /* R16911 - HDBASS_N4_0 */ - [16912] = 0x0072, /* R16912 - HDBASS_N5_1 */ - [16913] = 0x1483, /* R16913 - HDBASS_N5_0 */ - [16914] = 0x0043, /* R16914 - HDBASS_X1_1 */ - [16915] = 0x3525, /* R16915 - HDBASS_X1_0 */ - [16916] = 0x0006, /* R16916 - HDBASS_X2_1 */ - [16917] = 0x6A4A, /* R16917 - HDBASS_X2_0 */ - [16918] = 0x0043, /* R16918 - HDBASS_X3_1 */ - [16919] = 0x6079, /* R16919 - HDBASS_X3_0 */ - [16920] = 0x0008, /* R16920 - HDBASS_ATK_1 */ - [16921] = 0x0000, /* R16921 - HDBASS_ATK_0 */ - [16922] = 0x0001, /* R16922 - HDBASS_DCY_1 */ - [16923] = 0x0000, /* R16923 - HDBASS_DCY_0 */ - [16924] = 0x0059, /* R16924 - HDBASS_PG_1 */ - [16925] = 0x999A, /* R16925 - HDBASS_PG_0 */ - - [17048] = 0x0083, /* R17408 - HPF_C_1 */ - [17049] = 0x98AD, /* R17409 - HPF_C_0 */ - - [17920] = 0x007F, /* R17920 - ADCL_RETUNE_C1_1 */ - [17921] = 0xFFFF, /* R17921 - ADCL_RETUNE_C1_0 */ - [17922] = 0x0000, /* R17922 - ADCL_RETUNE_C2_1 */ - [17923] = 0x0000, /* R17923 - ADCL_RETUNE_C2_0 */ - [17924] = 0x0000, /* R17924 - ADCL_RETUNE_C3_1 */ - [17925] = 0x0000, /* R17925 - ADCL_RETUNE_C3_0 */ - [17926] = 0x0000, /* R17926 - ADCL_RETUNE_C4_1 */ - [17927] = 0x0000, /* R17927 - ADCL_RETUNE_C4_0 */ - [17928] = 0x0000, /* R17928 - ADCL_RETUNE_C5_1 */ - [17929] = 0x0000, /* R17929 - ADCL_RETUNE_C5_0 */ - [17930] = 0x0000, /* R17930 - ADCL_RETUNE_C6_1 */ - [17931] = 0x0000, /* R17931 - ADCL_RETUNE_C6_0 */ - [17932] = 0x0000, /* R17932 - ADCL_RETUNE_C7_1 */ - [17933] = 0x0000, /* R17933 - ADCL_RETUNE_C7_0 */ - [17934] = 0x0000, /* R17934 - ADCL_RETUNE_C8_1 */ - [17935] = 0x0000, /* R17935 - ADCL_RETUNE_C8_0 */ - [17936] = 0x0000, /* R17936 - ADCL_RETUNE_C9_1 */ - [17937] = 0x0000, /* R17937 - ADCL_RETUNE_C9_0 */ - [17938] = 0x0000, /* R17938 - ADCL_RETUNE_C10_1 */ - [17939] = 0x0000, /* R17939 - ADCL_RETUNE_C10_0 */ - [17940] = 0x0000, /* R17940 - ADCL_RETUNE_C11_1 */ - [17941] = 0x0000, /* R17941 - ADCL_RETUNE_C11_0 */ - [17942] = 0x0000, /* R17942 - ADCL_RETUNE_C12_1 */ - [17943] = 0x0000, /* R17943 - ADCL_RETUNE_C12_0 */ - [17944] = 0x0000, /* R17944 - ADCL_RETUNE_C13_1 */ - [17945] = 0x0000, /* R17945 - ADCL_RETUNE_C13_0 */ - [17946] = 0x0000, /* R17946 - ADCL_RETUNE_C14_1 */ - [17947] = 0x0000, /* R17947 - ADCL_RETUNE_C14_0 */ - [17948] = 0x0000, /* R17948 - ADCL_RETUNE_C15_1 */ - [17949] = 0x0000, /* R17949 - ADCL_RETUNE_C15_0 */ - [17950] = 0x0000, /* R17950 - ADCL_RETUNE_C16_1 */ - [17951] = 0x0000, /* R17951 - ADCL_RETUNE_C16_0 */ - [17952] = 0x0000, /* R17952 - ADCL_RETUNE_C17_1 */ - [17953] = 0x0000, /* R17953 - ADCL_RETUNE_C17_0 */ - [17954] = 0x0000, /* R17954 - ADCL_RETUNE_C18_1 */ - [17955] = 0x0000, /* R17955 - ADCL_RETUNE_C18_0 */ - [17956] = 0x0000, /* R17956 - ADCL_RETUNE_C19_1 */ - [17957] = 0x0000, /* R17957 - ADCL_RETUNE_C19_0 */ - [17958] = 0x0000, /* R17958 - ADCL_RETUNE_C20_1 */ - [17959] = 0x0000, /* R17959 - ADCL_RETUNE_C20_0 */ - [17960] = 0x0000, /* R17960 - ADCL_RETUNE_C21_1 */ - [17961] = 0x0000, /* R17961 - ADCL_RETUNE_C21_0 */ - [17962] = 0x0000, /* R17962 - ADCL_RETUNE_C22_1 */ - [17963] = 0x0000, /* R17963 - ADCL_RETUNE_C22_0 */ - [17964] = 0x0000, /* R17964 - ADCL_RETUNE_C23_1 */ - [17965] = 0x0000, /* R17965 - ADCL_RETUNE_C23_0 */ - [17966] = 0x0000, /* R17966 - ADCL_RETUNE_C24_1 */ - [17967] = 0x0000, /* R17967 - ADCL_RETUNE_C24_0 */ - [17968] = 0x0000, /* R17968 - ADCL_RETUNE_C25_1 */ - [17969] = 0x0000, /* R17969 - ADCL_RETUNE_C25_0 */ - [17970] = 0x0000, /* R17970 - ADCL_RETUNE_C26_1 */ - [17971] = 0x0000, /* R17971 - ADCL_RETUNE_C26_0 */ - [17972] = 0x0000, /* R17972 - ADCL_RETUNE_C27_1 */ - [17973] = 0x0000, /* R17973 - ADCL_RETUNE_C27_0 */ - [17974] = 0x0000, /* R17974 - ADCL_RETUNE_C28_1 */ - [17975] = 0x0000, /* R17975 - ADCL_RETUNE_C28_0 */ - [17976] = 0x0000, /* R17976 - ADCL_RETUNE_C29_1 */ - [17977] = 0x0000, /* R17977 - ADCL_RETUNE_C29_0 */ - [17978] = 0x0000, /* R17978 - ADCL_RETUNE_C30_1 */ - [17979] = 0x0000, /* R17979 - ADCL_RETUNE_C30_0 */ - [17980] = 0x0000, /* R17980 - ADCL_RETUNE_C31_1 */ - [17981] = 0x0000, /* R17981 - ADCL_RETUNE_C31_0 */ - [17982] = 0x0000, /* R17982 - ADCL_RETUNE_C32_1 */ - [17983] = 0x0000, /* R17983 - ADCL_RETUNE_C32_0 */ - - [18432] = 0x0020, /* R18432 - RETUNEADC_PG2_1 */ - [18433] = 0x0000, /* R18433 - RETUNEADC_PG2_0 */ - [18434] = 0x0040, /* R18434 - RETUNEADC_PG_1 */ - [18435] = 0x0000, /* R18435 - RETUNEADC_PG_0 */ - - [18944] = 0x007F, /* R18944 - ADCR_RETUNE_C1_1 */ - [18945] = 0xFFFF, /* R18945 - ADCR_RETUNE_C1_0 */ - [18946] = 0x0000, /* R18946 - ADCR_RETUNE_C2_1 */ - [18947] = 0x0000, /* R18947 - ADCR_RETUNE_C2_0 */ - [18948] = 0x0000, /* R18948 - ADCR_RETUNE_C3_1 */ - [18949] = 0x0000, /* R18949 - ADCR_RETUNE_C3_0 */ - [18950] = 0x0000, /* R18950 - ADCR_RETUNE_C4_1 */ - [18951] = 0x0000, /* R18951 - ADCR_RETUNE_C4_0 */ - [18952] = 0x0000, /* R18952 - ADCR_RETUNE_C5_1 */ - [18953] = 0x0000, /* R18953 - ADCR_RETUNE_C5_0 */ - [18954] = 0x0000, /* R18954 - ADCR_RETUNE_C6_1 */ - [18955] = 0x0000, /* R18955 - ADCR_RETUNE_C6_0 */ - [18956] = 0x0000, /* R18956 - ADCR_RETUNE_C7_1 */ - [18957] = 0x0000, /* R18957 - ADCR_RETUNE_C7_0 */ - [18958] = 0x0000, /* R18958 - ADCR_RETUNE_C8_1 */ - [18959] = 0x0000, /* R18959 - ADCR_RETUNE_C8_0 */ - [18960] = 0x0000, /* R18960 - ADCR_RETUNE_C9_1 */ - [18961] = 0x0000, /* R18961 - ADCR_RETUNE_C9_0 */ - [18962] = 0x0000, /* R18962 - ADCR_RETUNE_C10_1 */ - [18963] = 0x0000, /* R18963 - ADCR_RETUNE_C10_0 */ - [18964] = 0x0000, /* R18964 - ADCR_RETUNE_C11_1 */ - [18965] = 0x0000, /* R18965 - ADCR_RETUNE_C11_0 */ - [18966] = 0x0000, /* R18966 - ADCR_RETUNE_C12_1 */ - [18967] = 0x0000, /* R18967 - ADCR_RETUNE_C12_0 */ - [18968] = 0x0000, /* R18968 - ADCR_RETUNE_C13_1 */ - [18969] = 0x0000, /* R18969 - ADCR_RETUNE_C13_0 */ - [18970] = 0x0000, /* R18970 - ADCR_RETUNE_C14_1 */ - [18971] = 0x0000, /* R18971 - ADCR_RETUNE_C14_0 */ - [18972] = 0x0000, /* R18972 - ADCR_RETUNE_C15_1 */ - [18973] = 0x0000, /* R18973 - ADCR_RETUNE_C15_0 */ - [18974] = 0x0000, /* R18974 - ADCR_RETUNE_C16_1 */ - [18975] = 0x0000, /* R18975 - ADCR_RETUNE_C16_0 */ - [18976] = 0x0000, /* R18976 - ADCR_RETUNE_C17_1 */ - [18977] = 0x0000, /* R18977 - ADCR_RETUNE_C17_0 */ - [18978] = 0x0000, /* R18978 - ADCR_RETUNE_C18_1 */ - [18979] = 0x0000, /* R18979 - ADCR_RETUNE_C18_0 */ - [18980] = 0x0000, /* R18980 - ADCR_RETUNE_C19_1 */ - [18981] = 0x0000, /* R18981 - ADCR_RETUNE_C19_0 */ - [18982] = 0x0000, /* R18982 - ADCR_RETUNE_C20_1 */ - [18983] = 0x0000, /* R18983 - ADCR_RETUNE_C20_0 */ - [18984] = 0x0000, /* R18984 - ADCR_RETUNE_C21_1 */ - [18985] = 0x0000, /* R18985 - ADCR_RETUNE_C21_0 */ - [18986] = 0x0000, /* R18986 - ADCR_RETUNE_C22_1 */ - [18987] = 0x0000, /* R18987 - ADCR_RETUNE_C22_0 */ - [18988] = 0x0000, /* R18988 - ADCR_RETUNE_C23_1 */ - [18989] = 0x0000, /* R18989 - ADCR_RETUNE_C23_0 */ - [18990] = 0x0000, /* R18990 - ADCR_RETUNE_C24_1 */ - [18991] = 0x0000, /* R18991 - ADCR_RETUNE_C24_0 */ - [18992] = 0x0000, /* R18992 - ADCR_RETUNE_C25_1 */ - [18993] = 0x0000, /* R18993 - ADCR_RETUNE_C25_0 */ - [18994] = 0x0000, /* R18994 - ADCR_RETUNE_C26_1 */ - [18995] = 0x0000, /* R18995 - ADCR_RETUNE_C26_0 */ - [18996] = 0x0000, /* R18996 - ADCR_RETUNE_C27_1 */ - [18997] = 0x0000, /* R18997 - ADCR_RETUNE_C27_0 */ - [18998] = 0x0000, /* R18998 - ADCR_RETUNE_C28_1 */ - [18999] = 0x0000, /* R18999 - ADCR_RETUNE_C28_0 */ - [19000] = 0x0000, /* R19000 - ADCR_RETUNE_C29_1 */ - [19001] = 0x0000, /* R19001 - ADCR_RETUNE_C29_0 */ - [19002] = 0x0000, /* R19002 - ADCR_RETUNE_C30_1 */ - [19003] = 0x0000, /* R19003 - ADCR_RETUNE_C30_0 */ - [19004] = 0x0000, /* R19004 - ADCR_RETUNE_C31_1 */ - [19005] = 0x0000, /* R19005 - ADCR_RETUNE_C31_0 */ - [19006] = 0x0000, /* R19006 - ADCR_RETUNE_C32_1 */ - [19007] = 0x0000, /* R19007 - ADCR_RETUNE_C32_0 */ - - [19456] = 0x007F, /* R19456 - DACL_RETUNE_C1_1 */ - [19457] = 0xFFFF, /* R19457 - DACL_RETUNE_C1_0 */ - [19458] = 0x0000, /* R19458 - DACL_RETUNE_C2_1 */ - [19459] = 0x0000, /* R19459 - DACL_RETUNE_C2_0 */ - [19460] = 0x0000, /* R19460 - DACL_RETUNE_C3_1 */ - [19461] = 0x0000, /* R19461 - DACL_RETUNE_C3_0 */ - [19462] = 0x0000, /* R19462 - DACL_RETUNE_C4_1 */ - [19463] = 0x0000, /* R19463 - DACL_RETUNE_C4_0 */ - [19464] = 0x0000, /* R19464 - DACL_RETUNE_C5_1 */ - [19465] = 0x0000, /* R19465 - DACL_RETUNE_C5_0 */ - [19466] = 0x0000, /* R19466 - DACL_RETUNE_C6_1 */ - [19467] = 0x0000, /* R19467 - DACL_RETUNE_C6_0 */ - [19468] = 0x0000, /* R19468 - DACL_RETUNE_C7_1 */ - [19469] = 0x0000, /* R19469 - DACL_RETUNE_C7_0 */ - [19470] = 0x0000, /* R19470 - DACL_RETUNE_C8_1 */ - [19471] = 0x0000, /* R19471 - DACL_RETUNE_C8_0 */ - [19472] = 0x0000, /* R19472 - DACL_RETUNE_C9_1 */ - [19473] = 0x0000, /* R19473 - DACL_RETUNE_C9_0 */ - [19474] = 0x0000, /* R19474 - DACL_RETUNE_C10_1 */ - [19475] = 0x0000, /* R19475 - DACL_RETUNE_C10_0 */ - [19476] = 0x0000, /* R19476 - DACL_RETUNE_C11_1 */ - [19477] = 0x0000, /* R19477 - DACL_RETUNE_C11_0 */ - [19478] = 0x0000, /* R19478 - DACL_RETUNE_C12_1 */ - [19479] = 0x0000, /* R19479 - DACL_RETUNE_C12_0 */ - [19480] = 0x0000, /* R19480 - DACL_RETUNE_C13_1 */ - [19481] = 0x0000, /* R19481 - DACL_RETUNE_C13_0 */ - [19482] = 0x0000, /* R19482 - DACL_RETUNE_C14_1 */ - [19483] = 0x0000, /* R19483 - DACL_RETUNE_C14_0 */ - [19484] = 0x0000, /* R19484 - DACL_RETUNE_C15_1 */ - [19485] = 0x0000, /* R19485 - DACL_RETUNE_C15_0 */ - [19486] = 0x0000, /* R19486 - DACL_RETUNE_C16_1 */ - [19487] = 0x0000, /* R19487 - DACL_RETUNE_C16_0 */ - [19488] = 0x0000, /* R19488 - DACL_RETUNE_C17_1 */ - [19489] = 0x0000, /* R19489 - DACL_RETUNE_C17_0 */ - [19490] = 0x0000, /* R19490 - DACL_RETUNE_C18_1 */ - [19491] = 0x0000, /* R19491 - DACL_RETUNE_C18_0 */ - [19492] = 0x0000, /* R19492 - DACL_RETUNE_C19_1 */ - [19493] = 0x0000, /* R19493 - DACL_RETUNE_C19_0 */ - [19494] = 0x0000, /* R19494 - DACL_RETUNE_C20_1 */ - [19495] = 0x0000, /* R19495 - DACL_RETUNE_C20_0 */ - [19496] = 0x0000, /* R19496 - DACL_RETUNE_C21_1 */ - [19497] = 0x0000, /* R19497 - DACL_RETUNE_C21_0 */ - [19498] = 0x0000, /* R19498 - DACL_RETUNE_C22_1 */ - [19499] = 0x0000, /* R19499 - DACL_RETUNE_C22_0 */ - [19500] = 0x0000, /* R19500 - DACL_RETUNE_C23_1 */ - [19501] = 0x0000, /* R19501 - DACL_RETUNE_C23_0 */ - [19502] = 0x0000, /* R19502 - DACL_RETUNE_C24_1 */ - [19503] = 0x0000, /* R19503 - DACL_RETUNE_C24_0 */ - [19504] = 0x0000, /* R19504 - DACL_RETUNE_C25_1 */ - [19505] = 0x0000, /* R19505 - DACL_RETUNE_C25_0 */ - [19506] = 0x0000, /* R19506 - DACL_RETUNE_C26_1 */ - [19507] = 0x0000, /* R19507 - DACL_RETUNE_C26_0 */ - [19508] = 0x0000, /* R19508 - DACL_RETUNE_C27_1 */ - [19509] = 0x0000, /* R19509 - DACL_RETUNE_C27_0 */ - [19510] = 0x0000, /* R19510 - DACL_RETUNE_C28_1 */ - [19511] = 0x0000, /* R19511 - DACL_RETUNE_C28_0 */ - [19512] = 0x0000, /* R19512 - DACL_RETUNE_C29_1 */ - [19513] = 0x0000, /* R19513 - DACL_RETUNE_C29_0 */ - [19514] = 0x0000, /* R19514 - DACL_RETUNE_C30_1 */ - [19515] = 0x0000, /* R19515 - DACL_RETUNE_C30_0 */ - [19516] = 0x0000, /* R19516 - DACL_RETUNE_C31_1 */ - [19517] = 0x0000, /* R19517 - DACL_RETUNE_C31_0 */ - [19518] = 0x0000, /* R19518 - DACL_RETUNE_C32_1 */ - [19519] = 0x0000, /* R19519 - DACL_RETUNE_C32_0 */ - - [19968] = 0x0020, /* R19968 - RETUNEDAC_PG2_1 */ - [19969] = 0x0000, /* R19969 - RETUNEDAC_PG2_0 */ - [19970] = 0x0040, /* R19970 - RETUNEDAC_PG_1 */ - [19971] = 0x0000, /* R19971 - RETUNEDAC_PG_0 */ - - [20480] = 0x007F, /* R20480 - DACR_RETUNE_C1_1 */ - [20481] = 0xFFFF, /* R20481 - DACR_RETUNE_C1_0 */ - [20482] = 0x0000, /* R20482 - DACR_RETUNE_C2_1 */ - [20483] = 0x0000, /* R20483 - DACR_RETUNE_C2_0 */ - [20484] = 0x0000, /* R20484 - DACR_RETUNE_C3_1 */ - [20485] = 0x0000, /* R20485 - DACR_RETUNE_C3_0 */ - [20486] = 0x0000, /* R20486 - DACR_RETUNE_C4_1 */ - [20487] = 0x0000, /* R20487 - DACR_RETUNE_C4_0 */ - [20488] = 0x0000, /* R20488 - DACR_RETUNE_C5_1 */ - [20489] = 0x0000, /* R20489 - DACR_RETUNE_C5_0 */ - [20490] = 0x0000, /* R20490 - DACR_RETUNE_C6_1 */ - [20491] = 0x0000, /* R20491 - DACR_RETUNE_C6_0 */ - [20492] = 0x0000, /* R20492 - DACR_RETUNE_C7_1 */ - [20493] = 0x0000, /* R20493 - DACR_RETUNE_C7_0 */ - [20494] = 0x0000, /* R20494 - DACR_RETUNE_C8_1 */ - [20495] = 0x0000, /* R20495 - DACR_RETUNE_C8_0 */ - [20496] = 0x0000, /* R20496 - DACR_RETUNE_C9_1 */ - [20497] = 0x0000, /* R20497 - DACR_RETUNE_C9_0 */ - [20498] = 0x0000, /* R20498 - DACR_RETUNE_C10_1 */ - [20499] = 0x0000, /* R20499 - DACR_RETUNE_C10_0 */ - [20500] = 0x0000, /* R20500 - DACR_RETUNE_C11_1 */ - [20501] = 0x0000, /* R20501 - DACR_RETUNE_C11_0 */ - [20502] = 0x0000, /* R20502 - DACR_RETUNE_C12_1 */ - [20503] = 0x0000, /* R20503 - DACR_RETUNE_C12_0 */ - [20504] = 0x0000, /* R20504 - DACR_RETUNE_C13_1 */ - [20505] = 0x0000, /* R20505 - DACR_RETUNE_C13_0 */ - [20506] = 0x0000, /* R20506 - DACR_RETUNE_C14_1 */ - [20507] = 0x0000, /* R20507 - DACR_RETUNE_C14_0 */ - [20508] = 0x0000, /* R20508 - DACR_RETUNE_C15_1 */ - [20509] = 0x0000, /* R20509 - DACR_RETUNE_C15_0 */ - [20510] = 0x0000, /* R20510 - DACR_RETUNE_C16_1 */ - [20511] = 0x0000, /* R20511 - DACR_RETUNE_C16_0 */ - [20512] = 0x0000, /* R20512 - DACR_RETUNE_C17_1 */ - [20513] = 0x0000, /* R20513 - DACR_RETUNE_C17_0 */ - [20514] = 0x0000, /* R20514 - DACR_RETUNE_C18_1 */ - [20515] = 0x0000, /* R20515 - DACR_RETUNE_C18_0 */ - [20516] = 0x0000, /* R20516 - DACR_RETUNE_C19_1 */ - [20517] = 0x0000, /* R20517 - DACR_RETUNE_C19_0 */ - [20518] = 0x0000, /* R20518 - DACR_RETUNE_C20_1 */ - [20519] = 0x0000, /* R20519 - DACR_RETUNE_C20_0 */ - [20520] = 0x0000, /* R20520 - DACR_RETUNE_C21_1 */ - [20521] = 0x0000, /* R20521 - DACR_RETUNE_C21_0 */ - [20522] = 0x0000, /* R20522 - DACR_RETUNE_C22_1 */ - [20523] = 0x0000, /* R20523 - DACR_RETUNE_C22_0 */ - [20524] = 0x0000, /* R20524 - DACR_RETUNE_C23_1 */ - [20525] = 0x0000, /* R20525 - DACR_RETUNE_C23_0 */ - [20526] = 0x0000, /* R20526 - DACR_RETUNE_C24_1 */ - [20527] = 0x0000, /* R20527 - DACR_RETUNE_C24_0 */ - [20528] = 0x0000, /* R20528 - DACR_RETUNE_C25_1 */ - [20529] = 0x0000, /* R20529 - DACR_RETUNE_C25_0 */ - [20530] = 0x0000, /* R20530 - DACR_RETUNE_C26_1 */ - [20531] = 0x0000, /* R20531 - DACR_RETUNE_C26_0 */ - [20532] = 0x0000, /* R20532 - DACR_RETUNE_C27_1 */ - [20533] = 0x0000, /* R20533 - DACR_RETUNE_C27_0 */ - [20534] = 0x0000, /* R20534 - DACR_RETUNE_C28_1 */ - [20535] = 0x0000, /* R20535 - DACR_RETUNE_C28_0 */ - [20536] = 0x0000, /* R20536 - DACR_RETUNE_C29_1 */ - [20537] = 0x0000, /* R20537 - DACR_RETUNE_C29_0 */ - [20538] = 0x0000, /* R20538 - DACR_RETUNE_C30_1 */ - [20539] = 0x0000, /* R20539 - DACR_RETUNE_C30_0 */ - [20540] = 0x0000, /* R20540 - DACR_RETUNE_C31_1 */ - [20541] = 0x0000, /* R20541 - DACR_RETUNE_C31_0 */ - [20542] = 0x0000, /* R20542 - DACR_RETUNE_C32_1 */ - [20543] = 0x0000, /* R20543 - DACR_RETUNE_C32_0 */ - - [20992] = 0x008C, /* R20992 - VSS_XHD2_1 */ - [20993] = 0x0200, /* R20993 - VSS_XHD2_0 */ - [20994] = 0x0035, /* R20994 - VSS_XHD3_1 */ - [20995] = 0x0700, /* R20995 - VSS_XHD3_0 */ - [20996] = 0x003A, /* R20996 - VSS_XHN1_1 */ - [20997] = 0x4100, /* R20997 - VSS_XHN1_0 */ - [20998] = 0x008B, /* R20998 - VSS_XHN2_1 */ - [20999] = 0x7D00, /* R20999 - VSS_XHN2_0 */ - [21000] = 0x003A, /* R21000 - VSS_XHN3_1 */ - [21001] = 0x4100, /* R21001 - VSS_XHN3_0 */ - [21002] = 0x008C, /* R21002 - VSS_XLA_1 */ - [21003] = 0xFEE8, /* R21003 - VSS_XLA_0 */ - [21004] = 0x0078, /* R21004 - VSS_XLB_1 */ - [21005] = 0x0000, /* R21005 - VSS_XLB_0 */ - [21006] = 0x003F, /* R21006 - VSS_XLG_1 */ - [21007] = 0xB260, /* R21007 - VSS_XLG_0 */ - [21008] = 0x002D, /* R21008 - VSS_PG2_1 */ - [21009] = 0x1818, /* R21009 - VSS_PG2_0 */ - [21010] = 0x0020, /* R21010 - VSS_PG_1 */ - [21011] = 0x0000, /* R21011 - VSS_PG_0 */ - [21012] = 0x00F1, /* R21012 - VSS_XTD1_1 */ - [21013] = 0x8340, /* R21013 - VSS_XTD1_0 */ - [21014] = 0x00FB, /* R21014 - VSS_XTD2_1 */ - [21015] = 0x8300, /* R21015 - VSS_XTD2_0 */ - [21016] = 0x00EE, /* R21016 - VSS_XTD3_1 */ - [21017] = 0xAEC0, /* R21017 - VSS_XTD3_0 */ - [21018] = 0x00FB, /* R21018 - VSS_XTD4_1 */ - [21019] = 0xAC40, /* R21019 - VSS_XTD4_0 */ - [21020] = 0x00F1, /* R21020 - VSS_XTD5_1 */ - [21021] = 0x7F80, /* R21021 - VSS_XTD5_0 */ - [21022] = 0x00F4, /* R21022 - VSS_XTD6_1 */ - [21023] = 0x3B40, /* R21023 - VSS_XTD6_0 */ - [21024] = 0x00F5, /* R21024 - VSS_XTD7_1 */ - [21025] = 0xFB00, /* R21025 - VSS_XTD7_0 */ - [21026] = 0x00EA, /* R21026 - VSS_XTD8_1 */ - [21027] = 0x10C0, /* R21027 - VSS_XTD8_0 */ - [21028] = 0x00FC, /* R21028 - VSS_XTD9_1 */ - [21029] = 0xC580, /* R21029 - VSS_XTD9_0 */ - [21030] = 0x00E2, /* R21030 - VSS_XTD10_1 */ - [21031] = 0x75C0, /* R21031 - VSS_XTD10_0 */ - [21032] = 0x0004, /* R21032 - VSS_XTD11_1 */ - [21033] = 0xB480, /* R21033 - VSS_XTD11_0 */ - [21034] = 0x00D4, /* R21034 - VSS_XTD12_1 */ - [21035] = 0xF980, /* R21035 - VSS_XTD12_0 */ - [21036] = 0x0004, /* R21036 - VSS_XTD13_1 */ - [21037] = 0x9140, /* R21037 - VSS_XTD13_0 */ - [21038] = 0x00D8, /* R21038 - VSS_XTD14_1 */ - [21039] = 0xA480, /* R21039 - VSS_XTD14_0 */ - [21040] = 0x0002, /* R21040 - VSS_XTD15_1 */ - [21041] = 0x3DC0, /* R21041 - VSS_XTD15_0 */ - [21042] = 0x00CF, /* R21042 - VSS_XTD16_1 */ - [21043] = 0x7A80, /* R21043 - VSS_XTD16_0 */ - [21044] = 0x00DC, /* R21044 - VSS_XTD17_1 */ - [21045] = 0x0600, /* R21045 - VSS_XTD17_0 */ - [21046] = 0x00F2, /* R21046 - VSS_XTD18_1 */ - [21047] = 0xDAC0, /* R21047 - VSS_XTD18_0 */ - [21048] = 0x00BA, /* R21048 - VSS_XTD19_1 */ - [21049] = 0xF340, /* R21049 - VSS_XTD19_0 */ - [21050] = 0x000A, /* R21050 - VSS_XTD20_1 */ - [21051] = 0x7940, /* R21051 - VSS_XTD20_0 */ - [21052] = 0x001C, /* R21052 - VSS_XTD21_1 */ - [21053] = 0x0680, /* R21053 - VSS_XTD21_0 */ - [21054] = 0x00FD, /* R21054 - VSS_XTD22_1 */ - [21055] = 0x2D00, /* R21055 - VSS_XTD22_0 */ - [21056] = 0x001C, /* R21056 - VSS_XTD23_1 */ - [21057] = 0xE840, /* R21057 - VSS_XTD23_0 */ - [21058] = 0x000D, /* R21058 - VSS_XTD24_1 */ - [21059] = 0xDC40, /* R21059 - VSS_XTD24_0 */ - [21060] = 0x00FC, /* R21060 - VSS_XTD25_1 */ - [21061] = 0x9D00, /* R21061 - VSS_XTD25_0 */ - [21062] = 0x0009, /* R21062 - VSS_XTD26_1 */ - [21063] = 0x5580, /* R21063 - VSS_XTD26_0 */ - [21064] = 0x00FE, /* R21064 - VSS_XTD27_1 */ - [21065] = 0x7E80, /* R21065 - VSS_XTD27_0 */ - [21066] = 0x000E, /* R21066 - VSS_XTD28_1 */ - [21067] = 0xAB40, /* R21067 - VSS_XTD28_0 */ - [21068] = 0x00F9, /* R21068 - VSS_XTD29_1 */ - [21069] = 0x9880, /* R21069 - VSS_XTD29_0 */ - [21070] = 0x0009, /* R21070 - VSS_XTD30_1 */ - [21071] = 0x87C0, /* R21071 - VSS_XTD30_0 */ - [21072] = 0x00FD, /* R21072 - VSS_XTD31_1 */ - [21073] = 0x2C40, /* R21073 - VSS_XTD31_0 */ - [21074] = 0x0009, /* R21074 - VSS_XTD32_1 */ - [21075] = 0x4800, /* R21075 - VSS_XTD32_0 */ - [21076] = 0x0003, /* R21076 - VSS_XTS1_1 */ - [21077] = 0x5F40, /* R21077 - VSS_XTS1_0 */ - [21078] = 0x0000, /* R21078 - VSS_XTS2_1 */ - [21079] = 0x8700, /* R21079 - VSS_XTS2_0 */ - [21080] = 0x00FA, /* R21080 - VSS_XTS3_1 */ - [21081] = 0xE4C0, /* R21081 - VSS_XTS3_0 */ - [21082] = 0x0000, /* R21082 - VSS_XTS4_1 */ - [21083] = 0x0B40, /* R21083 - VSS_XTS4_0 */ - [21084] = 0x0004, /* R21084 - VSS_XTS5_1 */ - [21085] = 0xE180, /* R21085 - VSS_XTS5_0 */ - [21086] = 0x0001, /* R21086 - VSS_XTS6_1 */ - [21087] = 0x1F40, /* R21087 - VSS_XTS6_0 */ - [21088] = 0x00F8, /* R21088 - VSS_XTS7_1 */ - [21089] = 0xB000, /* R21089 - VSS_XTS7_0 */ - [21090] = 0x00FB, /* R21090 - VSS_XTS8_1 */ - [21091] = 0xCBC0, /* R21091 - VSS_XTS8_0 */ - [21092] = 0x0004, /* R21092 - VSS_XTS9_1 */ - [21093] = 0xF380, /* R21093 - VSS_XTS9_0 */ - [21094] = 0x0007, /* R21094 - VSS_XTS10_1 */ - [21095] = 0xDF40, /* R21095 - VSS_XTS10_0 */ - [21096] = 0x00FF, /* R21096 - VSS_XTS11_1 */ - [21097] = 0x0700, /* R21097 - VSS_XTS11_0 */ - [21098] = 0x00EF, /* R21098 - VSS_XTS12_1 */ - [21099] = 0xD700, /* R21099 - VSS_XTS12_0 */ - [21100] = 0x00FB, /* R21100 - VSS_XTS13_1 */ - [21101] = 0xAF40, /* R21101 - VSS_XTS13_0 */ - [21102] = 0x0010, /* R21102 - VSS_XTS14_1 */ - [21103] = 0x8A80, /* R21103 - VSS_XTS14_0 */ - [21104] = 0x0011, /* R21104 - VSS_XTS15_1 */ - [21105] = 0x07C0, /* R21105 - VSS_XTS15_0 */ - [21106] = 0x00E0, /* R21106 - VSS_XTS16_1 */ - [21107] = 0x0800, /* R21107 - VSS_XTS16_0 */ - [21108] = 0x00D2, /* R21108 - VSS_XTS17_1 */ - [21109] = 0x7600, /* R21109 - VSS_XTS17_0 */ - [21110] = 0x0020, /* R21110 - VSS_XTS18_1 */ - [21111] = 0xCF40, /* R21111 - VSS_XTS18_0 */ - [21112] = 0x0030, /* R21112 - VSS_XTS19_1 */ - [21113] = 0x2340, /* R21113 - VSS_XTS19_0 */ - [21114] = 0x00FD, /* R21114 - VSS_XTS20_1 */ - [21115] = 0x69C0, /* R21115 - VSS_XTS20_0 */ - [21116] = 0x0028, /* R21116 - VSS_XTS21_1 */ - [21117] = 0x3500, /* R21117 - VSS_XTS21_0 */ - [21118] = 0x0006, /* R21118 - VSS_XTS22_1 */ - [21119] = 0x3300, /* R21119 - VSS_XTS22_0 */ - [21120] = 0x00D9, /* R21120 - VSS_XTS23_1 */ - [21121] = 0xF6C0, /* R21121 - VSS_XTS23_0 */ - [21122] = 0x00F3, /* R21122 - VSS_XTS24_1 */ - [21123] = 0x3340, /* R21123 - VSS_XTS24_0 */ - [21124] = 0x000F, /* R21124 - VSS_XTS25_1 */ - [21125] = 0x4200, /* R21125 - VSS_XTS25_0 */ - [21126] = 0x0004, /* R21126 - VSS_XTS26_1 */ - [21127] = 0x0C80, /* R21127 - VSS_XTS26_0 */ - [21128] = 0x00FB, /* R21128 - VSS_XTS27_1 */ - [21129] = 0x3F80, /* R21129 - VSS_XTS27_0 */ - [21130] = 0x00F7, /* R21130 - VSS_XTS28_1 */ - [21131] = 0x57C0, /* R21131 - VSS_XTS28_0 */ - [21132] = 0x0003, /* R21132 - VSS_XTS29_1 */ - [21133] = 0x5400, /* R21133 - VSS_XTS29_0 */ - [21134] = 0x0000, /* R21134 - VSS_XTS30_1 */ - [21135] = 0xC6C0, /* R21135 - VSS_XTS30_0 */ - [21136] = 0x0003, /* R21136 - VSS_XTS31_1 */ - [21137] = 0x12C0, /* R21137 - VSS_XTS31_0 */ - [21138] = 0x00FD, /* R21138 - VSS_XTS32_1 */ - [21139] = 0x8580, /* R21139 - VSS_XTS32_0 */ +static struct reg_default wm8962_reg[] = { + { 0, 0x009F }, /* R0 - Left Input volume */ + { 1, 0x049F }, /* R1 - Right Input volume */ + { 2, 0x0000 }, /* R2 - HPOUTL volume */ + { 3, 0x0000 }, /* R3 - HPOUTR volume */ + { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0018 }, /* R5 - ADC & DAC Control 1 */ + { 6, 0x2008 }, /* R6 - ADC & DAC Control 2 */ + { 7, 0x000A }, /* R7 - Audio Interface 0 */ + { 8, 0x01E4 }, /* R8 - Clocking2 */ + { 9, 0x0300 }, /* R9 - Audio Interface 1 */ + { 10, 0x00C0 }, /* R10 - Left DAC volume */ + { 11, 0x00C0 }, /* R11 - Right DAC volume */ + + { 14, 0x0040 }, /* R14 - Audio Interface 2 */ + { 15, 0x6243 }, /* R15 - Software Reset */ + + { 17, 0x007B }, /* R17 - ALC1 */ + { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x1C32 }, /* R19 - ALC3 */ + { 20, 0x3200 }, /* R20 - Noise Gate */ + { 21, 0x00C0 }, /* R21 - Left ADC volume */ + { 22, 0x00C0 }, /* R22 - Right ADC volume */ + { 23, 0x0160 }, /* R23 - Additional control(1) */ + { 24, 0x0000 }, /* R24 - Additional control(2) */ + { 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */ + { 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */ + { 27, 0x0010 }, /* R27 - Additional Control (3) */ + { 28, 0x0000 }, /* R28 - Anti-pop */ + + { 30, 0x005E }, /* R30 - Clocking 3 */ + { 31, 0x0000 }, /* R31 - Input mixer control (1) */ + { 32, 0x0145 }, /* R32 - Left input mixer volume */ + { 33, 0x0145 }, /* R33 - Right input mixer volume */ + { 34, 0x0009 }, /* R34 - Input mixer control (2) */ + { 35, 0x0003 }, /* R35 - Input bias control */ + { 37, 0x0008 }, /* R37 - Left input PGA control */ + { 38, 0x0008 }, /* R38 - Right input PGA control */ + + { 40, 0x0000 }, /* R40 - SPKOUTL volume */ + { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + + { 47, 0x0000 }, /* R47 - Thermal Shutdown Status */ + { 48, 0x8027 }, /* R48 - Additional Control (4) */ + { 49, 0x0010 }, /* R49 - Class D Control 1 */ + + { 51, 0x0003 }, /* R51 - Class D Control 2 */ + + { 56, 0x0506 }, /* R56 - Clocking 4 */ + { 57, 0x0000 }, /* R57 - DAC DSP Mixing (1) */ + { 58, 0x0000 }, /* R58 - DAC DSP Mixing (2) */ + + { 60, 0x0300 }, /* R60 - DC Servo 0 */ + { 61, 0x0300 }, /* R61 - DC Servo 1 */ + + { 64, 0x0810 }, /* R64 - DC Servo 4 */ + + { 66, 0x0000 }, /* R66 - DC Servo 6 */ + + { 68, 0x001B }, /* R68 - Analogue PGA Bias */ + { 69, 0x0000 }, /* R69 - Analogue HP 0 */ + + { 71, 0x01FB }, /* R71 - Analogue HP 2 */ + { 72, 0x0000 }, /* R72 - Charge Pump 1 */ + + { 82, 0x0004 }, /* R82 - Charge Pump B */ + + { 87, 0x0000 }, /* R87 - Write Sequencer Control 1 */ + + { 90, 0x0000 }, /* R90 - Write Sequencer Control 2 */ + + { 93, 0x0000 }, /* R93 - Write Sequencer Control 3 */ + { 94, 0x0000 }, /* R94 - Control Interface */ + + { 99, 0x0000 }, /* R99 - Mixer Enables */ + { 100, 0x0000 }, /* R100 - Headphone Mixer (1) */ + { 101, 0x0000 }, /* R101 - Headphone Mixer (2) */ + { 102, 0x013F }, /* R102 - Headphone Mixer (3) */ + { 103, 0x013F }, /* R103 - Headphone Mixer (4) */ + + { 105, 0x0000 }, /* R105 - Speaker Mixer (1) */ + { 106, 0x0000 }, /* R106 - Speaker Mixer (2) */ + { 107, 0x013F }, /* R107 - Speaker Mixer (3) */ + { 108, 0x013F }, /* R108 - Speaker Mixer (4) */ + { 109, 0x0003 }, /* R109 - Speaker Mixer (5) */ + { 110, 0x0002 }, /* R110 - Beep Generator (1) */ + + { 115, 0x0006 }, /* R115 - Oscillator Trim (3) */ + { 116, 0x0026 }, /* R116 - Oscillator Trim (4) */ + + { 119, 0x0000 }, /* R119 - Oscillator Trim (7) */ + + { 124, 0x0011 }, /* R124 - Analogue Clocking1 */ + { 125, 0x004B }, /* R125 - Analogue Clocking2 */ + { 126, 0x000D }, /* R126 - Analogue Clocking3 */ + { 127, 0x0000 }, /* R127 - PLL Software Reset */ + + { 129, 0x0000 }, /* R129 - PLL2 */ + + { 131, 0x0000 }, /* R131 - PLL 4 */ + + { 136, 0x0067 }, /* R136 - PLL 9 */ + { 137, 0x001C }, /* R137 - PLL 10 */ + { 138, 0x0071 }, /* R138 - PLL 11 */ + { 139, 0x00C7 }, /* R139 - PLL 12 */ + { 140, 0x0067 }, /* R140 - PLL 13 */ + { 141, 0x0048 }, /* R141 - PLL 14 */ + { 142, 0x0022 }, /* R142 - PLL 15 */ + { 143, 0x0097 }, /* R143 - PLL 16 */ + + { 155, 0x000C }, /* R155 - FLL Control (1) */ + { 156, 0x0039 }, /* R156 - FLL Control (2) */ + { 157, 0x0180 }, /* R157 - FLL Control (3) */ + + { 159, 0x0032 }, /* R159 - FLL Control (5) */ + { 160, 0x0018 }, /* R160 - FLL Control (6) */ + { 161, 0x007D }, /* R161 - FLL Control (7) */ + { 162, 0x0008 }, /* R162 - FLL Control (8) */ + + { 252, 0x0005 }, /* R252 - General test 1 */ + + { 256, 0x0000 }, /* R256 - DF1 */ + { 257, 0x0000 }, /* R257 - DF2 */ + { 258, 0x0000 }, /* R258 - DF3 */ + { 259, 0x0000 }, /* R259 - DF4 */ + { 260, 0x0000 }, /* R260 - DF5 */ + { 261, 0x0000 }, /* R261 - DF6 */ + { 262, 0x0000 }, /* R262 - DF7 */ + + { 264, 0x0000 }, /* R264 - LHPF1 */ + { 265, 0x0000 }, /* R265 - LHPF2 */ + + { 268, 0x0000 }, /* R268 - THREED1 */ + { 269, 0x0000 }, /* R269 - THREED2 */ + { 270, 0x0000 }, /* R270 - THREED3 */ + { 271, 0x0000 }, /* R271 - THREED4 */ + + { 276, 0x000C }, /* R276 - DRC 1 */ + { 277, 0x0925 }, /* R277 - DRC 2 */ + { 278, 0x0000 }, /* R278 - DRC 3 */ + { 279, 0x0000 }, /* R279 - DRC 4 */ + { 280, 0x0000 }, /* R280 - DRC 5 */ + + { 285, 0x0000 }, /* R285 - Tloopback */ + + { 335, 0x0004 }, /* R335 - EQ1 */ + { 336, 0x6318 }, /* R336 - EQ2 */ + { 337, 0x6300 }, /* R337 - EQ3 */ + { 338, 0x0FCA }, /* R338 - EQ4 */ + { 339, 0x0400 }, /* R339 - EQ5 */ + { 340, 0x00D8 }, /* R340 - EQ6 */ + { 341, 0x1EB5 }, /* R341 - EQ7 */ + { 342, 0xF145 }, /* R342 - EQ8 */ + { 343, 0x0B75 }, /* R343 - EQ9 */ + { 344, 0x01C5 }, /* R344 - EQ10 */ + { 345, 0x1C58 }, /* R345 - EQ11 */ + { 346, 0xF373 }, /* R346 - EQ12 */ + { 347, 0x0A54 }, /* R347 - EQ13 */ + { 348, 0x0558 }, /* R348 - EQ14 */ + { 349, 0x168E }, /* R349 - EQ15 */ + { 350, 0xF829 }, /* R350 - EQ16 */ + { 351, 0x07AD }, /* R351 - EQ17 */ + { 352, 0x1103 }, /* R352 - EQ18 */ + { 353, 0x0564 }, /* R353 - EQ19 */ + { 354, 0x0559 }, /* R354 - EQ20 */ + { 355, 0x4000 }, /* R355 - EQ21 */ + { 356, 0x6318 }, /* R356 - EQ22 */ + { 357, 0x6300 }, /* R357 - EQ23 */ + { 358, 0x0FCA }, /* R358 - EQ24 */ + { 359, 0x0400 }, /* R359 - EQ25 */ + { 360, 0x00D8 }, /* R360 - EQ26 */ + { 361, 0x1EB5 }, /* R361 - EQ27 */ + { 362, 0xF145 }, /* R362 - EQ28 */ + { 363, 0x0B75 }, /* R363 - EQ29 */ + { 364, 0x01C5 }, /* R364 - EQ30 */ + { 365, 0x1C58 }, /* R365 - EQ31 */ + { 366, 0xF373 }, /* R366 - EQ32 */ + { 367, 0x0A54 }, /* R367 - EQ33 */ + { 368, 0x0558 }, /* R368 - EQ34 */ + { 369, 0x168E }, /* R369 - EQ35 */ + { 370, 0xF829 }, /* R370 - EQ36 */ + { 371, 0x07AD }, /* R371 - EQ37 */ + { 372, 0x1103 }, /* R372 - EQ38 */ + { 373, 0x0564 }, /* R373 - EQ39 */ + { 374, 0x0559 }, /* R374 - EQ40 */ + { 375, 0x4000 }, /* R375 - EQ41 */ + + { 513, 0x0000 }, /* R513 - GPIO 2 */ + { 514, 0x0000 }, /* R514 - GPIO 3 */ + + { 516, 0x8100 }, /* R516 - GPIO 5 */ + { 517, 0x8100 }, /* R517 - GPIO 6 */ + + { 560, 0x0000 }, /* R560 - Interrupt Status 1 */ + { 561, 0x0000 }, /* R561 - Interrupt Status 2 */ + + { 568, 0x0030 }, /* R568 - Interrupt Status 1 Mask */ + { 569, 0xFFED }, /* R569 - Interrupt Status 2 Mask */ + + { 576, 0x0000 }, /* R576 - Interrupt Control */ + + { 584, 0x002D }, /* R584 - IRQ Debounce */ + + { 586, 0x0000 }, /* R586 - MICINT Source Pol */ + + { 768, 0x1C00 }, /* R768 - DSP2 Power Management */ + + { 1037, 0x0000 }, /* R1037 - DSP2_ExecControl */ + + { 8192, 0x0000 }, /* R8192 - DSP2 Instruction RAM 0 */ + + { 9216, 0x0030 }, /* R9216 - DSP2 Address RAM 2 */ + { 9217, 0x0000 }, /* R9217 - DSP2 Address RAM 1 */ + { 9218, 0x0000 }, /* R9218 - DSP2 Address RAM 0 */ + + { 12288, 0x0000 }, /* R12288 - DSP2 Data1 RAM 1 */ + { 12289, 0x0000 }, /* R12289 - DSP2 Data1 RAM 0 */ + + { 13312, 0x0000 }, /* R13312 - DSP2 Data2 RAM 1 */ + { 13313, 0x0000 }, /* R13313 - DSP2 Data2 RAM 0 */ + + { 14336, 0x0000 }, /* R14336 - DSP2 Data3 RAM 1 */ + { 14337, 0x0000 }, /* R14337 - DSP2 Data3 RAM 0 */ + + { 15360, 0x000A }, /* R15360 - DSP2 Coeff RAM 0 */ + + { 16384, 0x0000 }, /* R16384 - RETUNEADC_SHARED_COEFF_1 */ + { 16385, 0x0000 }, /* R16385 - RETUNEADC_SHARED_COEFF_0 */ + { 16386, 0x0000 }, /* R16386 - RETUNEDAC_SHARED_COEFF_1 */ + { 16387, 0x0000 }, /* R16387 - RETUNEDAC_SHARED_COEFF_0 */ + { 16388, 0x0000 }, /* R16388 - SOUNDSTAGE_ENABLES_1 */ + { 16389, 0x0000 }, /* R16389 - SOUNDSTAGE_ENABLES_0 */ + + { 16896, 0x0002 }, /* R16896 - HDBASS_AI_1 */ + { 16897, 0xBD12 }, /* R16897 - HDBASS_AI_0 */ + { 16898, 0x007C }, /* R16898 - HDBASS_AR_1 */ + { 16899, 0x586C }, /* R16899 - HDBASS_AR_0 */ + { 16900, 0x0053 }, /* R16900 - HDBASS_B_1 */ + { 16901, 0x8121 }, /* R16901 - HDBASS_B_0 */ + { 16902, 0x003F }, /* R16902 - HDBASS_K_1 */ + { 16903, 0x8BD8 }, /* R16903 - HDBASS_K_0 */ + { 16904, 0x0032 }, /* R16904 - HDBASS_N1_1 */ + { 16905, 0xF52D }, /* R16905 - HDBASS_N1_0 */ + { 16906, 0x0065 }, /* R16906 - HDBASS_N2_1 */ + { 16907, 0xAC8C }, /* R16907 - HDBASS_N2_0 */ + { 16908, 0x006B }, /* R16908 - HDBASS_N3_1 */ + { 16909, 0xE087 }, /* R16909 - HDBASS_N3_0 */ + { 16910, 0x0072 }, /* R16910 - HDBASS_N4_1 */ + { 16911, 0x1483 }, /* R16911 - HDBASS_N4_0 */ + { 16912, 0x0072 }, /* R16912 - HDBASS_N5_1 */ + { 16913, 0x1483 }, /* R16913 - HDBASS_N5_0 */ + { 16914, 0x0043 }, /* R16914 - HDBASS_X1_1 */ + { 16915, 0x3525 }, /* R16915 - HDBASS_X1_0 */ + { 16916, 0x0006 }, /* R16916 - HDBASS_X2_1 */ + { 16917, 0x6A4A }, /* R16917 - HDBASS_X2_0 */ + { 16918, 0x0043 }, /* R16918 - HDBASS_X3_1 */ + { 16919, 0x6079 }, /* R16919 - HDBASS_X3_0 */ + { 16920, 0x0008 }, /* R16920 - HDBASS_ATK_1 */ + { 16921, 0x0000 }, /* R16921 - HDBASS_ATK_0 */ + { 16922, 0x0001 }, /* R16922 - HDBASS_DCY_1 */ + { 16923, 0x0000 }, /* R16923 - HDBASS_DCY_0 */ + { 16924, 0x0059 }, /* R16924 - HDBASS_PG_1 */ + { 16925, 0x999A }, /* R16925 - HDBASS_PG_0 */ + + { 17048, 0x0083 }, /* R17408 - HPF_C_1 */ + { 17049, 0x98AD }, /* R17409 - HPF_C_0 */ + + { 17920, 0x007F }, /* R17920 - ADCL_RETUNE_C1_1 */ + { 17921, 0xFFFF }, /* R17921 - ADCL_RETUNE_C1_0 */ + { 17922, 0x0000 }, /* R17922 - ADCL_RETUNE_C2_1 */ + { 17923, 0x0000 }, /* R17923 - ADCL_RETUNE_C2_0 */ + { 17924, 0x0000 }, /* R17924 - ADCL_RETUNE_C3_1 */ + { 17925, 0x0000 }, /* R17925 - ADCL_RETUNE_C3_0 */ + { 17926, 0x0000 }, /* R17926 - ADCL_RETUNE_C4_1 */ + { 17927, 0x0000 }, /* R17927 - ADCL_RETUNE_C4_0 */ + { 17928, 0x0000 }, /* R17928 - ADCL_RETUNE_C5_1 */ + { 17929, 0x0000 }, /* R17929 - ADCL_RETUNE_C5_0 */ + { 17930, 0x0000 }, /* R17930 - ADCL_RETUNE_C6_1 */ + { 17931, 0x0000 }, /* R17931 - ADCL_RETUNE_C6_0 */ + { 17932, 0x0000 }, /* R17932 - ADCL_RETUNE_C7_1 */ + { 17933, 0x0000 }, /* R17933 - ADCL_RETUNE_C7_0 */ + { 17934, 0x0000 }, /* R17934 - ADCL_RETUNE_C8_1 */ + { 17935, 0x0000 }, /* R17935 - ADCL_RETUNE_C8_0 */ + { 17936, 0x0000 }, /* R17936 - ADCL_RETUNE_C9_1 */ + { 17937, 0x0000 }, /* R17937 - ADCL_RETUNE_C9_0 */ + { 17938, 0x0000 }, /* R17938 - ADCL_RETUNE_C10_1 */ + { 17939, 0x0000 }, /* R17939 - ADCL_RETUNE_C10_0 */ + { 17940, 0x0000 }, /* R17940 - ADCL_RETUNE_C11_1 */ + { 17941, 0x0000 }, /* R17941 - ADCL_RETUNE_C11_0 */ + { 17942, 0x0000 }, /* R17942 - ADCL_RETUNE_C12_1 */ + { 17943, 0x0000 }, /* R17943 - ADCL_RETUNE_C12_0 */ + { 17944, 0x0000 }, /* R17944 - ADCL_RETUNE_C13_1 */ + { 17945, 0x0000 }, /* R17945 - ADCL_RETUNE_C13_0 */ + { 17946, 0x0000 }, /* R17946 - ADCL_RETUNE_C14_1 */ + { 17947, 0x0000 }, /* R17947 - ADCL_RETUNE_C14_0 */ + { 17948, 0x0000 }, /* R17948 - ADCL_RETUNE_C15_1 */ + { 17949, 0x0000 }, /* R17949 - ADCL_RETUNE_C15_0 */ + { 17950, 0x0000 }, /* R17950 - ADCL_RETUNE_C16_1 */ + { 17951, 0x0000 }, /* R17951 - ADCL_RETUNE_C16_0 */ + { 17952, 0x0000 }, /* R17952 - ADCL_RETUNE_C17_1 */ + { 17953, 0x0000 }, /* R17953 - ADCL_RETUNE_C17_0 */ + { 17954, 0x0000 }, /* R17954 - ADCL_RETUNE_C18_1 */ + { 17955, 0x0000 }, /* R17955 - ADCL_RETUNE_C18_0 */ + { 17956, 0x0000 }, /* R17956 - ADCL_RETUNE_C19_1 */ + { 17957, 0x0000 }, /* R17957 - ADCL_RETUNE_C19_0 */ + { 17958, 0x0000 }, /* R17958 - ADCL_RETUNE_C20_1 */ + { 17959, 0x0000 }, /* R17959 - ADCL_RETUNE_C20_0 */ + { 17960, 0x0000 }, /* R17960 - ADCL_RETUNE_C21_1 */ + { 17961, 0x0000 }, /* R17961 - ADCL_RETUNE_C21_0 */ + { 17962, 0x0000 }, /* R17962 - ADCL_RETUNE_C22_1 */ + { 17963, 0x0000 }, /* R17963 - ADCL_RETUNE_C22_0 */ + { 17964, 0x0000 }, /* R17964 - ADCL_RETUNE_C23_1 */ + { 17965, 0x0000 }, /* R17965 - ADCL_RETUNE_C23_0 */ + { 17966, 0x0000 }, /* R17966 - ADCL_RETUNE_C24_1 */ + { 17967, 0x0000 }, /* R17967 - ADCL_RETUNE_C24_0 */ + { 17968, 0x0000 }, /* R17968 - ADCL_RETUNE_C25_1 */ + { 17969, 0x0000 }, /* R17969 - ADCL_RETUNE_C25_0 */ + { 17970, 0x0000 }, /* R17970 - ADCL_RETUNE_C26_1 */ + { 17971, 0x0000 }, /* R17971 - ADCL_RETUNE_C26_0 */ + { 17972, 0x0000 }, /* R17972 - ADCL_RETUNE_C27_1 */ + { 17973, 0x0000 }, /* R17973 - ADCL_RETUNE_C27_0 */ + { 17974, 0x0000 }, /* R17974 - ADCL_RETUNE_C28_1 */ + { 17975, 0x0000 }, /* R17975 - ADCL_RETUNE_C28_0 */ + { 17976, 0x0000 }, /* R17976 - ADCL_RETUNE_C29_1 */ + { 17977, 0x0000 }, /* R17977 - ADCL_RETUNE_C29_0 */ + { 17978, 0x0000 }, /* R17978 - ADCL_RETUNE_C30_1 */ + { 17979, 0x0000 }, /* R17979 - ADCL_RETUNE_C30_0 */ + { 17980, 0x0000 }, /* R17980 - ADCL_RETUNE_C31_1 */ + { 17981, 0x0000 }, /* R17981 - ADCL_RETUNE_C31_0 */ + { 17982, 0x0000 }, /* R17982 - ADCL_RETUNE_C32_1 */ + { 17983, 0x0000 }, /* R17983 - ADCL_RETUNE_C32_0 */ + + { 18432, 0x0020 }, /* R18432 - RETUNEADC_PG2_1 */ + { 18433, 0x0000 }, /* R18433 - RETUNEADC_PG2_0 */ + { 18434, 0x0040 }, /* R18434 - RETUNEADC_PG_1 */ + { 18435, 0x0000 }, /* R18435 - RETUNEADC_PG_0 */ + + { 18944, 0x007F }, /* R18944 - ADCR_RETUNE_C1_1 */ + { 18945, 0xFFFF }, /* R18945 - ADCR_RETUNE_C1_0 */ + { 18946, 0x0000 }, /* R18946 - ADCR_RETUNE_C2_1 */ + { 18947, 0x0000 }, /* R18947 - ADCR_RETUNE_C2_0 */ + { 18948, 0x0000 }, /* R18948 - ADCR_RETUNE_C3_1 */ + { 18949, 0x0000 }, /* R18949 - ADCR_RETUNE_C3_0 */ + { 18950, 0x0000 }, /* R18950 - ADCR_RETUNE_C4_1 */ + { 18951, 0x0000 }, /* R18951 - ADCR_RETUNE_C4_0 */ + { 18952, 0x0000 }, /* R18952 - ADCR_RETUNE_C5_1 */ + { 18953, 0x0000 }, /* R18953 - ADCR_RETUNE_C5_0 */ + { 18954, 0x0000 }, /* R18954 - ADCR_RETUNE_C6_1 */ + { 18955, 0x0000 }, /* R18955 - ADCR_RETUNE_C6_0 */ + { 18956, 0x0000 }, /* R18956 - ADCR_RETUNE_C7_1 */ + { 18957, 0x0000 }, /* R18957 - ADCR_RETUNE_C7_0 */ + { 18958, 0x0000 }, /* R18958 - ADCR_RETUNE_C8_1 */ + { 18959, 0x0000 }, /* R18959 - ADCR_RETUNE_C8_0 */ + { 18960, 0x0000 }, /* R18960 - ADCR_RETUNE_C9_1 */ + { 18961, 0x0000 }, /* R18961 - ADCR_RETUNE_C9_0 */ + { 18962, 0x0000 }, /* R18962 - ADCR_RETUNE_C10_1 */ + { 18963, 0x0000 }, /* R18963 - ADCR_RETUNE_C10_0 */ + { 18964, 0x0000 }, /* R18964 - ADCR_RETUNE_C11_1 */ + { 18965, 0x0000 }, /* R18965 - ADCR_RETUNE_C11_0 */ + { 18966, 0x0000 }, /* R18966 - ADCR_RETUNE_C12_1 */ + { 18967, 0x0000 }, /* R18967 - ADCR_RETUNE_C12_0 */ + { 18968, 0x0000 }, /* R18968 - ADCR_RETUNE_C13_1 */ + { 18969, 0x0000 }, /* R18969 - ADCR_RETUNE_C13_0 */ + { 18970, 0x0000 }, /* R18970 - ADCR_RETUNE_C14_1 */ + { 18971, 0x0000 }, /* R18971 - ADCR_RETUNE_C14_0 */ + { 18972, 0x0000 }, /* R18972 - ADCR_RETUNE_C15_1 */ + { 18973, 0x0000 }, /* R18973 - ADCR_RETUNE_C15_0 */ + { 18974, 0x0000 }, /* R18974 - ADCR_RETUNE_C16_1 */ + { 18975, 0x0000 }, /* R18975 - ADCR_RETUNE_C16_0 */ + { 18976, 0x0000 }, /* R18976 - ADCR_RETUNE_C17_1 */ + { 18977, 0x0000 }, /* R18977 - ADCR_RETUNE_C17_0 */ + { 18978, 0x0000 }, /* R18978 - ADCR_RETUNE_C18_1 */ + { 18979, 0x0000 }, /* R18979 - ADCR_RETUNE_C18_0 */ + { 18980, 0x0000 }, /* R18980 - ADCR_RETUNE_C19_1 */ + { 18981, 0x0000 }, /* R18981 - ADCR_RETUNE_C19_0 */ + { 18982, 0x0000 }, /* R18982 - ADCR_RETUNE_C20_1 */ + { 18983, 0x0000 }, /* R18983 - ADCR_RETUNE_C20_0 */ + { 18984, 0x0000 }, /* R18984 - ADCR_RETUNE_C21_1 */ + { 18985, 0x0000 }, /* R18985 - ADCR_RETUNE_C21_0 */ + { 18986, 0x0000 }, /* R18986 - ADCR_RETUNE_C22_1 */ + { 18987, 0x0000 }, /* R18987 - ADCR_RETUNE_C22_0 */ + { 18988, 0x0000 }, /* R18988 - ADCR_RETUNE_C23_1 */ + { 18989, 0x0000 }, /* R18989 - ADCR_RETUNE_C23_0 */ + { 18990, 0x0000 }, /* R18990 - ADCR_RETUNE_C24_1 */ + { 18991, 0x0000 }, /* R18991 - ADCR_RETUNE_C24_0 */ + { 18992, 0x0000 }, /* R18992 - ADCR_RETUNE_C25_1 */ + { 18993, 0x0000 }, /* R18993 - ADCR_RETUNE_C25_0 */ + { 18994, 0x0000 }, /* R18994 - ADCR_RETUNE_C26_1 */ + { 18995, 0x0000 }, /* R18995 - ADCR_RETUNE_C26_0 */ + { 18996, 0x0000 }, /* R18996 - ADCR_RETUNE_C27_1 */ + { 18997, 0x0000 }, /* R18997 - ADCR_RETUNE_C27_0 */ + { 18998, 0x0000 }, /* R18998 - ADCR_RETUNE_C28_1 */ + { 18999, 0x0000 }, /* R18999 - ADCR_RETUNE_C28_0 */ + { 19000, 0x0000 }, /* R19000 - ADCR_RETUNE_C29_1 */ + { 19001, 0x0000 }, /* R19001 - ADCR_RETUNE_C29_0 */ + { 19002, 0x0000 }, /* R19002 - ADCR_RETUNE_C30_1 */ + { 19003, 0x0000 }, /* R19003 - ADCR_RETUNE_C30_0 */ + { 19004, 0x0000 }, /* R19004 - ADCR_RETUNE_C31_1 */ + { 19005, 0x0000 }, /* R19005 - ADCR_RETUNE_C31_0 */ + { 19006, 0x0000 }, /* R19006 - ADCR_RETUNE_C32_1 */ + { 19007, 0x0000 }, /* R19007 - ADCR_RETUNE_C32_0 */ + + { 19456, 0x007F }, /* R19456 - DACL_RETUNE_C1_1 */ + { 19457, 0xFFFF }, /* R19457 - DACL_RETUNE_C1_0 */ + { 19458, 0x0000 }, /* R19458 - DACL_RETUNE_C2_1 */ + { 19459, 0x0000 }, /* R19459 - DACL_RETUNE_C2_0 */ + { 19460, 0x0000 }, /* R19460 - DACL_RETUNE_C3_1 */ + { 19461, 0x0000 }, /* R19461 - DACL_RETUNE_C3_0 */ + { 19462, 0x0000 }, /* R19462 - DACL_RETUNE_C4_1 */ + { 19463, 0x0000 }, /* R19463 - DACL_RETUNE_C4_0 */ + { 19464, 0x0000 }, /* R19464 - DACL_RETUNE_C5_1 */ + { 19465, 0x0000 }, /* R19465 - DACL_RETUNE_C5_0 */ + { 19466, 0x0000 }, /* R19466 - DACL_RETUNE_C6_1 */ + { 19467, 0x0000 }, /* R19467 - DACL_RETUNE_C6_0 */ + { 19468, 0x0000 }, /* R19468 - DACL_RETUNE_C7_1 */ + { 19469, 0x0000 }, /* R19469 - DACL_RETUNE_C7_0 */ + { 19470, 0x0000 }, /* R19470 - DACL_RETUNE_C8_1 */ + { 19471, 0x0000 }, /* R19471 - DACL_RETUNE_C8_0 */ + { 19472, 0x0000 }, /* R19472 - DACL_RETUNE_C9_1 */ + { 19473, 0x0000 }, /* R19473 - DACL_RETUNE_C9_0 */ + { 19474, 0x0000 }, /* R19474 - DACL_RETUNE_C10_1 */ + { 19475, 0x0000 }, /* R19475 - DACL_RETUNE_C10_0 */ + { 19476, 0x0000 }, /* R19476 - DACL_RETUNE_C11_1 */ + { 19477, 0x0000 }, /* R19477 - DACL_RETUNE_C11_0 */ + { 19478, 0x0000 }, /* R19478 - DACL_RETUNE_C12_1 */ + { 19479, 0x0000 }, /* R19479 - DACL_RETUNE_C12_0 */ + { 19480, 0x0000 }, /* R19480 - DACL_RETUNE_C13_1 */ + { 19481, 0x0000 }, /* R19481 - DACL_RETUNE_C13_0 */ + { 19482, 0x0000 }, /* R19482 - DACL_RETUNE_C14_1 */ + { 19483, 0x0000 }, /* R19483 - DACL_RETUNE_C14_0 */ + { 19484, 0x0000 }, /* R19484 - DACL_RETUNE_C15_1 */ + { 19485, 0x0000 }, /* R19485 - DACL_RETUNE_C15_0 */ + { 19486, 0x0000 }, /* R19486 - DACL_RETUNE_C16_1 */ + { 19487, 0x0000 }, /* R19487 - DACL_RETUNE_C16_0 */ + { 19488, 0x0000 }, /* R19488 - DACL_RETUNE_C17_1 */ + { 19489, 0x0000 }, /* R19489 - DACL_RETUNE_C17_0 */ + { 19490, 0x0000 }, /* R19490 - DACL_RETUNE_C18_1 */ + { 19491, 0x0000 }, /* R19491 - DACL_RETUNE_C18_0 */ + { 19492, 0x0000 }, /* R19492 - DACL_RETUNE_C19_1 */ + { 19493, 0x0000 }, /* R19493 - DACL_RETUNE_C19_0 */ + { 19494, 0x0000 }, /* R19494 - DACL_RETUNE_C20_1 */ + { 19495, 0x0000 }, /* R19495 - DACL_RETUNE_C20_0 */ + { 19496, 0x0000 }, /* R19496 - DACL_RETUNE_C21_1 */ + { 19497, 0x0000 }, /* R19497 - DACL_RETUNE_C21_0 */ + { 19498, 0x0000 }, /* R19498 - DACL_RETUNE_C22_1 */ + { 19499, 0x0000 }, /* R19499 - DACL_RETUNE_C22_0 */ + { 19500, 0x0000 }, /* R19500 - DACL_RETUNE_C23_1 */ + { 19501, 0x0000 }, /* R19501 - DACL_RETUNE_C23_0 */ + { 19502, 0x0000 }, /* R19502 - DACL_RETUNE_C24_1 */ + { 19503, 0x0000 }, /* R19503 - DACL_RETUNE_C24_0 */ + { 19504, 0x0000 }, /* R19504 - DACL_RETUNE_C25_1 */ + { 19505, 0x0000 }, /* R19505 - DACL_RETUNE_C25_0 */ + { 19506, 0x0000 }, /* R19506 - DACL_RETUNE_C26_1 */ + { 19507, 0x0000 }, /* R19507 - DACL_RETUNE_C26_0 */ + { 19508, 0x0000 }, /* R19508 - DACL_RETUNE_C27_1 */ + { 19509, 0x0000 }, /* R19509 - DACL_RETUNE_C27_0 */ + { 19510, 0x0000 }, /* R19510 - DACL_RETUNE_C28_1 */ + { 19511, 0x0000 }, /* R19511 - DACL_RETUNE_C28_0 */ + { 19512, 0x0000 }, /* R19512 - DACL_RETUNE_C29_1 */ + { 19513, 0x0000 }, /* R19513 - DACL_RETUNE_C29_0 */ + { 19514, 0x0000 }, /* R19514 - DACL_RETUNE_C30_1 */ + { 19515, 0x0000 }, /* R19515 - DACL_RETUNE_C30_0 */ + { 19516, 0x0000 }, /* R19516 - DACL_RETUNE_C31_1 */ + { 19517, 0x0000 }, /* R19517 - DACL_RETUNE_C31_0 */ + { 19518, 0x0000 }, /* R19518 - DACL_RETUNE_C32_1 */ + { 19519, 0x0000 }, /* R19519 - DACL_RETUNE_C32_0 */ + + { 19968, 0x0020 }, /* R19968 - RETUNEDAC_PG2_1 */ + { 19969, 0x0000 }, /* R19969 - RETUNEDAC_PG2_0 */ + { 19970, 0x0040 }, /* R19970 - RETUNEDAC_PG_1 */ + { 19971, 0x0000 }, /* R19971 - RETUNEDAC_PG_0 */ + + { 20480, 0x007F }, /* R20480 - DACR_RETUNE_C1_1 */ + { 20481, 0xFFFF }, /* R20481 - DACR_RETUNE_C1_0 */ + { 20482, 0x0000 }, /* R20482 - DACR_RETUNE_C2_1 */ + { 20483, 0x0000 }, /* R20483 - DACR_RETUNE_C2_0 */ + { 20484, 0x0000 }, /* R20484 - DACR_RETUNE_C3_1 */ + { 20485, 0x0000 }, /* R20485 - DACR_RETUNE_C3_0 */ + { 20486, 0x0000 }, /* R20486 - DACR_RETUNE_C4_1 */ + { 20487, 0x0000 }, /* R20487 - DACR_RETUNE_C4_0 */ + { 20488, 0x0000 }, /* R20488 - DACR_RETUNE_C5_1 */ + { 20489, 0x0000 }, /* R20489 - DACR_RETUNE_C5_0 */ + { 20490, 0x0000 }, /* R20490 - DACR_RETUNE_C6_1 */ + { 20491, 0x0000 }, /* R20491 - DACR_RETUNE_C6_0 */ + { 20492, 0x0000 }, /* R20492 - DACR_RETUNE_C7_1 */ + { 20493, 0x0000 }, /* R20493 - DACR_RETUNE_C7_0 */ + { 20494, 0x0000 }, /* R20494 - DACR_RETUNE_C8_1 */ + { 20495, 0x0000 }, /* R20495 - DACR_RETUNE_C8_0 */ + { 20496, 0x0000 }, /* R20496 - DACR_RETUNE_C9_1 */ + { 20497, 0x0000 }, /* R20497 - DACR_RETUNE_C9_0 */ + { 20498, 0x0000 }, /* R20498 - DACR_RETUNE_C10_1 */ + { 20499, 0x0000 }, /* R20499 - DACR_RETUNE_C10_0 */ + { 20500, 0x0000 }, /* R20500 - DACR_RETUNE_C11_1 */ + { 20501, 0x0000 }, /* R20501 - DACR_RETUNE_C11_0 */ + { 20502, 0x0000 }, /* R20502 - DACR_RETUNE_C12_1 */ + { 20503, 0x0000 }, /* R20503 - DACR_RETUNE_C12_0 */ + { 20504, 0x0000 }, /* R20504 - DACR_RETUNE_C13_1 */ + { 20505, 0x0000 }, /* R20505 - DACR_RETUNE_C13_0 */ + { 20506, 0x0000 }, /* R20506 - DACR_RETUNE_C14_1 */ + { 20507, 0x0000 }, /* R20507 - DACR_RETUNE_C14_0 */ + { 20508, 0x0000 }, /* R20508 - DACR_RETUNE_C15_1 */ + { 20509, 0x0000 }, /* R20509 - DACR_RETUNE_C15_0 */ + { 20510, 0x0000 }, /* R20510 - DACR_RETUNE_C16_1 */ + { 20511, 0x0000 }, /* R20511 - DACR_RETUNE_C16_0 */ + { 20512, 0x0000 }, /* R20512 - DACR_RETUNE_C17_1 */ + { 20513, 0x0000 }, /* R20513 - DACR_RETUNE_C17_0 */ + { 20514, 0x0000 }, /* R20514 - DACR_RETUNE_C18_1 */ + { 20515, 0x0000 }, /* R20515 - DACR_RETUNE_C18_0 */ + { 20516, 0x0000 }, /* R20516 - DACR_RETUNE_C19_1 */ + { 20517, 0x0000 }, /* R20517 - DACR_RETUNE_C19_0 */ + { 20518, 0x0000 }, /* R20518 - DACR_RETUNE_C20_1 */ + { 20519, 0x0000 }, /* R20519 - DACR_RETUNE_C20_0 */ + { 20520, 0x0000 }, /* R20520 - DACR_RETUNE_C21_1 */ + { 20521, 0x0000 }, /* R20521 - DACR_RETUNE_C21_0 */ + { 20522, 0x0000 }, /* R20522 - DACR_RETUNE_C22_1 */ + { 20523, 0x0000 }, /* R20523 - DACR_RETUNE_C22_0 */ + { 20524, 0x0000 }, /* R20524 - DACR_RETUNE_C23_1 */ + { 20525, 0x0000 }, /* R20525 - DACR_RETUNE_C23_0 */ + { 20526, 0x0000 }, /* R20526 - DACR_RETUNE_C24_1 */ + { 20527, 0x0000 }, /* R20527 - DACR_RETUNE_C24_0 */ + { 20528, 0x0000 }, /* R20528 - DACR_RETUNE_C25_1 */ + { 20529, 0x0000 }, /* R20529 - DACR_RETUNE_C25_0 */ + { 20530, 0x0000 }, /* R20530 - DACR_RETUNE_C26_1 */ + { 20531, 0x0000 }, /* R20531 - DACR_RETUNE_C26_0 */ + { 20532, 0x0000 }, /* R20532 - DACR_RETUNE_C27_1 */ + { 20533, 0x0000 }, /* R20533 - DACR_RETUNE_C27_0 */ + { 20534, 0x0000 }, /* R20534 - DACR_RETUNE_C28_1 */ + { 20535, 0x0000 }, /* R20535 - DACR_RETUNE_C28_0 */ + { 20536, 0x0000 }, /* R20536 - DACR_RETUNE_C29_1 */ + { 20537, 0x0000 }, /* R20537 - DACR_RETUNE_C29_0 */ + { 20538, 0x0000 }, /* R20538 - DACR_RETUNE_C30_1 */ + { 20539, 0x0000 }, /* R20539 - DACR_RETUNE_C30_0 */ + { 20540, 0x0000 }, /* R20540 - DACR_RETUNE_C31_1 */ + { 20541, 0x0000 }, /* R20541 - DACR_RETUNE_C31_0 */ + { 20542, 0x0000 }, /* R20542 - DACR_RETUNE_C32_1 */ + { 20543, 0x0000 }, /* R20543 - DACR_RETUNE_C32_0 */ + + { 20992, 0x008C }, /* R20992 - VSS_XHD2_1 */ + { 20993, 0x0200 }, /* R20993 - VSS_XHD2_0 */ + { 20994, 0x0035 }, /* R20994 - VSS_XHD3_1 */ + { 20995, 0x0700 }, /* R20995 - VSS_XHD3_0 */ + { 20996, 0x003A }, /* R20996 - VSS_XHN1_1 */ + { 20997, 0x4100 }, /* R20997 - VSS_XHN1_0 */ + { 20998, 0x008B }, /* R20998 - VSS_XHN2_1 */ + { 20999, 0x7D00 }, /* R20999 - VSS_XHN2_0 */ + { 21000, 0x003A }, /* R21000 - VSS_XHN3_1 */ + { 21001, 0x4100 }, /* R21001 - VSS_XHN3_0 */ + { 21002, 0x008C }, /* R21002 - VSS_XLA_1 */ + { 21003, 0xFEE8 }, /* R21003 - VSS_XLA_0 */ + { 21004, 0x0078 }, /* R21004 - VSS_XLB_1 */ + { 21005, 0x0000 }, /* R21005 - VSS_XLB_0 */ + { 21006, 0x003F }, /* R21006 - VSS_XLG_1 */ + { 21007, 0xB260 }, /* R21007 - VSS_XLG_0 */ + { 21008, 0x002D }, /* R21008 - VSS_PG2_1 */ + { 21009, 0x1818 }, /* R21009 - VSS_PG2_0 */ + { 21010, 0x0020 }, /* R21010 - VSS_PG_1 */ + { 21011, 0x0000 }, /* R21011 - VSS_PG_0 */ + { 21012, 0x00F1 }, /* R21012 - VSS_XTD1_1 */ + { 21013, 0x8340 }, /* R21013 - VSS_XTD1_0 */ + { 21014, 0x00FB }, /* R21014 - VSS_XTD2_1 */ + { 21015, 0x8300 }, /* R21015 - VSS_XTD2_0 */ + { 21016, 0x00EE }, /* R21016 - VSS_XTD3_1 */ + { 21017, 0xAEC0 }, /* R21017 - VSS_XTD3_0 */ + { 21018, 0x00FB }, /* R21018 - VSS_XTD4_1 */ + { 21019, 0xAC40 }, /* R21019 - VSS_XTD4_0 */ + { 21020, 0x00F1 }, /* R21020 - VSS_XTD5_1 */ + { 21021, 0x7F80 }, /* R21021 - VSS_XTD5_0 */ + { 21022, 0x00F4 }, /* R21022 - VSS_XTD6_1 */ + { 21023, 0x3B40 }, /* R21023 - VSS_XTD6_0 */ + { 21024, 0x00F5 }, /* R21024 - VSS_XTD7_1 */ + { 21025, 0xFB00 }, /* R21025 - VSS_XTD7_0 */ + { 21026, 0x00EA }, /* R21026 - VSS_XTD8_1 */ + { 21027, 0x10C0 }, /* R21027 - VSS_XTD8_0 */ + { 21028, 0x00FC }, /* R21028 - VSS_XTD9_1 */ + { 21029, 0xC580 }, /* R21029 - VSS_XTD9_0 */ + { 21030, 0x00E2 }, /* R21030 - VSS_XTD10_1 */ + { 21031, 0x75C0 }, /* R21031 - VSS_XTD10_0 */ + { 21032, 0x0004 }, /* R21032 - VSS_XTD11_1 */ + { 21033, 0xB480 }, /* R21033 - VSS_XTD11_0 */ + { 21034, 0x00D4 }, /* R21034 - VSS_XTD12_1 */ + { 21035, 0xF980 }, /* R21035 - VSS_XTD12_0 */ + { 21036, 0x0004 }, /* R21036 - VSS_XTD13_1 */ + { 21037, 0x9140 }, /* R21037 - VSS_XTD13_0 */ + { 21038, 0x00D8 }, /* R21038 - VSS_XTD14_1 */ + { 21039, 0xA480 }, /* R21039 - VSS_XTD14_0 */ + { 21040, 0x0002 }, /* R21040 - VSS_XTD15_1 */ + { 21041, 0x3DC0 }, /* R21041 - VSS_XTD15_0 */ + { 21042, 0x00CF }, /* R21042 - VSS_XTD16_1 */ + { 21043, 0x7A80 }, /* R21043 - VSS_XTD16_0 */ + { 21044, 0x00DC }, /* R21044 - VSS_XTD17_1 */ + { 21045, 0x0600 }, /* R21045 - VSS_XTD17_0 */ + { 21046, 0x00F2 }, /* R21046 - VSS_XTD18_1 */ + { 21047, 0xDAC0 }, /* R21047 - VSS_XTD18_0 */ + { 21048, 0x00BA }, /* R21048 - VSS_XTD19_1 */ + { 21049, 0xF340 }, /* R21049 - VSS_XTD19_0 */ + { 21050, 0x000A }, /* R21050 - VSS_XTD20_1 */ + { 21051, 0x7940 }, /* R21051 - VSS_XTD20_0 */ + { 21052, 0x001C }, /* R21052 - VSS_XTD21_1 */ + { 21053, 0x0680 }, /* R21053 - VSS_XTD21_0 */ + { 21054, 0x00FD }, /* R21054 - VSS_XTD22_1 */ + { 21055, 0x2D00 }, /* R21055 - VSS_XTD22_0 */ + { 21056, 0x001C }, /* R21056 - VSS_XTD23_1 */ + { 21057, 0xE840 }, /* R21057 - VSS_XTD23_0 */ + { 21058, 0x000D }, /* R21058 - VSS_XTD24_1 */ + { 21059, 0xDC40 }, /* R21059 - VSS_XTD24_0 */ + { 21060, 0x00FC }, /* R21060 - VSS_XTD25_1 */ + { 21061, 0x9D00 }, /* R21061 - VSS_XTD25_0 */ + { 21062, 0x0009 }, /* R21062 - VSS_XTD26_1 */ + { 21063, 0x5580 }, /* R21063 - VSS_XTD26_0 */ + { 21064, 0x00FE }, /* R21064 - VSS_XTD27_1 */ + { 21065, 0x7E80 }, /* R21065 - VSS_XTD27_0 */ + { 21066, 0x000E }, /* R21066 - VSS_XTD28_1 */ + { 21067, 0xAB40 }, /* R21067 - VSS_XTD28_0 */ + { 21068, 0x00F9 }, /* R21068 - VSS_XTD29_1 */ + { 21069, 0x9880 }, /* R21069 - VSS_XTD29_0 */ + { 21070, 0x0009 }, /* R21070 - VSS_XTD30_1 */ + { 21071, 0x87C0 }, /* R21071 - VSS_XTD30_0 */ + { 21072, 0x00FD }, /* R21072 - VSS_XTD31_1 */ + { 21073, 0x2C40 }, /* R21073 - VSS_XTD31_0 */ + { 21074, 0x0009 }, /* R21074 - VSS_XTD32_1 */ + { 21075, 0x4800 }, /* R21075 - VSS_XTD32_0 */ + { 21076, 0x0003 }, /* R21076 - VSS_XTS1_1 */ + { 21077, 0x5F40 }, /* R21077 - VSS_XTS1_0 */ + { 21078, 0x0000 }, /* R21078 - VSS_XTS2_1 */ + { 21079, 0x8700 }, /* R21079 - VSS_XTS2_0 */ + { 21080, 0x00FA }, /* R21080 - VSS_XTS3_1 */ + { 21081, 0xE4C0 }, /* R21081 - VSS_XTS3_0 */ + { 21082, 0x0000 }, /* R21082 - VSS_XTS4_1 */ + { 21083, 0x0B40 }, /* R21083 - VSS_XTS4_0 */ + { 21084, 0x0004 }, /* R21084 - VSS_XTS5_1 */ + { 21085, 0xE180 }, /* R21085 - VSS_XTS5_0 */ + { 21086, 0x0001 }, /* R21086 - VSS_XTS6_1 */ + { 21087, 0x1F40 }, /* R21087 - VSS_XTS6_0 */ + { 21088, 0x00F8 }, /* R21088 - VSS_XTS7_1 */ + { 21089, 0xB000 }, /* R21089 - VSS_XTS7_0 */ + { 21090, 0x00FB }, /* R21090 - VSS_XTS8_1 */ + { 21091, 0xCBC0 }, /* R21091 - VSS_XTS8_0 */ + { 21092, 0x0004 }, /* R21092 - VSS_XTS9_1 */ + { 21093, 0xF380 }, /* R21093 - VSS_XTS9_0 */ + { 21094, 0x0007 }, /* R21094 - VSS_XTS10_1 */ + { 21095, 0xDF40 }, /* R21095 - VSS_XTS10_0 */ + { 21096, 0x00FF }, /* R21096 - VSS_XTS11_1 */ + { 21097, 0x0700 }, /* R21097 - VSS_XTS11_0 */ + { 21098, 0x00EF }, /* R21098 - VSS_XTS12_1 */ + { 21099, 0xD700 }, /* R21099 - VSS_XTS12_0 */ + { 21100, 0x00FB }, /* R21100 - VSS_XTS13_1 */ + { 21101, 0xAF40 }, /* R21101 - VSS_XTS13_0 */ + { 21102, 0x0010 }, /* R21102 - VSS_XTS14_1 */ + { 21103, 0x8A80 }, /* R21103 - VSS_XTS14_0 */ + { 21104, 0x0011 }, /* R21104 - VSS_XTS15_1 */ + { 21105, 0x07C0 }, /* R21105 - VSS_XTS15_0 */ + { 21106, 0x00E0 }, /* R21106 - VSS_XTS16_1 */ + { 21107, 0x0800 }, /* R21107 - VSS_XTS16_0 */ + { 21108, 0x00D2 }, /* R21108 - VSS_XTS17_1 */ + { 21109, 0x7600 }, /* R21109 - VSS_XTS17_0 */ + { 21110, 0x0020 }, /* R21110 - VSS_XTS18_1 */ + { 21111, 0xCF40 }, /* R21111 - VSS_XTS18_0 */ + { 21112, 0x0030 }, /* R21112 - VSS_XTS19_1 */ + { 21113, 0x2340 }, /* R21113 - VSS_XTS19_0 */ + { 21114, 0x00FD }, /* R21114 - VSS_XTS20_1 */ + { 21115, 0x69C0 }, /* R21115 - VSS_XTS20_0 */ + { 21116, 0x0028 }, /* R21116 - VSS_XTS21_1 */ + { 21117, 0x3500 }, /* R21117 - VSS_XTS21_0 */ + { 21118, 0x0006 }, /* R21118 - VSS_XTS22_1 */ + { 21119, 0x3300 }, /* R21119 - VSS_XTS22_0 */ + { 21120, 0x00D9 }, /* R21120 - VSS_XTS23_1 */ + { 21121, 0xF6C0 }, /* R21121 - VSS_XTS23_0 */ + { 21122, 0x00F3 }, /* R21122 - VSS_XTS24_1 */ + { 21123, 0x3340 }, /* R21123 - VSS_XTS24_0 */ + { 21124, 0x000F }, /* R21124 - VSS_XTS25_1 */ + { 21125, 0x4200 }, /* R21125 - VSS_XTS25_0 */ + { 21126, 0x0004 }, /* R21126 - VSS_XTS26_1 */ + { 21127, 0x0C80 }, /* R21127 - VSS_XTS26_0 */ + { 21128, 0x00FB }, /* R21128 - VSS_XTS27_1 */ + { 21129, 0x3F80 }, /* R21129 - VSS_XTS27_0 */ + { 21130, 0x00F7 }, /* R21130 - VSS_XTS28_1 */ + { 21131, 0x57C0 }, /* R21131 - VSS_XTS28_0 */ + { 21132, 0x0003 }, /* R21132 - VSS_XTS29_1 */ + { 21133, 0x5400 }, /* R21133 - VSS_XTS29_0 */ + { 21134, 0x0000 }, /* R21134 - VSS_XTS30_1 */ + { 21135, 0xC6C0 }, /* R21135 - VSS_XTS30_0 */ + { 21136, 0x0003 }, /* R21136 - VSS_XTS31_1 */ + { 21137, 0x12C0 }, /* R21137 - VSS_XTS31_0 */ + { 21138, 0x00FD }, /* R21138 - VSS_XTS32_1 */ + { 21139, 0x8580 }, /* R21139 - VSS_XTS32_0 */ }; static const struct wm8962_reg_access { @@ -802,7 +804,7 @@ static const struct wm8962_reg_access { u16 vol; } wm8962_reg_access[WM8962_MAX_REGISTER + 1] = { [0] = { 0x00FF, 0x01FF, 0x0000 }, /* R0 - Left Input volume */ - [1] = { 0xFEFF, 0x01FF, 0xFFFF }, /* R1 - Right Input volume */ + [1] = { 0xFEFF, 0x01FF, 0x0000 }, /* R1 - Right Input volume */ [2] = { 0x00FF, 0x01FF, 0x0000 }, /* R2 - HPOUTL volume */ [3] = { 0x00FF, 0x01FF, 0x0000 }, /* R3 - HPOUTR volume */ [4] = { 0x07FE, 0x07FE, 0xFFFF }, /* R4 - Clocking1 */ @@ -1943,7 +1945,7 @@ static const struct wm8962_reg_access { [21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */ }; -static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8962_volatile_register(struct device *dev, unsigned int reg) { if (wm8962_reg_access[reg].vol) return 1; @@ -1951,7 +1953,7 @@ static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int re return 0; } -static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8962_readable_register(struct device *dev, unsigned int reg) { if (wm8962_reg_access[reg].read) return 1; @@ -1959,15 +1961,15 @@ static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int re return 0; } -static int wm8962_reset(struct snd_soc_codec *codec) +static int wm8962_reset(struct wm8962_priv *wm8962) { int ret; - ret = snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243); + ret = regmap_write(wm8962->regmap, WM8962_SOFTWARE_RESET, 0x6243); if (ret != 0) return ret; - return snd_soc_write(codec, WM8962_PLL_SOFTWARE_RESET, 0); + return regmap_write(wm8962->regmap, WM8962_PLL_SOFTWARE_RESET, 0); } static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); @@ -2345,6 +2347,10 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, int src; int fll; + /* Ignore attempts to run the event during startup */ + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + return 0; + src = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_SRC_MASK; switch (src) { @@ -2939,33 +2945,6 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) return 0; } -static void wm8962_sync_cache(struct snd_soc_codec *codec) -{ - u16 *reg_cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - dev_dbg(codec->dev, "Syncing cache\n"); - - codec->cache_only = 0; - - /* Sync back cached values if they're different from the - * hardware default. - */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (i == WM8962_SOFTWARE_RESET) - continue; - if (reg_cache[i] == wm8962_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } - - codec->cache_sync = 0; -} - /* -1 for reserved values */ static const int bclk_divs[] = { 1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32 @@ -3093,7 +3072,8 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8962_sync_cache(codec); + regcache_cache_only(wm8962->regmap, false); + regcache_sync(wm8962->regmap); snd_soc_update_bits(codec, WM8962_ANTI_POP, WM8962_STARTUP_BIAS_ENA | @@ -3966,26 +3946,12 @@ static int wm8962_probe(struct snd_soc_codec *codec) bool dmicclk, dmicdat; wm8962->codec = codec; - INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); - init_completion(&wm8962->fll_lock); - - codec->cache_sync = 1; - codec->dapm.idle_bias_off = 1; + codec->control_data = wm8962->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; - } - - for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) - wm8962->supplies[i].supply = wm8962_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8962->supplies), - wm8962->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; + return ret; } wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0; @@ -4008,43 +3974,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) } } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), - wm8962->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - ret = snd_soc_read(codec, WM8962_SOFTWARE_RESET); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_enable; - } - if (ret != wm8962_reg[WM8962_SOFTWARE_RESET]) { - dev_err(codec->dev, "Device is not a WM8962, ID %x != %x\n", - ret, wm8962_reg[WM8962_SOFTWARE_RESET]); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8962_RIGHT_INPUT_VOLUME); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_enable; - } - - dev_info(codec->dev, "customer id %x revision %c\n", - (ret & WM8962_CUST_ID_MASK) >> WM8962_CUST_ID_SHIFT, - ((ret & WM8962_CHIP_REV_MASK) >> WM8962_CHIP_REV_SHIFT) - + 'A'); - - ret = wm8962_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - /* SYSCLK defaults to on; make sure it is off so we can safely * write to registers if the device is declocked. */ @@ -4059,8 +3988,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, 0); - regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); - if (pdata) { /* Apply static configuration for GPIOs */ for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) @@ -4170,13 +4097,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) } return 0; - -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); -err: - return ret; } static int wm8962_remove(struct snd_soc_codec *codec) @@ -4194,7 +4114,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) regulator_unregister_notifier(wm8962->supplies[i].consumer, &wm8962->disable_nb[i]); - regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); return 0; } @@ -4203,20 +4122,28 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, .set_bias_level = wm8962_set_bias_level, - .reg_cache_size = WM8962_MAX_REGISTER + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8962_reg, - .volatile_register = wm8962_volatile_register, - .readable_register = wm8962_readable_register, .set_pll = wm8962_set_fll, }; +static const struct regmap_config wm8962_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM8962_MAX_REGISTER, + .reg_defaults = wm8962_reg, + .num_reg_defaults = ARRAY_SIZE(wm8962_reg), + .volatile_reg = wm8962_volatile_register, + .readable_reg = wm8962_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8962_priv *wm8962; - int ret; + unsigned int reg; + int ret, i; wm8962 = kzalloc(sizeof(struct wm8962_priv), GFP_KERNEL); if (wm8962 == NULL) @@ -4224,19 +4151,103 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8962); + INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); + init_completion(&wm8962->fll_lock); wm8962->irq = i2c->irq; + for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) + wm8962->supplies[i].supply = wm8962_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8962->supplies), + wm8962->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err_alloc; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), + wm8962->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + wm8962->regmap = regmap_init_i2c(i2c, &wm8962_regmap); + if (IS_ERR(wm8962->regmap)) { + ret = PTR_ERR(wm8962->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(wm8962->regmap, true); + + ret = regmap_read(wm8962->regmap, WM8962_SOFTWARE_RESET, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register\n"); + goto err_regmap; + } + if (reg != 0x6243) { + dev_err(&i2c->dev, + "Device is not a WM8962, ID %x != 0x6243\n", ret); + ret = -EINVAL; + goto err_regmap; + } + + ret = regmap_read(wm8962->regmap, WM8962_RIGHT_INPUT_VOLUME, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read device revision: %d\n", + ret); + goto err_regmap; + } + + dev_info(&i2c->dev, "customer id %x revision %c\n", + (reg & WM8962_CUST_ID_MASK) >> WM8962_CUST_ID_SHIFT, + ((reg & WM8962_CHIP_REV_MASK) >> WM8962_CHIP_REV_SHIFT) + + 'A'); + + regcache_cache_bypass(wm8962->regmap, false); + + ret = wm8962_reset(wm8962); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_regmap; + } + + regcache_cache_only(wm8962->regmap, true); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8962, &wm8962_dai, 1); if (ret < 0) - kfree(wm8962); + goto err_regmap; + + /* The drivers should power up as needed */ + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + + return 0; +err_regmap: + regmap_exit(wm8962->regmap); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); +err_alloc: + kfree(wm8962); return ret; } static __devexit int wm8962_i2c_remove(struct i2c_client *client) { + struct wm8962_priv *wm8962 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8962->regmap); + regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From bd132ec585c498ee27d7eedf8569703606743928 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 23 Oct 2011 11:10:45 +0100 Subject: ASoC: Convert wm5100 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 1488 +++++++++++++++++++------------------- sound/soc/codecs/wm5100.c | 49 +- sound/soc/codecs/wm5100.h | 7 +- 3 files changed, 786 insertions(+), 758 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e9ce81a57b85..6b2ab65735de 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -13,7 +13,7 @@ #include "wm5100.h" -int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +bool wm5100_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM5100_SOFTWARE_RESET: @@ -36,7 +36,7 @@ int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg) } } -int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg) +bool wm5100_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case WM5100_SOFTWARE_RESET: @@ -786,746 +786,746 @@ int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg) } } -u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { - [0x0000] = 0x0000, /* R0 - software reset */ - [0x0001] = 0x0000, /* R1 - Device Revision */ - [0x0010] = 0x0801, /* R16 - Ctrl IF 1 */ - [0x0020] = 0x0000, /* R32 - Tone Generator 1 */ - [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ - [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ - [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ - [0x0100] = 0x0002, /* R256 - Clocking 1 */ - [0x0101] = 0x0000, /* R257 - Clocking 3 */ - [0x0102] = 0x0011, /* R258 - Clocking 4 */ - [0x0103] = 0x0011, /* R259 - Clocking 5 */ - [0x0104] = 0x0011, /* R260 - Clocking 6 */ - [0x0107] = 0x0000, /* R263 - Clocking 7 */ - [0x0108] = 0x0000, /* R264 - Clocking 8 */ - [0x0120] = 0x0000, /* R288 - ASRC_ENABLE */ - [0x0121] = 0x0000, /* R289 - ASRC_STATUS */ - [0x0122] = 0x0000, /* R290 - ASRC_RATE1 */ - [0x0141] = 0x8000, /* R321 - ISRC 1 CTRL 1 */ - [0x0142] = 0x0000, /* R322 - ISRC 1 CTRL 2 */ - [0x0143] = 0x8000, /* R323 - ISRC 2 CTRL1 */ - [0x0144] = 0x0000, /* R324 - ISRC 2 CTRL 2 */ - [0x0182] = 0x0000, /* R386 - FLL1 Control 1 */ - [0x0183] = 0x0000, /* R387 - FLL1 Control 2 */ - [0x0184] = 0x0000, /* R388 - FLL1 Control 3 */ - [0x0186] = 0x0177, /* R390 - FLL1 Control 5 */ - [0x0187] = 0x0001, /* R391 - FLL1 Control 6 */ - [0x0188] = 0x0000, /* R392 - FLL1 EFS 1 */ - [0x01A2] = 0x0000, /* R418 - FLL2 Control 1 */ - [0x01A3] = 0x0000, /* R419 - FLL2 Control 2 */ - [0x01A4] = 0x0000, /* R420 - FLL2 Control 3 */ - [0x01A6] = 0x0177, /* R422 - FLL2 Control 5 */ - [0x01A7] = 0x0001, /* R423 - FLL2 Control 6 */ - [0x01A8] = 0x0000, /* R424 - FLL2 EFS 1 */ - [0x0200] = 0x0020, /* R512 - Mic Charge Pump 1 */ - [0x0201] = 0xB084, /* R513 - Mic Charge Pump 2 */ - [0x0202] = 0xBBDE, /* R514 - HP Charge Pump 1 */ - [0x0211] = 0x20D4, /* R529 - LDO1 Control */ - [0x0215] = 0x0062, /* R533 - Mic Bias Ctrl 1 */ - [0x0216] = 0x0062, /* R534 - Mic Bias Ctrl 2 */ - [0x0217] = 0x0062, /* R535 - Mic Bias Ctrl 3 */ - [0x0280] = 0x0004, /* R640 - Accessory Detect Mode 1 */ - [0x0288] = 0x0020, /* R648 - Headphone Detect 1 */ - [0x0289] = 0x0000, /* R649 - Headphone Detect 2 */ - [0x0290] = 0x1100, /* R656 - Mic Detect 1 */ - [0x0291] = 0x009F, /* R657 - Mic Detect 2 */ - [0x0292] = 0x0000, /* R658 - Mic Detect 3 */ - [0x0301] = 0x0000, /* R769 - Input Enables */ - [0x0302] = 0x0000, /* R770 - Input Enables Status */ - [0x0310] = 0x2280, /* R784 - Status */ - [0x0311] = 0x0080, /* R785 - IN1R Control */ - [0x0312] = 0x2280, /* R786 - IN2L Control */ - [0x0313] = 0x0080, /* R787 - IN2R Control */ - [0x0314] = 0x2280, /* R788 - IN3L Control */ - [0x0315] = 0x0080, /* R789 - IN3R Control */ - [0x0316] = 0x2280, /* R790 - IN4L Control */ - [0x0317] = 0x0080, /* R791 - IN4R Control */ - [0x0318] = 0x0000, /* R792 - RXANC_SRC */ - [0x0319] = 0x0022, /* R793 - Input Volume Ramp */ - [0x0320] = 0x0180, /* R800 - ADC Digital Volume 1L */ - [0x0321] = 0x0180, /* R801 - ADC Digital Volume 1R */ - [0x0322] = 0x0180, /* R802 - ADC Digital Volume 2L */ - [0x0323] = 0x0180, /* R803 - ADC Digital Volume 2R */ - [0x0324] = 0x0180, /* R804 - ADC Digital Volume 3L */ - [0x0325] = 0x0180, /* R805 - ADC Digital Volume 3R */ - [0x0326] = 0x0180, /* R806 - ADC Digital Volume 4L */ - [0x0327] = 0x0180, /* R807 - ADC Digital Volume 4R */ - [0x0401] = 0x0000, /* R1025 - Output Enables 2 */ - [0x0402] = 0x0000, /* R1026 - Output Status 1 */ - [0x0403] = 0x0000, /* R1027 - Output Status 2 */ - [0x0408] = 0x0000, /* R1032 - Channel Enables 1 */ - [0x0410] = 0x0080, /* R1040 - Out Volume 1L */ - [0x0411] = 0x0080, /* R1041 - Out Volume 1R */ - [0x0412] = 0x0080, /* R1042 - DAC Volume Limit 1L */ - [0x0413] = 0x0080, /* R1043 - DAC Volume Limit 1R */ - [0x0414] = 0x0080, /* R1044 - Out Volume 2L */ - [0x0415] = 0x0080, /* R1045 - Out Volume 2R */ - [0x0416] = 0x0080, /* R1046 - DAC Volume Limit 2L */ - [0x0417] = 0x0080, /* R1047 - DAC Volume Limit 2R */ - [0x0418] = 0x0080, /* R1048 - Out Volume 3L */ - [0x0419] = 0x0080, /* R1049 - Out Volume 3R */ - [0x041A] = 0x0080, /* R1050 - DAC Volume Limit 3L */ - [0x041B] = 0x0080, /* R1051 - DAC Volume Limit 3R */ - [0x041C] = 0x0080, /* R1052 - Out Volume 4L */ - [0x041D] = 0x0080, /* R1053 - Out Volume 4R */ - [0x041E] = 0x0080, /* R1054 - DAC Volume Limit 5L */ - [0x041F] = 0x0080, /* R1055 - DAC Volume Limit 5R */ - [0x0420] = 0x0080, /* R1056 - DAC Volume Limit 6L */ - [0x0421] = 0x0080, /* R1057 - DAC Volume Limit 6R */ - [0x0440] = 0x0000, /* R1088 - DAC AEC Control 1 */ - [0x0441] = 0x0022, /* R1089 - Output Volume Ramp */ - [0x0480] = 0x0180, /* R1152 - DAC Digital Volume 1L */ - [0x0481] = 0x0180, /* R1153 - DAC Digital Volume 1R */ - [0x0482] = 0x0180, /* R1154 - DAC Digital Volume 2L */ - [0x0483] = 0x0180, /* R1155 - DAC Digital Volume 2R */ - [0x0484] = 0x0180, /* R1156 - DAC Digital Volume 3L */ - [0x0485] = 0x0180, /* R1157 - DAC Digital Volume 3R */ - [0x0486] = 0x0180, /* R1158 - DAC Digital Volume 4L */ - [0x0487] = 0x0180, /* R1159 - DAC Digital Volume 4R */ - [0x0488] = 0x0180, /* R1160 - DAC Digital Volume 5L */ - [0x0489] = 0x0180, /* R1161 - DAC Digital Volume 5R */ - [0x048A] = 0x0180, /* R1162 - DAC Digital Volume 6L */ - [0x048B] = 0x0180, /* R1163 - DAC Digital Volume 6R */ - [0x04C0] = 0x0069, /* R1216 - PDM SPK1 CTRL 1 */ - [0x04C1] = 0x0000, /* R1217 - PDM SPK1 CTRL 2 */ - [0x04C2] = 0x0069, /* R1218 - PDM SPK2 CTRL 1 */ - [0x04C3] = 0x0000, /* R1219 - PDM SPK2 CTRL 2 */ - [0x0500] = 0x000C, /* R1280 - Audio IF 1_1 */ - [0x0501] = 0x0008, /* R1281 - Audio IF 1_2 */ - [0x0502] = 0x0000, /* R1282 - Audio IF 1_3 */ - [0x0503] = 0x0000, /* R1283 - Audio IF 1_4 */ - [0x0504] = 0x0000, /* R1284 - Audio IF 1_5 */ - [0x0505] = 0x0300, /* R1285 - Audio IF 1_6 */ - [0x0506] = 0x0300, /* R1286 - Audio IF 1_7 */ - [0x0507] = 0x1820, /* R1287 - Audio IF 1_8 */ - [0x0508] = 0x1820, /* R1288 - Audio IF 1_9 */ - [0x0509] = 0x0000, /* R1289 - Audio IF 1_10 */ - [0x050A] = 0x0001, /* R1290 - Audio IF 1_11 */ - [0x050B] = 0x0002, /* R1291 - Audio IF 1_12 */ - [0x050C] = 0x0003, /* R1292 - Audio IF 1_13 */ - [0x050D] = 0x0004, /* R1293 - Audio IF 1_14 */ - [0x050E] = 0x0005, /* R1294 - Audio IF 1_15 */ - [0x050F] = 0x0006, /* R1295 - Audio IF 1_16 */ - [0x0510] = 0x0007, /* R1296 - Audio IF 1_17 */ - [0x0511] = 0x0000, /* R1297 - Audio IF 1_18 */ - [0x0512] = 0x0001, /* R1298 - Audio IF 1_19 */ - [0x0513] = 0x0002, /* R1299 - Audio IF 1_20 */ - [0x0514] = 0x0003, /* R1300 - Audio IF 1_21 */ - [0x0515] = 0x0004, /* R1301 - Audio IF 1_22 */ - [0x0516] = 0x0005, /* R1302 - Audio IF 1_23 */ - [0x0517] = 0x0006, /* R1303 - Audio IF 1_24 */ - [0x0518] = 0x0007, /* R1304 - Audio IF 1_25 */ - [0x0519] = 0x0000, /* R1305 - Audio IF 1_26 */ - [0x051A] = 0x0000, /* R1306 - Audio IF 1_27 */ - [0x0540] = 0x000C, /* R1344 - Audio IF 2_1 */ - [0x0541] = 0x0008, /* R1345 - Audio IF 2_2 */ - [0x0542] = 0x0000, /* R1346 - Audio IF 2_3 */ - [0x0543] = 0x0000, /* R1347 - Audio IF 2_4 */ - [0x0544] = 0x0000, /* R1348 - Audio IF 2_5 */ - [0x0545] = 0x0300, /* R1349 - Audio IF 2_6 */ - [0x0546] = 0x0300, /* R1350 - Audio IF 2_7 */ - [0x0547] = 0x1820, /* R1351 - Audio IF 2_8 */ - [0x0548] = 0x1820, /* R1352 - Audio IF 2_9 */ - [0x0549] = 0x0000, /* R1353 - Audio IF 2_10 */ - [0x054A] = 0x0001, /* R1354 - Audio IF 2_11 */ - [0x0551] = 0x0000, /* R1361 - Audio IF 2_18 */ - [0x0552] = 0x0001, /* R1362 - Audio IF 2_19 */ - [0x0559] = 0x0000, /* R1369 - Audio IF 2_26 */ - [0x055A] = 0x0000, /* R1370 - Audio IF 2_27 */ - [0x0580] = 0x000C, /* R1408 - Audio IF 3_1 */ - [0x0581] = 0x0008, /* R1409 - Audio IF 3_2 */ - [0x0582] = 0x0000, /* R1410 - Audio IF 3_3 */ - [0x0583] = 0x0000, /* R1411 - Audio IF 3_4 */ - [0x0584] = 0x0000, /* R1412 - Audio IF 3_5 */ - [0x0585] = 0x0300, /* R1413 - Audio IF 3_6 */ - [0x0586] = 0x0300, /* R1414 - Audio IF 3_7 */ - [0x0587] = 0x1820, /* R1415 - Audio IF 3_8 */ - [0x0588] = 0x1820, /* R1416 - Audio IF 3_9 */ - [0x0589] = 0x0000, /* R1417 - Audio IF 3_10 */ - [0x058A] = 0x0001, /* R1418 - Audio IF 3_11 */ - [0x0591] = 0x0000, /* R1425 - Audio IF 3_18 */ - [0x0592] = 0x0001, /* R1426 - Audio IF 3_19 */ - [0x0599] = 0x0000, /* R1433 - Audio IF 3_26 */ - [0x059A] = 0x0000, /* R1434 - Audio IF 3_27 */ - [0x0640] = 0x0000, /* R1600 - PWM1MIX Input 1 Source */ - [0x0641] = 0x0080, /* R1601 - PWM1MIX Input 1 Volume */ - [0x0642] = 0x0000, /* R1602 - PWM1MIX Input 2 Source */ - [0x0643] = 0x0080, /* R1603 - PWM1MIX Input 2 Volume */ - [0x0644] = 0x0000, /* R1604 - PWM1MIX Input 3 Source */ - [0x0645] = 0x0080, /* R1605 - PWM1MIX Input 3 Volume */ - [0x0646] = 0x0000, /* R1606 - PWM1MIX Input 4 Source */ - [0x0647] = 0x0080, /* R1607 - PWM1MIX Input 4 Volume */ - [0x0648] = 0x0000, /* R1608 - PWM2MIX Input 1 Source */ - [0x0649] = 0x0080, /* R1609 - PWM2MIX Input 1 Volume */ - [0x064A] = 0x0000, /* R1610 - PWM2MIX Input 2 Source */ - [0x064B] = 0x0080, /* R1611 - PWM2MIX Input 2 Volume */ - [0x064C] = 0x0000, /* R1612 - PWM2MIX Input 3 Source */ - [0x064D] = 0x0080, /* R1613 - PWM2MIX Input 3 Volume */ - [0x064E] = 0x0000, /* R1614 - PWM2MIX Input 4 Source */ - [0x064F] = 0x0080, /* R1615 - PWM2MIX Input 4 Volume */ - [0x0680] = 0x0000, /* R1664 - OUT1LMIX Input 1 Source */ - [0x0681] = 0x0080, /* R1665 - OUT1LMIX Input 1 Volume */ - [0x0682] = 0x0000, /* R1666 - OUT1LMIX Input 2 Source */ - [0x0683] = 0x0080, /* R1667 - OUT1LMIX Input 2 Volume */ - [0x0684] = 0x0000, /* R1668 - OUT1LMIX Input 3 Source */ - [0x0685] = 0x0080, /* R1669 - OUT1LMIX Input 3 Volume */ - [0x0686] = 0x0000, /* R1670 - OUT1LMIX Input 4 Source */ - [0x0687] = 0x0080, /* R1671 - OUT1LMIX Input 4 Volume */ - [0x0688] = 0x0000, /* R1672 - OUT1RMIX Input 1 Source */ - [0x0689] = 0x0080, /* R1673 - OUT1RMIX Input 1 Volume */ - [0x068A] = 0x0000, /* R1674 - OUT1RMIX Input 2 Source */ - [0x068B] = 0x0080, /* R1675 - OUT1RMIX Input 2 Volume */ - [0x068C] = 0x0000, /* R1676 - OUT1RMIX Input 3 Source */ - [0x068D] = 0x0080, /* R1677 - OUT1RMIX Input 3 Volume */ - [0x068E] = 0x0000, /* R1678 - OUT1RMIX Input 4 Source */ - [0x068F] = 0x0080, /* R1679 - OUT1RMIX Input 4 Volume */ - [0x0690] = 0x0000, /* R1680 - OUT2LMIX Input 1 Source */ - [0x0691] = 0x0080, /* R1681 - OUT2LMIX Input 1 Volume */ - [0x0692] = 0x0000, /* R1682 - OUT2LMIX Input 2 Source */ - [0x0693] = 0x0080, /* R1683 - OUT2LMIX Input 2 Volume */ - [0x0694] = 0x0000, /* R1684 - OUT2LMIX Input 3 Source */ - [0x0695] = 0x0080, /* R1685 - OUT2LMIX Input 3 Volume */ - [0x0696] = 0x0000, /* R1686 - OUT2LMIX Input 4 Source */ - [0x0697] = 0x0080, /* R1687 - OUT2LMIX Input 4 Volume */ - [0x0698] = 0x0000, /* R1688 - OUT2RMIX Input 1 Source */ - [0x0699] = 0x0080, /* R1689 - OUT2RMIX Input 1 Volume */ - [0x069A] = 0x0000, /* R1690 - OUT2RMIX Input 2 Source */ - [0x069B] = 0x0080, /* R1691 - OUT2RMIX Input 2 Volume */ - [0x069C] = 0x0000, /* R1692 - OUT2RMIX Input 3 Source */ - [0x069D] = 0x0080, /* R1693 - OUT2RMIX Input 3 Volume */ - [0x069E] = 0x0000, /* R1694 - OUT2RMIX Input 4 Source */ - [0x069F] = 0x0080, /* R1695 - OUT2RMIX Input 4 Volume */ - [0x06A0] = 0x0000, /* R1696 - OUT3LMIX Input 1 Source */ - [0x06A1] = 0x0080, /* R1697 - OUT3LMIX Input 1 Volume */ - [0x06A2] = 0x0000, /* R1698 - OUT3LMIX Input 2 Source */ - [0x06A3] = 0x0080, /* R1699 - OUT3LMIX Input 2 Volume */ - [0x06A4] = 0x0000, /* R1700 - OUT3LMIX Input 3 Source */ - [0x06A5] = 0x0080, /* R1701 - OUT3LMIX Input 3 Volume */ - [0x06A6] = 0x0000, /* R1702 - OUT3LMIX Input 4 Source */ - [0x06A7] = 0x0080, /* R1703 - OUT3LMIX Input 4 Volume */ - [0x06A8] = 0x0000, /* R1704 - OUT3RMIX Input 1 Source */ - [0x06A9] = 0x0080, /* R1705 - OUT3RMIX Input 1 Volume */ - [0x06AA] = 0x0000, /* R1706 - OUT3RMIX Input 2 Source */ - [0x06AB] = 0x0080, /* R1707 - OUT3RMIX Input 2 Volume */ - [0x06AC] = 0x0000, /* R1708 - OUT3RMIX Input 3 Source */ - [0x06AD] = 0x0080, /* R1709 - OUT3RMIX Input 3 Volume */ - [0x06AE] = 0x0000, /* R1710 - OUT3RMIX Input 4 Source */ - [0x06AF] = 0x0080, /* R1711 - OUT3RMIX Input 4 Volume */ - [0x06B0] = 0x0000, /* R1712 - OUT4LMIX Input 1 Source */ - [0x06B1] = 0x0080, /* R1713 - OUT4LMIX Input 1 Volume */ - [0x06B2] = 0x0000, /* R1714 - OUT4LMIX Input 2 Source */ - [0x06B3] = 0x0080, /* R1715 - OUT4LMIX Input 2 Volume */ - [0x06B4] = 0x0000, /* R1716 - OUT4LMIX Input 3 Source */ - [0x06B5] = 0x0080, /* R1717 - OUT4LMIX Input 3 Volume */ - [0x06B6] = 0x0000, /* R1718 - OUT4LMIX Input 4 Source */ - [0x06B7] = 0x0080, /* R1719 - OUT4LMIX Input 4 Volume */ - [0x06B8] = 0x0000, /* R1720 - OUT4RMIX Input 1 Source */ - [0x06B9] = 0x0080, /* R1721 - OUT4RMIX Input 1 Volume */ - [0x06BA] = 0x0000, /* R1722 - OUT4RMIX Input 2 Source */ - [0x06BB] = 0x0080, /* R1723 - OUT4RMIX Input 2 Volume */ - [0x06BC] = 0x0000, /* R1724 - OUT4RMIX Input 3 Source */ - [0x06BD] = 0x0080, /* R1725 - OUT4RMIX Input 3 Volume */ - [0x06BE] = 0x0000, /* R1726 - OUT4RMIX Input 4 Source */ - [0x06BF] = 0x0080, /* R1727 - OUT4RMIX Input 4 Volume */ - [0x06C0] = 0x0000, /* R1728 - OUT5LMIX Input 1 Source */ - [0x06C1] = 0x0080, /* R1729 - OUT5LMIX Input 1 Volume */ - [0x06C2] = 0x0000, /* R1730 - OUT5LMIX Input 2 Source */ - [0x06C3] = 0x0080, /* R1731 - OUT5LMIX Input 2 Volume */ - [0x06C4] = 0x0000, /* R1732 - OUT5LMIX Input 3 Source */ - [0x06C5] = 0x0080, /* R1733 - OUT5LMIX Input 3 Volume */ - [0x06C6] = 0x0000, /* R1734 - OUT5LMIX Input 4 Source */ - [0x06C7] = 0x0080, /* R1735 - OUT5LMIX Input 4 Volume */ - [0x06C8] = 0x0000, /* R1736 - OUT5RMIX Input 1 Source */ - [0x06C9] = 0x0080, /* R1737 - OUT5RMIX Input 1 Volume */ - [0x06CA] = 0x0000, /* R1738 - OUT5RMIX Input 2 Source */ - [0x06CB] = 0x0080, /* R1739 - OUT5RMIX Input 2 Volume */ - [0x06CC] = 0x0000, /* R1740 - OUT5RMIX Input 3 Source */ - [0x06CD] = 0x0080, /* R1741 - OUT5RMIX Input 3 Volume */ - [0x06CE] = 0x0000, /* R1742 - OUT5RMIX Input 4 Source */ - [0x06CF] = 0x0080, /* R1743 - OUT5RMIX Input 4 Volume */ - [0x06D0] = 0x0000, /* R1744 - OUT6LMIX Input 1 Source */ - [0x06D1] = 0x0080, /* R1745 - OUT6LMIX Input 1 Volume */ - [0x06D2] = 0x0000, /* R1746 - OUT6LMIX Input 2 Source */ - [0x06D3] = 0x0080, /* R1747 - OUT6LMIX Input 2 Volume */ - [0x06D4] = 0x0000, /* R1748 - OUT6LMIX Input 3 Source */ - [0x06D5] = 0x0080, /* R1749 - OUT6LMIX Input 3 Volume */ - [0x06D6] = 0x0000, /* R1750 - OUT6LMIX Input 4 Source */ - [0x06D7] = 0x0080, /* R1751 - OUT6LMIX Input 4 Volume */ - [0x06D8] = 0x0000, /* R1752 - OUT6RMIX Input 1 Source */ - [0x06D9] = 0x0080, /* R1753 - OUT6RMIX Input 1 Volume */ - [0x06DA] = 0x0000, /* R1754 - OUT6RMIX Input 2 Source */ - [0x06DB] = 0x0080, /* R1755 - OUT6RMIX Input 2 Volume */ - [0x06DC] = 0x0000, /* R1756 - OUT6RMIX Input 3 Source */ - [0x06DD] = 0x0080, /* R1757 - OUT6RMIX Input 3 Volume */ - [0x06DE] = 0x0000, /* R1758 - OUT6RMIX Input 4 Source */ - [0x06DF] = 0x0080, /* R1759 - OUT6RMIX Input 4 Volume */ - [0x0700] = 0x0000, /* R1792 - AIF1TX1MIX Input 1 Source */ - [0x0701] = 0x0080, /* R1793 - AIF1TX1MIX Input 1 Volume */ - [0x0702] = 0x0000, /* R1794 - AIF1TX1MIX Input 2 Source */ - [0x0703] = 0x0080, /* R1795 - AIF1TX1MIX Input 2 Volume */ - [0x0704] = 0x0000, /* R1796 - AIF1TX1MIX Input 3 Source */ - [0x0705] = 0x0080, /* R1797 - AIF1TX1MIX Input 3 Volume */ - [0x0706] = 0x0000, /* R1798 - AIF1TX1MIX Input 4 Source */ - [0x0707] = 0x0080, /* R1799 - AIF1TX1MIX Input 4 Volume */ - [0x0708] = 0x0000, /* R1800 - AIF1TX2MIX Input 1 Source */ - [0x0709] = 0x0080, /* R1801 - AIF1TX2MIX Input 1 Volume */ - [0x070A] = 0x0000, /* R1802 - AIF1TX2MIX Input 2 Source */ - [0x070B] = 0x0080, /* R1803 - AIF1TX2MIX Input 2 Volume */ - [0x070C] = 0x0000, /* R1804 - AIF1TX2MIX Input 3 Source */ - [0x070D] = 0x0080, /* R1805 - AIF1TX2MIX Input 3 Volume */ - [0x070E] = 0x0000, /* R1806 - AIF1TX2MIX Input 4 Source */ - [0x070F] = 0x0080, /* R1807 - AIF1TX2MIX Input 4 Volume */ - [0x0710] = 0x0000, /* R1808 - AIF1TX3MIX Input 1 Source */ - [0x0711] = 0x0080, /* R1809 - AIF1TX3MIX Input 1 Volume */ - [0x0712] = 0x0000, /* R1810 - AIF1TX3MIX Input 2 Source */ - [0x0713] = 0x0080, /* R1811 - AIF1TX3MIX Input 2 Volume */ - [0x0714] = 0x0000, /* R1812 - AIF1TX3MIX Input 3 Source */ - [0x0715] = 0x0080, /* R1813 - AIF1TX3MIX Input 3 Volume */ - [0x0716] = 0x0000, /* R1814 - AIF1TX3MIX Input 4 Source */ - [0x0717] = 0x0080, /* R1815 - AIF1TX3MIX Input 4 Volume */ - [0x0718] = 0x0000, /* R1816 - AIF1TX4MIX Input 1 Source */ - [0x0719] = 0x0080, /* R1817 - AIF1TX4MIX Input 1 Volume */ - [0x071A] = 0x0000, /* R1818 - AIF1TX4MIX Input 2 Source */ - [0x071B] = 0x0080, /* R1819 - AIF1TX4MIX Input 2 Volume */ - [0x071C] = 0x0000, /* R1820 - AIF1TX4MIX Input 3 Source */ - [0x071D] = 0x0080, /* R1821 - AIF1TX4MIX Input 3 Volume */ - [0x071E] = 0x0000, /* R1822 - AIF1TX4MIX Input 4 Source */ - [0x071F] = 0x0080, /* R1823 - AIF1TX4MIX Input 4 Volume */ - [0x0720] = 0x0000, /* R1824 - AIF1TX5MIX Input 1 Source */ - [0x0721] = 0x0080, /* R1825 - AIF1TX5MIX Input 1 Volume */ - [0x0722] = 0x0000, /* R1826 - AIF1TX5MIX Input 2 Source */ - [0x0723] = 0x0080, /* R1827 - AIF1TX5MIX Input 2 Volume */ - [0x0724] = 0x0000, /* R1828 - AIF1TX5MIX Input 3 Source */ - [0x0725] = 0x0080, /* R1829 - AIF1TX5MIX Input 3 Volume */ - [0x0726] = 0x0000, /* R1830 - AIF1TX5MIX Input 4 Source */ - [0x0727] = 0x0080, /* R1831 - AIF1TX5MIX Input 4 Volume */ - [0x0728] = 0x0000, /* R1832 - AIF1TX6MIX Input 1 Source */ - [0x0729] = 0x0080, /* R1833 - AIF1TX6MIX Input 1 Volume */ - [0x072A] = 0x0000, /* R1834 - AIF1TX6MIX Input 2 Source */ - [0x072B] = 0x0080, /* R1835 - AIF1TX6MIX Input 2 Volume */ - [0x072C] = 0x0000, /* R1836 - AIF1TX6MIX Input 3 Source */ - [0x072D] = 0x0080, /* R1837 - AIF1TX6MIX Input 3 Volume */ - [0x072E] = 0x0000, /* R1838 - AIF1TX6MIX Input 4 Source */ - [0x072F] = 0x0080, /* R1839 - AIF1TX6MIX Input 4 Volume */ - [0x0730] = 0x0000, /* R1840 - AIF1TX7MIX Input 1 Source */ - [0x0731] = 0x0080, /* R1841 - AIF1TX7MIX Input 1 Volume */ - [0x0732] = 0x0000, /* R1842 - AIF1TX7MIX Input 2 Source */ - [0x0733] = 0x0080, /* R1843 - AIF1TX7MIX Input 2 Volume */ - [0x0734] = 0x0000, /* R1844 - AIF1TX7MIX Input 3 Source */ - [0x0735] = 0x0080, /* R1845 - AIF1TX7MIX Input 3 Volume */ - [0x0736] = 0x0000, /* R1846 - AIF1TX7MIX Input 4 Source */ - [0x0737] = 0x0080, /* R1847 - AIF1TX7MIX Input 4 Volume */ - [0x0738] = 0x0000, /* R1848 - AIF1TX8MIX Input 1 Source */ - [0x0739] = 0x0080, /* R1849 - AIF1TX8MIX Input 1 Volume */ - [0x073A] = 0x0000, /* R1850 - AIF1TX8MIX Input 2 Source */ - [0x073B] = 0x0080, /* R1851 - AIF1TX8MIX Input 2 Volume */ - [0x073C] = 0x0000, /* R1852 - AIF1TX8MIX Input 3 Source */ - [0x073D] = 0x0080, /* R1853 - AIF1TX8MIX Input 3 Volume */ - [0x073E] = 0x0000, /* R1854 - AIF1TX8MIX Input 4 Source */ - [0x073F] = 0x0080, /* R1855 - AIF1TX8MIX Input 4 Volume */ - [0x0740] = 0x0000, /* R1856 - AIF2TX1MIX Input 1 Source */ - [0x0741] = 0x0080, /* R1857 - AIF2TX1MIX Input 1 Volume */ - [0x0742] = 0x0000, /* R1858 - AIF2TX1MIX Input 2 Source */ - [0x0743] = 0x0080, /* R1859 - AIF2TX1MIX Input 2 Volume */ - [0x0744] = 0x0000, /* R1860 - AIF2TX1MIX Input 3 Source */ - [0x0745] = 0x0080, /* R1861 - AIF2TX1MIX Input 3 Volume */ - [0x0746] = 0x0000, /* R1862 - AIF2TX1MIX Input 4 Source */ - [0x0747] = 0x0080, /* R1863 - AIF2TX1MIX Input 4 Volume */ - [0x0748] = 0x0000, /* R1864 - AIF2TX2MIX Input 1 Source */ - [0x0749] = 0x0080, /* R1865 - AIF2TX2MIX Input 1 Volume */ - [0x074A] = 0x0000, /* R1866 - AIF2TX2MIX Input 2 Source */ - [0x074B] = 0x0080, /* R1867 - AIF2TX2MIX Input 2 Volume */ - [0x074C] = 0x0000, /* R1868 - AIF2TX2MIX Input 3 Source */ - [0x074D] = 0x0080, /* R1869 - AIF2TX2MIX Input 3 Volume */ - [0x074E] = 0x0000, /* R1870 - AIF2TX2MIX Input 4 Source */ - [0x074F] = 0x0080, /* R1871 - AIF2TX2MIX Input 4 Volume */ - [0x0780] = 0x0000, /* R1920 - AIF3TX1MIX Input 1 Source */ - [0x0781] = 0x0080, /* R1921 - AIF3TX1MIX Input 1 Volume */ - [0x0782] = 0x0000, /* R1922 - AIF3TX1MIX Input 2 Source */ - [0x0783] = 0x0080, /* R1923 - AIF3TX1MIX Input 2 Volume */ - [0x0784] = 0x0000, /* R1924 - AIF3TX1MIX Input 3 Source */ - [0x0785] = 0x0080, /* R1925 - AIF3TX1MIX Input 3 Volume */ - [0x0786] = 0x0000, /* R1926 - AIF3TX1MIX Input 4 Source */ - [0x0787] = 0x0080, /* R1927 - AIF3TX1MIX Input 4 Volume */ - [0x0788] = 0x0000, /* R1928 - AIF3TX2MIX Input 1 Source */ - [0x0789] = 0x0080, /* R1929 - AIF3TX2MIX Input 1 Volume */ - [0x078A] = 0x0000, /* R1930 - AIF3TX2MIX Input 2 Source */ - [0x078B] = 0x0080, /* R1931 - AIF3TX2MIX Input 2 Volume */ - [0x078C] = 0x0000, /* R1932 - AIF3TX2MIX Input 3 Source */ - [0x078D] = 0x0080, /* R1933 - AIF3TX2MIX Input 3 Volume */ - [0x078E] = 0x0000, /* R1934 - AIF3TX2MIX Input 4 Source */ - [0x078F] = 0x0080, /* R1935 - AIF3TX2MIX Input 4 Volume */ - [0x0880] = 0x0000, /* R2176 - EQ1MIX Input 1 Source */ - [0x0881] = 0x0080, /* R2177 - EQ1MIX Input 1 Volume */ - [0x0882] = 0x0000, /* R2178 - EQ1MIX Input 2 Source */ - [0x0883] = 0x0080, /* R2179 - EQ1MIX Input 2 Volume */ - [0x0884] = 0x0000, /* R2180 - EQ1MIX Input 3 Source */ - [0x0885] = 0x0080, /* R2181 - EQ1MIX Input 3 Volume */ - [0x0886] = 0x0000, /* R2182 - EQ1MIX Input 4 Source */ - [0x0887] = 0x0080, /* R2183 - EQ1MIX Input 4 Volume */ - [0x0888] = 0x0000, /* R2184 - EQ2MIX Input 1 Source */ - [0x0889] = 0x0080, /* R2185 - EQ2MIX Input 1 Volume */ - [0x088A] = 0x0000, /* R2186 - EQ2MIX Input 2 Source */ - [0x088B] = 0x0080, /* R2187 - EQ2MIX Input 2 Volume */ - [0x088C] = 0x0000, /* R2188 - EQ2MIX Input 3 Source */ - [0x088D] = 0x0080, /* R2189 - EQ2MIX Input 3 Volume */ - [0x088E] = 0x0000, /* R2190 - EQ2MIX Input 4 Source */ - [0x088F] = 0x0080, /* R2191 - EQ2MIX Input 4 Volume */ - [0x0890] = 0x0000, /* R2192 - EQ3MIX Input 1 Source */ - [0x0891] = 0x0080, /* R2193 - EQ3MIX Input 1 Volume */ - [0x0892] = 0x0000, /* R2194 - EQ3MIX Input 2 Source */ - [0x0893] = 0x0080, /* R2195 - EQ3MIX Input 2 Volume */ - [0x0894] = 0x0000, /* R2196 - EQ3MIX Input 3 Source */ - [0x0895] = 0x0080, /* R2197 - EQ3MIX Input 3 Volume */ - [0x0896] = 0x0000, /* R2198 - EQ3MIX Input 4 Source */ - [0x0897] = 0x0080, /* R2199 - EQ3MIX Input 4 Volume */ - [0x0898] = 0x0000, /* R2200 - EQ4MIX Input 1 Source */ - [0x0899] = 0x0080, /* R2201 - EQ4MIX Input 1 Volume */ - [0x089A] = 0x0000, /* R2202 - EQ4MIX Input 2 Source */ - [0x089B] = 0x0080, /* R2203 - EQ4MIX Input 2 Volume */ - [0x089C] = 0x0000, /* R2204 - EQ4MIX Input 3 Source */ - [0x089D] = 0x0080, /* R2205 - EQ4MIX Input 3 Volume */ - [0x089E] = 0x0000, /* R2206 - EQ4MIX Input 4 Source */ - [0x089F] = 0x0080, /* R2207 - EQ4MIX Input 4 Volume */ - [0x08C0] = 0x0000, /* R2240 - DRC1LMIX Input 1 Source */ - [0x08C1] = 0x0080, /* R2241 - DRC1LMIX Input 1 Volume */ - [0x08C2] = 0x0000, /* R2242 - DRC1LMIX Input 2 Source */ - [0x08C3] = 0x0080, /* R2243 - DRC1LMIX Input 2 Volume */ - [0x08C4] = 0x0000, /* R2244 - DRC1LMIX Input 3 Source */ - [0x08C5] = 0x0080, /* R2245 - DRC1LMIX Input 3 Volume */ - [0x08C6] = 0x0000, /* R2246 - DRC1LMIX Input 4 Source */ - [0x08C7] = 0x0080, /* R2247 - DRC1LMIX Input 4 Volume */ - [0x08C8] = 0x0000, /* R2248 - DRC1RMIX Input 1 Source */ - [0x08C9] = 0x0080, /* R2249 - DRC1RMIX Input 1 Volume */ - [0x08CA] = 0x0000, /* R2250 - DRC1RMIX Input 2 Source */ - [0x08CB] = 0x0080, /* R2251 - DRC1RMIX Input 2 Volume */ - [0x08CC] = 0x0000, /* R2252 - DRC1RMIX Input 3 Source */ - [0x08CD] = 0x0080, /* R2253 - DRC1RMIX Input 3 Volume */ - [0x08CE] = 0x0000, /* R2254 - DRC1RMIX Input 4 Source */ - [0x08CF] = 0x0080, /* R2255 - DRC1RMIX Input 4 Volume */ - [0x0900] = 0x0000, /* R2304 - HPLP1MIX Input 1 Source */ - [0x0901] = 0x0080, /* R2305 - HPLP1MIX Input 1 Volume */ - [0x0902] = 0x0000, /* R2306 - HPLP1MIX Input 2 Source */ - [0x0903] = 0x0080, /* R2307 - HPLP1MIX Input 2 Volume */ - [0x0904] = 0x0000, /* R2308 - HPLP1MIX Input 3 Source */ - [0x0905] = 0x0080, /* R2309 - HPLP1MIX Input 3 Volume */ - [0x0906] = 0x0000, /* R2310 - HPLP1MIX Input 4 Source */ - [0x0907] = 0x0080, /* R2311 - HPLP1MIX Input 4 Volume */ - [0x0908] = 0x0000, /* R2312 - HPLP2MIX Input 1 Source */ - [0x0909] = 0x0080, /* R2313 - HPLP2MIX Input 1 Volume */ - [0x090A] = 0x0000, /* R2314 - HPLP2MIX Input 2 Source */ - [0x090B] = 0x0080, /* R2315 - HPLP2MIX Input 2 Volume */ - [0x090C] = 0x0000, /* R2316 - HPLP2MIX Input 3 Source */ - [0x090D] = 0x0080, /* R2317 - HPLP2MIX Input 3 Volume */ - [0x090E] = 0x0000, /* R2318 - HPLP2MIX Input 4 Source */ - [0x090F] = 0x0080, /* R2319 - HPLP2MIX Input 4 Volume */ - [0x0910] = 0x0000, /* R2320 - HPLP3MIX Input 1 Source */ - [0x0911] = 0x0080, /* R2321 - HPLP3MIX Input 1 Volume */ - [0x0912] = 0x0000, /* R2322 - HPLP3MIX Input 2 Source */ - [0x0913] = 0x0080, /* R2323 - HPLP3MIX Input 2 Volume */ - [0x0914] = 0x0000, /* R2324 - HPLP3MIX Input 3 Source */ - [0x0915] = 0x0080, /* R2325 - HPLP3MIX Input 3 Volume */ - [0x0916] = 0x0000, /* R2326 - HPLP3MIX Input 4 Source */ - [0x0917] = 0x0080, /* R2327 - HPLP3MIX Input 4 Volume */ - [0x0918] = 0x0000, /* R2328 - HPLP4MIX Input 1 Source */ - [0x0919] = 0x0080, /* R2329 - HPLP4MIX Input 1 Volume */ - [0x091A] = 0x0000, /* R2330 - HPLP4MIX Input 2 Source */ - [0x091B] = 0x0080, /* R2331 - HPLP4MIX Input 2 Volume */ - [0x091C] = 0x0000, /* R2332 - HPLP4MIX Input 3 Source */ - [0x091D] = 0x0080, /* R2333 - HPLP4MIX Input 3 Volume */ - [0x091E] = 0x0000, /* R2334 - HPLP4MIX Input 4 Source */ - [0x091F] = 0x0080, /* R2335 - HPLP4MIX Input 4 Volume */ - [0x0940] = 0x0000, /* R2368 - DSP1LMIX Input 1 Source */ - [0x0941] = 0x0080, /* R2369 - DSP1LMIX Input 1 Volume */ - [0x0942] = 0x0000, /* R2370 - DSP1LMIX Input 2 Source */ - [0x0943] = 0x0080, /* R2371 - DSP1LMIX Input 2 Volume */ - [0x0944] = 0x0000, /* R2372 - DSP1LMIX Input 3 Source */ - [0x0945] = 0x0080, /* R2373 - DSP1LMIX Input 3 Volume */ - [0x0946] = 0x0000, /* R2374 - DSP1LMIX Input 4 Source */ - [0x0947] = 0x0080, /* R2375 - DSP1LMIX Input 4 Volume */ - [0x0948] = 0x0000, /* R2376 - DSP1RMIX Input 1 Source */ - [0x0949] = 0x0080, /* R2377 - DSP1RMIX Input 1 Volume */ - [0x094A] = 0x0000, /* R2378 - DSP1RMIX Input 2 Source */ - [0x094B] = 0x0080, /* R2379 - DSP1RMIX Input 2 Volume */ - [0x094C] = 0x0000, /* R2380 - DSP1RMIX Input 3 Source */ - [0x094D] = 0x0080, /* R2381 - DSP1RMIX Input 3 Volume */ - [0x094E] = 0x0000, /* R2382 - DSP1RMIX Input 4 Source */ - [0x094F] = 0x0080, /* R2383 - DSP1RMIX Input 4 Volume */ - [0x0950] = 0x0000, /* R2384 - DSP1AUX1MIX Input 1 Source */ - [0x0958] = 0x0000, /* R2392 - DSP1AUX2MIX Input 1 Source */ - [0x0960] = 0x0000, /* R2400 - DSP1AUX3MIX Input 1 Source */ - [0x0968] = 0x0000, /* R2408 - DSP1AUX4MIX Input 1 Source */ - [0x0970] = 0x0000, /* R2416 - DSP1AUX5MIX Input 1 Source */ - [0x0978] = 0x0000, /* R2424 - DSP1AUX6MIX Input 1 Source */ - [0x0980] = 0x0000, /* R2432 - DSP2LMIX Input 1 Source */ - [0x0981] = 0x0080, /* R2433 - DSP2LMIX Input 1 Volume */ - [0x0982] = 0x0000, /* R2434 - DSP2LMIX Input 2 Source */ - [0x0983] = 0x0080, /* R2435 - DSP2LMIX Input 2 Volume */ - [0x0984] = 0x0000, /* R2436 - DSP2LMIX Input 3 Source */ - [0x0985] = 0x0080, /* R2437 - DSP2LMIX Input 3 Volume */ - [0x0986] = 0x0000, /* R2438 - DSP2LMIX Input 4 Source */ - [0x0987] = 0x0080, /* R2439 - DSP2LMIX Input 4 Volume */ - [0x0988] = 0x0000, /* R2440 - DSP2RMIX Input 1 Source */ - [0x0989] = 0x0080, /* R2441 - DSP2RMIX Input 1 Volume */ - [0x098A] = 0x0000, /* R2442 - DSP2RMIX Input 2 Source */ - [0x098B] = 0x0080, /* R2443 - DSP2RMIX Input 2 Volume */ - [0x098C] = 0x0000, /* R2444 - DSP2RMIX Input 3 Source */ - [0x098D] = 0x0080, /* R2445 - DSP2RMIX Input 3 Volume */ - [0x098E] = 0x0000, /* R2446 - DSP2RMIX Input 4 Source */ - [0x098F] = 0x0080, /* R2447 - DSP2RMIX Input 4 Volume */ - [0x0990] = 0x0000, /* R2448 - DSP2AUX1MIX Input 1 Source */ - [0x0998] = 0x0000, /* R2456 - DSP2AUX2MIX Input 1 Source */ - [0x09A0] = 0x0000, /* R2464 - DSP2AUX3MIX Input 1 Source */ - [0x09A8] = 0x0000, /* R2472 - DSP2AUX4MIX Input 1 Source */ - [0x09B0] = 0x0000, /* R2480 - DSP2AUX5MIX Input 1 Source */ - [0x09B8] = 0x0000, /* R2488 - DSP2AUX6MIX Input 1 Source */ - [0x09C0] = 0x0000, /* R2496 - DSP3LMIX Input 1 Source */ - [0x09C1] = 0x0080, /* R2497 - DSP3LMIX Input 1 Volume */ - [0x09C2] = 0x0000, /* R2498 - DSP3LMIX Input 2 Source */ - [0x09C3] = 0x0080, /* R2499 - DSP3LMIX Input 2 Volume */ - [0x09C4] = 0x0000, /* R2500 - DSP3LMIX Input 3 Source */ - [0x09C5] = 0x0080, /* R2501 - DSP3LMIX Input 3 Volume */ - [0x09C6] = 0x0000, /* R2502 - DSP3LMIX Input 4 Source */ - [0x09C7] = 0x0080, /* R2503 - DSP3LMIX Input 4 Volume */ - [0x09C8] = 0x0000, /* R2504 - DSP3RMIX Input 1 Source */ - [0x09C9] = 0x0080, /* R2505 - DSP3RMIX Input 1 Volume */ - [0x09CA] = 0x0000, /* R2506 - DSP3RMIX Input 2 Source */ - [0x09CB] = 0x0080, /* R2507 - DSP3RMIX Input 2 Volume */ - [0x09CC] = 0x0000, /* R2508 - DSP3RMIX Input 3 Source */ - [0x09CD] = 0x0080, /* R2509 - DSP3RMIX Input 3 Volume */ - [0x09CE] = 0x0000, /* R2510 - DSP3RMIX Input 4 Source */ - [0x09CF] = 0x0080, /* R2511 - DSP3RMIX Input 4 Volume */ - [0x09D0] = 0x0000, /* R2512 - DSP3AUX1MIX Input 1 Source */ - [0x09D8] = 0x0000, /* R2520 - DSP3AUX2MIX Input 1 Source */ - [0x09E0] = 0x0000, /* R2528 - DSP3AUX3MIX Input 1 Source */ - [0x09E8] = 0x0000, /* R2536 - DSP3AUX4MIX Input 1 Source */ - [0x09F0] = 0x0000, /* R2544 - DSP3AUX5MIX Input 1 Source */ - [0x09F8] = 0x0000, /* R2552 - DSP3AUX6MIX Input 1 Source */ - [0x0A80] = 0x0000, /* R2688 - ASRC1LMIX Input 1 Source */ - [0x0A88] = 0x0000, /* R2696 - ASRC1RMIX Input 1 Source */ - [0x0A90] = 0x0000, /* R2704 - ASRC2LMIX Input 1 Source */ - [0x0A98] = 0x0000, /* R2712 - ASRC2RMIX Input 1 Source */ - [0x0B00] = 0x0000, /* R2816 - ISRC1DEC1MIX Input 1 Source */ - [0x0B08] = 0x0000, /* R2824 - ISRC1DEC2MIX Input 1 Source */ - [0x0B10] = 0x0000, /* R2832 - ISRC1DEC3MIX Input 1 Source */ - [0x0B18] = 0x0000, /* R2840 - ISRC1DEC4MIX Input 1 Source */ - [0x0B20] = 0x0000, /* R2848 - ISRC1INT1MIX Input 1 Source */ - [0x0B28] = 0x0000, /* R2856 - ISRC1INT2MIX Input 1 Source */ - [0x0B30] = 0x0000, /* R2864 - ISRC1INT3MIX Input 1 Source */ - [0x0B38] = 0x0000, /* R2872 - ISRC1INT4MIX Input 1 Source */ - [0x0B40] = 0x0000, /* R2880 - ISRC2DEC1MIX Input 1 Source */ - [0x0B48] = 0x0000, /* R2888 - ISRC2DEC2MIX Input 1 Source */ - [0x0B50] = 0x0000, /* R2896 - ISRC2DEC3MIX Input 1 Source */ - [0x0B58] = 0x0000, /* R2904 - ISRC2DEC4MIX Input 1 Source */ - [0x0B60] = 0x0000, /* R2912 - ISRC2INT1MIX Input 1 Source */ - [0x0B68] = 0x0000, /* R2920 - ISRC2INT2MIX Input 1 Source */ - [0x0B70] = 0x0000, /* R2928 - ISRC2INT3MIX Input 1 Source */ - [0x0B78] = 0x0000, /* R2936 - ISRC2INT4MIX Input 1 Source */ - [0x0C00] = 0xA001, /* R3072 - GPIO CTRL 1 */ - [0x0C01] = 0xA001, /* R3073 - GPIO CTRL 2 */ - [0x0C02] = 0xA001, /* R3074 - GPIO CTRL 3 */ - [0x0C03] = 0xA001, /* R3075 - GPIO CTRL 4 */ - [0x0C04] = 0xA001, /* R3076 - GPIO CTRL 5 */ - [0x0C05] = 0xA001, /* R3077 - GPIO CTRL 6 */ - [0x0C23] = 0x4003, /* R3107 - Misc Pad Ctrl 1 */ - [0x0C24] = 0x0000, /* R3108 - Misc Pad Ctrl 2 */ - [0x0C25] = 0x0000, /* R3109 - Misc Pad Ctrl 3 */ - [0x0C26] = 0x0000, /* R3110 - Misc Pad Ctrl 4 */ - [0x0C27] = 0x0000, /* R3111 - Misc Pad Ctrl 5 */ - [0x0C28] = 0x0000, /* R3112 - Misc GPIO 1 */ - [0x0D00] = 0x0000, /* R3328 - Interrupt Status 1 */ - [0x0D01] = 0x0000, /* R3329 - Interrupt Status 2 */ - [0x0D02] = 0x0000, /* R3330 - Interrupt Status 3 */ - [0x0D03] = 0x0000, /* R3331 - Interrupt Status 4 */ - [0x0D04] = 0x0000, /* R3332 - Interrupt Raw Status 2 */ - [0x0D05] = 0x0000, /* R3333 - Interrupt Raw Status 3 */ - [0x0D06] = 0x0000, /* R3334 - Interrupt Raw Status 4 */ - [0x0D07] = 0xFFFF, /* R3335 - Interrupt Status 1 Mask */ - [0x0D08] = 0xFFFF, /* R3336 - Interrupt Status 2 Mask */ - [0x0D09] = 0xFFFF, /* R3337 - Interrupt Status 3 Mask */ - [0x0D0A] = 0xFFFF, /* R3338 - Interrupt Status 4 Mask */ - [0x0D1F] = 0x0000, /* R3359 - Interrupt Control */ - [0x0D20] = 0xFFFF, /* R3360 - IRQ Debounce 1 */ - [0x0D21] = 0xFFFF, /* R3361 - IRQ Debounce 2 */ - [0x0E00] = 0x0000, /* R3584 - FX_Ctrl */ - [0x0E10] = 0x6318, /* R3600 - EQ1_1 */ - [0x0E11] = 0x6300, /* R3601 - EQ1_2 */ - [0x0E12] = 0x0FC8, /* R3602 - EQ1_3 */ - [0x0E13] = 0x03FE, /* R3603 - EQ1_4 */ - [0x0E14] = 0x00E0, /* R3604 - EQ1_5 */ - [0x0E15] = 0x1EC4, /* R3605 - EQ1_6 */ - [0x0E16] = 0xF136, /* R3606 - EQ1_7 */ - [0x0E17] = 0x0409, /* R3607 - EQ1_8 */ - [0x0E18] = 0x04CC, /* R3608 - EQ1_9 */ - [0x0E19] = 0x1C9B, /* R3609 - EQ1_10 */ - [0x0E1A] = 0xF337, /* R3610 - EQ1_11 */ - [0x0E1B] = 0x040B, /* R3611 - EQ1_12 */ - [0x0E1C] = 0x0CBB, /* R3612 - EQ1_13 */ - [0x0E1D] = 0x16F8, /* R3613 - EQ1_14 */ - [0x0E1E] = 0xF7D9, /* R3614 - EQ1_15 */ - [0x0E1F] = 0x040A, /* R3615 - EQ1_16 */ - [0x0E20] = 0x1F14, /* R3616 - EQ1_17 */ - [0x0E21] = 0x058C, /* R3617 - EQ1_18 */ - [0x0E22] = 0x0563, /* R3618 - EQ1_19 */ - [0x0E23] = 0x4000, /* R3619 - EQ1_20 */ - [0x0E26] = 0x6318, /* R3622 - EQ2_1 */ - [0x0E27] = 0x6300, /* R3623 - EQ2_2 */ - [0x0E28] = 0x0FC8, /* R3624 - EQ2_3 */ - [0x0E29] = 0x03FE, /* R3625 - EQ2_4 */ - [0x0E2A] = 0x00E0, /* R3626 - EQ2_5 */ - [0x0E2B] = 0x1EC4, /* R3627 - EQ2_6 */ - [0x0E2C] = 0xF136, /* R3628 - EQ2_7 */ - [0x0E2D] = 0x0409, /* R3629 - EQ2_8 */ - [0x0E2E] = 0x04CC, /* R3630 - EQ2_9 */ - [0x0E2F] = 0x1C9B, /* R3631 - EQ2_10 */ - [0x0E30] = 0xF337, /* R3632 - EQ2_11 */ - [0x0E31] = 0x040B, /* R3633 - EQ2_12 */ - [0x0E32] = 0x0CBB, /* R3634 - EQ2_13 */ - [0x0E33] = 0x16F8, /* R3635 - EQ2_14 */ - [0x0E34] = 0xF7D9, /* R3636 - EQ2_15 */ - [0x0E35] = 0x040A, /* R3637 - EQ2_16 */ - [0x0E36] = 0x1F14, /* R3638 - EQ2_17 */ - [0x0E37] = 0x058C, /* R3639 - EQ2_18 */ - [0x0E38] = 0x0563, /* R3640 - EQ2_19 */ - [0x0E39] = 0x4000, /* R3641 - EQ2_20 */ - [0x0E3C] = 0x6318, /* R3644 - EQ3_1 */ - [0x0E3D] = 0x6300, /* R3645 - EQ3_2 */ - [0x0E3E] = 0x0FC8, /* R3646 - EQ3_3 */ - [0x0E3F] = 0x03FE, /* R3647 - EQ3_4 */ - [0x0E40] = 0x00E0, /* R3648 - EQ3_5 */ - [0x0E41] = 0x1EC4, /* R3649 - EQ3_6 */ - [0x0E42] = 0xF136, /* R3650 - EQ3_7 */ - [0x0E43] = 0x0409, /* R3651 - EQ3_8 */ - [0x0E44] = 0x04CC, /* R3652 - EQ3_9 */ - [0x0E45] = 0x1C9B, /* R3653 - EQ3_10 */ - [0x0E46] = 0xF337, /* R3654 - EQ3_11 */ - [0x0E47] = 0x040B, /* R3655 - EQ3_12 */ - [0x0E48] = 0x0CBB, /* R3656 - EQ3_13 */ - [0x0E49] = 0x16F8, /* R3657 - EQ3_14 */ - [0x0E4A] = 0xF7D9, /* R3658 - EQ3_15 */ - [0x0E4B] = 0x040A, /* R3659 - EQ3_16 */ - [0x0E4C] = 0x1F14, /* R3660 - EQ3_17 */ - [0x0E4D] = 0x058C, /* R3661 - EQ3_18 */ - [0x0E4E] = 0x0563, /* R3662 - EQ3_19 */ - [0x0E4F] = 0x4000, /* R3663 - EQ3_20 */ - [0x0E52] = 0x6318, /* R3666 - EQ4_1 */ - [0x0E53] = 0x6300, /* R3667 - EQ4_2 */ - [0x0E54] = 0x0FC8, /* R3668 - EQ4_3 */ - [0x0E55] = 0x03FE, /* R3669 - EQ4_4 */ - [0x0E56] = 0x00E0, /* R3670 - EQ4_5 */ - [0x0E57] = 0x1EC4, /* R3671 - EQ4_6 */ - [0x0E58] = 0xF136, /* R3672 - EQ4_7 */ - [0x0E59] = 0x0409, /* R3673 - EQ4_8 */ - [0x0E5A] = 0x04CC, /* R3674 - EQ4_9 */ - [0x0E5B] = 0x1C9B, /* R3675 - EQ4_10 */ - [0x0E5C] = 0xF337, /* R3676 - EQ4_11 */ - [0x0E5D] = 0x040B, /* R3677 - EQ4_12 */ - [0x0E5E] = 0x0CBB, /* R3678 - EQ4_13 */ - [0x0E5F] = 0x16F8, /* R3679 - EQ4_14 */ - [0x0E60] = 0xF7D9, /* R3680 - EQ4_15 */ - [0x0E61] = 0x040A, /* R3681 - EQ4_16 */ - [0x0E62] = 0x1F14, /* R3682 - EQ4_17 */ - [0x0E63] = 0x058C, /* R3683 - EQ4_18 */ - [0x0E64] = 0x0563, /* R3684 - EQ4_19 */ - [0x0E65] = 0x4000, /* R3685 - EQ4_20 */ - [0x0E80] = 0x0018, /* R3712 - DRC1 ctrl1 */ - [0x0E81] = 0x0933, /* R3713 - DRC1 ctrl2 */ - [0x0E82] = 0x0018, /* R3714 - DRC1 ctrl3 */ - [0x0E83] = 0x0000, /* R3715 - DRC1 ctrl4 */ - [0x0E84] = 0x0000, /* R3716 - DRC1 ctrl5 */ - [0x0EC0] = 0x0000, /* R3776 - HPLPF1_1 */ - [0x0EC1] = 0x0000, /* R3777 - HPLPF1_2 */ - [0x0EC4] = 0x0000, /* R3780 - HPLPF2_1 */ - [0x0EC5] = 0x0000, /* R3781 - HPLPF2_2 */ - [0x0EC8] = 0x0000, /* R3784 - HPLPF3_1 */ - [0x0EC9] = 0x0000, /* R3785 - HPLPF3_2 */ - [0x0ECC] = 0x0000, /* R3788 - HPLPF4_1 */ - [0x0ECD] = 0x0000, /* R3789 - HPLPF4_2 */ - [0x4000] = 0x0000, /* R16384 - DSP1 DM 0 */ - [0x4001] = 0x0000, /* R16385 - DSP1 DM 1 */ - [0x4002] = 0x0000, /* R16386 - DSP1 DM 2 */ - [0x4003] = 0x0000, /* R16387 - DSP1 DM 3 */ - [0x41FC] = 0x0000, /* R16892 - DSP1 DM 508 */ - [0x41FD] = 0x0000, /* R16893 - DSP1 DM 509 */ - [0x41FE] = 0x0000, /* R16894 - DSP1 DM 510 */ - [0x41FF] = 0x0000, /* R16895 - DSP1 DM 511 */ - [0x4800] = 0x0000, /* R18432 - DSP1 PM 0 */ - [0x4801] = 0x0000, /* R18433 - DSP1 PM 1 */ - [0x4802] = 0x0000, /* R18434 - DSP1 PM 2 */ - [0x4803] = 0x0000, /* R18435 - DSP1 PM 3 */ - [0x4804] = 0x0000, /* R18436 - DSP1 PM 4 */ - [0x4805] = 0x0000, /* R18437 - DSP1 PM 5 */ - [0x4DFA] = 0x0000, /* R19962 - DSP1 PM 1530 */ - [0x4DFB] = 0x0000, /* R19963 - DSP1 PM 1531 */ - [0x4DFC] = 0x0000, /* R19964 - DSP1 PM 1532 */ - [0x4DFD] = 0x0000, /* R19965 - DSP1 PM 1533 */ - [0x4DFE] = 0x0000, /* R19966 - DSP1 PM 1534 */ - [0x4DFF] = 0x0000, /* R19967 - DSP1 PM 1535 */ - [0x5000] = 0x0000, /* R20480 - DSP1 ZM 0 */ - [0x5001] = 0x0000, /* R20481 - DSP1 ZM 1 */ - [0x5002] = 0x0000, /* R20482 - DSP1 ZM 2 */ - [0x5003] = 0x0000, /* R20483 - DSP1 ZM 3 */ - [0x57FC] = 0x0000, /* R22524 - DSP1 ZM 2044 */ - [0x57FD] = 0x0000, /* R22525 - DSP1 ZM 2045 */ - [0x57FE] = 0x0000, /* R22526 - DSP1 ZM 2046 */ - [0x57FF] = 0x0000, /* R22527 - DSP1 ZM 2047 */ - [0x6000] = 0x0000, /* R24576 - DSP2 DM 0 */ - [0x6001] = 0x0000, /* R24577 - DSP2 DM 1 */ - [0x6002] = 0x0000, /* R24578 - DSP2 DM 2 */ - [0x6003] = 0x0000, /* R24579 - DSP2 DM 3 */ - [0x61FC] = 0x0000, /* R25084 - DSP2 DM 508 */ - [0x61FD] = 0x0000, /* R25085 - DSP2 DM 509 */ - [0x61FE] = 0x0000, /* R25086 - DSP2 DM 510 */ - [0x61FF] = 0x0000, /* R25087 - DSP2 DM 511 */ - [0x6800] = 0x0000, /* R26624 - DSP2 PM 0 */ - [0x6801] = 0x0000, /* R26625 - DSP2 PM 1 */ - [0x6802] = 0x0000, /* R26626 - DSP2 PM 2 */ - [0x6803] = 0x0000, /* R26627 - DSP2 PM 3 */ - [0x6804] = 0x0000, /* R26628 - DSP2 PM 4 */ - [0x6805] = 0x0000, /* R26629 - DSP2 PM 5 */ - [0x6DFA] = 0x0000, /* R28154 - DSP2 PM 1530 */ - [0x6DFB] = 0x0000, /* R28155 - DSP2 PM 1531 */ - [0x6DFC] = 0x0000, /* R28156 - DSP2 PM 1532 */ - [0x6DFD] = 0x0000, /* R28157 - DSP2 PM 1533 */ - [0x6DFE] = 0x0000, /* R28158 - DSP2 PM 1534 */ - [0x6DFF] = 0x0000, /* R28159 - DSP2 PM 1535 */ - [0x7000] = 0x0000, /* R28672 - DSP2 ZM 0 */ - [0x7001] = 0x0000, /* R28673 - DSP2 ZM 1 */ - [0x7002] = 0x0000, /* R28674 - DSP2 ZM 2 */ - [0x7003] = 0x0000, /* R28675 - DSP2 ZM 3 */ - [0x77FC] = 0x0000, /* R30716 - DSP2 ZM 2044 */ - [0x77FD] = 0x0000, /* R30717 - DSP2 ZM 2045 */ - [0x77FE] = 0x0000, /* R30718 - DSP2 ZM 2046 */ - [0x77FF] = 0x0000, /* R30719 - DSP2 ZM 2047 */ - [0x8000] = 0x0000, /* R32768 - DSP3 DM 0 */ - [0x8001] = 0x0000, /* R32769 - DSP3 DM 1 */ - [0x8002] = 0x0000, /* R32770 - DSP3 DM 2 */ - [0x8003] = 0x0000, /* R32771 - DSP3 DM 3 */ - [0x81FC] = 0x0000, /* R33276 - DSP3 DM 508 */ - [0x81FD] = 0x0000, /* R33277 - DSP3 DM 509 */ - [0x81FE] = 0x0000, /* R33278 - DSP3 DM 510 */ - [0x81FF] = 0x0000, /* R33279 - DSP3 DM 511 */ - [0x8800] = 0x0000, /* R34816 - DSP3 PM 0 */ - [0x8801] = 0x0000, /* R34817 - DSP3 PM 1 */ - [0x8802] = 0x0000, /* R34818 - DSP3 PM 2 */ - [0x8803] = 0x0000, /* R34819 - DSP3 PM 3 */ - [0x8804] = 0x0000, /* R34820 - DSP3 PM 4 */ - [0x8805] = 0x0000, /* R34821 - DSP3 PM 5 */ - [0x8DFA] = 0x0000, /* R36346 - DSP3 PM 1530 */ - [0x8DFB] = 0x0000, /* R36347 - DSP3 PM 1531 */ - [0x8DFC] = 0x0000, /* R36348 - DSP3 PM 1532 */ - [0x8DFD] = 0x0000, /* R36349 - DSP3 PM 1533 */ - [0x8DFE] = 0x0000, /* R36350 - DSP3 PM 1534 */ - [0x8DFF] = 0x0000, /* R36351 - DSP3 PM 1535 */ - [0x9000] = 0x0000, /* R36864 - DSP3 ZM 0 */ - [0x9001] = 0x0000, /* R36865 - DSP3 ZM 1 */ - [0x9002] = 0x0000, /* R36866 - DSP3 ZM 2 */ - [0x9003] = 0x0000, /* R36867 - DSP3 ZM 3 */ - [0x97FC] = 0x0000, /* R38908 - DSP3 ZM 2044 */ - [0x97FD] = 0x0000, /* R38909 - DSP3 ZM 2045 */ - [0x97FE] = 0x0000, /* R38910 - DSP3 ZM 2046 */ - [0x97FF] = 0x0000 /* R38911 - DSP3 ZM 2047 */ +struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { + { 0x0000, 0x0000 }, /* R0 - software reset */ + { 0x0001, 0x0000 }, /* R1 - Device Revision */ + { 0x0010, 0x0801 }, /* R16 - Ctrl IF 1 */ + { 0x0020, 0x0000 }, /* R32 - Tone Generator 1 */ + { 0x0030, 0x0000 }, /* R48 - PWM Drive 1 */ + { 0x0031, 0x0100 }, /* R49 - PWM Drive 2 */ + { 0x0032, 0x0100 }, /* R50 - PWM Drive 3 */ + { 0x0100, 0x0002 }, /* R256 - Clocking 1 */ + { 0x0101, 0x0000 }, /* R257 - Clocking 3 */ + { 0x0102, 0x0011 }, /* R258 - Clocking 4 */ + { 0x0103, 0x0011 }, /* R259 - Clocking 5 */ + { 0x0104, 0x0011 }, /* R260 - Clocking 6 */ + { 0x0107, 0x0000 }, /* R263 - Clocking 7 */ + { 0x0108, 0x0000 }, /* R264 - Clocking 8 */ + { 0x0120, 0x0000 }, /* R288 - ASRC_ENABLE */ + { 0x0121, 0x0000 }, /* R289 - ASRC_STATUS */ + { 0x0122, 0x0000 }, /* R290 - ASRC_RATE1 */ + { 0x0141, 0x8000 }, /* R321 - ISRC 1 CTRL 1 */ + { 0x0142, 0x0000 }, /* R322 - ISRC 1 CTRL 2 */ + { 0x0143, 0x8000 }, /* R323 - ISRC 2 CTRL1 */ + { 0x0144, 0x0000 }, /* R324 - ISRC 2 CTRL 2 */ + { 0x0182, 0x0000 }, /* R386 - FLL1 Control 1 */ + { 0x0183, 0x0000 }, /* R387 - FLL1 Control 2 */ + { 0x0184, 0x0000 }, /* R388 - FLL1 Control 3 */ + { 0x0186, 0x0177 }, /* R390 - FLL1 Control 5 */ + { 0x0187, 0x0001 }, /* R391 - FLL1 Control 6 */ + { 0x0188, 0x0000 }, /* R392 - FLL1 EFS 1 */ + { 0x01A2, 0x0000 }, /* R418 - FLL2 Control 1 */ + { 0x01A3, 0x0000 }, /* R419 - FLL2 Control 2 */ + { 0x01A4, 0x0000 }, /* R420 - FLL2 Control 3 */ + { 0x01A6, 0x0177 }, /* R422 - FLL2 Control 5 */ + { 0x01A7, 0x0001 }, /* R423 - FLL2 Control 6 */ + { 0x01A8, 0x0000 }, /* R424 - FLL2 EFS 1 */ + { 0x0200, 0x0020 }, /* R512 - Mic Charge Pump 1 */ + { 0x0201, 0xB084 }, /* R513 - Mic Charge Pump 2 */ + { 0x0202, 0xBBDE }, /* R514 - HP Charge Pump 1 */ + { 0x0211, 0x20D4 }, /* R529 - LDO1 Control */ + { 0x0215, 0x0062 }, /* R533 - Mic Bias Ctrl 1 */ + { 0x0216, 0x0062 }, /* R534 - Mic Bias Ctrl 2 */ + { 0x0217, 0x0062 }, /* R535 - Mic Bias Ctrl 3 */ + { 0x0280, 0x0004 }, /* R640 - Accessory Detect Mode 1 */ + { 0x0288, 0x0020 }, /* R648 - Headphone Detect 1 */ + { 0x0289, 0x0000 }, /* R649 - Headphone Detect 2 */ + { 0x0290, 0x1100 }, /* R656 - Mic Detect 1 */ + { 0x0291, 0x009F }, /* R657 - Mic Detect 2 */ + { 0x0292, 0x0000 }, /* R658 - Mic Detect 3 */ + { 0x0301, 0x0000 }, /* R769 - Input Enables */ + { 0x0302, 0x0000 }, /* R770 - Input Enables Status */ + { 0x0310, 0x2280 }, /* R784 - Status */ + { 0x0311, 0x0080 }, /* R785 - IN1R Control */ + { 0x0312, 0x2280 }, /* R786 - IN2L Control */ + { 0x0313, 0x0080 }, /* R787 - IN2R Control */ + { 0x0314, 0x2280 }, /* R788 - IN3L Control */ + { 0x0315, 0x0080 }, /* R789 - IN3R Control */ + { 0x0316, 0x2280 }, /* R790 - IN4L Control */ + { 0x0317, 0x0080 }, /* R791 - IN4R Control */ + { 0x0318, 0x0000 }, /* R792 - RXANC_SRC */ + { 0x0319, 0x0022 }, /* R793 - Input Volume Ramp */ + { 0x0320, 0x0180 }, /* R800 - ADC Digital Volume 1L */ + { 0x0321, 0x0180 }, /* R801 - ADC Digital Volume 1R */ + { 0x0322, 0x0180 }, /* R802 - ADC Digital Volume 2L */ + { 0x0323, 0x0180 }, /* R803 - ADC Digital Volume 2R */ + { 0x0324, 0x0180 }, /* R804 - ADC Digital Volume 3L */ + { 0x0325, 0x0180 }, /* R805 - ADC Digital Volume 3R */ + { 0x0326, 0x0180 }, /* R806 - ADC Digital Volume 4L */ + { 0x0327, 0x0180 }, /* R807 - ADC Digital Volume 4R */ + { 0x0401, 0x0000 }, /* R1025 - Output Enables 2 */ + { 0x0402, 0x0000 }, /* R1026 - Output Status 1 */ + { 0x0403, 0x0000 }, /* R1027 - Output Status 2 */ + { 0x0408, 0x0000 }, /* R1032 - Channel Enables 1 */ + { 0x0410, 0x0080 }, /* R1040 - Out Volume 1L */ + { 0x0411, 0x0080 }, /* R1041 - Out Volume 1R */ + { 0x0412, 0x0080 }, /* R1042 - DAC Volume Limit 1L */ + { 0x0413, 0x0080 }, /* R1043 - DAC Volume Limit 1R */ + { 0x0414, 0x0080 }, /* R1044 - Out Volume 2L */ + { 0x0415, 0x0080 }, /* R1045 - Out Volume 2R */ + { 0x0416, 0x0080 }, /* R1046 - DAC Volume Limit 2L */ + { 0x0417, 0x0080 }, /* R1047 - DAC Volume Limit 2R */ + { 0x0418, 0x0080 }, /* R1048 - Out Volume 3L */ + { 0x0419, 0x0080 }, /* R1049 - Out Volume 3R */ + { 0x041A, 0x0080 }, /* R1050 - DAC Volume Limit 3L */ + { 0x041B, 0x0080 }, /* R1051 - DAC Volume Limit 3R */ + { 0x041C, 0x0080 }, /* R1052 - Out Volume 4L */ + { 0x041D, 0x0080 }, /* R1053 - Out Volume 4R */ + { 0x041E, 0x0080 }, /* R1054 - DAC Volume Limit 5L */ + { 0x041F, 0x0080 }, /* R1055 - DAC Volume Limit 5R */ + { 0x0420, 0x0080 }, /* R1056 - DAC Volume Limit 6L */ + { 0x0421, 0x0080 }, /* R1057 - DAC Volume Limit 6R */ + { 0x0440, 0x0000 }, /* R1088 - DAC AEC Control 1 */ + { 0x0441, 0x0022 }, /* R1089 - Output Volume Ramp */ + { 0x0480, 0x0180 }, /* R1152 - DAC Digital Volume 1L */ + { 0x0481, 0x0180 }, /* R1153 - DAC Digital Volume 1R */ + { 0x0482, 0x0180 }, /* R1154 - DAC Digital Volume 2L */ + { 0x0483, 0x0180 }, /* R1155 - DAC Digital Volume 2R */ + { 0x0484, 0x0180 }, /* R1156 - DAC Digital Volume 3L */ + { 0x0485, 0x0180 }, /* R1157 - DAC Digital Volume 3R */ + { 0x0486, 0x0180 }, /* R1158 - DAC Digital Volume 4L */ + { 0x0487, 0x0180 }, /* R1159 - DAC Digital Volume 4R */ + { 0x0488, 0x0180 }, /* R1160 - DAC Digital Volume 5L */ + { 0x0489, 0x0180 }, /* R1161 - DAC Digital Volume 5R */ + { 0x048A, 0x0180 }, /* R1162 - DAC Digital Volume 6L */ + { 0x048B, 0x0180 }, /* R1163 - DAC Digital Volume 6R */ + { 0x04C0, 0x0069 }, /* R1216 - PDM SPK1 CTRL 1 */ + { 0x04C1, 0x0000 }, /* R1217 - PDM SPK1 CTRL 2 */ + { 0x04C2, 0x0069 }, /* R1218 - PDM SPK2 CTRL 1 */ + { 0x04C3, 0x0000 }, /* R1219 - PDM SPK2 CTRL 2 */ + { 0x0500, 0x000C }, /* R1280 - Audio IF 1_1 */ + { 0x0501, 0x0008 }, /* R1281 - Audio IF 1_2 */ + { 0x0502, 0x0000 }, /* R1282 - Audio IF 1_3 */ + { 0x0503, 0x0000 }, /* R1283 - Audio IF 1_4 */ + { 0x0504, 0x0000 }, /* R1284 - Audio IF 1_5 */ + { 0x0505, 0x0300 }, /* R1285 - Audio IF 1_6 */ + { 0x0506, 0x0300 }, /* R1286 - Audio IF 1_7 */ + { 0x0507, 0x1820 }, /* R1287 - Audio IF 1_8 */ + { 0x0508, 0x1820 }, /* R1288 - Audio IF 1_9 */ + { 0x0509, 0x0000 }, /* R1289 - Audio IF 1_10 */ + { 0x050A, 0x0001 }, /* R1290 - Audio IF 1_11 */ + { 0x050B, 0x0002 }, /* R1291 - Audio IF 1_12 */ + { 0x050C, 0x0003 }, /* R1292 - Audio IF 1_13 */ + { 0x050D, 0x0004 }, /* R1293 - Audio IF 1_14 */ + { 0x050E, 0x0005 }, /* R1294 - Audio IF 1_15 */ + { 0x050F, 0x0006 }, /* R1295 - Audio IF 1_16 */ + { 0x0510, 0x0007 }, /* R1296 - Audio IF 1_17 */ + { 0x0511, 0x0000 }, /* R1297 - Audio IF 1_18 */ + { 0x0512, 0x0001 }, /* R1298 - Audio IF 1_19 */ + { 0x0513, 0x0002 }, /* R1299 - Audio IF 1_20 */ + { 0x0514, 0x0003 }, /* R1300 - Audio IF 1_21 */ + { 0x0515, 0x0004 }, /* R1301 - Audio IF 1_22 */ + { 0x0516, 0x0005 }, /* R1302 - Audio IF 1_23 */ + { 0x0517, 0x0006 }, /* R1303 - Audio IF 1_24 */ + { 0x0518, 0x0007 }, /* R1304 - Audio IF 1_25 */ + { 0x0519, 0x0000 }, /* R1305 - Audio IF 1_26 */ + { 0x051A, 0x0000 }, /* R1306 - Audio IF 1_27 */ + { 0x0540, 0x000C }, /* R1344 - Audio IF 2_1 */ + { 0x0541, 0x0008 }, /* R1345 - Audio IF 2_2 */ + { 0x0542, 0x0000 }, /* R1346 - Audio IF 2_3 */ + { 0x0543, 0x0000 }, /* R1347 - Audio IF 2_4 */ + { 0x0544, 0x0000 }, /* R1348 - Audio IF 2_5 */ + { 0x0545, 0x0300 }, /* R1349 - Audio IF 2_6 */ + { 0x0546, 0x0300 }, /* R1350 - Audio IF 2_7 */ + { 0x0547, 0x1820 }, /* R1351 - Audio IF 2_8 */ + { 0x0548, 0x1820 }, /* R1352 - Audio IF 2_9 */ + { 0x0549, 0x0000 }, /* R1353 - Audio IF 2_10 */ + { 0x054A, 0x0001 }, /* R1354 - Audio IF 2_11 */ + { 0x0551, 0x0000 }, /* R1361 - Audio IF 2_18 */ + { 0x0552, 0x0001 }, /* R1362 - Audio IF 2_19 */ + { 0x0559, 0x0000 }, /* R1369 - Audio IF 2_26 */ + { 0x055A, 0x0000 }, /* R1370 - Audio IF 2_27 */ + { 0x0580, 0x000C }, /* R1408 - Audio IF 3_1 */ + { 0x0581, 0x0008 }, /* R1409 - Audio IF 3_2 */ + { 0x0582, 0x0000 }, /* R1410 - Audio IF 3_3 */ + { 0x0583, 0x0000 }, /* R1411 - Audio IF 3_4 */ + { 0x0584, 0x0000 }, /* R1412 - Audio IF 3_5 */ + { 0x0585, 0x0300 }, /* R1413 - Audio IF 3_6 */ + { 0x0586, 0x0300 }, /* R1414 - Audio IF 3_7 */ + { 0x0587, 0x1820 }, /* R1415 - Audio IF 3_8 */ + { 0x0588, 0x1820 }, /* R1416 - Audio IF 3_9 */ + { 0x0589, 0x0000 }, /* R1417 - Audio IF 3_10 */ + { 0x058A, 0x0001 }, /* R1418 - Audio IF 3_11 */ + { 0x0591, 0x0000 }, /* R1425 - Audio IF 3_18 */ + { 0x0592, 0x0001 }, /* R1426 - Audio IF 3_19 */ + { 0x0599, 0x0000 }, /* R1433 - Audio IF 3_26 */ + { 0x059A, 0x0000 }, /* R1434 - Audio IF 3_27 */ + { 0x0640, 0x0000 }, /* R1600 - PWM1MIX Input 1 Source */ + { 0x0641, 0x0080 }, /* R1601 - PWM1MIX Input 1 Volume */ + { 0x0642, 0x0000 }, /* R1602 - PWM1MIX Input 2 Source */ + { 0x0643, 0x0080 }, /* R1603 - PWM1MIX Input 2 Volume */ + { 0x0644, 0x0000 }, /* R1604 - PWM1MIX Input 3 Source */ + { 0x0645, 0x0080 }, /* R1605 - PWM1MIX Input 3 Volume */ + { 0x0646, 0x0000 }, /* R1606 - PWM1MIX Input 4 Source */ + { 0x0647, 0x0080 }, /* R1607 - PWM1MIX Input 4 Volume */ + { 0x0648, 0x0000 }, /* R1608 - PWM2MIX Input 1 Source */ + { 0x0649, 0x0080 }, /* R1609 - PWM2MIX Input 1 Volume */ + { 0x064A, 0x0000 }, /* R1610 - PWM2MIX Input 2 Source */ + { 0x064B, 0x0080 }, /* R1611 - PWM2MIX Input 2 Volume */ + { 0x064C, 0x0000 }, /* R1612 - PWM2MIX Input 3 Source */ + { 0x064D, 0x0080 }, /* R1613 - PWM2MIX Input 3 Volume */ + { 0x064E, 0x0000 }, /* R1614 - PWM2MIX Input 4 Source */ + { 0x064F, 0x0080 }, /* R1615 - PWM2MIX Input 4 Volume */ + { 0x0680, 0x0000 }, /* R1664 - OUT1LMIX Input 1 Source */ + { 0x0681, 0x0080 }, /* R1665 - OUT1LMIX Input 1 Volume */ + { 0x0682, 0x0000 }, /* R1666 - OUT1LMIX Input 2 Source */ + { 0x0683, 0x0080 }, /* R1667 - OUT1LMIX Input 2 Volume */ + { 0x0684, 0x0000 }, /* R1668 - OUT1LMIX Input 3 Source */ + { 0x0685, 0x0080 }, /* R1669 - OUT1LMIX Input 3 Volume */ + { 0x0686, 0x0000 }, /* R1670 - OUT1LMIX Input 4 Source */ + { 0x0687, 0x0080 }, /* R1671 - OUT1LMIX Input 4 Volume */ + { 0x0688, 0x0000 }, /* R1672 - OUT1RMIX Input 1 Source */ + { 0x0689, 0x0080 }, /* R1673 - OUT1RMIX Input 1 Volume */ + { 0x068A, 0x0000 }, /* R1674 - OUT1RMIX Input 2 Source */ + { 0x068B, 0x0080 }, /* R1675 - OUT1RMIX Input 2 Volume */ + { 0x068C, 0x0000 }, /* R1676 - OUT1RMIX Input 3 Source */ + { 0x068D, 0x0080 }, /* R1677 - OUT1RMIX Input 3 Volume */ + { 0x068E, 0x0000 }, /* R1678 - OUT1RMIX Input 4 Source */ + { 0x068F, 0x0080 }, /* R1679 - OUT1RMIX Input 4 Volume */ + { 0x0690, 0x0000 }, /* R1680 - OUT2LMIX Input 1 Source */ + { 0x0691, 0x0080 }, /* R1681 - OUT2LMIX Input 1 Volume */ + { 0x0692, 0x0000 }, /* R1682 - OUT2LMIX Input 2 Source */ + { 0x0693, 0x0080 }, /* R1683 - OUT2LMIX Input 2 Volume */ + { 0x0694, 0x0000 }, /* R1684 - OUT2LMIX Input 3 Source */ + { 0x0695, 0x0080 }, /* R1685 - OUT2LMIX Input 3 Volume */ + { 0x0696, 0x0000 }, /* R1686 - OUT2LMIX Input 4 Source */ + { 0x0697, 0x0080 }, /* R1687 - OUT2LMIX Input 4 Volume */ + { 0x0698, 0x0000 }, /* R1688 - OUT2RMIX Input 1 Source */ + { 0x0699, 0x0080 }, /* R1689 - OUT2RMIX Input 1 Volume */ + { 0x069A, 0x0000 }, /* R1690 - OUT2RMIX Input 2 Source */ + { 0x069B, 0x0080 }, /* R1691 - OUT2RMIX Input 2 Volume */ + { 0x069C, 0x0000 }, /* R1692 - OUT2RMIX Input 3 Source */ + { 0x069D, 0x0080 }, /* R1693 - OUT2RMIX Input 3 Volume */ + { 0x069E, 0x0000 }, /* R1694 - OUT2RMIX Input 4 Source */ + { 0x069F, 0x0080 }, /* R1695 - OUT2RMIX Input 4 Volume */ + { 0x06A0, 0x0000 }, /* R1696 - OUT3LMIX Input 1 Source */ + { 0x06A1, 0x0080 }, /* R1697 - OUT3LMIX Input 1 Volume */ + { 0x06A2, 0x0000 }, /* R1698 - OUT3LMIX Input 2 Source */ + { 0x06A3, 0x0080 }, /* R1699 - OUT3LMIX Input 2 Volume */ + { 0x06A4, 0x0000 }, /* R1700 - OUT3LMIX Input 3 Source */ + { 0x06A5, 0x0080 }, /* R1701 - OUT3LMIX Input 3 Volume */ + { 0x06A6, 0x0000 }, /* R1702 - OUT3LMIX Input 4 Source */ + { 0x06A7, 0x0080 }, /* R1703 - OUT3LMIX Input 4 Volume */ + { 0x06A8, 0x0000 }, /* R1704 - OUT3RMIX Input 1 Source */ + { 0x06A9, 0x0080 }, /* R1705 - OUT3RMIX Input 1 Volume */ + { 0x06AA, 0x0000 }, /* R1706 - OUT3RMIX Input 2 Source */ + { 0x06AB, 0x0080 }, /* R1707 - OUT3RMIX Input 2 Volume */ + { 0x06AC, 0x0000 }, /* R1708 - OUT3RMIX Input 3 Source */ + { 0x06AD, 0x0080 }, /* R1709 - OUT3RMIX Input 3 Volume */ + { 0x06AE, 0x0000 }, /* R1710 - OUT3RMIX Input 4 Source */ + { 0x06AF, 0x0080 }, /* R1711 - OUT3RMIX Input 4 Volume */ + { 0x06B0, 0x0000 }, /* R1712 - OUT4LMIX Input 1 Source */ + { 0x06B1, 0x0080 }, /* R1713 - OUT4LMIX Input 1 Volume */ + { 0x06B2, 0x0000 }, /* R1714 - OUT4LMIX Input 2 Source */ + { 0x06B3, 0x0080 }, /* R1715 - OUT4LMIX Input 2 Volume */ + { 0x06B4, 0x0000 }, /* R1716 - OUT4LMIX Input 3 Source */ + { 0x06B5, 0x0080 }, /* R1717 - OUT4LMIX Input 3 Volume */ + { 0x06B6, 0x0000 }, /* R1718 - OUT4LMIX Input 4 Source */ + { 0x06B7, 0x0080 }, /* R1719 - OUT4LMIX Input 4 Volume */ + { 0x06B8, 0x0000 }, /* R1720 - OUT4RMIX Input 1 Source */ + { 0x06B9, 0x0080 }, /* R1721 - OUT4RMIX Input 1 Volume */ + { 0x06BA, 0x0000 }, /* R1722 - OUT4RMIX Input 2 Source */ + { 0x06BB, 0x0080 }, /* R1723 - OUT4RMIX Input 2 Volume */ + { 0x06BC, 0x0000 }, /* R1724 - OUT4RMIX Input 3 Source */ + { 0x06BD, 0x0080 }, /* R1725 - OUT4RMIX Input 3 Volume */ + { 0x06BE, 0x0000 }, /* R1726 - OUT4RMIX Input 4 Source */ + { 0x06BF, 0x0080 }, /* R1727 - OUT4RMIX Input 4 Volume */ + { 0x06C0, 0x0000 }, /* R1728 - OUT5LMIX Input 1 Source */ + { 0x06C1, 0x0080 }, /* R1729 - OUT5LMIX Input 1 Volume */ + { 0x06C2, 0x0000 }, /* R1730 - OUT5LMIX Input 2 Source */ + { 0x06C3, 0x0080 }, /* R1731 - OUT5LMIX Input 2 Volume */ + { 0x06C4, 0x0000 }, /* R1732 - OUT5LMIX Input 3 Source */ + { 0x06C5, 0x0080 }, /* R1733 - OUT5LMIX Input 3 Volume */ + { 0x06C6, 0x0000 }, /* R1734 - OUT5LMIX Input 4 Source */ + { 0x06C7, 0x0080 }, /* R1735 - OUT5LMIX Input 4 Volume */ + { 0x06C8, 0x0000 }, /* R1736 - OUT5RMIX Input 1 Source */ + { 0x06C9, 0x0080 }, /* R1737 - OUT5RMIX Input 1 Volume */ + { 0x06CA, 0x0000 }, /* R1738 - OUT5RMIX Input 2 Source */ + { 0x06CB, 0x0080 }, /* R1739 - OUT5RMIX Input 2 Volume */ + { 0x06CC, 0x0000 }, /* R1740 - OUT5RMIX Input 3 Source */ + { 0x06CD, 0x0080 }, /* R1741 - OUT5RMIX Input 3 Volume */ + { 0x06CE, 0x0000 }, /* R1742 - OUT5RMIX Input 4 Source */ + { 0x06CF, 0x0080 }, /* R1743 - OUT5RMIX Input 4 Volume */ + { 0x06D0, 0x0000 }, /* R1744 - OUT6LMIX Input 1 Source */ + { 0x06D1, 0x0080 }, /* R1745 - OUT6LMIX Input 1 Volume */ + { 0x06D2, 0x0000 }, /* R1746 - OUT6LMIX Input 2 Source */ + { 0x06D3, 0x0080 }, /* R1747 - OUT6LMIX Input 2 Volume */ + { 0x06D4, 0x0000 }, /* R1748 - OUT6LMIX Input 3 Source */ + { 0x06D5, 0x0080 }, /* R1749 - OUT6LMIX Input 3 Volume */ + { 0x06D6, 0x0000 }, /* R1750 - OUT6LMIX Input 4 Source */ + { 0x06D7, 0x0080 }, /* R1751 - OUT6LMIX Input 4 Volume */ + { 0x06D8, 0x0000 }, /* R1752 - OUT6RMIX Input 1 Source */ + { 0x06D9, 0x0080 }, /* R1753 - OUT6RMIX Input 1 Volume */ + { 0x06DA, 0x0000 }, /* R1754 - OUT6RMIX Input 2 Source */ + { 0x06DB, 0x0080 }, /* R1755 - OUT6RMIX Input 2 Volume */ + { 0x06DC, 0x0000 }, /* R1756 - OUT6RMIX Input 3 Source */ + { 0x06DD, 0x0080 }, /* R1757 - OUT6RMIX Input 3 Volume */ + { 0x06DE, 0x0000 }, /* R1758 - OUT6RMIX Input 4 Source */ + { 0x06DF, 0x0080 }, /* R1759 - OUT6RMIX Input 4 Volume */ + { 0x0700, 0x0000 }, /* R1792 - AIF1TX1MIX Input 1 Source */ + { 0x0701, 0x0080 }, /* R1793 - AIF1TX1MIX Input 1 Volume */ + { 0x0702, 0x0000 }, /* R1794 - AIF1TX1MIX Input 2 Source */ + { 0x0703, 0x0080 }, /* R1795 - AIF1TX1MIX Input 2 Volume */ + { 0x0704, 0x0000 }, /* R1796 - AIF1TX1MIX Input 3 Source */ + { 0x0705, 0x0080 }, /* R1797 - AIF1TX1MIX Input 3 Volume */ + { 0x0706, 0x0000 }, /* R1798 - AIF1TX1MIX Input 4 Source */ + { 0x0707, 0x0080 }, /* R1799 - AIF1TX1MIX Input 4 Volume */ + { 0x0708, 0x0000 }, /* R1800 - AIF1TX2MIX Input 1 Source */ + { 0x0709, 0x0080 }, /* R1801 - AIF1TX2MIX Input 1 Volume */ + { 0x070A, 0x0000 }, /* R1802 - AIF1TX2MIX Input 2 Source */ + { 0x070B, 0x0080 }, /* R1803 - AIF1TX2MIX Input 2 Volume */ + { 0x070C, 0x0000 }, /* R1804 - AIF1TX2MIX Input 3 Source */ + { 0x070D, 0x0080 }, /* R1805 - AIF1TX2MIX Input 3 Volume */ + { 0x070E, 0x0000 }, /* R1806 - AIF1TX2MIX Input 4 Source */ + { 0x070F, 0x0080 }, /* R1807 - AIF1TX2MIX Input 4 Volume */ + { 0x0710, 0x0000 }, /* R1808 - AIF1TX3MIX Input 1 Source */ + { 0x0711, 0x0080 }, /* R1809 - AIF1TX3MIX Input 1 Volume */ + { 0x0712, 0x0000 }, /* R1810 - AIF1TX3MIX Input 2 Source */ + { 0x0713, 0x0080 }, /* R1811 - AIF1TX3MIX Input 2 Volume */ + { 0x0714, 0x0000 }, /* R1812 - AIF1TX3MIX Input 3 Source */ + { 0x0715, 0x0080 }, /* R1813 - AIF1TX3MIX Input 3 Volume */ + { 0x0716, 0x0000 }, /* R1814 - AIF1TX3MIX Input 4 Source */ + { 0x0717, 0x0080 }, /* R1815 - AIF1TX3MIX Input 4 Volume */ + { 0x0718, 0x0000 }, /* R1816 - AIF1TX4MIX Input 1 Source */ + { 0x0719, 0x0080 }, /* R1817 - AIF1TX4MIX Input 1 Volume */ + { 0x071A, 0x0000 }, /* R1818 - AIF1TX4MIX Input 2 Source */ + { 0x071B, 0x0080 }, /* R1819 - AIF1TX4MIX Input 2 Volume */ + { 0x071C, 0x0000 }, /* R1820 - AIF1TX4MIX Input 3 Source */ + { 0x071D, 0x0080 }, /* R1821 - AIF1TX4MIX Input 3 Volume */ + { 0x071E, 0x0000 }, /* R1822 - AIF1TX4MIX Input 4 Source */ + { 0x071F, 0x0080 }, /* R1823 - AIF1TX4MIX Input 4 Volume */ + { 0x0720, 0x0000 }, /* R1824 - AIF1TX5MIX Input 1 Source */ + { 0x0721, 0x0080 }, /* R1825 - AIF1TX5MIX Input 1 Volume */ + { 0x0722, 0x0000 }, /* R1826 - AIF1TX5MIX Input 2 Source */ + { 0x0723, 0x0080 }, /* R1827 - AIF1TX5MIX Input 2 Volume */ + { 0x0724, 0x0000 }, /* R1828 - AIF1TX5MIX Input 3 Source */ + { 0x0725, 0x0080 }, /* R1829 - AIF1TX5MIX Input 3 Volume */ + { 0x0726, 0x0000 }, /* R1830 - AIF1TX5MIX Input 4 Source */ + { 0x0727, 0x0080 }, /* R1831 - AIF1TX5MIX Input 4 Volume */ + { 0x0728, 0x0000 }, /* R1832 - AIF1TX6MIX Input 1 Source */ + { 0x0729, 0x0080 }, /* R1833 - AIF1TX6MIX Input 1 Volume */ + { 0x072A, 0x0000 }, /* R1834 - AIF1TX6MIX Input 2 Source */ + { 0x072B, 0x0080 }, /* R1835 - AIF1TX6MIX Input 2 Volume */ + { 0x072C, 0x0000 }, /* R1836 - AIF1TX6MIX Input 3 Source */ + { 0x072D, 0x0080 }, /* R1837 - AIF1TX6MIX Input 3 Volume */ + { 0x072E, 0x0000 }, /* R1838 - AIF1TX6MIX Input 4 Source */ + { 0x072F, 0x0080 }, /* R1839 - AIF1TX6MIX Input 4 Volume */ + { 0x0730, 0x0000 }, /* R1840 - AIF1TX7MIX Input 1 Source */ + { 0x0731, 0x0080 }, /* R1841 - AIF1TX7MIX Input 1 Volume */ + { 0x0732, 0x0000 }, /* R1842 - AIF1TX7MIX Input 2 Source */ + { 0x0733, 0x0080 }, /* R1843 - AIF1TX7MIX Input 2 Volume */ + { 0x0734, 0x0000 }, /* R1844 - AIF1TX7MIX Input 3 Source */ + { 0x0735, 0x0080 }, /* R1845 - AIF1TX7MIX Input 3 Volume */ + { 0x0736, 0x0000 }, /* R1846 - AIF1TX7MIX Input 4 Source */ + { 0x0737, 0x0080 }, /* R1847 - AIF1TX7MIX Input 4 Volume */ + { 0x0738, 0x0000 }, /* R1848 - AIF1TX8MIX Input 1 Source */ + { 0x0739, 0x0080 }, /* R1849 - AIF1TX8MIX Input 1 Volume */ + { 0x073A, 0x0000 }, /* R1850 - AIF1TX8MIX Input 2 Source */ + { 0x073B, 0x0080 }, /* R1851 - AIF1TX8MIX Input 2 Volume */ + { 0x073C, 0x0000 }, /* R1852 - AIF1TX8MIX Input 3 Source */ + { 0x073D, 0x0080 }, /* R1853 - AIF1TX8MIX Input 3 Volume */ + { 0x073E, 0x0000 }, /* R1854 - AIF1TX8MIX Input 4 Source */ + { 0x073F, 0x0080 }, /* R1855 - AIF1TX8MIX Input 4 Volume */ + { 0x0740, 0x0000 }, /* R1856 - AIF2TX1MIX Input 1 Source */ + { 0x0741, 0x0080 }, /* R1857 - AIF2TX1MIX Input 1 Volume */ + { 0x0742, 0x0000 }, /* R1858 - AIF2TX1MIX Input 2 Source */ + { 0x0743, 0x0080 }, /* R1859 - AIF2TX1MIX Input 2 Volume */ + { 0x0744, 0x0000 }, /* R1860 - AIF2TX1MIX Input 3 Source */ + { 0x0745, 0x0080 }, /* R1861 - AIF2TX1MIX Input 3 Volume */ + { 0x0746, 0x0000 }, /* R1862 - AIF2TX1MIX Input 4 Source */ + { 0x0747, 0x0080 }, /* R1863 - AIF2TX1MIX Input 4 Volume */ + { 0x0748, 0x0000 }, /* R1864 - AIF2TX2MIX Input 1 Source */ + { 0x0749, 0x0080 }, /* R1865 - AIF2TX2MIX Input 1 Volume */ + { 0x074A, 0x0000 }, /* R1866 - AIF2TX2MIX Input 2 Source */ + { 0x074B, 0x0080 }, /* R1867 - AIF2TX2MIX Input 2 Volume */ + { 0x074C, 0x0000 }, /* R1868 - AIF2TX2MIX Input 3 Source */ + { 0x074D, 0x0080 }, /* R1869 - AIF2TX2MIX Input 3 Volume */ + { 0x074E, 0x0000 }, /* R1870 - AIF2TX2MIX Input 4 Source */ + { 0x074F, 0x0080 }, /* R1871 - AIF2TX2MIX Input 4 Volume */ + { 0x0780, 0x0000 }, /* R1920 - AIF3TX1MIX Input 1 Source */ + { 0x0781, 0x0080 }, /* R1921 - AIF3TX1MIX Input 1 Volume */ + { 0x0782, 0x0000 }, /* R1922 - AIF3TX1MIX Input 2 Source */ + { 0x0783, 0x0080 }, /* R1923 - AIF3TX1MIX Input 2 Volume */ + { 0x0784, 0x0000 }, /* R1924 - AIF3TX1MIX Input 3 Source */ + { 0x0785, 0x0080 }, /* R1925 - AIF3TX1MIX Input 3 Volume */ + { 0x0786, 0x0000 }, /* R1926 - AIF3TX1MIX Input 4 Source */ + { 0x0787, 0x0080 }, /* R1927 - AIF3TX1MIX Input 4 Volume */ + { 0x0788, 0x0000 }, /* R1928 - AIF3TX2MIX Input 1 Source */ + { 0x0789, 0x0080 }, /* R1929 - AIF3TX2MIX Input 1 Volume */ + { 0x078A, 0x0000 }, /* R1930 - AIF3TX2MIX Input 2 Source */ + { 0x078B, 0x0080 }, /* R1931 - AIF3TX2MIX Input 2 Volume */ + { 0x078C, 0x0000 }, /* R1932 - AIF3TX2MIX Input 3 Source */ + { 0x078D, 0x0080 }, /* R1933 - AIF3TX2MIX Input 3 Volume */ + { 0x078E, 0x0000 }, /* R1934 - AIF3TX2MIX Input 4 Source */ + { 0x078F, 0x0080 }, /* R1935 - AIF3TX2MIX Input 4 Volume */ + { 0x0880, 0x0000 }, /* R2176 - EQ1MIX Input 1 Source */ + { 0x0881, 0x0080 }, /* R2177 - EQ1MIX Input 1 Volume */ + { 0x0882, 0x0000 }, /* R2178 - EQ1MIX Input 2 Source */ + { 0x0883, 0x0080 }, /* R2179 - EQ1MIX Input 2 Volume */ + { 0x0884, 0x0000 }, /* R2180 - EQ1MIX Input 3 Source */ + { 0x0885, 0x0080 }, /* R2181 - EQ1MIX Input 3 Volume */ + { 0x0886, 0x0000 }, /* R2182 - EQ1MIX Input 4 Source */ + { 0x0887, 0x0080 }, /* R2183 - EQ1MIX Input 4 Volume */ + { 0x0888, 0x0000 }, /* R2184 - EQ2MIX Input 1 Source */ + { 0x0889, 0x0080 }, /* R2185 - EQ2MIX Input 1 Volume */ + { 0x088A, 0x0000 }, /* R2186 - EQ2MIX Input 2 Source */ + { 0x088B, 0x0080 }, /* R2187 - EQ2MIX Input 2 Volume */ + { 0x088C, 0x0000 }, /* R2188 - EQ2MIX Input 3 Source */ + { 0x088D, 0x0080 }, /* R2189 - EQ2MIX Input 3 Volume */ + { 0x088E, 0x0000 }, /* R2190 - EQ2MIX Input 4 Source */ + { 0x088F, 0x0080 }, /* R2191 - EQ2MIX Input 4 Volume */ + { 0x0890, 0x0000 }, /* R2192 - EQ3MIX Input 1 Source */ + { 0x0891, 0x0080 }, /* R2193 - EQ3MIX Input 1 Volume */ + { 0x0892, 0x0000 }, /* R2194 - EQ3MIX Input 2 Source */ + { 0x0893, 0x0080 }, /* R2195 - EQ3MIX Input 2 Volume */ + { 0x0894, 0x0000 }, /* R2196 - EQ3MIX Input 3 Source */ + { 0x0895, 0x0080 }, /* R2197 - EQ3MIX Input 3 Volume */ + { 0x0896, 0x0000 }, /* R2198 - EQ3MIX Input 4 Source */ + { 0x0897, 0x0080 }, /* R2199 - EQ3MIX Input 4 Volume */ + { 0x0898, 0x0000 }, /* R2200 - EQ4MIX Input 1 Source */ + { 0x0899, 0x0080 }, /* R2201 - EQ4MIX Input 1 Volume */ + { 0x089A, 0x0000 }, /* R2202 - EQ4MIX Input 2 Source */ + { 0x089B, 0x0080 }, /* R2203 - EQ4MIX Input 2 Volume */ + { 0x089C, 0x0000 }, /* R2204 - EQ4MIX Input 3 Source */ + { 0x089D, 0x0080 }, /* R2205 - EQ4MIX Input 3 Volume */ + { 0x089E, 0x0000 }, /* R2206 - EQ4MIX Input 4 Source */ + { 0x089F, 0x0080 }, /* R2207 - EQ4MIX Input 4 Volume */ + { 0x08C0, 0x0000 }, /* R2240 - DRC1LMIX Input 1 Source */ + { 0x08C1, 0x0080 }, /* R2241 - DRC1LMIX Input 1 Volume */ + { 0x08C2, 0x0000 }, /* R2242 - DRC1LMIX Input 2 Source */ + { 0x08C3, 0x0080 }, /* R2243 - DRC1LMIX Input 2 Volume */ + { 0x08C4, 0x0000 }, /* R2244 - DRC1LMIX Input 3 Source */ + { 0x08C5, 0x0080 }, /* R2245 - DRC1LMIX Input 3 Volume */ + { 0x08C6, 0x0000 }, /* R2246 - DRC1LMIX Input 4 Source */ + { 0x08C7, 0x0080 }, /* R2247 - DRC1LMIX Input 4 Volume */ + { 0x08C8, 0x0000 }, /* R2248 - DRC1RMIX Input 1 Source */ + { 0x08C9, 0x0080 }, /* R2249 - DRC1RMIX Input 1 Volume */ + { 0x08CA, 0x0000 }, /* R2250 - DRC1RMIX Input 2 Source */ + { 0x08CB, 0x0080 }, /* R2251 - DRC1RMIX Input 2 Volume */ + { 0x08CC, 0x0000 }, /* R2252 - DRC1RMIX Input 3 Source */ + { 0x08CD, 0x0080 }, /* R2253 - DRC1RMIX Input 3 Volume */ + { 0x08CE, 0x0000 }, /* R2254 - DRC1RMIX Input 4 Source */ + { 0x08CF, 0x0080 }, /* R2255 - DRC1RMIX Input 4 Volume */ + { 0x0900, 0x0000 }, /* R2304 - HPLP1MIX Input 1 Source */ + { 0x0901, 0x0080 }, /* R2305 - HPLP1MIX Input 1 Volume */ + { 0x0902, 0x0000 }, /* R2306 - HPLP1MIX Input 2 Source */ + { 0x0903, 0x0080 }, /* R2307 - HPLP1MIX Input 2 Volume */ + { 0x0904, 0x0000 }, /* R2308 - HPLP1MIX Input 3 Source */ + { 0x0905, 0x0080 }, /* R2309 - HPLP1MIX Input 3 Volume */ + { 0x0906, 0x0000 }, /* R2310 - HPLP1MIX Input 4 Source */ + { 0x0907, 0x0080 }, /* R2311 - HPLP1MIX Input 4 Volume */ + { 0x0908, 0x0000 }, /* R2312 - HPLP2MIX Input 1 Source */ + { 0x0909, 0x0080 }, /* R2313 - HPLP2MIX Input 1 Volume */ + { 0x090A, 0x0000 }, /* R2314 - HPLP2MIX Input 2 Source */ + { 0x090B, 0x0080 }, /* R2315 - HPLP2MIX Input 2 Volume */ + { 0x090C, 0x0000 }, /* R2316 - HPLP2MIX Input 3 Source */ + { 0x090D, 0x0080 }, /* R2317 - HPLP2MIX Input 3 Volume */ + { 0x090E, 0x0000 }, /* R2318 - HPLP2MIX Input 4 Source */ + { 0x090F, 0x0080 }, /* R2319 - HPLP2MIX Input 4 Volume */ + { 0x0910, 0x0000 }, /* R2320 - HPLP3MIX Input 1 Source */ + { 0x0911, 0x0080 }, /* R2321 - HPLP3MIX Input 1 Volume */ + { 0x0912, 0x0000 }, /* R2322 - HPLP3MIX Input 2 Source */ + { 0x0913, 0x0080 }, /* R2323 - HPLP3MIX Input 2 Volume */ + { 0x0914, 0x0000 }, /* R2324 - HPLP3MIX Input 3 Source */ + { 0x0915, 0x0080 }, /* R2325 - HPLP3MIX Input 3 Volume */ + { 0x0916, 0x0000 }, /* R2326 - HPLP3MIX Input 4 Source */ + { 0x0917, 0x0080 }, /* R2327 - HPLP3MIX Input 4 Volume */ + { 0x0918, 0x0000 }, /* R2328 - HPLP4MIX Input 1 Source */ + { 0x0919, 0x0080 }, /* R2329 - HPLP4MIX Input 1 Volume */ + { 0x091A, 0x0000 }, /* R2330 - HPLP4MIX Input 2 Source */ + { 0x091B, 0x0080 }, /* R2331 - HPLP4MIX Input 2 Volume */ + { 0x091C, 0x0000 }, /* R2332 - HPLP4MIX Input 3 Source */ + { 0x091D, 0x0080 }, /* R2333 - HPLP4MIX Input 3 Volume */ + { 0x091E, 0x0000 }, /* R2334 - HPLP4MIX Input 4 Source */ + { 0x091F, 0x0080 }, /* R2335 - HPLP4MIX Input 4 Volume */ + { 0x0940, 0x0000 }, /* R2368 - DSP1LMIX Input 1 Source */ + { 0x0941, 0x0080 }, /* R2369 - DSP1LMIX Input 1 Volume */ + { 0x0942, 0x0000 }, /* R2370 - DSP1LMIX Input 2 Source */ + { 0x0943, 0x0080 }, /* R2371 - DSP1LMIX Input 2 Volume */ + { 0x0944, 0x0000 }, /* R2372 - DSP1LMIX Input 3 Source */ + { 0x0945, 0x0080 }, /* R2373 - DSP1LMIX Input 3 Volume */ + { 0x0946, 0x0000 }, /* R2374 - DSP1LMIX Input 4 Source */ + { 0x0947, 0x0080 }, /* R2375 - DSP1LMIX Input 4 Volume */ + { 0x0948, 0x0000 }, /* R2376 - DSP1RMIX Input 1 Source */ + { 0x0949, 0x0080 }, /* R2377 - DSP1RMIX Input 1 Volume */ + { 0x094A, 0x0000 }, /* R2378 - DSP1RMIX Input 2 Source */ + { 0x094B, 0x0080 }, /* R2379 - DSP1RMIX Input 2 Volume */ + { 0x094C, 0x0000 }, /* R2380 - DSP1RMIX Input 3 Source */ + { 0x094D, 0x0080 }, /* R2381 - DSP1RMIX Input 3 Volume */ + { 0x094E, 0x0000 }, /* R2382 - DSP1RMIX Input 4 Source */ + { 0x094F, 0x0080 }, /* R2383 - DSP1RMIX Input 4 Volume */ + { 0x0950, 0x0000 }, /* R2384 - DSP1AUX1MIX Input 1 Source */ + { 0x0958, 0x0000 }, /* R2392 - DSP1AUX2MIX Input 1 Source */ + { 0x0960, 0x0000 }, /* R2400 - DSP1AUX3MIX Input 1 Source */ + { 0x0968, 0x0000 }, /* R2408 - DSP1AUX4MIX Input 1 Source */ + { 0x0970, 0x0000 }, /* R2416 - DSP1AUX5MIX Input 1 Source */ + { 0x0978, 0x0000 }, /* R2424 - DSP1AUX6MIX Input 1 Source */ + { 0x0980, 0x0000 }, /* R2432 - DSP2LMIX Input 1 Source */ + { 0x0981, 0x0080 }, /* R2433 - DSP2LMIX Input 1 Volume */ + { 0x0982, 0x0000 }, /* R2434 - DSP2LMIX Input 2 Source */ + { 0x0983, 0x0080 }, /* R2435 - DSP2LMIX Input 2 Volume */ + { 0x0984, 0x0000 }, /* R2436 - DSP2LMIX Input 3 Source */ + { 0x0985, 0x0080 }, /* R2437 - DSP2LMIX Input 3 Volume */ + { 0x0986, 0x0000 }, /* R2438 - DSP2LMIX Input 4 Source */ + { 0x0987, 0x0080 }, /* R2439 - DSP2LMIX Input 4 Volume */ + { 0x0988, 0x0000 }, /* R2440 - DSP2RMIX Input 1 Source */ + { 0x0989, 0x0080 }, /* R2441 - DSP2RMIX Input 1 Volume */ + { 0x098A, 0x0000 }, /* R2442 - DSP2RMIX Input 2 Source */ + { 0x098B, 0x0080 }, /* R2443 - DSP2RMIX Input 2 Volume */ + { 0x098C, 0x0000 }, /* R2444 - DSP2RMIX Input 3 Source */ + { 0x098D, 0x0080 }, /* R2445 - DSP2RMIX Input 3 Volume */ + { 0x098E, 0x0000 }, /* R2446 - DSP2RMIX Input 4 Source */ + { 0x098F, 0x0080 }, /* R2447 - DSP2RMIX Input 4 Volume */ + { 0x0990, 0x0000 }, /* R2448 - DSP2AUX1MIX Input 1 Source */ + { 0x0998, 0x0000 }, /* R2456 - DSP2AUX2MIX Input 1 Source */ + { 0x09A0, 0x0000 }, /* R2464 - DSP2AUX3MIX Input 1 Source */ + { 0x09A8, 0x0000 }, /* R2472 - DSP2AUX4MIX Input 1 Source */ + { 0x09B0, 0x0000 }, /* R2480 - DSP2AUX5MIX Input 1 Source */ + { 0x09B8, 0x0000 }, /* R2488 - DSP2AUX6MIX Input 1 Source */ + { 0x09C0, 0x0000 }, /* R2496 - DSP3LMIX Input 1 Source */ + { 0x09C1, 0x0080 }, /* R2497 - DSP3LMIX Input 1 Volume */ + { 0x09C2, 0x0000 }, /* R2498 - DSP3LMIX Input 2 Source */ + { 0x09C3, 0x0080 }, /* R2499 - DSP3LMIX Input 2 Volume */ + { 0x09C4, 0x0000 }, /* R2500 - DSP3LMIX Input 3 Source */ + { 0x09C5, 0x0080 }, /* R2501 - DSP3LMIX Input 3 Volume */ + { 0x09C6, 0x0000 }, /* R2502 - DSP3LMIX Input 4 Source */ + { 0x09C7, 0x0080 }, /* R2503 - DSP3LMIX Input 4 Volume */ + { 0x09C8, 0x0000 }, /* R2504 - DSP3RMIX Input 1 Source */ + { 0x09C9, 0x0080 }, /* R2505 - DSP3RMIX Input 1 Volume */ + { 0x09CA, 0x0000 }, /* R2506 - DSP3RMIX Input 2 Source */ + { 0x09CB, 0x0080 }, /* R2507 - DSP3RMIX Input 2 Volume */ + { 0x09CC, 0x0000 }, /* R2508 - DSP3RMIX Input 3 Source */ + { 0x09CD, 0x0080 }, /* R2509 - DSP3RMIX Input 3 Volume */ + { 0x09CE, 0x0000 }, /* R2510 - DSP3RMIX Input 4 Source */ + { 0x09CF, 0x0080 }, /* R2511 - DSP3RMIX Input 4 Volume */ + { 0x09D0, 0x0000 }, /* R2512 - DSP3AUX1MIX Input 1 Source */ + { 0x09D8, 0x0000 }, /* R2520 - DSP3AUX2MIX Input 1 Source */ + { 0x09E0, 0x0000 }, /* R2528 - DSP3AUX3MIX Input 1 Source */ + { 0x09E8, 0x0000 }, /* R2536 - DSP3AUX4MIX Input 1 Source */ + { 0x09F0, 0x0000 }, /* R2544 - DSP3AUX5MIX Input 1 Source */ + { 0x09F8, 0x0000 }, /* R2552 - DSP3AUX6MIX Input 1 Source */ + { 0x0A80, 0x0000 }, /* R2688 - ASRC1LMIX Input 1 Source */ + { 0x0A88, 0x0000 }, /* R2696 - ASRC1RMIX Input 1 Source */ + { 0x0A90, 0x0000 }, /* R2704 - ASRC2LMIX Input 1 Source */ + { 0x0A98, 0x0000 }, /* R2712 - ASRC2RMIX Input 1 Source */ + { 0x0B00, 0x0000 }, /* R2816 - ISRC1DEC1MIX Input 1 Source */ + { 0x0B08, 0x0000 }, /* R2824 - ISRC1DEC2MIX Input 1 Source */ + { 0x0B10, 0x0000 }, /* R2832 - ISRC1DEC3MIX Input 1 Source */ + { 0x0B18, 0x0000 }, /* R2840 - ISRC1DEC4MIX Input 1 Source */ + { 0x0B20, 0x0000 }, /* R2848 - ISRC1INT1MIX Input 1 Source */ + { 0x0B28, 0x0000 }, /* R2856 - ISRC1INT2MIX Input 1 Source */ + { 0x0B30, 0x0000 }, /* R2864 - ISRC1INT3MIX Input 1 Source */ + { 0x0B38, 0x0000 }, /* R2872 - ISRC1INT4MIX Input 1 Source */ + { 0x0B40, 0x0000 }, /* R2880 - ISRC2DEC1MIX Input 1 Source */ + { 0x0B48, 0x0000 }, /* R2888 - ISRC2DEC2MIX Input 1 Source */ + { 0x0B50, 0x0000 }, /* R2896 - ISRC2DEC3MIX Input 1 Source */ + { 0x0B58, 0x0000 }, /* R2904 - ISRC2DEC4MIX Input 1 Source */ + { 0x0B60, 0x0000 }, /* R2912 - ISRC2INT1MIX Input 1 Source */ + { 0x0B68, 0x0000 }, /* R2920 - ISRC2INT2MIX Input 1 Source */ + { 0x0B70, 0x0000 }, /* R2928 - ISRC2INT3MIX Input 1 Source */ + { 0x0B78, 0x0000 }, /* R2936 - ISRC2INT4MIX Input 1 Source */ + { 0x0C00, 0xA001 }, /* R3072 - GPIO CTRL 1 */ + { 0x0C01, 0xA001 }, /* R3073 - GPIO CTRL 2 */ + { 0x0C02, 0xA001 }, /* R3074 - GPIO CTRL 3 */ + { 0x0C03, 0xA001 }, /* R3075 - GPIO CTRL 4 */ + { 0x0C04, 0xA001 }, /* R3076 - GPIO CTRL 5 */ + { 0x0C05, 0xA001 }, /* R3077 - GPIO CTRL 6 */ + { 0x0C23, 0x4003 }, /* R3107 - Misc Pad Ctrl 1 */ + { 0x0C24, 0x0000 }, /* R3108 - Misc Pad Ctrl 2 */ + { 0x0C25, 0x0000 }, /* R3109 - Misc Pad Ctrl 3 */ + { 0x0C26, 0x0000 }, /* R3110 - Misc Pad Ctrl 4 */ + { 0x0C27, 0x0000 }, /* R3111 - Misc Pad Ctrl 5 */ + { 0x0C28, 0x0000 }, /* R3112 - Misc GPIO 1 */ + { 0x0D00, 0x0000 }, /* R3328 - Interrupt Status 1 */ + { 0x0D01, 0x0000 }, /* R3329 - Interrupt Status 2 */ + { 0x0D02, 0x0000 }, /* R3330 - Interrupt Status 3 */ + { 0x0D03, 0x0000 }, /* R3331 - Interrupt Status 4 */ + { 0x0D04, 0x0000 }, /* R3332 - Interrupt Raw Status 2 */ + { 0x0D05, 0x0000 }, /* R3333 - Interrupt Raw Status 3 */ + { 0x0D06, 0x0000 }, /* R3334 - Interrupt Raw Status 4 */ + { 0x0D07, 0xFFFF }, /* R3335 - Interrupt Status 1 Mask */ + { 0x0D08, 0xFFFF }, /* R3336 - Interrupt Status 2 Mask */ + { 0x0D09, 0xFFFF }, /* R3337 - Interrupt Status 3 Mask */ + { 0x0D0A, 0xFFFF }, /* R3338 - Interrupt Status 4 Mask */ + { 0x0D1F, 0x0000 }, /* R3359 - Interrupt Control */ + { 0x0D20, 0xFFFF }, /* R3360 - IRQ Debounce 1 */ + { 0x0D21, 0xFFFF }, /* R3361 - IRQ Debounce 2 */ + { 0x0E00, 0x0000 }, /* R3584 - FX_Ctrl */ + { 0x0E10, 0x6318 }, /* R3600 - EQ1_1 */ + { 0x0E11, 0x6300 }, /* R3601 - EQ1_2 */ + { 0x0E12, 0x0FC8 }, /* R3602 - EQ1_3 */ + { 0x0E13, 0x03FE }, /* R3603 - EQ1_4 */ + { 0x0E14, 0x00E0 }, /* R3604 - EQ1_5 */ + { 0x0E15, 0x1EC4 }, /* R3605 - EQ1_6 */ + { 0x0E16, 0xF136 }, /* R3606 - EQ1_7 */ + { 0x0E17, 0x0409 }, /* R3607 - EQ1_8 */ + { 0x0E18, 0x04CC }, /* R3608 - EQ1_9 */ + { 0x0E19, 0x1C9B }, /* R3609 - EQ1_10 */ + { 0x0E1A, 0xF337 }, /* R3610 - EQ1_11 */ + { 0x0E1B, 0x040B }, /* R3611 - EQ1_12 */ + { 0x0E1C, 0x0CBB }, /* R3612 - EQ1_13 */ + { 0x0E1D, 0x16F8 }, /* R3613 - EQ1_14 */ + { 0x0E1E, 0xF7D9 }, /* R3614 - EQ1_15 */ + { 0x0E1F, 0x040A }, /* R3615 - EQ1_16 */ + { 0x0E20, 0x1F14 }, /* R3616 - EQ1_17 */ + { 0x0E21, 0x058C }, /* R3617 - EQ1_18 */ + { 0x0E22, 0x0563 }, /* R3618 - EQ1_19 */ + { 0x0E23, 0x4000 }, /* R3619 - EQ1_20 */ + { 0x0E26, 0x6318 }, /* R3622 - EQ2_1 */ + { 0x0E27, 0x6300 }, /* R3623 - EQ2_2 */ + { 0x0E28, 0x0FC8 }, /* R3624 - EQ2_3 */ + { 0x0E29, 0x03FE }, /* R3625 - EQ2_4 */ + { 0x0E2A, 0x00E0 }, /* R3626 - EQ2_5 */ + { 0x0E2B, 0x1EC4 }, /* R3627 - EQ2_6 */ + { 0x0E2C, 0xF136 }, /* R3628 - EQ2_7 */ + { 0x0E2D, 0x0409 }, /* R3629 - EQ2_8 */ + { 0x0E2E, 0x04CC }, /* R3630 - EQ2_9 */ + { 0x0E2F, 0x1C9B }, /* R3631 - EQ2_10 */ + { 0x0E30, 0xF337 }, /* R3632 - EQ2_11 */ + { 0x0E31, 0x040B }, /* R3633 - EQ2_12 */ + { 0x0E32, 0x0CBB }, /* R3634 - EQ2_13 */ + { 0x0E33, 0x16F8 }, /* R3635 - EQ2_14 */ + { 0x0E34, 0xF7D9 }, /* R3636 - EQ2_15 */ + { 0x0E35, 0x040A }, /* R3637 - EQ2_16 */ + { 0x0E36, 0x1F14 }, /* R3638 - EQ2_17 */ + { 0x0E37, 0x058C }, /* R3639 - EQ2_18 */ + { 0x0E38, 0x0563 }, /* R3640 - EQ2_19 */ + { 0x0E39, 0x4000 }, /* R3641 - EQ2_20 */ + { 0x0E3C, 0x6318 }, /* R3644 - EQ3_1 */ + { 0x0E3D, 0x6300 }, /* R3645 - EQ3_2 */ + { 0x0E3E, 0x0FC8 }, /* R3646 - EQ3_3 */ + { 0x0E3F, 0x03FE }, /* R3647 - EQ3_4 */ + { 0x0E40, 0x00E0 }, /* R3648 - EQ3_5 */ + { 0x0E41, 0x1EC4 }, /* R3649 - EQ3_6 */ + { 0x0E42, 0xF136 }, /* R3650 - EQ3_7 */ + { 0x0E43, 0x0409 }, /* R3651 - EQ3_8 */ + { 0x0E44, 0x04CC }, /* R3652 - EQ3_9 */ + { 0x0E45, 0x1C9B }, /* R3653 - EQ3_10 */ + { 0x0E46, 0xF337 }, /* R3654 - EQ3_11 */ + { 0x0E47, 0x040B }, /* R3655 - EQ3_12 */ + { 0x0E48, 0x0CBB }, /* R3656 - EQ3_13 */ + { 0x0E49, 0x16F8 }, /* R3657 - EQ3_14 */ + { 0x0E4A, 0xF7D9 }, /* R3658 - EQ3_15 */ + { 0x0E4B, 0x040A }, /* R3659 - EQ3_16 */ + { 0x0E4C, 0x1F14 }, /* R3660 - EQ3_17 */ + { 0x0E4D, 0x058C }, /* R3661 - EQ3_18 */ + { 0x0E4E, 0x0563 }, /* R3662 - EQ3_19 */ + { 0x0E4F, 0x4000 }, /* R3663 - EQ3_20 */ + { 0x0E52, 0x6318 }, /* R3666 - EQ4_1 */ + { 0x0E53, 0x6300 }, /* R3667 - EQ4_2 */ + { 0x0E54, 0x0FC8 }, /* R3668 - EQ4_3 */ + { 0x0E55, 0x03FE }, /* R3669 - EQ4_4 */ + { 0x0E56, 0x00E0 }, /* R3670 - EQ4_5 */ + { 0x0E57, 0x1EC4 }, /* R3671 - EQ4_6 */ + { 0x0E58, 0xF136 }, /* R3672 - EQ4_7 */ + { 0x0E59, 0x0409 }, /* R3673 - EQ4_8 */ + { 0x0E5A, 0x04CC }, /* R3674 - EQ4_9 */ + { 0x0E5B, 0x1C9B }, /* R3675 - EQ4_10 */ + { 0x0E5C, 0xF337 }, /* R3676 - EQ4_11 */ + { 0x0E5D, 0x040B }, /* R3677 - EQ4_12 */ + { 0x0E5E, 0x0CBB }, /* R3678 - EQ4_13 */ + { 0x0E5F, 0x16F8 }, /* R3679 - EQ4_14 */ + { 0x0E60, 0xF7D9 }, /* R3680 - EQ4_15 */ + { 0x0E61, 0x040A }, /* R3681 - EQ4_16 */ + { 0x0E62, 0x1F14 }, /* R3682 - EQ4_17 */ + { 0x0E63, 0x058C }, /* R3683 - EQ4_18 */ + { 0x0E64, 0x0563 }, /* R3684 - EQ4_19 */ + { 0x0E65, 0x4000 }, /* R3685 - EQ4_20 */ + { 0x0E80, 0x0018 }, /* R3712 - DRC1 ctrl1 */ + { 0x0E81, 0x0933 }, /* R3713 - DRC1 ctrl2 */ + { 0x0E82, 0x0018 }, /* R3714 - DRC1 ctrl3 */ + { 0x0E83, 0x0000 }, /* R3715 - DRC1 ctrl4 */ + { 0x0E84, 0x0000 }, /* R3716 - DRC1 ctrl5 */ + { 0x0EC0, 0x0000 }, /* R3776 - HPLPF1_1 */ + { 0x0EC1, 0x0000 }, /* R3777 - HPLPF1_2 */ + { 0x0EC4, 0x0000 }, /* R3780 - HPLPF2_1 */ + { 0x0EC5, 0x0000 }, /* R3781 - HPLPF2_2 */ + { 0x0EC8, 0x0000 }, /* R3784 - HPLPF3_1 */ + { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ + { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ + { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ + { 0x4000, 0x0000 }, /* R16384 - DSP1 DM 0 */ + { 0x4001, 0x0000 }, /* R16385 - DSP1 DM 1 */ + { 0x4002, 0x0000 }, /* R16386 - DSP1 DM 2 */ + { 0x4003, 0x0000 }, /* R16387 - DSP1 DM 3 */ + { 0x41FC, 0x0000 }, /* R16892 - DSP1 DM 508 */ + { 0x41FD, 0x0000 }, /* R16893 - DSP1 DM 509 */ + { 0x41FE, 0x0000 }, /* R16894 - DSP1 DM 510 */ + { 0x41FF, 0x0000 }, /* R16895 - DSP1 DM 511 */ + { 0x4800, 0x0000 }, /* R18432 - DSP1 PM 0 */ + { 0x4801, 0x0000 }, /* R18433 - DSP1 PM 1 */ + { 0x4802, 0x0000 }, /* R18434 - DSP1 PM 2 */ + { 0x4803, 0x0000 }, /* R18435 - DSP1 PM 3 */ + { 0x4804, 0x0000 }, /* R18436 - DSP1 PM 4 */ + { 0x4805, 0x0000 }, /* R18437 - DSP1 PM 5 */ + { 0x4DFA, 0x0000 }, /* R19962 - DSP1 PM 1530 */ + { 0x4DFB, 0x0000 }, /* R19963 - DSP1 PM 1531 */ + { 0x4DFC, 0x0000 }, /* R19964 - DSP1 PM 1532 */ + { 0x4DFD, 0x0000 }, /* R19965 - DSP1 PM 1533 */ + { 0x4DFE, 0x0000 }, /* R19966 - DSP1 PM 1534 */ + { 0x4DFF, 0x0000 }, /* R19967 - DSP1 PM 1535 */ + { 0x5000, 0x0000 }, /* R20480 - DSP1 ZM 0 */ + { 0x5001, 0x0000 }, /* R20481 - DSP1 ZM 1 */ + { 0x5002, 0x0000 }, /* R20482 - DSP1 ZM 2 */ + { 0x5003, 0x0000 }, /* R20483 - DSP1 ZM 3 */ + { 0x57FC, 0x0000 }, /* R22524 - DSP1 ZM 2044 */ + { 0x57FD, 0x0000 }, /* R22525 - DSP1 ZM 2045 */ + { 0x57FE, 0x0000 }, /* R22526 - DSP1 ZM 2046 */ + { 0x57FF, 0x0000 }, /* R22527 - DSP1 ZM 2047 */ + { 0x6000, 0x0000 }, /* R24576 - DSP2 DM 0 */ + { 0x6001, 0x0000 }, /* R24577 - DSP2 DM 1 */ + { 0x6002, 0x0000 }, /* R24578 - DSP2 DM 2 */ + { 0x6003, 0x0000 }, /* R24579 - DSP2 DM 3 */ + { 0x61FC, 0x0000 }, /* R25084 - DSP2 DM 508 */ + { 0x61FD, 0x0000 }, /* R25085 - DSP2 DM 509 */ + { 0x61FE, 0x0000 }, /* R25086 - DSP2 DM 510 */ + { 0x61FF, 0x0000 }, /* R25087 - DSP2 DM 511 */ + { 0x6800, 0x0000 }, /* R26624 - DSP2 PM 0 */ + { 0x6801, 0x0000 }, /* R26625 - DSP2 PM 1 */ + { 0x6802, 0x0000 }, /* R26626 - DSP2 PM 2 */ + { 0x6803, 0x0000 }, /* R26627 - DSP2 PM 3 */ + { 0x6804, 0x0000 }, /* R26628 - DSP2 PM 4 */ + { 0x6805, 0x0000 }, /* R26629 - DSP2 PM 5 */ + { 0x6DFA, 0x0000 }, /* R28154 - DSP2 PM 1530 */ + { 0x6DFB, 0x0000 }, /* R28155 - DSP2 PM 1531 */ + { 0x6DFC, 0x0000 }, /* R28156 - DSP2 PM 1532 */ + { 0x6DFD, 0x0000 }, /* R28157 - DSP2 PM 1533 */ + { 0x6DFE, 0x0000 }, /* R28158 - DSP2 PM 1534 */ + { 0x6DFF, 0x0000 }, /* R28159 - DSP2 PM 1535 */ + { 0x7000, 0x0000 }, /* R28672 - DSP2 ZM 0 */ + { 0x7001, 0x0000 }, /* R28673 - DSP2 ZM 1 */ + { 0x7002, 0x0000 }, /* R28674 - DSP2 ZM 2 */ + { 0x7003, 0x0000 }, /* R28675 - DSP2 ZM 3 */ + { 0x77FC, 0x0000 }, /* R30716 - DSP2 ZM 2044 */ + { 0x77FD, 0x0000 }, /* R30717 - DSP2 ZM 2045 */ + { 0x77FE, 0x0000 }, /* R30718 - DSP2 ZM 2046 */ + { 0x77FF, 0x0000 }, /* R30719 - DSP2 ZM 2047 */ + { 0x8000, 0x0000 }, /* R32768 - DSP3 DM 0 */ + { 0x8001, 0x0000 }, /* R32769 - DSP3 DM 1 */ + { 0x8002, 0x0000 }, /* R32770 - DSP3 DM 2 */ + { 0x8003, 0x0000 }, /* R32771 - DSP3 DM 3 */ + { 0x81FC, 0x0000 }, /* R33276 - DSP3 DM 508 */ + { 0x81FD, 0x0000 }, /* R33277 - DSP3 DM 509 */ + { 0x81FE, 0x0000 }, /* R33278 - DSP3 DM 510 */ + { 0x81FF, 0x0000 }, /* R33279 - DSP3 DM 511 */ + { 0x8800, 0x0000 }, /* R34816 - DSP3 PM 0 */ + { 0x8801, 0x0000 }, /* R34817 - DSP3 PM 1 */ + { 0x8802, 0x0000 }, /* R34818 - DSP3 PM 2 */ + { 0x8803, 0x0000 }, /* R34819 - DSP3 PM 3 */ + { 0x8804, 0x0000 }, /* R34820 - DSP3 PM 4 */ + { 0x8805, 0x0000 }, /* R34821 - DSP3 PM 5 */ + { 0x8DFA, 0x0000 }, /* R36346 - DSP3 PM 1530 */ + { 0x8DFB, 0x0000 }, /* R36347 - DSP3 PM 1531 */ + { 0x8DFC, 0x0000 }, /* R36348 - DSP3 PM 1532 */ + { 0x8DFD, 0x0000 }, /* R36349 - DSP3 PM 1533 */ + { 0x8DFE, 0x0000 }, /* R36350 - DSP3 PM 1534 */ + { 0x8DFF, 0x0000 }, /* R36351 - DSP3 PM 1535 */ + { 0x9000, 0x0000 }, /* R36864 - DSP3 ZM 0 */ + { 0x9001, 0x0000 }, /* R36865 - DSP3 ZM 1 */ + { 0x9002, 0x0000 }, /* R36866 - DSP3 ZM 2 */ + { 0x9003, 0x0000 }, /* R36867 - DSP3 ZM 3 */ + { 0x97FC, 0x0000 }, /* R38908 - DSP3 ZM 2044 */ + { 0x97FD, 0x0000 }, /* R38909 - DSP3 ZM 2045 */ + { 0x97FE, 0x0000 }, /* R38910 - DSP3 ZM 2046 */ + { 0x97FF, 0x0000 }, /* R38911 - DSP3 ZM 2047 */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 42d9039a49e9..b2d1f80648ff 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -51,6 +51,7 @@ struct wm5100_fll { /* codec private data */ struct wm5100_priv { + struct regmap *regmap; struct snd_soc_codec *codec; struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; @@ -1375,7 +1376,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, msleep(2); } - codec->cache_only = false; + regcache_cache_only(wm5100->regmap, false); switch (wm5100->rev) { case 0: @@ -1993,6 +1994,9 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = 50; + snd_soc_update_bits(codec, WM5100_CLOCKING_3, WM5100_SYSCLK_ENA, + WM5100_SYSCLK_ENA); + /* Poll for the lock; will use interrupt when we can test */ for (i = 0; i < timeout; i++) { if (i2c->irq) { @@ -2453,8 +2457,9 @@ static int wm5100_probe(struct snd_soc_codec *codec) int ret, i, irq_flags; wm5100->codec = codec; + codec->control_data = wm5100->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2552,7 +2557,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) goto err_reset; } - codec->cache_only = true; + regcache_cache_only(wm5100->regmap, true); wm5100_init_gpio(codec); @@ -2733,14 +2738,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .num_dapm_widgets = ARRAY_SIZE(wm5100_dapm_widgets), .dapm_routes = wm5100_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm5100_dapm_routes), +}; - .reg_cache_size = ARRAY_SIZE(wm5100_reg_defaults), - .reg_word_size = sizeof(u16), - .compress_type = SND_SOC_RBTREE_COMPRESSION, - .reg_cache_default = wm5100_reg_defaults, +static const struct regmap_config wm5100_regmap = { + .reg_bits = 16, + .val_bits = 16, - .volatile_register = wm5100_volatile_register, - .readable_register = wm5100_readable_register, + .max_register = WM5100_MAX_REGISTER, + .reg_defaults = wm5100_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm5100_reg_defaults), + .volatile_reg = wm5100_volatile_register, + .readable_reg = wm5100_readable_register, + .cache_type = REGCACHE_RBTREE, }; static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, @@ -2754,6 +2763,14 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (wm5100 == NULL) return -ENOMEM; + wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); + if (IS_ERR(wm5100->regmap)) { + ret = PTR_ERR(wm5100->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err_alloc; + } + for (i = 0; i < ARRAY_SIZE(wm5100->fll); i++) init_completion(&wm5100->fll[i].lock); @@ -2767,16 +2784,26 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(wm5100_dai)); if (ret < 0) { dev_err(&i2c->dev, "Failed to register WM5100: %d\n", ret); - kfree(wm5100); + goto err_regmap; } return ret; + +err_regmap: + regmap_exit(wm5100->regmap); +err_alloc: + kfree(wm5100); + return ret; } static __devexit int wm5100_i2c_remove(struct i2c_client *client) { + struct wm5100_priv *wm5100 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm5100->regmap); + kfree(wm5100); + return 0; } diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 970759636bdc..25cb6016f9d7 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -15,6 +15,7 @@ #define WM5100_ASOC_H #include +#include int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); @@ -5147,9 +5148,9 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_DSP3_ZM_END_SHIFT 0 /* DSP3_ZM_END - [15:0] */ #define WM5100_DSP3_ZM_END_WIDTH 16 /* DSP3_ZM_END - [15:0] */ -int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg); -int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg); +bool wm5100_readable_register(struct device *dev, unsigned int reg); +bool wm5100_volatile_register(struct device *dev, unsigned int reg); -extern u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1]; +extern struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT]; #endif -- cgit v1.2.1 From 7cfa467b74bb252cc3b74a1a1995c54fe43f90d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Oct 2011 19:49:03 +0100 Subject: ASoC: Convert WM9081 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 278 ++++++++++++++++++++++++++++++---------------- 1 file changed, 180 insertions(+), 98 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7563a91c9ed3..a8fc0659e4ac 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -30,69 +31,61 @@ #include #include "wm9081.h" -static u16 wm9081_reg_defaults[] = { - 0x0000, /* R0 - Software Reset */ - 0x0000, /* R1 */ - 0x00B9, /* R2 - Analogue Lineout */ - 0x00B9, /* R3 - Analogue Speaker PGA */ - 0x0001, /* R4 - VMID Control */ - 0x0068, /* R5 - Bias Control 1 */ - 0x0000, /* R6 */ - 0x0000, /* R7 - Analogue Mixer */ - 0x0000, /* R8 - Anti Pop Control */ - 0x01DB, /* R9 - Analogue Speaker 1 */ - 0x0018, /* R10 - Analogue Speaker 2 */ - 0x0180, /* R11 - Power Management */ - 0x0000, /* R12 - Clock Control 1 */ - 0x0038, /* R13 - Clock Control 2 */ - 0x4000, /* R14 - Clock Control 3 */ - 0x0000, /* R15 */ - 0x0000, /* R16 - FLL Control 1 */ - 0x0200, /* R17 - FLL Control 2 */ - 0x0000, /* R18 - FLL Control 3 */ - 0x0204, /* R19 - FLL Control 4 */ - 0x0000, /* R20 - FLL Control 5 */ - 0x0000, /* R21 */ - 0x0000, /* R22 - Audio Interface 1 */ - 0x0002, /* R23 - Audio Interface 2 */ - 0x0008, /* R24 - Audio Interface 3 */ - 0x0022, /* R25 - Audio Interface 4 */ - 0x0000, /* R26 - Interrupt Status */ - 0x0006, /* R27 - Interrupt Status Mask */ - 0x0000, /* R28 - Interrupt Polarity */ - 0x0000, /* R29 - Interrupt Control */ - 0x00C0, /* R30 - DAC Digital 1 */ - 0x0008, /* R31 - DAC Digital 2 */ - 0x09AF, /* R32 - DRC 1 */ - 0x4201, /* R33 - DRC 2 */ - 0x0000, /* R34 - DRC 3 */ - 0x0000, /* R35 - DRC 4 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 - Write Sequencer 1 */ - 0x0000, /* R39 - Write Sequencer 2 */ - 0x0002, /* R40 - MW Slave 1 */ - 0x0000, /* R41 */ - 0x0000, /* R42 - EQ 1 */ - 0x0000, /* R43 - EQ 2 */ - 0x0FCA, /* R44 - EQ 3 */ - 0x0400, /* R45 - EQ 4 */ - 0x00B8, /* R46 - EQ 5 */ - 0x1EB5, /* R47 - EQ 6 */ - 0xF145, /* R48 - EQ 7 */ - 0x0B75, /* R49 - EQ 8 */ - 0x01C5, /* R50 - EQ 9 */ - 0x169E, /* R51 - EQ 10 */ - 0xF829, /* R52 - EQ 11 */ - 0x07AD, /* R53 - EQ 12 */ - 0x1103, /* R54 - EQ 13 */ - 0x1C58, /* R55 - EQ 14 */ - 0xF373, /* R56 - EQ 15 */ - 0x0A54, /* R57 - EQ 16 */ - 0x0558, /* R58 - EQ 17 */ - 0x0564, /* R59 - EQ 18 */ - 0x0559, /* R60 - EQ 19 */ - 0x4000, /* R61 - EQ 20 */ +static struct reg_default wm9081_reg[] = { + { 0, 0x9081 }, /* R0 - Software Reset */ + { 2, 0x00B9 }, /* R2 - Analogue Lineout */ + { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ + { 4, 0x0001 }, /* R4 - VMID Control */ + { 5, 0x0068 }, /* R5 - Bias Control 1 */ + { 7, 0x0000 }, /* R7 - Analogue Mixer */ + { 8, 0x0000 }, /* R8 - Anti Pop Control */ + { 9, 0x01DB }, /* R9 - Analogue Speaker 1 */ + { 10, 0x0018 }, /* R10 - Analogue Speaker 2 */ + { 11, 0x0180 }, /* R11 - Power Management */ + { 12, 0x0000 }, /* R12 - Clock Control 1 */ + { 13, 0x0038 }, /* R13 - Clock Control 2 */ + { 14, 0x4000 }, /* R14 - Clock Control 3 */ + { 16, 0x0000 }, /* R16 - FLL Control 1 */ + { 17, 0x0200 }, /* R17 - FLL Control 2 */ + { 18, 0x0000 }, /* R18 - FLL Control 3 */ + { 19, 0x0204 }, /* R19 - FLL Control 4 */ + { 20, 0x0000 }, /* R20 - FLL Control 5 */ + { 22, 0x0000 }, /* R22 - Audio Interface 1 */ + { 23, 0x0002 }, /* R23 - Audio Interface 2 */ + { 24, 0x0008 }, /* R24 - Audio Interface 3 */ + { 25, 0x0022 }, /* R25 - Audio Interface 4 */ + { 27, 0x0006 }, /* R27 - Interrupt Status Mask */ + { 28, 0x0000 }, /* R28 - Interrupt Polarity */ + { 29, 0x0000 }, /* R29 - Interrupt Control */ + { 30, 0x00C0 }, /* R30 - DAC Digital 1 */ + { 31, 0x0008 }, /* R31 - DAC Digital 2 */ + { 32, 0x09AF }, /* R32 - DRC 1 */ + { 33, 0x4201 }, /* R33 - DRC 2 */ + { 34, 0x0000 }, /* R34 - DRC 3 */ + { 35, 0x0000 }, /* R35 - DRC 4 */ + { 38, 0x0000 }, /* R38 - Write Sequencer 1 */ + { 39, 0x0000 }, /* R39 - Write Sequencer 2 */ + { 40, 0x0002 }, /* R40 - MW Slave 1 */ + { 42, 0x0000 }, /* R42 - EQ 1 */ + { 43, 0x0000 }, /* R43 - EQ 2 */ + { 44, 0x0FCA }, /* R44 - EQ 3 */ + { 45, 0x0400 }, /* R45 - EQ 4 */ + { 46, 0x00B8 }, /* R46 - EQ 5 */ + { 47, 0x1EB5 }, /* R47 - EQ 6 */ + { 48, 0xF145 }, /* R48 - EQ 7 */ + { 49, 0x0B75 }, /* R49 - EQ 8 */ + { 50, 0x01C5 }, /* R50 - EQ 9 */ + { 51, 0x169E }, /* R51 - EQ 10 */ + { 52, 0xF829 }, /* R52 - EQ 11 */ + { 53, 0x07AD }, /* R53 - EQ 12 */ + { 54, 0x1103 }, /* R54 - EQ 13 */ + { 55, 0x1C58 }, /* R55 - EQ 14 */ + { 56, 0xF373 }, /* R56 - EQ 15 */ + { 57, 0x0A54 }, /* R57 - EQ 16 */ + { 58, 0x0558 }, /* R58 - EQ 17 */ + { 59, 0x0564 }, /* R59 - EQ 18 */ + { 60, 0x0559 }, /* R60 - EQ 19 */ + { 61, 0x4000 }, /* R61 - EQ 20 */ }; static struct { @@ -156,7 +149,7 @@ static struct { }; struct wm9081_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk_source; int mclk_rate; int sysclk_rate; @@ -169,20 +162,84 @@ struct wm9081_priv { struct wm9081_pdata pdata; }; -static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm9081_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: case WM9081_INTERRUPT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -static int wm9081_reset(struct snd_soc_codec *codec) +static bool wm9081_readable_register(struct device *dev, unsigned int reg) { - return snd_soc_write(codec, WM9081_SOFTWARE_RESET, 0); + switch (reg) { + case WM9081_SOFTWARE_RESET: + case WM9081_ANALOGUE_LINEOUT: + case WM9081_ANALOGUE_SPEAKER_PGA: + case WM9081_VMID_CONTROL: + case WM9081_BIAS_CONTROL_1: + case WM9081_ANALOGUE_MIXER: + case WM9081_ANTI_POP_CONTROL: + case WM9081_ANALOGUE_SPEAKER_1: + case WM9081_ANALOGUE_SPEAKER_2: + case WM9081_POWER_MANAGEMENT: + case WM9081_CLOCK_CONTROL_1: + case WM9081_CLOCK_CONTROL_2: + case WM9081_CLOCK_CONTROL_3: + case WM9081_FLL_CONTROL_1: + case WM9081_FLL_CONTROL_2: + case WM9081_FLL_CONTROL_3: + case WM9081_FLL_CONTROL_4: + case WM9081_FLL_CONTROL_5: + case WM9081_AUDIO_INTERFACE_1: + case WM9081_AUDIO_INTERFACE_2: + case WM9081_AUDIO_INTERFACE_3: + case WM9081_AUDIO_INTERFACE_4: + case WM9081_INTERRUPT_STATUS: + case WM9081_INTERRUPT_STATUS_MASK: + case WM9081_INTERRUPT_POLARITY: + case WM9081_INTERRUPT_CONTROL: + case WM9081_DAC_DIGITAL_1: + case WM9081_DAC_DIGITAL_2: + case WM9081_DRC_1: + case WM9081_DRC_2: + case WM9081_DRC_3: + case WM9081_DRC_4: + case WM9081_WRITE_SEQUENCER_1: + case WM9081_WRITE_SEQUENCER_2: + case WM9081_MW_SLAVE_1: + case WM9081_EQ_1: + case WM9081_EQ_2: + case WM9081_EQ_3: + case WM9081_EQ_4: + case WM9081_EQ_5: + case WM9081_EQ_6: + case WM9081_EQ_7: + case WM9081_EQ_8: + case WM9081_EQ_9: + case WM9081_EQ_10: + case WM9081_EQ_11: + case WM9081_EQ_12: + case WM9081_EQ_13: + case WM9081_EQ_14: + case WM9081_EQ_15: + case WM9081_EQ_16: + case WM9081_EQ_17: + case WM9081_EQ_18: + case WM9081_EQ_19: + case WM9081_EQ_20: + return true; + default: + return false; + } +} + +static int wm9081_reset(struct regmap *map) +{ + return regmap_write(map, WM9081_SOFTWARE_RESET, 0x9081); } static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); @@ -1215,25 +1272,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm9081->control_type); + codec->control_data = wm9081->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); - if (reg != 0x9081) { - dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); - ret = -EINVAL; - return ret; - } - - ret = wm9081_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - reg = 0; if (wm9081->pdata.irq_high) reg |= WM9081_IRQ_POL; @@ -1277,15 +1323,9 @@ static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm9081_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM9081_SOFTWARE_RESET) - continue; + struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - snd_soc_write(codec, i, reg_cache[i]); - } + regcache_sync(wm9081->regmap); wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1305,11 +1345,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm9081_reg_defaults, - .volatile_register = wm9081_volatile_register, - .controls = wm9081_snd_controls, .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, @@ -1318,11 +1353,24 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), }; +static const struct regmap_config wm9081_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM9081_MAX_REGISTER, + .reg_defaults = wm9081_reg, + .num_reg_defaults = ARRAY_SIZE(wm9081_reg), + .volatile_reg = wm9081_volatile_register, + .readable_reg = wm9081_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm9081_priv *wm9081; + unsigned int reg; int ret; wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); @@ -1330,7 +1378,30 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm9081); - wm9081->control_type = SND_SOC_I2C; + + wm9081->regmap = regmap_init_i2c(i2c, &wm9081_regmap); + if (IS_ERR(wm9081->regmap)) { + ret = PTR_ERR(wm9081->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = regmap_read(wm9081->regmap, WM9081_SOFTWARE_RESET, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + goto err_regmap; + } + if (reg != 0x9081) { + dev_err(&i2c->dev, "Device is not a WM9081: ID=0x%x\n", reg); + ret = -EINVAL; + goto err_regmap; + } + + ret = wm9081_reset(wm9081->regmap); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_regmap; + } if (dev_get_platdata(&i2c->dev)) memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), @@ -1339,13 +1410,24 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) - kfree(wm9081); + goto err_regmap; + + return 0; + +err_regmap: + regmap_exit(wm9081->regmap); +err: + kfree(wm9081); + return ret; } static __devexit int wm9081_i2c_remove(struct i2c_client *client) { + struct wm9081_priv *wm9081 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm9081->regmap); kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 0469e7b98cde0579d16ce5868eccccfec1bc043e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Nov 2011 15:22:09 +0000 Subject: ASoC: Disable debounce on some WM8962 interrupts Allow them to work when the device is unclocked. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6d82b35a70d0..2ae04ba4ab60 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4038,6 +4038,12 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Stereo control for EQ */ snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); + /* Don't debouce interrupts so we don't need SYSCLK */ + snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE, + WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | + WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, + 0); + wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ -- cgit v1.2.1 From 7d6f6b0f39c41f681e152cd9b1e54246ddb73028 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 27 Oct 2011 09:46:53 +0200 Subject: ASoC: Convert wm8971 MICBIAS to a supply widget Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b444b297d0b2..3a06a95dd96f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -224,7 +224,7 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0), SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", WM8971_PWR1, 1, 0, NULL, 0), SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0), SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), -- cgit v1.2.1 From b402735883c95c270ac42c40370a2663c2c81371 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 9 Nov 2011 17:00:05 +0800 Subject: ASoC: wm9081: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 81 ++++++++++++++++++++--------------------------- 1 file changed, 35 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b491ae19b84b..f7c0738a9da6 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -826,84 +826,74 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP, 0); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source and disable bias */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - reg &= ~WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC); /* Disable VMID with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1291,11 +1281,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, -- cgit v1.2.1 From 60bf5b072826cd76537071d7464e9fd74ea49350 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Nov 2011 16:29:07 +0000 Subject: ASoC: Need to convert wm5100 cache sync to direct regmap usage too ASoC knows nothing about the cache now. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b2d1f80648ff..340ffe20b53b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1400,7 +1400,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, break; } - snd_soc_cache_sync(codec); + regcache_sync(wm5100->regmap); } break; -- cgit v1.2.1 From d926b5a3d921decc0fc537ba8a5ad53350c78f82 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Nov 2011 16:08:48 +0000 Subject: ASoC: Mark WM5100 MISC CONTROL as readable Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 6b2ab65735de..3e90dea4e267 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -85,6 +85,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_MIC_DETECT_1: case WM5100_MIC_DETECT_2: case WM5100_MIC_DETECT_3: + case WM5100_MISC_CONTROL: case WM5100_INPUT_ENABLES: case WM5100_INPUT_ENABLES_STATUS: case WM5100_IN1L_CONTROL: -- cgit v1.2.1 From 588ac5e0b63da9cdef8b1b1d71dbcd95a8a94131 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Nov 2011 16:12:04 +0000 Subject: ASoC: Move most WM5100 resource allocation to I2C probe More standard Linuxish. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 263 +++++++++++++++++++++++----------------------- 1 file changed, 131 insertions(+), 132 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 340ffe20b53b..08bf073ce19f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -205,17 +205,15 @@ static void wm5100_free_sr(struct snd_soc_codec *codec, int rate) } } -static int wm5100_reset(struct snd_soc_codec *codec) +static int wm5100_reset(struct wm5100_priv *wm5100) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_set_value_cansleep(wm5100->pdata.reset, 1); return 0; } else { - return snd_soc_write(codec, WM5100_SOFTWARE_RESET, 0); + return regmap_write(wm5100->regmap, WM5100_SOFTWARE_RESET, 0); } } @@ -2465,98 +2463,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) - wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; - - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request core supplies: %d\n", - ret); - return ret; - } - - wm5100->cpvdd = regulator_get(&i2c->dev, "CPVDD"); - if (IS_ERR(wm5100->cpvdd)) { - ret = PTR_ERR(wm5100->cpvdd); - dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); - goto err_core; - } - - wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); - if (IS_ERR(wm5100->dbvdd2)) { - ret = PTR_ERR(wm5100->dbvdd2); - dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); - goto err_cpvdd; - } - - wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); - if (IS_ERR(wm5100->dbvdd3)) { - ret = PTR_ERR(wm5100->dbvdd3); - dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); - goto err_dbvdd2; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable core supplies: %d\n", - ret); - goto err_dbvdd3; - } - - if (wm5100->pdata.ldo_ena) { - ret = gpio_request_one(wm5100->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM5100 LDOENA"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", - wm5100->pdata.ldo_ena, ret); - goto err_enable; - } - msleep(2); - } - - if (wm5100->pdata.reset) { - ret = gpio_request_one(wm5100->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM5100 /RESET"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", - wm5100->pdata.reset, ret); - goto err_ldo; - } - } - - ret = snd_soc_read(codec, WM5100_SOFTWARE_RESET); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_reset; - } - switch (ret) { - case 0x8997: - case 0x5100: - break; - - default: - dev_err(codec->dev, "Device is not a WM5100, ID is %x\n", ret); - ret = -EINVAL; - goto err_reset; - } - - ret = snd_soc_read(codec, WM5100_DEVICE_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read revision register\n"); - goto err_reset; - } - wm5100->rev = ret & WM5100_DEVICE_REVISION_MASK; - - dev_info(codec->dev, "revision %c\n", wm5100->rev + 'A'); - - ret = wm5100_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_reset; - } - regcache_cache_only(wm5100->regmap, true); wm5100_init_gpio(codec); @@ -2668,28 +2574,6 @@ err_gpio: if (i2c->irq) free_irq(i2c->irq, codec); wm5100_free_gpio(codec); -err_reset: - if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); - gpio_free(wm5100->pdata.reset); - } -err_ldo: - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - gpio_free(wm5100->pdata.ldo_ena); - } -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); -err_dbvdd3: - regulator_put(wm5100->dbvdd3); -err_dbvdd2: - regulator_put(wm5100->dbvdd2); -err_cpvdd: - regulator_put(wm5100->cpvdd); -err_core: - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); return ret; } @@ -2706,19 +2590,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); wm5100_free_gpio(codec); - if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); - gpio_free(wm5100->pdata.reset); - } - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - gpio_free(wm5100->pdata.ldo_ena); - } - regulator_put(wm5100->dbvdd3); - regulator_put(wm5100->dbvdd2); - regulator_put(wm5100->cpvdd); - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); return 0; } @@ -2757,6 +2628,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, { struct wm5100_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm5100_priv *wm5100; + unsigned int reg; int ret, i; wm5100 = kzalloc(sizeof(struct wm5100_priv), GFP_KERNEL); @@ -2779,16 +2651,130 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm5100); + for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) + wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request core supplies: %d\n", + ret); + goto err_regmap; + } + + wm5100->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm5100->cpvdd)) { + ret = PTR_ERR(wm5100->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_core; + } + + wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); + if (IS_ERR(wm5100->dbvdd2)) { + ret = PTR_ERR(wm5100->dbvdd2); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_cpvdd; + } + + wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); + if (IS_ERR(wm5100->dbvdd3)) { + ret = PTR_ERR(wm5100->dbvdd3); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_dbvdd2; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", + ret); + goto err_dbvdd3; + } + + if (wm5100->pdata.ldo_ena) { + ret = gpio_request_one(wm5100->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, "WM5100 LDOENA"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", + wm5100->pdata.ldo_ena, ret); + goto err_enable; + } + msleep(2); + } + + if (wm5100->pdata.reset) { + ret = gpio_request_one(wm5100->pdata.reset, + GPIOF_OUT_INIT_HIGH, "WM5100 /RESET"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", + wm5100->pdata.reset, ret); + goto err_ldo; + } + } + + ret = regmap_read(wm5100->regmap, WM5100_SOFTWARE_RESET, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register\n"); + goto err_reset; + } + switch (reg) { + case 0x8997: + case 0x5100: + break; + + default: + dev_err(&i2c->dev, "Device is not a WM5100, ID is %x\n", reg); + ret = -EINVAL; + goto err_reset; + } + + ret = regmap_read(wm5100->regmap, WM5100_DEVICE_REVISION, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read revision register\n"); + goto err_reset; + } + wm5100->rev = reg & WM5100_DEVICE_REVISION_MASK; + + dev_info(&i2c->dev, "revision %c\n", wm5100->rev + 'A'); + + ret = wm5100_reset(wm5100); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_reset; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); if (ret < 0) { dev_err(&i2c->dev, "Failed to register WM5100: %d\n", ret); - goto err_regmap; + goto err_reset; } return ret; +err_reset: + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } +err_ldo: + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); +err_dbvdd3: + regulator_put(wm5100->dbvdd3); +err_dbvdd2: + regulator_put(wm5100->dbvdd2); +err_cpvdd: + regulator_put(wm5100->cpvdd); +err_core: + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); err_alloc: @@ -2801,6 +2787,19 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) struct wm5100_priv *wm5100 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } + regulator_put(wm5100->dbvdd3); + regulator_put(wm5100->dbvdd2); + regulator_put(wm5100->cpvdd); + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); regmap_exit(wm5100->regmap); kfree(wm5100); -- cgit v1.2.1 From abda5dfdd56e548a7c569a40c404d8679c4f35f1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 23 Aug 2011 17:40:01 +0100 Subject: ASoC: Add Lowland machine driver The Lowland platform is based on the Cragganmore system like Speyside but uses the WM5100 audio CODEC. Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 7 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/lowland.c | 246 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 255 insertions(+) create mode 100644 sound/soc/samsung/lowland.c (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 53aaa69eda03..71f38de18222 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -198,3 +198,10 @@ config SND_SOC_SPEYSIDE_WM8962 depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8962 + +config SND_SOC_LOWLAND + tristate "Audio support for Wolfson Lowland" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM5100 + select SND_SOC_WM9081 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8509d3c4366e..7802c25db775 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -40,6 +40,7 @@ snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-speyside-wm8962-objs := speyside_wm8962.o +snd-soc-lowland-objs := lowland.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -61,3 +62,4 @@ obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o +obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c new file mode 100644 index 000000000000..eff1b4b65df4 --- /dev/null +++ b/sound/soc/samsung/lowland.c @@ -0,0 +1,246 @@ +/* + * Lowland audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include + +#include "../codecs/wm5100.h" +#include "../codecs/wm9081.h" + +#define MCLK1_RATE (44100 * 512) +#define CLKOUT_RATE (44100 * 256) + +static int lowland_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops lowland_ops = { + .hw_params = lowland_hw_params, +}; + +static struct snd_soc_jack lowland_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin lowland_headset_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_codec_set_sysclk(codec, WM5100_CLK_SYSCLK, + WM5100_CLKSRC_MCLK1, MCLK1_RATE, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK clock source: %d\n", ret); + return ret; + } + + /* Clock OPCLK, used by the other audio components. */ + ret = snd_soc_codec_set_sysclk(codec, WM5100_CLK_OPCLK, 0, + CLKOUT_RATE, 0); + if (ret < 0) { + pr_err("Failed to set OPCLK rate: %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0, + &lowland_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&lowland_headset, + ARRAY_SIZE(lowland_headset_pins), + lowland_headset_pins); + if (ret) + return ret; + + wm5100_detect(codec, &lowland_headset); + + return 0; +} + +static struct snd_soc_dai_link lowland_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5100-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5100.1-001a", + .ops = &lowland_ops, + .init = lowland_wm5100_init, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5100-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .ops = &lowland_ops, + .ignore_suspend = 1, + }, +}; + +static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm) +{ + snd_soc_dapm_nc_pin(dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + CLKOUT_RATE, 0); +} + +static struct snd_soc_aux_dev lowland_aux_dev[] = { + { + .name = "wm9081", + .codec_name = "wm9081.1-006c", + .init = lowland_wm9081_init, + }, +}; + +static struct snd_soc_codec_conf lowland_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("Main DMIC"), + SOC_DAPM_PIN_SWITCH("Main AMIC"), + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), + + SND_SOC_DAPM_MIC("Main AMIC", NULL), + SND_SOC_DAPM_MIC("Main DMIC", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Sub IN1", NULL, "HPOUT2L" }, + { "Sub IN2", NULL, "HPOUT2R" }, + + { "Main Speaker", NULL, "Sub SPKN" }, + { "Main Speaker", NULL, "Sub SPKP" }, + { "Main Speaker", NULL, "SPKDAT1" }, +}; + +static struct snd_soc_card lowland = { + .name = "Lowland", + .dai_link = lowland_dai, + .num_links = ARRAY_SIZE(lowland_dai), + .aux_dev = lowland_aux_dev, + .num_aux_devs = ARRAY_SIZE(lowland_aux_dev), + .codec_conf = lowland_codec_conf, + .num_configs = ARRAY_SIZE(lowland_codec_conf), + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), +}; + +static __devinit int lowland_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &lowland; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit lowland_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver lowland_driver = { + .driver = { + .name = "lowland", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = lowland_probe, + .remove = __devexit_p(lowland_remove), +}; + +static int __init lowland_audio_init(void) +{ + return platform_driver_register(&lowland_driver); +} +module_init(lowland_audio_init); + +static void __exit lowland_audio_exit(void) +{ + platform_driver_unregister(&lowland_driver); +} +module_exit(lowland_audio_exit); + +MODULE_DESCRIPTION("Lowland audio support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:lowland"); -- cgit v1.2.1 From 9db16e4c1b21abe5bfc15b6a14824acc0ce0d594 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Nov 2011 17:27:28 +0000 Subject: ASoC: Convert WM5100 gpiolib support to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 52 +++++++++++++++++++++++------------------------ 1 file changed, 25 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 08bf073ce19f..0077086d8e5b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2352,24 +2352,22 @@ static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; - snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); } static int wm5100_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); - ret = snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_FN_MASK | WM5100_GP1_DIR | - WM5100_GP1_LVL, val); + ret = regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR | + WM5100_GP1_LVL, val); if (ret < 0) return ret; else @@ -2379,25 +2377,24 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; + unsigned int reg; int ret; - ret = snd_soc_read(codec, WM5100_GPIO_CTRL_1 + offset); + ret = regmap_read(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, ®); if (ret < 0) return ret; - return (ret & WM5100_GP1_LVL) != 0; + return (reg & WM5100_GP1_LVL) != 0; } static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; - return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_FN_MASK | WM5100_GP1_DIR, - (1 << WM5100_GP1_FN_SHIFT) | - (1 << WM5100_GP1_DIR_SHIFT)); + return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR, + (1 << WM5100_GP1_FN_SHIFT) | + (1 << WM5100_GP1_DIR_SHIFT)); } static struct gpio_chip wm5100_template_chip = { @@ -2410,14 +2407,14 @@ static struct gpio_chip wm5100_template_chip = { .can_sleep = 1, }; -static void wm5100_init_gpio(struct snd_soc_codec *codec) +static void wm5100_init_gpio(struct i2c_client *i2c) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); int ret; wm5100->gpio_chip = wm5100_template_chip; wm5100->gpio_chip.ngpio = 6; - wm5100->gpio_chip.dev = codec->dev; + wm5100->gpio_chip.dev = &i2c->dev; if (wm5100->pdata.gpio_base) wm5100->gpio_chip.base = wm5100->pdata.gpio_base; @@ -2426,24 +2423,24 @@ static void wm5100_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm5100->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm5100_free_gpio(struct snd_soc_codec *codec) +static void wm5100_free_gpio(struct i2c_client *i2c) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); int ret; ret = gpiochip_remove(&wm5100->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm5100_init_gpio(struct snd_soc_codec *codec) +static void wm5100_init_gpio(struct i2c_client *i2c) { } -static void wm5100_free_gpio(struct snd_soc_codec *codec) +static void wm5100_free_gpio(struct i2c_client *i2c) { } #endif @@ -2465,7 +2462,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) regcache_cache_only(wm5100->regmap, true); - wm5100_init_gpio(codec); for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, @@ -2573,7 +2569,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) err_gpio: if (i2c->irq) free_irq(i2c->irq, codec); - wm5100_free_gpio(codec); return ret; } @@ -2589,7 +2584,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) } if (i2c->irq) free_irq(i2c->irq, codec); - wm5100_free_gpio(codec); return 0; } @@ -2743,6 +2737,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, goto err_reset; } + wm5100_init_gpio(i2c); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2754,6 +2750,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 1); gpio_free(wm5100->pdata.reset); @@ -2787,6 +2784,7 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) struct wm5100_priv *wm5100 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + wm5100_free_gpio(client); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 1); gpio_free(wm5100->pdata.reset); -- cgit v1.2.1 From f4034147259f72cb7c4870a4188bd8beb592f87d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Nov 2011 23:15:26 +0000 Subject: ASoC: Fix duplicate const warnings in da7210.c Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index b545b7d37222..8b5848a6374c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -240,7 +240,7 @@ static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ -static const char const *da7210_hpf_cutoff_txt[] = { +static const char * const da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; @@ -251,7 +251,7 @@ static const struct soc_enum da7210_adc_hpf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ -static const char const *da7210_vf_cutoff_txt[] = { +static const char * const da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -- cgit v1.2.1 From 94d5f7c0255bd712d68732a0180558d45fe6eac5 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Sat, 5 Nov 2011 12:38:02 +0200 Subject: ASoC: Add new Realtek ALC5632 CODEC driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky Signed-off-by: Andrey Danin Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1153 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5632.h | 249 ++++++++++ 4 files changed, 1407 insertions(+) create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514d93d4..684cc1570689 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -168,6 +169,8 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a7c415dc22fe..af64905f36ca 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -31,6 +31,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 000000000000..ee6a497b5e71 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1153 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team +* +* Authors: Leon Romanovsky +* Andrey Danin +* Ilya Petrov +* Marc Dietrich +* +* Based on alc5623.c by Arnaud Patard +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +*/ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "alc5632.h" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +/* -34.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +/* -46.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +/* -16.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +/* 0db min scale, 6 db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); +/* 0db min scalem 0.75db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + /* left starts at bit 8, right at bit 0 */ + /* 31 steps (5 bit), -46.5db scale */ + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Playback Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +/* speaker amplifier */ +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); + + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("DAC Right Channel", + ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0, + &alc5632_amp_mux_controls), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + + +static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* DAC */ + {"DAC Right Channel", NULL, "I2S Mix"}, + {"DAC Left Channel", NULL, "I2S Mix"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, + {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + + + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"Left Speaker", NULL, "AB-D Amp Mux"}, + {"Right Speaker", NULL, "AB-D Amp Mux"}, + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, + +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -EINVAL; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + /* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); + /* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); + /* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); + /* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); + /* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + |ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) + +#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5632_ADD1_POWER_EN \ + (ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_ADD3_POWER_EN, + ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + /* "normal" mode: 0 @ 26 */ + /* set all PR0-7 mixers to 0 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0); + + msleep(500); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_ADD2_POWER_EN, + ALC5632_ADD2_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_ADD1_POWER_EN, + ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, + ALC5632_PWR_ADD2_VREF); + /* "normal" mode: 0 @ 26 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0xffff ^ (ALC5632_PWR_VREF_PR3 + | ALC5632_PWR_VREF_PR2)); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_PWR_MANAG_ADD3_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, + .symmetric_rates = 1, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int ret; + + /* mark cache as needed to sync */ + codec->cache_sync = 1; + + ret = snd_soc_cache_sync(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, + .controls = alc5632_snd_controls, + .num_controls = ARRAY_SIZE(alc5632_snd_controls), + .dapm_widgets = alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets), + .dapm_routes = alc5632_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + alc5632 = devm_kzalloc(&client->dev, + sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 000000000000..ff4c0fd0d2ec --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,249 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team +* +* Authors: Leon Romanovsky +* Andrey Danin +* Ilya Petrov +* Marc Dietrich +* +* Based on alc5623.h by Arnaud Patard +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +*/ + +#ifndef _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 /* spe out vol */ +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 /* hp out vol */ +#define ALC5632_AUX_OUT_VOL 0x06 /* aux out vol */ +#define ALC5632_PHONE_IN_VOL 0x08 /* phone in vol */ +#define ALC5632_LINE_IN_VOL 0x0A /* line in vol */ +#define ALC5632_STEREO_DAC_IN_VOL 0x0C /* stereo dac in vol */ +#define ALC5632_MIC_VOL 0x0E /* mic in vol */ +/* stero dac/mic routing */ +#define ALC5632_MIC_ROUTING_CTRL 0x10 +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 /* rec gain */ +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE (-16.5) +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 /* mixer control */ +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 /* mic phone ctrl */ +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 /* digi mic / bost ctl */ +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_DOWN_CTRL_STATUS_MASK 0xEF00 +#define ALC5632_PWR_VREF_PR3 (1 << 11) +#define ALC5632_PWR_VREF_PR2 (1 << 10) +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +/* stereo/voice DAC / stereo adc func ctrl */ +#define ALC5632_DAC_FUNC_SELECT 0x2E + +/* Main serial data port ctrl (i2s) */ +#define ALC5632_DAI_CONTROL 0x34 + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +/* 0:voice, 1:main */ +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) +/* extend serial data port control (VoDAC_i2c/pcm) */ +#define ALC5632_DAI_CONTROL2 0x36 +/* 0:gpio func, 1:voice pcm */ +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) +/* 0:master, 1:slave */ +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) +/* 0:disable, 1:enable */ +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) +/* 0:main, 1:voice */ +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_MANAG_ADD1_MASK 0xEFFF +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_MANAG_ADD2_MASK 0x7FFF +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_MANAG_ADD3_MASK 0x7CFF +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- cgit v1.2.1 From c9016a7937122b72d87ff2037664b7bd717d3e4b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Nov 2011 13:06:52 +0000 Subject: ASoC: Remove LZO cache type There are no current users and new drivers ought to be using the regmap API and its cache implementation directly so just delete the ASoC copy. Signed-off-by: Mark Brown --- sound/soc/Kconfig | 15 -- sound/soc/soc-cache.c | 384 -------------------------------------------------- 2 files changed, 399 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 1381db853ef0..35e662d270e6 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,21 +22,6 @@ menuconfig SND_SOC if SND_SOC -config SND_SOC_CACHE_LZO - bool "Support LZO compression for register caches" - select LZO_COMPRESS - select LZO_DECOMPRESS - ---help--- - Select this to enable LZO compression for register caches. - This will allow machine or CODEC drivers to compress register - caches in memory, reducing the memory consumption at the - expense of performance. If this is not present and is used - the system will fall back to uncompressed caches. - - Usually it is safe to disable this option, where cache - compression in used the rbtree option will typically perform - better. - config SND_SOC_AC97_BUS bool diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4b3b4e..18bb6b3335e0 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include @@ -439,378 +438,6 @@ err: return ret; } -#ifdef CONFIG_SND_SOC_CACHE_LZO -struct snd_soc_lzo_ctx { - void *wmem; - void *dst; - const void *src; - size_t src_len; - size_t dst_len; - size_t decompressed_size; - unsigned long *sync_bmp; - int sync_bmp_nbits; -}; - -#define LZO_BLOCK_NUM 8 -static int snd_soc_lzo_block_count(void) -{ - return LZO_BLOCK_NUM; -} - -static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx) -{ - lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL); - if (!lzo_ctx->wmem) - return -ENOMEM; - return 0; -} - -static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t compress_size; - int ret; - - ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &compress_size, lzo_ctx->wmem); - if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len) - return -EINVAL; - lzo_ctx->dst_len = compress_size; - return 0; -} - -static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t dst_len; - int ret; - - dst_len = lzo_ctx->dst_len; - ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &dst_len); - if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len) - return -EINVAL; - return 0; -} - -static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE); - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_compress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo_ctx->decompressed_size; - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_decompress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / - codec_drv->reg_word_size); -} - -static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) -{ - return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - unsigned int val; - int i; - int ret; - - lzo_blocks = codec->reg_cache; - for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(!snd_soc_codec_writable_register(codec, i)); - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - codec->cache_bypass = 1; - ret = snd_soc_write(codec, i, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - i, val); - } - - return 0; -} - -static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - goto out; - } - - /* write the new value to the cache */ - if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value, - codec->driver->reg_word_size)) { - kfree(lzo_block->dst); - goto out; - } - - /* prepare the source to be the decompressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* compress the block */ - ret = snd_soc_lzo_compress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - kfree(lzo_block->src); - goto out; - } - - /* set the bit so we know we have to sync this register */ - set_bit(reg, lzo_block->sync_bmp); - kfree(tmp_dst); - kfree(lzo_block->src); - return 0; -out: - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return ret; -} - -static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - *value = 0; - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret >= 0) - /* fetch the value from the cache */ - *value = snd_soc_get_cache_val(lzo_block->dst, blkpos, - codec->driver->reg_word_size); - - kfree(lzo_block->dst); - /* restore the pointer and length of the compressed block */ - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return 0; -} - -static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - int i, blkcount; - - lzo_blocks = codec->reg_cache; - if (!lzo_blocks) - return 0; - - blkcount = snd_soc_lzo_block_count(); - /* - * the pointer to the bitmap used for syncing the cache - * is shared amongst all lzo_blocks. Ensure it is freed - * only once. - */ - if (lzo_blocks[0]) - kfree(lzo_blocks[0]->sync_bmp); - for (i = 0; i < blkcount; ++i) { - if (lzo_blocks[i]) { - kfree(lzo_blocks[i]->wmem); - kfree(lzo_blocks[i]->dst); - } - /* each lzo_block is a pointer returned by kmalloc or NULL */ - kfree(lzo_blocks[i]); - } - kfree(lzo_blocks); - codec->reg_cache = NULL; - return 0; -} - -static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - size_t bmp_size; - const struct snd_soc_codec_driver *codec_drv; - int ret, tofree, i, blksize, blkcount; - const char *p, *end; - unsigned long *sync_bmp; - - ret = 0; - codec_drv = codec->driver; - - /* - * If we have not been given a default register cache - * then allocate a dummy zero-ed out region, compress it - * and remember to free it afterwards. - */ - tofree = 0; - if (!codec->reg_def_copy) - tofree = 1; - - if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) - return -ENOMEM; - } - - blkcount = snd_soc_lzo_block_count(); - codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks, - GFP_KERNEL); - if (!codec->reg_cache) { - ret = -ENOMEM; - goto err_tofree; - } - lzo_blocks = codec->reg_cache; - - /* - * allocate a bitmap to be used when syncing the cache with - * the hardware. Each time a register is modified, the corresponding - * bit is set in the bitmap, so we know that we have to sync - * that register. - */ - bmp_size = codec_drv->reg_cache_size; - sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long), - GFP_KERNEL); - if (!sync_bmp) { - ret = -ENOMEM; - goto err; - } - bitmap_zero(sync_bmp, bmp_size); - - /* allocate the lzo blocks and initialize them */ - for (i = 0; i < blkcount; ++i) { - lzo_blocks[i] = kzalloc(sizeof **lzo_blocks, - GFP_KERNEL); - if (!lzo_blocks[i]) { - kfree(sync_bmp); - ret = -ENOMEM; - goto err; - } - lzo_blocks[i]->sync_bmp = sync_bmp; - lzo_blocks[i]->sync_bmp_nbits = bmp_size; - /* alloc the working space for the compressed block */ - ret = snd_soc_lzo_prepare(lzo_blocks[i]); - if (ret < 0) - goto err; - } - - blksize = snd_soc_lzo_get_blksize(codec); - p = codec->reg_def_copy; - end = codec->reg_def_copy + codec->reg_size; - /* compress the register map and fill the lzo blocks */ - for (i = 0; i < blkcount; ++i, p += blksize) { - lzo_blocks[i]->src = p; - if (p + blksize > end) - lzo_blocks[i]->src_len = end - p; - else - lzo_blocks[i]->src_len = blksize; - ret = snd_soc_lzo_compress_cache_block(codec, - lzo_blocks[i]); - if (ret < 0) - goto err; - lzo_blocks[i]->decompressed_size = - lzo_blocks[i]->src_len; - } - - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return 0; -err: - snd_soc_cache_exit(codec); -err_tofree: - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return ret; -} -#endif - static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; @@ -889,17 +516,6 @@ static const struct snd_soc_cache_ops cache_types[] = { .write = snd_soc_flat_cache_write, .sync = snd_soc_flat_cache_sync }, -#ifdef CONFIG_SND_SOC_CACHE_LZO - { - .id = SND_SOC_LZO_COMPRESSION, - .name = "LZO", - .init = snd_soc_lzo_cache_init, - .exit = snd_soc_lzo_cache_exit, - .read = snd_soc_lzo_cache_read, - .write = snd_soc_lzo_cache_write, - .sync = snd_soc_lzo_cache_sync - }, -#endif { .id = SND_SOC_RBTREE_COMPRESSION, .name = "rbtree", -- cgit v1.2.1 From d9b5e9c6bccc3850b91ddaac11b49f2510375f5b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Nov 2011 16:14:04 +0000 Subject: ASoC: Move WM5100 platform data based setup into I2C probe Get things configured as early as possible, especially useful for the GPIOs which might be useful anyway. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 0077086d8e5b..f37d67f4058b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2467,24 +2467,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, WM5100_OUT_VU); - for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { - snd_soc_update_bits(codec, WM5100_IN1L_CONTROL, - WM5100_IN1_MODE_MASK | - WM5100_IN1_DMIC_SUP_MASK, - (wm5100->pdata.in_mode[i] << - WM5100_IN1_MODE_SHIFT) | - (wm5100->pdata.dmic_sup[i] << - WM5100_IN1_DMIC_SUP_SHIFT)); - } - - for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { - if (!wm5100->pdata.gpio_defaults[i]) - continue; - - snd_soc_write(codec, WM5100_GPIO_CTRL_1 + i, - wm5100->pdata.gpio_defaults[i]); - } - /* Don't debounce interrupts to support use of SYSCLK only */ snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_1, 0); snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_2, 0); @@ -2739,6 +2721,24 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100_init_gpio(i2c); + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { + if (!wm5100->pdata.gpio_defaults[i]) + continue; + + regmap_write(wm5100->regmap, WM5100_GPIO_CTRL_1 + i, + wm5100->pdata.gpio_defaults[i]); + } + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { + regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL, + WM5100_IN1_MODE_MASK | + WM5100_IN1_DMIC_SUP_MASK, + (wm5100->pdata.in_mode[i] << + WM5100_IN1_MODE_SHIFT) | + (wm5100->pdata.dmic_sup[i] << + WM5100_IN1_DMIC_SUP_SHIFT)); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); -- cgit v1.2.1 From c42da64293b81463e9d3d1a74254f3425509a29b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Nov 2011 17:15:54 +0000 Subject: ASoC: Convert WM8995 to direct regmap usage Large code size increase due to the addition of readability information and the reformatting of the defaults table. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 717 +++++++++++++++++++++++++++++++++++++++------- 1 file changed, 608 insertions(+), 109 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4d109b1ad124..3774acb69ddd 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -43,88 +44,331 @@ static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = { "MICVDD" }; -static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { - [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, - [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, - [28] = 0x000f, [32] = 0x0005, [33] = 0x0005, [40] = 0x0003, - [41] = 0x0013, [48] = 0x0004, [56] = 0x09f8, [64] = 0x1f25, - [69] = 0x0004, [82] = 0xaaaa, [84] = 0x2a2a, [146] = 0x0060, - [256] = 0x0002, [257] = 0x8004, [520] = 0x0010, [528] = 0x0083, - [529] = 0x0083, [548] = 0x0c80, [580] = 0x0c80, [768] = 0x4050, - [769] = 0x4000, [771] = 0x0040, [772] = 0x0040, [773] = 0x0040, - [774] = 0x0004, [775] = 0x0100, [784] = 0x4050, [785] = 0x4000, - [787] = 0x0040, [788] = 0x0040, [789] = 0x0040, [1024] = 0x00c0, - [1025] = 0x00c0, [1026] = 0x00c0, [1027] = 0x00c0, [1028] = 0x00c0, - [1029] = 0x00c0, [1030] = 0x00c0, [1031] = 0x00c0, [1056] = 0x0200, - [1057] = 0x0010, [1058] = 0x0200, [1059] = 0x0010, [1088] = 0x0098, - [1089] = 0x0845, [1104] = 0x0098, [1105] = 0x0845, [1152] = 0x6318, - [1153] = 0x6300, [1154] = 0x0fca, [1155] = 0x0400, [1156] = 0x00d8, - [1157] = 0x1eb5, [1158] = 0xf145, [1159] = 0x0b75, [1160] = 0x01c5, - [1161] = 0x1c58, [1162] = 0xf373, [1163] = 0x0a54, [1164] = 0x0558, - [1165] = 0x168e, [1166] = 0xf829, [1167] = 0x07ad, [1168] = 0x1103, - [1169] = 0x0564, [1170] = 0x0559, [1171] = 0x4000, [1184] = 0x6318, - [1185] = 0x6300, [1186] = 0x0fca, [1187] = 0x0400, [1188] = 0x00d8, - [1189] = 0x1eb5, [1190] = 0xf145, [1191] = 0x0b75, [1192] = 0x01c5, - [1193] = 0x1c58, [1194] = 0xf373, [1195] = 0x0a54, [1196] = 0x0558, - [1197] = 0x168e, [1198] = 0xf829, [1199] = 0x07ad, [1200] = 0x1103, - [1201] = 0x0564, [1202] = 0x0559, [1203] = 0x4000, [1280] = 0x00c0, - [1281] = 0x00c0, [1282] = 0x00c0, [1283] = 0x00c0, [1312] = 0x0200, - [1313] = 0x0010, [1344] = 0x0098, [1345] = 0x0845, [1408] = 0x6318, - [1409] = 0x6300, [1410] = 0x0fca, [1411] = 0x0400, [1412] = 0x00d8, - [1413] = 0x1eb5, [1414] = 0xf145, [1415] = 0x0b75, [1416] = 0x01c5, - [1417] = 0x1c58, [1418] = 0xf373, [1419] = 0x0a54, [1420] = 0x0558, - [1421] = 0x168e, [1422] = 0xf829, [1423] = 0x07ad, [1424] = 0x1103, - [1425] = 0x0564, [1426] = 0x0559, [1427] = 0x4000, [1568] = 0x0002, - [1792] = 0xa100, [1793] = 0xa101, [1794] = 0xa101, [1795] = 0xa101, - [1796] = 0xa101, [1797] = 0xa101, [1798] = 0xa101, [1799] = 0xa101, - [1800] = 0xa101, [1801] = 0xa101, [1802] = 0xa101, [1803] = 0xa101, - [1804] = 0xa101, [1805] = 0xa101, [1825] = 0x0055, [1848] = 0x3fff, - [1849] = 0x1fff, [2049] = 0x0001, [2050] = 0x0069, [2056] = 0x0002, - [2057] = 0x0003, [2058] = 0x0069, [12288] = 0x0001, [12289] = 0x0001, - [12291] = 0x0006, [12292] = 0x0040, [12293] = 0x0001, [12294] = 0x000f, - [12295] = 0x0006, [12296] = 0x0001, [12297] = 0x0003, [12298] = 0x0104, - [12300] = 0x0060, [12301] = 0x0011, [12302] = 0x0401, [12304] = 0x0050, - [12305] = 0x0003, [12306] = 0x0100, [12308] = 0x0051, [12309] = 0x0003, - [12310] = 0x0104, [12311] = 0x000a, [12312] = 0x0060, [12313] = 0x003b, - [12314] = 0x0502, [12315] = 0x0100, [12316] = 0x2fff, [12320] = 0x2fff, - [12324] = 0x2fff, [12328] = 0x2fff, [12332] = 0x2fff, [12336] = 0x2fff, - [12340] = 0x2fff, [12344] = 0x2fff, [12348] = 0x2fff, [12352] = 0x0001, - [12353] = 0x0001, [12355] = 0x0006, [12356] = 0x0040, [12357] = 0x0001, - [12358] = 0x000f, [12359] = 0x0006, [12360] = 0x0001, [12361] = 0x0003, - [12362] = 0x0104, [12364] = 0x0060, [12365] = 0x0011, [12366] = 0x0401, - [12368] = 0x0050, [12369] = 0x0003, [12370] = 0x0100, [12372] = 0x0060, - [12373] = 0x003b, [12374] = 0x0502, [12375] = 0x0100, [12376] = 0x2fff, - [12380] = 0x2fff, [12384] = 0x2fff, [12388] = 0x2fff, [12392] = 0x2fff, - [12396] = 0x2fff, [12400] = 0x2fff, [12404] = 0x2fff, [12408] = 0x2fff, - [12412] = 0x2fff, [12416] = 0x0001, [12417] = 0x0001, [12419] = 0x0006, - [12420] = 0x0040, [12421] = 0x0001, [12422] = 0x000f, [12423] = 0x0006, - [12424] = 0x0001, [12425] = 0x0003, [12426] = 0x0106, [12428] = 0x0061, - [12429] = 0x0011, [12430] = 0x0401, [12432] = 0x0050, [12433] = 0x0003, - [12434] = 0x0102, [12436] = 0x0051, [12437] = 0x0003, [12438] = 0x0106, - [12439] = 0x000a, [12440] = 0x0061, [12441] = 0x003b, [12442] = 0x0502, - [12443] = 0x0100, [12444] = 0x2fff, [12448] = 0x2fff, [12452] = 0x2fff, - [12456] = 0x2fff, [12460] = 0x2fff, [12464] = 0x2fff, [12468] = 0x2fff, - [12472] = 0x2fff, [12476] = 0x2fff, [12480] = 0x0001, [12481] = 0x0001, - [12483] = 0x0006, [12484] = 0x0040, [12485] = 0x0001, [12486] = 0x000f, - [12487] = 0x0006, [12488] = 0x0001, [12489] = 0x0003, [12490] = 0x0106, - [12492] = 0x0061, [12493] = 0x0011, [12494] = 0x0401, [12496] = 0x0050, - [12497] = 0x0003, [12498] = 0x0102, [12500] = 0x0061, [12501] = 0x003b, - [12502] = 0x0502, [12503] = 0x0100, [12504] = 0x2fff, [12508] = 0x2fff, - [12512] = 0x2fff, [12516] = 0x2fff, [12520] = 0x2fff, [12524] = 0x2fff, - [12528] = 0x2fff, [12532] = 0x2fff, [12536] = 0x2fff, [12540] = 0x2fff, - [12544] = 0x0060, [12546] = 0x0601, [12548] = 0x0050, [12550] = 0x0100, - [12552] = 0x0001, [12554] = 0x0104, [12555] = 0x0100, [12556] = 0x2fff, - [12560] = 0x2fff, [12564] = 0x2fff, [12568] = 0x2fff, [12572] = 0x2fff, - [12576] = 0x2fff, [12580] = 0x2fff, [12584] = 0x2fff, [12588] = 0x2fff, - [12592] = 0x2fff, [12596] = 0x2fff, [12600] = 0x2fff, [12604] = 0x2fff, - [12608] = 0x0061, [12610] = 0x0601, [12612] = 0x0050, [12614] = 0x0102, - [12616] = 0x0001, [12618] = 0x0106, [12619] = 0x0100, [12620] = 0x2fff, - [12624] = 0x2fff, [12628] = 0x2fff, [12632] = 0x2fff, [12636] = 0x2fff, - [12640] = 0x2fff, [12644] = 0x2fff, [12648] = 0x2fff, [12652] = 0x2fff, - [12656] = 0x2fff, [12660] = 0x2fff, [12664] = 0x2fff, [12668] = 0x2fff, - [12672] = 0x0060, [12674] = 0x0601, [12676] = 0x0061, [12678] = 0x0601, - [12680] = 0x0050, [12682] = 0x0300, [12684] = 0x0001, [12686] = 0x0304, - [12688] = 0x0040, [12690] = 0x000f, [12692] = 0x0001, [12695] = 0x0100 +static struct reg_default wm8995_reg_defaults[] = { + { 0, 0x8995 }, + { 5, 0x0100 }, + { 16, 0x000b }, + { 17, 0x000b }, + { 24, 0x02c0 }, + { 25, 0x02c0 }, + { 26, 0x02c0 }, + { 27, 0x02c0 }, + { 28, 0x000f }, + { 32, 0x0005 }, + { 33, 0x0005 }, + { 40, 0x0003 }, + { 41, 0x0013 }, + { 48, 0x0004 }, + { 56, 0x09f8 }, + { 64, 0x1f25 }, + { 69, 0x0004 }, + { 82, 0xaaaa }, + { 84, 0x2a2a }, + { 146, 0x0060 }, + { 256, 0x0002 }, + { 257, 0x8004 }, + { 520, 0x0010 }, + { 528, 0x0083 }, + { 529, 0x0083 }, + { 548, 0x0c80 }, + { 580, 0x0c80 }, + { 768, 0x4050 }, + { 769, 0x4000 }, + { 771, 0x0040 }, + { 772, 0x0040 }, + { 773, 0x0040 }, + { 774, 0x0004 }, + { 775, 0x0100 }, + { 784, 0x4050 }, + { 785, 0x4000 }, + { 787, 0x0040 }, + { 788, 0x0040 }, + { 789, 0x0040 }, + { 1024, 0x00c0 }, + { 1025, 0x00c0 }, + { 1026, 0x00c0 }, + { 1027, 0x00c0 }, + { 1028, 0x00c0 }, + { 1029, 0x00c0 }, + { 1030, 0x00c0 }, + { 1031, 0x00c0 }, + { 1056, 0x0200 }, + { 1057, 0x0010 }, + { 1058, 0x0200 }, + { 1059, 0x0010 }, + { 1088, 0x0098 }, + { 1089, 0x0845 }, + { 1104, 0x0098 }, + { 1105, 0x0845 }, + { 1152, 0x6318 }, + { 1153, 0x6300 }, + { 1154, 0x0fca }, + { 1155, 0x0400 }, + { 1156, 0x00d8 }, + { 1157, 0x1eb5 }, + { 1158, 0xf145 }, + { 1159, 0x0b75 }, + { 1160, 0x01c5 }, + { 1161, 0x1c58 }, + { 1162, 0xf373 }, + { 1163, 0x0a54 }, + { 1164, 0x0558 }, + { 1165, 0x168e }, + { 1166, 0xf829 }, + { 1167, 0x07ad }, + { 1168, 0x1103 }, + { 1169, 0x0564 }, + { 1170, 0x0559 }, + { 1171, 0x4000 }, + { 1184, 0x6318 }, + { 1185, 0x6300 }, + { 1186, 0x0fca }, + { 1187, 0x0400 }, + { 1188, 0x00d8 }, + { 1189, 0x1eb5 }, + { 1190, 0xf145 }, + { 1191, 0x0b75 }, + { 1192, 0x01c5 }, + { 1193, 0x1c58 }, + { 1194, 0xf373 }, + { 1195, 0x0a54 }, + { 1196, 0x0558 }, + { 1197, 0x168e }, + { 1198, 0xf829 }, + { 1199, 0x07ad }, + { 1200, 0x1103 }, + { 1201, 0x0564 }, + { 1202, 0x0559 }, + { 1203, 0x4000 }, + { 1280, 0x00c0 }, + { 1281, 0x00c0 }, + { 1282, 0x00c0 }, + { 1283, 0x00c0 }, + { 1312, 0x0200 }, + { 1313, 0x0010 }, + { 1344, 0x0098 }, + { 1345, 0x0845 }, + { 1408, 0x6318 }, + { 1409, 0x6300 }, + { 1410, 0x0fca }, + { 1411, 0x0400 }, + { 1412, 0x00d8 }, + { 1413, 0x1eb5 }, + { 1414, 0xf145 }, + { 1415, 0x0b75 }, + { 1416, 0x01c5 }, + { 1417, 0x1c58 }, + { 1418, 0xf373 }, + { 1419, 0x0a54 }, + { 1420, 0x0558 }, + { 1421, 0x168e }, + { 1422, 0xf829 }, + { 1423, 0x07ad }, + { 1424, 0x1103 }, + { 1425, 0x0564 }, + { 1426, 0x0559 }, + { 1427, 0x4000 }, + { 1568, 0x0002 }, + { 1792, 0xa100 }, + { 1793, 0xa101 }, + { 1794, 0xa101 }, + { 1795, 0xa101 }, + { 1796, 0xa101 }, + { 1797, 0xa101 }, + { 1798, 0xa101 }, + { 1799, 0xa101 }, + { 1800, 0xa101 }, + { 1801, 0xa101 }, + { 1802, 0xa101 }, + { 1803, 0xa101 }, + { 1804, 0xa101 }, + { 1805, 0xa101 }, + { 1825, 0x0055 }, + { 1848, 0x3fff }, + { 1849, 0x1fff }, + { 2049, 0x0001 }, + { 2050, 0x0069 }, + { 2056, 0x0002 }, + { 2057, 0x0003 }, + { 2058, 0x0069 }, + { 12288, 0x0001 }, + { 12289, 0x0001 }, + { 12291, 0x0006 }, + { 12292, 0x0040 }, + { 12293, 0x0001 }, + { 12294, 0x000f }, + { 12295, 0x0006 }, + { 12296, 0x0001 }, + { 12297, 0x0003 }, + { 12298, 0x0104 }, + { 12300, 0x0060 }, + { 12301, 0x0011 }, + { 12302, 0x0401 }, + { 12304, 0x0050 }, + { 12305, 0x0003 }, + { 12306, 0x0100 }, + { 12308, 0x0051 }, + { 12309, 0x0003 }, + { 12310, 0x0104 }, + { 12311, 0x000a }, + { 12312, 0x0060 }, + { 12313, 0x003b }, + { 12314, 0x0502 }, + { 12315, 0x0100 }, + { 12316, 0x2fff }, + { 12320, 0x2fff }, + { 12324, 0x2fff }, + { 12328, 0x2fff }, + { 12332, 0x2fff }, + { 12336, 0x2fff }, + { 12340, 0x2fff }, + { 12344, 0x2fff }, + { 12348, 0x2fff }, + { 12352, 0x0001 }, + { 12353, 0x0001 }, + { 12355, 0x0006 }, + { 12356, 0x0040 }, + { 12357, 0x0001 }, + { 12358, 0x000f }, + { 12359, 0x0006 }, + { 12360, 0x0001 }, + { 12361, 0x0003 }, + { 12362, 0x0104 }, + { 12364, 0x0060 }, + { 12365, 0x0011 }, + { 12366, 0x0401 }, + { 12368, 0x0050 }, + { 12369, 0x0003 }, + { 12370, 0x0100 }, + { 12372, 0x0060 }, + { 12373, 0x003b }, + { 12374, 0x0502 }, + { 12375, 0x0100 }, + { 12376, 0x2fff }, + { 12380, 0x2fff }, + { 12384, 0x2fff }, + { 12388, 0x2fff }, + { 12392, 0x2fff }, + { 12396, 0x2fff }, + { 12400, 0x2fff }, + { 12404, 0x2fff }, + { 12408, 0x2fff }, + { 12412, 0x2fff }, + { 12416, 0x0001 }, + { 12417, 0x0001 }, + { 12419, 0x0006 }, + { 12420, 0x0040 }, + { 12421, 0x0001 }, + { 12422, 0x000f }, + { 12423, 0x0006 }, + { 12424, 0x0001 }, + { 12425, 0x0003 }, + { 12426, 0x0106 }, + { 12428, 0x0061 }, + { 12429, 0x0011 }, + { 12430, 0x0401 }, + { 12432, 0x0050 }, + { 12433, 0x0003 }, + { 12434, 0x0102 }, + { 12436, 0x0051 }, + { 12437, 0x0003 }, + { 12438, 0x0106 }, + { 12439, 0x000a }, + { 12440, 0x0061 }, + { 12441, 0x003b }, + { 12442, 0x0502 }, + { 12443, 0x0100 }, + { 12444, 0x2fff }, + { 12448, 0x2fff }, + { 12452, 0x2fff }, + { 12456, 0x2fff }, + { 12460, 0x2fff }, + { 12464, 0x2fff }, + { 12468, 0x2fff }, + { 12472, 0x2fff }, + { 12476, 0x2fff }, + { 12480, 0x0001 }, + { 12481, 0x0001 }, + { 12483, 0x0006 }, + { 12484, 0x0040 }, + { 12485, 0x0001 }, + { 12486, 0x000f }, + { 12487, 0x0006 }, + { 12488, 0x0001 }, + { 12489, 0x0003 }, + { 12490, 0x0106 }, + { 12492, 0x0061 }, + { 12493, 0x0011 }, + { 12494, 0x0401 }, + { 12496, 0x0050 }, + { 12497, 0x0003 }, + { 12498, 0x0102 }, + { 12500, 0x0061 }, + { 12501, 0x003b }, + { 12502, 0x0502 }, + { 12503, 0x0100 }, + { 12504, 0x2fff }, + { 12508, 0x2fff }, + { 12512, 0x2fff }, + { 12516, 0x2fff }, + { 12520, 0x2fff }, + { 12524, 0x2fff }, + { 12528, 0x2fff }, + { 12532, 0x2fff }, + { 12536, 0x2fff }, + { 12540, 0x2fff }, + { 12544, 0x0060 }, + { 12546, 0x0601 }, + { 12548, 0x0050 }, + { 12550, 0x0100 }, + { 12552, 0x0001 }, + { 12554, 0x0104 }, + { 12555, 0x0100 }, + { 12556, 0x2fff }, + { 12560, 0x2fff }, + { 12564, 0x2fff }, + { 12568, 0x2fff }, + { 12572, 0x2fff }, + { 12576, 0x2fff }, + { 12580, 0x2fff }, + { 12584, 0x2fff }, + { 12588, 0x2fff }, + { 12592, 0x2fff }, + { 12596, 0x2fff }, + { 12600, 0x2fff }, + { 12604, 0x2fff }, + { 12608, 0x0061 }, + { 12610, 0x0601 }, + { 12612, 0x0050 }, + { 12614, 0x0102 }, + { 12616, 0x0001 }, + { 12618, 0x0106 }, + { 12619, 0x0100 }, + { 12620, 0x2fff }, + { 12624, 0x2fff }, + { 12628, 0x2fff }, + { 12632, 0x2fff }, + { 12636, 0x2fff }, + { 12640, 0x2fff }, + { 12644, 0x2fff }, + { 12648, 0x2fff }, + { 12652, 0x2fff }, + { 12656, 0x2fff }, + { 12660, 0x2fff }, + { 12664, 0x2fff }, + { 12668, 0x2fff }, + { 12672, 0x0060 }, + { 12674, 0x0601 }, + { 12676, 0x0061 }, + { 12678, 0x0601 }, + { 12680, 0x0050 }, + { 12682, 0x0300 }, + { 12684, 0x0001 }, + { 12686, 0x0304 }, + { 12688, 0x0040 }, + { 12690, 0x000f }, + { 12692, 0x0001 }, + { 12695, 0x0100 }, }; struct fll_config { @@ -134,7 +378,7 @@ struct fll_config { }; struct wm8995_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk[2]; int mclk[2]; int aifclk[2]; @@ -156,7 +400,7 @@ static int wm8995_regulator_event_##n(struct notifier_block *nb, \ struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8995->codec->cache_sync = 1; \ + regcache_mark_dirty(wm8995->regmap); \ } \ return 0; \ } @@ -949,31 +1193,244 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = { { "SPK2R", NULL, "SPK2R Driver" } }; -static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8995_readable(struct device *dev, unsigned int reg) { - /* out of bounds registers are generally considered - * volatile to support register banks that are partially - * owned by something else for e.g. a DSP - */ - if (reg > WM8995_MAX_CACHED_REGISTER) - return 1; - switch (reg) { case WM8995_SOFTWARE_RESET: + case WM8995_POWER_MANAGEMENT_1: + case WM8995_POWER_MANAGEMENT_2: + case WM8995_POWER_MANAGEMENT_3: + case WM8995_POWER_MANAGEMENT_4: + case WM8995_POWER_MANAGEMENT_5: + case WM8995_LEFT_LINE_INPUT_1_VOLUME: + case WM8995_RIGHT_LINE_INPUT_1_VOLUME: + case WM8995_LEFT_LINE_INPUT_CONTROL: + case WM8995_DAC1_LEFT_VOLUME: + case WM8995_DAC1_RIGHT_VOLUME: + case WM8995_DAC2_LEFT_VOLUME: + case WM8995_DAC2_RIGHT_VOLUME: + case WM8995_OUTPUT_VOLUME_ZC_1: + case WM8995_MICBIAS_1: + case WM8995_MICBIAS_2: + case WM8995_LDO_1: + case WM8995_LDO_2: + case WM8995_ACCESSORY_DETECT_MODE1: + case WM8995_ACCESSORY_DETECT_MODE2: + case WM8995_HEADPHONE_DETECT1: + case WM8995_HEADPHONE_DETECT2: + case WM8995_MIC_DETECT_1: + case WM8995_MIC_DETECT_2: + case WM8995_CHARGE_PUMP_1: + case WM8995_CLASS_W_1: + case WM8995_DC_SERVO_1: + case WM8995_DC_SERVO_2: + case WM8995_DC_SERVO_3: + case WM8995_DC_SERVO_5: + case WM8995_DC_SERVO_6: + case WM8995_DC_SERVO_7: case WM8995_DC_SERVO_READBACK_0: + case WM8995_ANALOGUE_HP_1: + case WM8995_ANALOGUE_HP_2: + case WM8995_CHIP_REVISION: + case WM8995_CONTROL_INTERFACE_1: + case WM8995_CONTROL_INTERFACE_2: + case WM8995_WRITE_SEQUENCER_CTRL_1: + case WM8995_WRITE_SEQUENCER_CTRL_2: + case WM8995_AIF1_CLOCKING_1: + case WM8995_AIF1_CLOCKING_2: + case WM8995_AIF2_CLOCKING_1: + case WM8995_AIF2_CLOCKING_2: + case WM8995_CLOCKING_1: + case WM8995_CLOCKING_2: + case WM8995_AIF1_RATE: + case WM8995_AIF2_RATE: + case WM8995_RATE_STATUS: + case WM8995_FLL1_CONTROL_1: + case WM8995_FLL1_CONTROL_2: + case WM8995_FLL1_CONTROL_3: + case WM8995_FLL1_CONTROL_4: + case WM8995_FLL1_CONTROL_5: + case WM8995_FLL2_CONTROL_1: + case WM8995_FLL2_CONTROL_2: + case WM8995_FLL2_CONTROL_3: + case WM8995_FLL2_CONTROL_4: + case WM8995_FLL2_CONTROL_5: + case WM8995_AIF1_CONTROL_1: + case WM8995_AIF1_CONTROL_2: + case WM8995_AIF1_MASTER_SLAVE: + case WM8995_AIF1_BCLK: + case WM8995_AIF1ADC_LRCLK: + case WM8995_AIF1DAC_LRCLK: + case WM8995_AIF1DAC_DATA: + case WM8995_AIF1ADC_DATA: + case WM8995_AIF2_CONTROL_1: + case WM8995_AIF2_CONTROL_2: + case WM8995_AIF2_MASTER_SLAVE: + case WM8995_AIF2_BCLK: + case WM8995_AIF2ADC_LRCLK: + case WM8995_AIF2DAC_LRCLK: + case WM8995_AIF2DAC_DATA: + case WM8995_AIF2ADC_DATA: + case WM8995_AIF1_ADC1_LEFT_VOLUME: + case WM8995_AIF1_ADC1_RIGHT_VOLUME: + case WM8995_AIF1_DAC1_LEFT_VOLUME: + case WM8995_AIF1_DAC1_RIGHT_VOLUME: + case WM8995_AIF1_ADC2_LEFT_VOLUME: + case WM8995_AIF1_ADC2_RIGHT_VOLUME: + case WM8995_AIF1_DAC2_LEFT_VOLUME: + case WM8995_AIF1_DAC2_RIGHT_VOLUME: + case WM8995_AIF1_ADC1_FILTERS: + case WM8995_AIF1_ADC2_FILTERS: + case WM8995_AIF1_DAC1_FILTERS_1: + case WM8995_AIF1_DAC1_FILTERS_2: + case WM8995_AIF1_DAC2_FILTERS_1: + case WM8995_AIF1_DAC2_FILTERS_2: + case WM8995_AIF1_DRC1_1: + case WM8995_AIF1_DRC1_2: + case WM8995_AIF1_DRC1_3: + case WM8995_AIF1_DRC1_4: + case WM8995_AIF1_DRC1_5: + case WM8995_AIF1_DRC2_1: + case WM8995_AIF1_DRC2_2: + case WM8995_AIF1_DRC2_3: + case WM8995_AIF1_DRC2_4: + case WM8995_AIF1_DRC2_5: + case WM8995_AIF1_DAC1_EQ_GAINS_1: + case WM8995_AIF1_DAC1_EQ_GAINS_2: + case WM8995_AIF1_DAC1_EQ_BAND_1_A: + case WM8995_AIF1_DAC1_EQ_BAND_1_B: + case WM8995_AIF1_DAC1_EQ_BAND_1_PG: + case WM8995_AIF1_DAC1_EQ_BAND_2_A: + case WM8995_AIF1_DAC1_EQ_BAND_2_B: + case WM8995_AIF1_DAC1_EQ_BAND_2_C: + case WM8995_AIF1_DAC1_EQ_BAND_2_PG: + case WM8995_AIF1_DAC1_EQ_BAND_3_A: + case WM8995_AIF1_DAC1_EQ_BAND_3_B: + case WM8995_AIF1_DAC1_EQ_BAND_3_C: + case WM8995_AIF1_DAC1_EQ_BAND_3_PG: + case WM8995_AIF1_DAC1_EQ_BAND_4_A: + case WM8995_AIF1_DAC1_EQ_BAND_4_B: + case WM8995_AIF1_DAC1_EQ_BAND_4_C: + case WM8995_AIF1_DAC1_EQ_BAND_4_PG: + case WM8995_AIF1_DAC1_EQ_BAND_5_A: + case WM8995_AIF1_DAC1_EQ_BAND_5_B: + case WM8995_AIF1_DAC1_EQ_BAND_5_PG: + case WM8995_AIF1_DAC2_EQ_GAINS_1: + case WM8995_AIF1_DAC2_EQ_GAINS_2: + case WM8995_AIF1_DAC2_EQ_BAND_1_A: + case WM8995_AIF1_DAC2_EQ_BAND_1_B: + case WM8995_AIF1_DAC2_EQ_BAND_1_PG: + case WM8995_AIF1_DAC2_EQ_BAND_2_A: + case WM8995_AIF1_DAC2_EQ_BAND_2_B: + case WM8995_AIF1_DAC2_EQ_BAND_2_C: + case WM8995_AIF1_DAC2_EQ_BAND_2_PG: + case WM8995_AIF1_DAC2_EQ_BAND_3_A: + case WM8995_AIF1_DAC2_EQ_BAND_3_B: + case WM8995_AIF1_DAC2_EQ_BAND_3_C: + case WM8995_AIF1_DAC2_EQ_BAND_3_PG: + case WM8995_AIF1_DAC2_EQ_BAND_4_A: + case WM8995_AIF1_DAC2_EQ_BAND_4_B: + case WM8995_AIF1_DAC2_EQ_BAND_4_C: + case WM8995_AIF1_DAC2_EQ_BAND_4_PG: + case WM8995_AIF1_DAC2_EQ_BAND_5_A: + case WM8995_AIF1_DAC2_EQ_BAND_5_B: + case WM8995_AIF1_DAC2_EQ_BAND_5_PG: + case WM8995_AIF2_ADC_LEFT_VOLUME: + case WM8995_AIF2_ADC_RIGHT_VOLUME: + case WM8995_AIF2_DAC_LEFT_VOLUME: + case WM8995_AIF2_DAC_RIGHT_VOLUME: + case WM8995_AIF2_ADC_FILTERS: + case WM8995_AIF2_DAC_FILTERS_1: + case WM8995_AIF2_DAC_FILTERS_2: + case WM8995_AIF2_DRC_1: + case WM8995_AIF2_DRC_2: + case WM8995_AIF2_DRC_3: + case WM8995_AIF2_DRC_4: + case WM8995_AIF2_DRC_5: + case WM8995_AIF2_EQ_GAINS_1: + case WM8995_AIF2_EQ_GAINS_2: + case WM8995_AIF2_EQ_BAND_1_A: + case WM8995_AIF2_EQ_BAND_1_B: + case WM8995_AIF2_EQ_BAND_1_PG: + case WM8995_AIF2_EQ_BAND_2_A: + case WM8995_AIF2_EQ_BAND_2_B: + case WM8995_AIF2_EQ_BAND_2_C: + case WM8995_AIF2_EQ_BAND_2_PG: + case WM8995_AIF2_EQ_BAND_3_A: + case WM8995_AIF2_EQ_BAND_3_B: + case WM8995_AIF2_EQ_BAND_3_C: + case WM8995_AIF2_EQ_BAND_3_PG: + case WM8995_AIF2_EQ_BAND_4_A: + case WM8995_AIF2_EQ_BAND_4_B: + case WM8995_AIF2_EQ_BAND_4_C: + case WM8995_AIF2_EQ_BAND_4_PG: + case WM8995_AIF2_EQ_BAND_5_A: + case WM8995_AIF2_EQ_BAND_5_B: + case WM8995_AIF2_EQ_BAND_5_PG: + case WM8995_DAC1_MIXER_VOLUMES: + case WM8995_DAC1_LEFT_MIXER_ROUTING: + case WM8995_DAC1_RIGHT_MIXER_ROUTING: + case WM8995_DAC2_MIXER_VOLUMES: + case WM8995_DAC2_LEFT_MIXER_ROUTING: + case WM8995_DAC2_RIGHT_MIXER_ROUTING: + case WM8995_AIF1_ADC1_LEFT_MIXER_ROUTING: + case WM8995_AIF1_ADC1_RIGHT_MIXER_ROUTING: + case WM8995_AIF1_ADC2_LEFT_MIXER_ROUTING: + case WM8995_AIF1_ADC2_RIGHT_MIXER_ROUTING: + case WM8995_DAC_SOFTMUTE: + case WM8995_OVERSAMPLING: + case WM8995_SIDETONE: + case WM8995_GPIO_1: + case WM8995_GPIO_2: + case WM8995_GPIO_3: + case WM8995_GPIO_4: + case WM8995_GPIO_5: + case WM8995_GPIO_6: + case WM8995_GPIO_7: + case WM8995_GPIO_8: + case WM8995_GPIO_9: + case WM8995_GPIO_10: + case WM8995_GPIO_11: + case WM8995_GPIO_12: + case WM8995_GPIO_13: + case WM8995_GPIO_14: + case WM8995_PULL_CONTROL_1: + case WM8995_PULL_CONTROL_2: case WM8995_INTERRUPT_STATUS_1: case WM8995_INTERRUPT_STATUS_2: + case WM8995_INTERRUPT_RAW_STATUS_2: case WM8995_INTERRUPT_STATUS_1_MASK: case WM8995_INTERRUPT_STATUS_2_MASK: case WM8995_INTERRUPT_CONTROL: + case WM8995_LEFT_PDM_SPEAKER_1: + case WM8995_RIGHT_PDM_SPEAKER_1: + case WM8995_PDM_SPEAKER_1_MUTE_SEQUENCE: + case WM8995_LEFT_PDM_SPEAKER_2: + case WM8995_RIGHT_PDM_SPEAKER_2: + case WM8995_PDM_SPEAKER_2_MUTE_SEQUENCE: + return true; + default: + return false; + } +} + +static bool wm8995_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8995_SOFTWARE_RESET: + case WM8995_DC_SERVO_READBACK_0: + case WM8995_INTERRUPT_STATUS_1: + case WM8995_INTERRUPT_STATUS_2: + case WM8995_INTERRUPT_CONTROL: case WM8995_ACCESSORY_DETECT_MODE1: case WM8995_ACCESSORY_DETECT_MODE2: case WM8995_HEADPHONE_DETECT1: case WM8995_HEADPHONE_DETECT2: - return 1; + case WM8995_RATE_STATUS: + return true; + default: + return false; } - - return 0; } static int wm8995_aif_mute(struct snd_soc_dai *dai, int mute) @@ -1528,7 +1985,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8995->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -1594,7 +2051,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type); + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1783,11 +2240,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .suspend = wm8995_suspend, .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8995_reg_defs), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8995_reg_defs, - .volatile_register = wm8995_volatile, - .compress_type = SND_SOC_RBTREE_COMPRESSION +}; + +static struct regmap_config wm8995_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM8995_MAX_REGISTER, + .reg_defaults = wm8995_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8995_reg_defaults), + .volatile_reg = wm8995_volatile, + .readable_reg = wm8995_readable, + .cache_type = REGCACHE_RBTREE, }; #if defined(CONFIG_SPI_MASTER) @@ -1800,21 +2264,37 @@ static int __devinit wm8995_spi_probe(struct spi_device *spi) if (!wm8995) return -ENOMEM; - wm8995->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8995); + wm8995->regmap = regmap_init_spi(spi, &wm8995_regmap); + if (IS_ERR(wm8995->regmap)) { + ret = PTR_ERR(wm8995->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + goto err_alloc; + } + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8995, wm8995_dai, ARRAY_SIZE(wm8995_dai)); if (ret < 0) - kfree(wm8995); + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(wm8995->regmap); +err_alloc: + kfree(wm8995); + return ret; } static int __devexit wm8995_spi_remove(struct spi_device *spi) { + struct wm8995_priv *wm8995 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(wm8995->regmap); + kfree(wm8995); return 0; } @@ -1839,21 +2319,40 @@ static __devinit int wm8995_i2c_probe(struct i2c_client *i2c, if (!wm8995) return -ENOMEM; - wm8995->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, wm8995); + wm8995->regmap = regmap_init_i2c(i2c, &wm8995_regmap); + if (IS_ERR(wm8995->regmap)) { + ret = PTR_ERR(wm8995->regmap); + dev_err(&i2c->dev, "Failed to register regmap: %d\n", ret); + goto err_alloc; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8995, wm8995_dai, ARRAY_SIZE(wm8995_dai)); - if (ret < 0) - kfree(wm8995); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } + + return ret; + +err_regmap: + regmap_exit(wm8995->regmap); +err_alloc: + kfree(wm8995); + return ret; } static __devexit int wm8995_i2c_remove(struct i2c_client *client) { + struct wm8995_priv *wm8995 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8995->regmap); + kfree(wm8995); return 0; } -- cgit v1.2.1 From d8c29e7f78a6c52fc5cfa956c4b72c797a468241 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 10 Nov 2011 21:22:15 +0200 Subject: ASoC: Remove unused defines in alc5632 codec Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 21 --------------------- 1 file changed, 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ee6a497b5e71..048c60e4cbac 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -964,27 +964,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) return 0; } -#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ - | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ - | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ - | ALC5632_ADC_REC_MONOMIX) - -#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ - | ALC5632_MIC_ROUTE_SPK \ - | ALC5632_MIC_ROUTE_MONOMIX) - -#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ - | ALC5632_PWR_DAC_STATUS \ - | ALC5632_PWR_AMIX_STATUS \ - | ALC5632_PWR_VREF_STATUS) - -#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ - / ALC5632_ADC_REC_GAIN_STEP) - -#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) - -#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) - static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); -- cgit v1.2.1 From 88c494b99a5873a46738c4c3f6f37ccce87b03e9 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 10 Nov 2011 21:22:16 +0200 Subject: ASoC: Remove unnecessary backslash from alc5632 codec Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 048c60e4cbac..8a3bf7194728 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -811,7 +811,7 @@ static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, static int alc5632_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L |ALC5632_MISC_HP_DEPOP_MUTE_R; u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; -- cgit v1.2.1 From ed2dd7da35cad3115c38fd42eecbecae899a1d7a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Nov 2011 16:21:01 -0800 Subject: ASoC: ak4642: add ak4642_set_bias_level() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b854eb0e6ad1..004a093547f9 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -196,8 +196,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); snd_soc_write(codec, L_IVC, 0x91); /* volume */ snd_soc_write(codec, R_IVC, 0x91); /* volume */ - snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, - PMVCM | PMMIN | PMDAC); + snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, + PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { @@ -217,8 +217,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); - snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, - PMVCM | PMADL); + snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); } @@ -376,6 +375,22 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int ak4642_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, PW_MGMT1, 0x00); + break; + default: + snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + static struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, @@ -425,12 +440,22 @@ static int ak4642_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); + ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int ak4642_remove(struct snd_soc_codec *codec) +{ + ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, + .remove = ak4642_remove, .resume = ak4642_resume, + .set_bias_level = ak4642_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, -- cgit v1.2.1 From 24747daea5610676fd1e2c2ca603c8822a085c87 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Nov 2011 16:21:31 -0800 Subject: ASoC: ak4642: add DAPM support for HeadPhone Output Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 44 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 35 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 004a093547f9..da9caf0d5317 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -152,6 +152,37 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { 0, 0xFF, 1, out_tlv), }; +static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { + SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + + SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, + &ak4642_hpout_mixer_controls[0], + ARRAY_SIZE(ak4642_hpout_mixer_controls)), + + SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, + &ak4642_hpout_mixer_controls[0], + ARRAY_SIZE(ak4642_hpout_mixer_controls)), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), +}; + +static const struct snd_soc_dapm_route ak4642_intercon[] = { + + /* Outputs */ + {"HPOUTL", NULL, "HPOUTL Mixer"}, + {"HPOUTR", NULL, "HPOUTR Mixer"}, + + {"HPOUTL Mixer", "DACH", "DAC"}, + {"HPOUTR Mixer", "DACH", "DAC"}, +}; /* codec private data */ struct ak4642_priv { @@ -192,13 +223,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ - snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); - snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); snd_soc_write(codec, L_IVC, 0x91); /* volume */ snd_soc_write(codec, R_IVC, 0x91); /* volume */ - snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, - PMMIN | PMDAC); - snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* @@ -233,10 +259,6 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { /* stop headphone output */ snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); - snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); - snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, 0); - snd_soc_update_bits(codec, MD_CTL3, BST1, 0); - snd_soc_update_bits(codec, MD_CTL4, DACH, 0); } else { /* stop stereo input */ snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); @@ -459,6 +481,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, + .dapm_widgets = ak4642_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), + .dapm_routes = ak4642_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.1 From 3c7035268c2c89942fe51a61833d1066b4a766eb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Nov 2011 16:21:42 -0800 Subject: ASoC: ak4642: add headphone mute switch control Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index da9caf0d5317..b2460c2eebe0 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -150,6 +150,8 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), + + SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { @@ -225,7 +227,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, */ snd_soc_write(codec, L_IVC, 0x91); /* volume */ snd_soc_write(codec, R_IVC, 0x91); /* volume */ - snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input @@ -257,8 +258,6 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* stop headphone output */ - snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); } else { /* stop stereo input */ snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); -- cgit v1.2.1 From e8c83dbfb7fc0c3cec141112524906b029a1f413 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Nov 2011 16:21:55 -0800 Subject: ASoC: ak4642: add Line out support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b2460c2eebe0..daec5f75def2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -158,11 +158,16 @@ static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), }; +static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), +}; + static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, &ak4642_hpout_mixer_controls[0], @@ -172,6 +177,10 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { &ak4642_hpout_mixer_controls[0], ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, + &ak4642_lout_mixer_controls[0], + ARRAY_SIZE(ak4642_lout_mixer_controls)), + /* DAC */ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), }; @@ -181,9 +190,11 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ {"HPOUTL", NULL, "HPOUTL Mixer"}, {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"LINEOUT", NULL, "LINEOUT Mixer"}, {"HPOUTL Mixer", "DACH", "DAC"}, {"HPOUTR Mixer", "DACH", "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; /* codec private data */ -- cgit v1.2.1 From a9317e8b6b53ab61d3ee764b6456596efd8c83b7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Nov 2011 16:22:05 -0800 Subject: ASoC: ak4642: add ak4648 support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 44 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index daec5f75def2..859e0155e18d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -20,6 +20,7 @@ * * AK4642 is tested. * AK4643 is tested. + * AK4648 is tested. */ #include @@ -71,8 +72,6 @@ #define HP_MS 0x23 #define SPK_MS 0x24 -#define AK4642_CACHEREGNUM 0x25 - /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ #define PMMIN (1 << 5) /* MIN Input Power Management */ @@ -206,7 +205,7 @@ struct ak4642_priv { /* * ak4642 register cache */ -static const u8 ak4642_reg[AK4642_CACHEREGNUM] = { +static const u8 ak4642_reg[] = { 0x00, 0x00, 0x01, 0x00, 0x02, 0x00, 0x00, 0x00, 0xe1, 0xe1, 0x18, 0x00, @@ -219,6 +218,19 @@ static const u8 ak4642_reg[AK4642_CACHEREGNUM] = { 0x00, }; +static const u8 ak4648_reg[] = { + 0x00, 0x00, 0x01, 0x00, + 0x02, 0x00, 0x00, 0x00, + 0xe1, 0xe1, 0x18, 0x00, + 0xe1, 0x18, 0x11, 0xb8, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x88, 0x88, 0x08, +}; + static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -488,9 +500,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .remove = ak4642_remove, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4642_reg), + .reg_cache_default = ak4642_reg, /* ak4642 reg */ + .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ + .reg_word_size = sizeof(u8), + .dapm_widgets = ak4642_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), + .dapm_routes = ak4642_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), +}; + +static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, + .set_bias_level = ak4642_set_bias_level, + .reg_cache_default = ak4648_reg, /* ak4648 reg */ + .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ .reg_word_size = sizeof(u8), - .reg_cache_default = ak4642_reg, .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, @@ -512,7 +538,8 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_ak4642, &ak4642_dai, 1); + (struct snd_soc_codec_driver *)id->driver_data, + &ak4642_dai, 1); if (ret < 0) kfree(ak4642); return ret; @@ -526,8 +553,9 @@ static __devexit int ak4642_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id ak4642_i2c_id[] = { - { "ak4642", 0 }, - { "ak4643", 0 }, + { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, + { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, + { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, { } }; MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); -- cgit v1.2.1 From e012ba249171a205c5735a76b947bdae9cf34c6e Mon Sep 17 00:00:00 2001 From: Johannes Stezenbach Date: Mon, 14 Nov 2011 17:23:17 +0100 Subject: ASoC: sta32x: add platform data definition Add a structure for platform specific configuration and use it, thereby removing a few FIXMEs which marked hard-coded values. Signed-off-by: Johannes Stezenbach Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 30 +++++++++++++++++++++--------- 1 file changed, 21 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index d2f37152f940..97091e3b9a0b 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -35,6 +35,7 @@ #include #include +#include #include "sta32x.h" #define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ @@ -73,6 +74,7 @@ static const char *sta32x_supply_names[] = { struct sta32x_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_codec *codec; + struct sta32x_platform_data *pdata; unsigned int mclk; unsigned int format; @@ -775,9 +777,10 @@ static int sta32x_resume(struct snd_soc_codec *codec) static int sta32x_probe(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); - int i, ret = 0; + int i, ret = 0, thermal = 0; sta32x->codec = codec; + sta32x->pdata = dev_get_platdata(codec->dev); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) @@ -820,25 +823,34 @@ static int sta32x_probe(struct snd_soc_codec *codec) snd_soc_cache_write(codec, STA32X_AUTO3, 0x00); snd_soc_cache_write(codec, STA32X_C3CFG, 0x40); - /* FIXME enable thermal warning adjustment and recovery */ + /* set thermal warning adjustment and recovery */ + if (!(sta32x->pdata->thermal_conf & STA32X_THERMAL_ADJUSTMENT_ENABLE)) + thermal |= STA32X_CONFA_TWAB; + if (!(sta32x->pdata->thermal_conf & STA32X_THERMAL_RECOVERY_ENABLE)) + thermal |= STA32X_CONFA_TWRB; snd_soc_update_bits(codec, STA32X_CONFA, - STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, + thermal); - /* FIXME select 2.1 mode */ + /* select output configuration */ snd_soc_update_bits(codec, STA32X_CONFF, STA32X_CONFF_OCFG_MASK, - 1 << STA32X_CONFF_OCFG_SHIFT); + sta32x->pdata->output_conf + << STA32X_CONFF_OCFG_SHIFT); - /* FIXME channel to output mapping */ + /* channel to output mapping */ snd_soc_update_bits(codec, STA32X_C1CFG, STA32X_CxCFG_OM_MASK, - 0 << STA32X_CxCFG_OM_SHIFT); + sta32x->pdata->ch1_output_mapping + << STA32X_CxCFG_OM_SHIFT); snd_soc_update_bits(codec, STA32X_C2CFG, STA32X_CxCFG_OM_MASK, - 1 << STA32X_CxCFG_OM_SHIFT); + sta32x->pdata->ch2_output_mapping + << STA32X_CxCFG_OM_SHIFT); snd_soc_update_bits(codec, STA32X_C3CFG, STA32X_CxCFG_OM_MASK, - 2 << STA32X_CxCFG_OM_SHIFT); + sta32x->pdata->ch3_output_mapping + << STA32X_CxCFG_OM_SHIFT); /* initialize coefficient shadow RAM with reset values */ for (i = 4; i <= 49; i += 5) -- cgit v1.2.1 From 3fb5eac50d66cab4a41177269432ffffcc3e67ac Mon Sep 17 00:00:00 2001 From: Johannes Stezenbach Date: Mon, 14 Nov 2011 17:23:18 +0100 Subject: ASoC: sta32x: add workaround for ESD reset issue sta32x resets and loses all configuration during ESD test. Work around by polling the CONFA register once a second and restore all coeffcients and registers when CONFA changes unexpectedly. Signed-off-by: Johannes Stezenbach Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 49 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 97091e3b9a0b..3b0deafd766b 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -80,6 +81,8 @@ struct sta32x_priv { unsigned int format; u32 coef_shadow[STA32X_COEF_COUNT]; + struct delayed_work watchdog_work; + int shutdown; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -304,6 +307,46 @@ int sta32x_cache_sync(struct snd_soc_codec *codec) return rc; } +/* work around ESD issue where sta32x resets and loses all configuration */ +static void sta32x_watchdog(struct work_struct *work) +{ + struct sta32x_priv *sta32x = container_of(work, struct sta32x_priv, + watchdog_work.work); + struct snd_soc_codec *codec = sta32x->codec; + unsigned int confa, confa_cached; + + /* check if sta32x has reset itself */ + confa_cached = snd_soc_read(codec, STA32X_CONFA); + codec->cache_bypass = 1; + confa = snd_soc_read(codec, STA32X_CONFA); + codec->cache_bypass = 0; + if (confa != confa_cached) { + codec->cache_sync = 1; + sta32x_cache_sync(codec); + } + + if (!sta32x->shutdown) + schedule_delayed_work(&sta32x->watchdog_work, + round_jiffies_relative(HZ)); +} + +static void sta32x_watchdog_start(struct sta32x_priv *sta32x) +{ + if (sta32x->pdata->needs_esd_watchdog) { + sta32x->shutdown = 0; + schedule_delayed_work(&sta32x->watchdog_work, + round_jiffies_relative(HZ)); + } +} + +static void sta32x_watchdog_stop(struct sta32x_priv *sta32x) +{ + if (sta32x->pdata->needs_esd_watchdog) { + sta32x->shutdown = 1; + cancel_delayed_work_sync(&sta32x->watchdog_work); + } +} + #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -714,6 +757,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, } sta32x_cache_sync(codec); + sta32x_watchdog_start(sta32x); } /* Power up to mute */ @@ -730,7 +774,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, STA32X_CONFF_PWDN); msleep(300); - + sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); break; @@ -863,6 +907,9 @@ static int sta32x_probe(struct snd_soc_codec *codec) sta32x->coef_shadow[60] = 0x400000; sta32x->coef_shadow[61] = 0x400000; + if (sta32x->pdata->needs_esd_watchdog) + INIT_DELAYED_WORK(&sta32x->watchdog_work, sta32x_watchdog); + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); @@ -879,6 +926,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + sta32x_watchdog_stop(sta32x); sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); -- cgit v1.2.1 From 6662ff5c3b8efe8c107118d9506ad65daf3e3a1b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 13 Nov 2011 11:56:28 +0800 Subject: ASoC: Remove unused control_data and mutex fields from struct alc5632_priv Signed-off-by: Axel Lin Acked-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 8a3bf7194728..07e958aeea5c 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -68,8 +68,6 @@ static const u16 alc5632_reg_defaults[] = { /* codec private data */ struct alc5632_priv { enum snd_soc_control_type control_type; - void *control_data; - struct mutex mutex; u8 id; unsigned int sysclk; }; @@ -1071,9 +1069,7 @@ static int alc5632_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5632); - alc5632->control_data = client; alc5632->control_type = SND_SOC_I2C; - mutex_init(&alc5632->mutex); ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5632, &alc5632_dai, 1); -- cgit v1.2.1 From c9be8427b1dbd5e9d0313762fb80b2633abb694b Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Wed, 16 Nov 2011 12:07:00 +0200 Subject: ASoC: alc5632: Fix compile without CONFIG_PM Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 07e958aeea5c..e560a2119b12 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -939,6 +939,7 @@ static struct snd_soc_dai_driver alc5632_dai = { .symmetric_rates = 1, }; +#ifdef CONFIG_PM static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -961,6 +962,10 @@ static int alc5632_resume(struct snd_soc_codec *codec) alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } +#else +#define alc5632_suspend NULL +#define alc5632_resume NULL +#endif static int alc5632_probe(struct snd_soc_codec *codec) { -- cgit v1.2.1 From bb39753c2ba69d4d9467a109b03861cf43a6dcf8 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Wed, 16 Nov 2011 12:06:58 +0200 Subject: ASoC: Convert ALC5632 codec to use regmap API Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 217 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/alc5632.h | 2 + 2 files changed, 161 insertions(+), 58 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e560a2119b12..c5055c1a2d55 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include @@ -34,45 +35,129 @@ /* * ALC5632 register cache */ -static const u16 alc5632_reg_defaults[] = { - 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ - 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ - 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ - 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ - 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ - 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ - 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ - 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ - 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ - 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ - 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ - 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ - 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ - 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ - 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ - 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ - 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ - 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +static struct reg_default alc5632_reg_defaults[] = { + { 0, 0x59B4 }, + { 1, 0x0000 }, + { 2, 0x8080 }, + { 3, 0x0000 }, + { 4, 0x8080 }, + { 5, 0x0000 }, + { 6, 0x8080 }, + { 7, 0x0000 }, + { 8, 0xC800 }, + { 9, 0x0000 }, + { 10, 0xE808 }, + { 11, 0x0000 }, + { 12, 0x1010 }, + { 13, 0x0000 }, + { 14, 0x0808 }, + { 15, 0x0000 }, + { 16, 0xEE0F }, + { 17, 0x0000 }, + { 18, 0xCBCB }, + { 19, 0x0000 }, + { 20, 0x7F7F }, + { 21, 0x0000 }, + { 22, 0x0000 }, + { 23, 0x0000 }, + { 24, 0xE010 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x8008 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0000 }, + { 35, 0x0000 }, + { 36, 0x00C0 }, + { 37, 0x0000 }, + { 38, 0xEF00 }, + { 39, 0x0000 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0000 }, + { 45, 0x0000 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0000 }, + { 50, 0x0000 }, + { 51, 0x0000 }, + { 52, 0x8000 }, + { 53, 0x0000 }, + { 54, 0x0000 }, + { 55, 0x0000 }, + { 56, 0x0000 }, + { 57, 0x0000 }, + { 58, 0x0000 }, + { 59, 0x0000 }, + { 60, 0x0000 }, + { 61, 0x0000 }, + { 62, 0x8000 }, + { 63, 0x0000 }, + { 64, 0x0C0A }, + { 65, 0x0000 }, + { 66, 0x0000 }, + { 67, 0x0000 }, + { 68, 0x0000 }, + { 69, 0x0000 }, + { 70, 0x0000 }, + { 71, 0x0000 }, + { 72, 0x0000 }, + { 73, 0x0000 }, + { 74, 0x0000 }, + { 75, 0x0000 }, + { 76, 0xBE3E }, + { 77, 0x0000 }, + { 78, 0xBE3E }, + { 79, 0x0000 }, + { 80, 0x0000 }, + { 81, 0x0000 }, + { 82, 0x0000 }, + { 83, 0x0000 }, + { 84, 0x803A }, + { 85, 0x0000 }, + { 86, 0x0000 }, + { 87, 0x0000 }, + { 88, 0x0000 }, + { 89, 0x0000 }, + { 90, 0x0009 }, + { 91, 0x0000 }, + { 92, 0x0000 }, + { 93, 0x0000 }, + { 94, 0x3000 }, + { 95, 0x0000 }, + { 96, 0x3075 }, + { 97, 0x0000 }, + { 98, 0x1010 }, + { 99, 0x0000 }, + { 100, 0x3110 }, + { 101, 0x0000 }, + { 102, 0x0000 }, + { 103, 0x0000 }, + { 104, 0x0553 }, + { 105, 0x0000 }, + { 106, 0x0000 }, + { 107, 0x0000 }, + { 108, 0x0000 }, + { 109, 0x0000 }, + { 110, 0x0000 }, + { 111, 0x0000 }, }; /* codec private data */ struct alc5632_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u8 id; unsigned int sysclk; }; -static int alc5632_volatile_register(struct snd_soc_codec *codec, +static bool alc5632_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -82,19 +167,18 @@ static int alc5632_volatile_register(struct snd_soc_codec *codec, case ALC5632_OVER_CURR_STATUS: case ALC5632_HID_CTRL_DATA: case ALC5632_EQ_CTRL: - return 1; + return true; default: break; } - return 0; + return false; } -static inline int alc5632_reset(struct snd_soc_codec *codec) +static inline int alc5632_reset(struct regmap *map) { - snd_soc_write(codec, ALC5632_RESET, 0); - return snd_soc_read(codec, ALC5632_RESET); + return regmap_write(map, ALC5632_RESET, 0x59B4); } static int amp_mixer_event(struct snd_soc_dapm_widget *w, @@ -948,16 +1032,9 @@ static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int alc5632_resume(struct snd_soc_codec *codec) { - int ret; - - /* mark cache as needed to sync */ - codec->cache_sync = 1; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_cache_sync(codec); - if (ret != 0) { - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - return ret; - } + regcache_sync(alc5632->regmap); alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -972,14 +1049,14 @@ static int alc5632_probe(struct snd_soc_codec *codec) struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); - if (ret < 0) { + codec->control_data = alc5632->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - alc5632_reset(codec); - /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1008,11 +1085,6 @@ static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .suspend = alc5632_suspend, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = alc5632_reg_defaults, - .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), - .volatile_register = alc5632_volatile_register, .controls = alc5632_snd_controls, .num_controls = ARRAY_SIZE(alc5632_snd_controls), .dapm_widgets = alc5632_dapm_widgets, @@ -1021,13 +1093,24 @@ static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), }; +static struct regmap_config alc5632_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = ALC5632_MAX_REGISTER, + .reg_defaults = alc5632_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_reg = alc5632_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + /* * alc5632 2 wire address is determined by A1 pin * state during powerup. * low = 0x1a * high = 0x1b */ -static int alc5632_i2c_probe(struct i2c_client *client, +static __devinit int alc5632_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5632_priv *alc5632; @@ -1074,20 +1157,38 @@ static int alc5632_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5632); - alc5632->control_type = SND_SOC_I2C; + + alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); + if (IS_ERR(alc5632->regmap)) { + ret = PTR_ERR(alc5632->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = alc5632_reset(alc5632->regmap); + if (ret < 0) { + dev_err(&client->dev, "Failed to issue reset\n"); + regmap_exit(alc5632->regmap); + return ret; + } ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5632, &alc5632_dai, 1); - if (ret != 0) + + if (ret < 0) { dev_err(&client->dev, "Failed to register codec: %d\n", ret); + regmap_exit(alc5632->regmap); + return ret; + } return ret; } static int alc5632_i2c_remove(struct i2c_client *client) { + struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - + regmap_exit(alc5632->regmap); return 0; } diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h index ff4c0fd0d2ec..357651ec074e 100644 --- a/sound/soc/codecs/alc5632.h +++ b/sound/soc/codecs/alc5632.h @@ -246,4 +246,6 @@ #define ALC5632_VENDOR_ID1 0x7C #define ALC5632_VENDOR_ID2 0x7E +#define ALC5632_MAX_REGISTER 0x7E + #endif -- cgit v1.2.1 From 1a083257eb95af8e1d6e0d03e960c34f0017ad31 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sun, 13 Nov 2011 21:53:13 +0200 Subject: ASoC: alc5632: rename volume/switch contols for master and speaker volumes. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index c5055c1a2d55..6bfbbc71154d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -226,10 +226,10 @@ static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */ /* 31 steps (5 bit), -46.5db scale */ - SOC_DOUBLE_TLV("Line Playback Volume", + SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), /* bit 15 mutes left, bit 7 right */ - SOC_DOUBLE("Line Playback Switch", + SOC_DOUBLE("Speaker Playback Switch", ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), @@ -248,9 +248,9 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), - SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), - SOC_DOUBLE("Stereo DAC Playback Switch", + SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Mic1 Capture Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), -- cgit v1.2.1 From 086834e2d2bdf74e4e53bee9ee5359dfe849da1a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Nov 2011 13:38:28 +0000 Subject: ASoC: Say how long short WM8958 DSP2 firmwares are Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 0293763debe5..39e9557bdfd7 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short\n", name); + dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + name, fw->size); goto err; } -- cgit v1.2.1 From 6d10c91493a0b32744f649776744f898d27ea303 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Wed, 16 Nov 2011 12:32:27 -0600 Subject: ASoC: Add support for CS42L73 codec This patch adds support for the Cirrus Logic CS42L73 low power stereo codec. Signed-off-by: Brian Austin Signed-off-by: Georgi Vlaev Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1457 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 +++++++ 4 files changed, 1690 insertions(+) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 684cc1570689..686f45a07f34 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -178,6 +179,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index af64905f36ca..62b01e4e7983 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -117,6 +118,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 000000000000..6fe259aceb7b --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1457 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, + * Brian Austin, Cirrus Logic Inc, + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + struct sp_config config[3]; + struct regmap *regmap; + u32 sysclk; + u8 mclksel; + u32 mclk; +}; + +struct reg_default cs42l73_reg_defaults[] = { + { 1, 0x42 }, /* r01 - Device ID A&B */ + { 2, 0xA7 }, /* r02 - Device ID C&D */ + { 3, 0x30 }, /* r03 - Device ID E */ + { 6, 0xF1 }, /* r06 - Power Ctl 1 */ + { 7, 0xDF }, /* r07 - Power Ctl 2 */ + { 8, 0x3F }, /* r08 - Power Ctl 3 */ + { 9, 0x50 }, /* r09 - Charge Pump Freq */ + { 10, 0x53 }, /* r0A - Output Load MicBias Short Detect */ + { 11, 0x00 }, /* r0B - DMIC Master Clock Ctl */ + { 12, 0x00 }, /* r0C - Aux PCM Ctl */ + { 13, 0x15 }, /* r0D - Aux PCM Master Clock Ctl */ + { 14, 0x00 }, /* r0E - Audio PCM Ctl */ + { 15, 0x15 }, /* r0F - Audio PCM Master Clock Ctl */ + { 16, 0x00 }, /* r10 - Voice PCM Ctl */ + { 17, 0x15 }, /* r11 - Voice PCM Master Clock Ctl */ + { 18, 0x00 }, /* r12 - Voice/Aux Sample Rate */ + { 19, 0x06 }, /* r13 - Misc I/O Path Ctl */ + { 20, 0x00 }, /* r14 - ADC Input Path Ctl */ + { 21, 0x00 }, /* r15 - MICA Preamp, PGA Volume */ + { 22, 0x00 }, /* r16 - MICB Preamp, PGA Volume */ + { 23, 0x00 }, /* r17 - Input Path A Digital Volume */ + { 24, 0x00 }, /* r18 - Input Path B Digital Volume */ + { 25, 0x00 }, /* r19 - Playback Digital Ctl */ + { 26, 0x00 }, /* r1A - HP/LO Left Digital Volume */ + { 27, 0x00 }, /* r1B - HP/LO Right Digital Volume */ + { 28, 0x00 }, /* r1C - Speakerphone Digital Volume */ + { 29, 0x00 }, /* r1D - Ear/SPKLO Digital Volume */ + { 30, 0x00 }, /* r1E - HP Left Analog Volume */ + { 31, 0x00 }, /* r1F - HP Right Analog Volume */ + { 32, 0x00 }, /* r20 - LO Left Analog Volume */ + { 33, 0x00 }, /* r21 - LO Right Analog Volume */ + { 34, 0x00 }, /* r22 - Stereo Input Path Advisory Volume */ + { 35, 0x00 }, /* r23 - Aux PCM Input Advisory Volume */ + { 36, 0x00 }, /* r24 - Audio PCM Input Advisory Volume */ + { 37, 0x00 }, /* r25 - Voice PCM Input Advisory Volume */ + { 38, 0x00 }, /* r26 - Limiter Attack Rate HP/LO */ + { 39, 0x7F }, /* r27 - Limter Ctl, Release Rate HP/LO */ + { 40, 0x00 }, /* r28 - Limter Threshold HP/LO */ + { 41, 0x00 }, /* r29 - Limiter Attack Rate Speakerphone */ + { 42, 0x3F }, /* r2A - Limter Ctl, Release Rate Speakerphone */ + { 43, 0x00 }, /* r2B - Limter Threshold Speakerphone */ + { 44, 0x00 }, /* r2C - Limiter Attack Rate Ear/SPKLO */ + { 45, 0x3F }, /* r2D - Limter Ctl, Release Rate Ear/SPKLO */ + { 46, 0x00 }, /* r2E - Limter Threshold Ear/SPKLO */ + { 47, 0x00 }, /* r2F - ALC Enable, Attack Rate Left/Right */ + { 48, 0x3F }, /* r30 - ALC Release Rate Left/Right */ + { 49, 0x00 }, /* r31 - ALC Threshold Left/Right */ + { 50, 0x00 }, /* r32 - Noise Gate Ctl Left/Right */ + { 51, 0x00 }, /* r33 - ALC/NG Misc Ctl */ + { 52, 0x18 }, /* r34 - Mixer Ctl */ + { 53, 0x3F }, /* r35 - HP/LO Left Mixer Input Path Volume */ + { 54, 0x3F }, /* r36 - HP/LO Right Mixer Input Path Volume */ + { 55, 0x3F }, /* r37 - HP/LO Left Mixer Aux PCM Volume */ + { 56, 0x3F }, /* r38 - HP/LO Right Mixer Aux PCM Volume */ + { 57, 0x3F }, /* r39 - HP/LO Left Mixer Audio PCM Volume */ + { 58, 0x3F }, /* r3A - HP/LO Right Mixer Audio PCM Volume */ + { 59, 0x3F }, /* r3B - HP/LO Left Mixer Voice PCM Mono Volume */ + { 60, 0x3F }, /* r3C - HP/LO Right Mixer Voice PCM Mono Volume */ + { 61, 0x3F }, /* r3D - Aux PCM Left Mixer Input Path Volume */ + { 62, 0x3F }, /* r3E - Aux PCM Right Mixer Input Path Volume */ + { 63, 0x3F }, /* r3F - Aux PCM Left Mixer Volume */ + { 64, 0x3F }, /* r40 - Aux PCM Left Mixer Volume */ + { 65, 0x3F }, /* r41 - Aux PCM Left Mixer Audio PCM L Volume */ + { 66, 0x3F }, /* r42 - Aux PCM Right Mixer Audio PCM R Volume */ + { 67, 0x3F }, /* r43 - Aux PCM Left Mixer Voice PCM Volume */ + { 68, 0x3F }, /* r44 - Aux PCM Right Mixer Voice PCM Volume */ + { 69, 0x3F }, /* r45 - Audio PCM Left Input Path Volume */ + { 70, 0x3F }, /* r46 - Audio PCM Right Input Path Volume */ + { 71, 0x3F }, /* r47 - Audio PCM Left Mixer Aux PCM L Volume */ + { 72, 0x3F }, /* r48 - Audio PCM Right Mixer Aux PCM R Volume */ + { 73, 0x3F }, /* r49 - Audio PCM Left Mixer Volume */ + { 74, 0x3F }, /* r4A - Audio PCM Right Mixer Volume */ + { 75, 0x3F }, /* r4B - Audio PCM Left Mixer Voice PCM Volume */ + { 76, 0x3F }, /* r4C - Audio PCM Right Mixer Voice PCM Volume */ + { 77, 0x3F }, /* r4D - Voice PCM Left Input Path Volume */ + { 78, 0x3F }, /* r4E - Voice PCM Right Input Path Volume */ + { 79, 0x3F }, /* r4F - Voice PCM Left Mixer Aux PCM L Volume */ + { 80, 0x3F }, /* r50 - Voice PCM Right Mixer Aux PCM R Volume */ + { 81, 0x3F }, /* r51 - Voice PCM Left Mixer Audio PCM L Volume */ + { 82, 0x3F }, /* r52 - Voice PCM Right Mixer Audio PCM R Volume */ + { 83, 0x3F }, /* r53 - Voice PCM Left Mixer Voice PCM Volume */ + { 84, 0x3F }, /* r54 - Voice PCM Right Mixer Voice PCM Volume */ + { 85, 0xAA }, /* r55 - Mono Mixer Ctl */ + { 86, 0x3F }, /* r56 - SPK Mono Mixer Input Path Volume */ + { 87, 0x3F }, /* r57 - SPK Mono Mixer Aux PCM Mono/L/R Volume */ + { 88, 0x3F }, /* r58 - SPK Mono Mixer Audio PCM Mono/L/R Volume */ + { 89, 0x3F }, /* r59 - SPK Mono Mixer Voice PCM Mono Volume */ + { 90, 0x3F }, /* r5A - SPKLO Mono Mixer Input Path Mono Volume */ + { 91, 0x3F }, /* r5B - SPKLO Mono Mixer Aux Mono/L/R Volume */ + { 92, 0x3F }, /* r5C - SPKLO Mono Mixer Audio Mono/L/R Volume */ + { 93, 0x3F }, /* r5D - SPKLO Mono Mixer Voice Mono Volume */ + { 94, 0x00 }, /* r5E - Interrupt Mask 1 */ + { 95, 0x00 }, /* r5F - Interrupt Mask 2 */ +}; + +static bool cs42l73_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_IS1: + case CS42L73_IS2: + return true; + default: + return false; + } +} + +static bool cs42l73_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_DEVID_AB: + case CS42L73_DEVID_CD: + case CS42L73_DEVID_E: + case CS42L73_REVID: + case CS42L73_PWRCTL1: + case CS42L73_PWRCTL2: + case CS42L73_PWRCTL3: + case CS42L73_CPFCHC: + case CS42L73_OLMBMSDC: + case CS42L73_DMMCC: + case CS42L73_XSPC: + case CS42L73_XSPMMCC: + case CS42L73_ASPC: + case CS42L73_ASPMMCC: + case CS42L73_VSPC: + case CS42L73_VSPMMCC: + case CS42L73_VXSPFS: + case CS42L73_MIOPC: + case CS42L73_ADCIPC: + case CS42L73_MICAPREPGAAVOL: + case CS42L73_MICBPREPGABVOL: + case CS42L73_IPADVOL: + case CS42L73_IPBDVOL: + case CS42L73_PBDC: + case CS42L73_HLADVOL: + case CS42L73_HLBDVOL: + case CS42L73_SPKDVOL: + case CS42L73_ESLDVOL: + case CS42L73_HPAAVOL: + case CS42L73_HPBAVOL: + case CS42L73_LOAAVOL: + case CS42L73_LOBAVOL: + case CS42L73_STRINV: + case CS42L73_XSPINV: + case CS42L73_ASPINV: + case CS42L73_VSPINV: + case CS42L73_LIMARATEHL: + case CS42L73_LIMRRATEHL: + case CS42L73_LMAXHL: + case CS42L73_LIMARATESPK: + case CS42L73_LIMRRATESPK: + case CS42L73_LMAXSPK: + case CS42L73_LIMARATEESL: + case CS42L73_LIMRRATEESL: + case CS42L73_LMAXESL: + case CS42L73_ALCARATE: + case CS42L73_ALCRRATE: + case CS42L73_ALCMINMAX: + case CS42L73_NGCAB: + case CS42L73_ALCNGMC: + case CS42L73_MIXERCTL: + case CS42L73_HLAIPAA: + case CS42L73_HLBIPBA: + case CS42L73_HLAXSPAA: + case CS42L73_HLBXSPBA: + case CS42L73_HLAASPAA: + case CS42L73_HLBASPBA: + case CS42L73_HLAVSPMA: + case CS42L73_HLBVSPMA: + case CS42L73_XSPAIPAA: + case CS42L73_XSPBIPBA: + case CS42L73_XSPAXSPAA: + case CS42L73_XSPBXSPBA: + case CS42L73_XSPAASPAA: + case CS42L73_XSPAASPBA: + case CS42L73_XSPAVSPMA: + case CS42L73_XSPBVSPMA: + case CS42L73_ASPAIPAA: + case CS42L73_ASPBIPBA: + case CS42L73_ASPAXSPAA: + case CS42L73_ASPBXSPBA: + case CS42L73_ASPAASPAA: + case CS42L73_ASPBASPBA: + case CS42L73_ASPAVSPMA: + case CS42L73_ASPBVSPMA: + case CS42L73_VSPAIPAA: + case CS42L73_VSPBIPBA: + case CS42L73_VSPAXSPAA: + case CS42L73_VSPBXSPBA: + case CS42L73_VSPAASPAA: + case CS42L73_VSPBASPBA: + case CS42L73_VSPAVSPMA: + case CS42L73_VSPBVSPMA: + case CS42L73_MMIXCTL: + case CS42L73_SPKMIPMA: + case CS42L73_SPKMXSPA: + case CS42L73_SPKMASPA: + case CS42L73_SPKMVSPMA: + case CS42L73_ESLMIPMA: + case CS42L73_ESLMXSPA: + case CS42L73_ESLMASPA: + case CS42L73_ESLMVSPMA: + case CS42L73_IM1: + case CS42L73_IM2: + return true; + default: + return false; + } +} + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const struct snd_kcontrol_new input_left_mixer[] = { + SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1, + 5, 1, 1), + SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1, + 4, 1, 1), +}; + +static const struct snd_kcontrol_new input_right_mixer[] = { + SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1, + 7, 1, 1), + SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1, + 6, 1, 1), +}; + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mix_texts[] = { + "Left", "Right", "Mono Mix"}; + +static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; + +static const struct soc_enum spk_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_asp_mixer = + SOC_DAPM_ENUM("Route", spk_asp_enum); + +static const struct soc_enum spk_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_xsp_mixer = + SOC_DAPM_ENUM("Route", spk_xsp_enum); + +static const struct soc_enum esl_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_asp_mixer = + SOC_DAPM_ENUM("Route", esl_asp_enum); + +static const struct soc_enum esl_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_xsp_mixer = + SOC_DAPM_ENUM("Route", esl_xsp_enum); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new vsp_output_mux = + SOC_DAPM_ENUM("Route", vsp_output_mux_enum); + +static const struct snd_kcontrol_new xsp_output_mux = + SOC_DAPM_ENUM("Route", xsp_output_mux_enum); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_ENUM("Charge Pump Frequency", charge_pump_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("VSPOUT Mono/Stereo Select", vsp_output_mux_enum), + SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum), +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1), + SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM, + 0, 0, input_left_mixer, + ARRAY_SIZE(input_left_mixer)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM, + 0, 0, input_right_mixer, + ARRAY_SIZE(input_right_mixer)), + + SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + + SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM, + 0, 0, &esl_xsp_mixer), + + SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM, + 0, 0, &esl_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM, + 0, 0, &spk_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM, + 0, 0, &spk_xsp_mixer), + + SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + /* SPKLO EARSPK Paths */ + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"EAR Amp", "Switch", "ESL DAC"}, + {"SPKLO Amp", "Switch", "ESL DAC"}, + + {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, + {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, + + {"ESL Mixer", NULL, "ESL-ASP Mux"}, + {"ESL Mixer", NULL, "ESL-XSP Mux"}, + + {"ESL-ASP Mux", "Left", "ASPINL"}, + {"ESL-ASP Mux", "Right", "ASPINR"}, + {"ESL-ASP Mux", "Mono Mix", "ASPINM"}, + + {"ESL-XSP Mux", "Left", "XSPINL"}, + {"ESL-XSP Mux", "Right", "XSPINR"}, + {"ESL-XSP Mux", "Mono Mix", "XSPINM"}, + + /* Speakerphone Paths */ + {"SPKOUT", NULL, "SPK Amp"}, + {"SPK Amp", "Switch", "SPK DAC"}, + + {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, + {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, + + {"SPK Mixer", NULL, "SPK-ASP Mux"}, + {"SPK Mixer", NULL, "SPK-XSP Mux"}, + + {"SPK-ASP Mux", "Left", "ASPINL"}, + {"SPK-ASP Mux", "Mono Mix", "ASPINM"}, + {"SPK-ASP Mux", "Right", "ASPINR"}, + + {"SPK-XSP Mux", "Left", "XSPINL"}, + {"SPK-XSP Mux", "Mono Mix", "XSPINM"}, + {"SPK-XSP Mux", "Right", "XSPINR"}, + + /* HP LineOUT Paths */ + {"HPOUTA", NULL, "HP Amp"}, + {"HPOUTB", NULL, "HP Amp"}, + {"LINEOUTA", NULL, "LO Amp"}, + {"LINEOUTB", NULL, "LO Amp"}, + + {"HP Amp", "Switch", "HL Left DAC"}, + {"HP Amp", "Switch", "HL Right DAC"}, + {"LO Amp", "Switch", "HL Left DAC"}, + {"LO Amp", "Switch", "HL Right DAC"}, + + {"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"}, + /* Loopback */ + {"HL Left DAC", "HL-IP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-IP Volume", "HL Right Mixer"}, + {"HL Left Mixer", NULL, "Input Left Capture"}, + {"HL Right Mixer", NULL, "Input Right Capture"}, + + {"HL Left Mixer", NULL, "ASPINL"}, + {"HL Right Mixer", NULL, "ASPINR"}, + {"HL Left Mixer", NULL, "XSPINL"}, + {"HL Right Mixer", NULL, "XSPINR"}, + {"HL Left Mixer", NULL, "VSPIN"}, + {"HL Right Mixer", NULL, "VSPIN"}, + + /* Capture Paths */ + {"MIC1", NULL, "MIC1 Bias"}, + {"PGA Left Mux", "Mic 1", "MIC1"}, + {"MIC2", NULL, "MIC2 Bias"}, + {"PGA Right Mux", "Mic 2", "MIC2"}, + + {"PGA Left Mux", "Line A", "LINEINA"}, + {"PGA Right Mux", "Line B", "LINEINB"}, + + {"PGA Left", NULL, "PGA Left Mux"}, + {"PGA Right", NULL, "PGA Right Mux"}, + + {"ADC Left", NULL, "PGA Left"}, + {"ADC Right", NULL, "PGA Right"}, + + {"Input Left Capture", "ADC Left Input", "ADC Left"}, + {"Input Right Capture", "ADC Right Input", "ADC Right"}, + {"Input Left Capture", "DMIC Left Input", "DMIC Left"}, + {"Input Right Capture", "DMIC Right Input", "DMIC Right"}, + + /* Audio Capture */ + {"ASPL Output Mixer", NULL, "Input Left Capture"}, + {"ASPR Output Mixer", NULL, "Input Right Capture"}, + + {"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"}, + {"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"}, + + /* Auxillary Capture */ + {"XSPL Output Mixer", NULL, "Input Left Capture"}, + {"XSPR Output Mixer", NULL, "Input Right Capture"}, + + {"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"}, + {"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"}, + + {"XSPOUTL", NULL, "XSPL Output Mixer"}, + {"XSPOUTR", NULL, "XSPR Output Mixer"}, + + /* Voice Capture */ + {"VSPL Output Mixer", NULL, "Input Left Capture"}, + {"VSPR Output Mixer", NULL, "Input Left Capture"}, + + {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, + {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + + {"VSPOUTL", NULL, "VSPL Output Mixer"}, + {"VSPOUTR", NULL, "VSPR Output Mixer"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc &= MCK_SCLK_64FS; + } else { + /* CS42L73 Slave */ + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + cs42l73_update_asrc(codec, id, srate); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l73->regmap, false); + regcache_sync(cs42l73->regmap); + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + regcache_sync(cs42l73->regmap); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs42l73->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + regcache_cache_only(cs42l73->regmap, true); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static struct regmap_config cs42l73_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L73_MAX_REGISTER, + .reg_defaults = cs42l73_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l73_reg_defaults), + .volatile_reg = cs42l73_volatile_register, + .readable_reg = cs42l73_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + if (IS_ERR(cs42l73->regmap)) { + ret = PTR_ERR(cs42l73->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_CD, ®); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, ®); + devid |= (reg & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(&i2c_client->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + goto err_regmap; + } + + ret = regmap_read(cs42l73->regmap, CS42L73_REVID, ®); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_regmap; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + regcache_cache_only(cs42l73->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l73->regmap); + +err: + kfree(cs42l73); + + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l73->regmap); + + kfree(cs42l73); + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, "); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 000000000000..7c3bf7fd2f99 --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev + * Brian Austin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ +#define CS42L73_MAX_REGISTER 0x61 /* Total Registers */ +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- cgit v1.2.1 From 43fa8e53379003c92e6aabaf7b3e19bd482947bb Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 17 Nov 2011 12:01:28 +0200 Subject: ASoC: alc5632: Remove unrelevant registers and name the relevant Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 161 ++++++++++++++------------------------------- 1 file changed, 49 insertions(+), 112 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 6bfbbc71154d..96605423437d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,118 +36,55 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, - { 1, 0x0000 }, - { 2, 0x8080 }, - { 3, 0x0000 }, - { 4, 0x8080 }, - { 5, 0x0000 }, - { 6, 0x8080 }, - { 7, 0x0000 }, - { 8, 0xC800 }, - { 9, 0x0000 }, - { 10, 0xE808 }, - { 11, 0x0000 }, - { 12, 0x1010 }, - { 13, 0x0000 }, - { 14, 0x0808 }, - { 15, 0x0000 }, - { 16, 0xEE0F }, - { 17, 0x0000 }, - { 18, 0xCBCB }, - { 19, 0x0000 }, - { 20, 0x7F7F }, - { 21, 0x0000 }, - { 22, 0x0000 }, - { 23, 0x0000 }, - { 24, 0xE010 }, - { 25, 0x0000 }, - { 26, 0x0000 }, - { 27, 0x0000 }, - { 28, 0x8008 }, - { 29, 0x0000 }, - { 30, 0x0000 }, - { 31, 0x0000 }, - { 32, 0x0000 }, - { 33, 0x0000 }, - { 34, 0x0000 }, - { 35, 0x0000 }, - { 36, 0x00C0 }, - { 37, 0x0000 }, - { 38, 0xEF00 }, - { 39, 0x0000 }, - { 40, 0x0000 }, - { 41, 0x0000 }, - { 42, 0x0000 }, - { 43, 0x0000 }, - { 44, 0x0000 }, - { 45, 0x0000 }, - { 46, 0x0000 }, - { 47, 0x0000 }, - { 48, 0x0000 }, - { 49, 0x0000 }, - { 50, 0x0000 }, - { 51, 0x0000 }, - { 52, 0x8000 }, - { 53, 0x0000 }, - { 54, 0x0000 }, - { 55, 0x0000 }, - { 56, 0x0000 }, - { 57, 0x0000 }, - { 58, 0x0000 }, - { 59, 0x0000 }, - { 60, 0x0000 }, - { 61, 0x0000 }, - { 62, 0x8000 }, - { 63, 0x0000 }, - { 64, 0x0C0A }, - { 65, 0x0000 }, - { 66, 0x0000 }, - { 67, 0x0000 }, - { 68, 0x0000 }, - { 69, 0x0000 }, - { 70, 0x0000 }, - { 71, 0x0000 }, - { 72, 0x0000 }, - { 73, 0x0000 }, - { 74, 0x0000 }, - { 75, 0x0000 }, - { 76, 0xBE3E }, - { 77, 0x0000 }, - { 78, 0xBE3E }, - { 79, 0x0000 }, - { 80, 0x0000 }, - { 81, 0x0000 }, - { 82, 0x0000 }, - { 83, 0x0000 }, - { 84, 0x803A }, - { 85, 0x0000 }, - { 86, 0x0000 }, - { 87, 0x0000 }, - { 88, 0x0000 }, - { 89, 0x0000 }, - { 90, 0x0009 }, - { 91, 0x0000 }, - { 92, 0x0000 }, - { 93, 0x0000 }, - { 94, 0x3000 }, - { 95, 0x0000 }, - { 96, 0x3075 }, - { 97, 0x0000 }, - { 98, 0x1010 }, - { 99, 0x0000 }, - { 100, 0x3110 }, - { 101, 0x0000 }, - { 102, 0x0000 }, - { 103, 0x0000 }, - { 104, 0x0553 }, - { 105, 0x0000 }, - { 106, 0x0000 }, - { 107, 0x0000 }, - { 108, 0x0000 }, - { 109, 0x0000 }, - { 110, 0x0000 }, - { 111, 0x0000 }, + { 0, 0x59B4 }, /* R0 - Reset */ + { 2, 0x8080 }, /* R2 - Speaker Output Volume */ + { 4, 0x8080 }, /* R4 - Headphone Output Volume */ + { 6, 0x8080 }, /* R6 - AUXOUT Volume */ + { 8, 0xC800 }, /* R8 - Phone Input */ + { 10, 0xE808 }, /* R10 - LINE_IN Volume */ + { 12, 0x1010 }, /* R12 - STEREO DAC Input Volume */ + { 14, 0x0808 }, /* R14 - MIC Input Volume */ + { 16, 0xEE0F }, /* R16 - Stereo DAC and MIC Routing Control */ + { 18, 0xCBCB }, /* R18 - ADC Record Gain */ + { 20, 0x7F7F }, /* R20 - ADC Record Mixer Control */ + { 24, 0xE010 }, /* R24 - Voice DAC Volume */ + { 28, 0x8008 }, /* R28 - Output Mixer Control */ + { 34, 0x0000 }, /* R34 - Microphone Control */ + { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost + Control */ + { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ + { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC + Function Select */ + { 52, 0x8000 }, /* R52 - Main Serial Data Port Control + (Stereo I2S) */ + { 54, 0x0000 }, /* R54 - Extend Serial Data Port Control + (VoDAC_I2S/PCM) */ + { 58, 0x0000 }, /* R58 - Power Management Addition 1 */ + { 60, 0x0000 }, /* R60 - Power Management Addition 2 */ + { 62, 0x8000 }, /* R62 - Power Management Addition 3 */ + { 64, 0x0C0A }, /* R64 - General Purpose Control Register 1 */ + { 66, 0x0000 }, /* R66 - General Purpose Control Register 2 */ + { 68, 0x0000 }, /* R68 - PLL1 Control */ + { 70, 0x0000 }, /* R70 - PLL2 Control */ + { 76, 0xBE3E }, /* R76 - GPIO Pin Configuration */ + { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ + { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ + { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ + { 84, 0x803A }, /* R84 - GPIO Pin Status */ + { 86, 0x0000 }, /* R86 - Pin Sharing */ + { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ + { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ + { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ + { 94, 0x3000 }, /* R94 - MISC Control */ + { 96, 0x3075 }, /* R96 - Stereo DAC Clock Control_1 */ + { 98, 0x1010 }, /* R98 - Stereo DAC Clock Control_2 */ + { 100, 0x3110 }, /* R100 - VoDAC_PCM Clock Control_1 */ + { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect + Block Control */ + { 106, 0x0000 }, /* R106 - Private Register Address */ + { 108, 0x0000 }, /* R108 - Private Register Data */ + { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC + HPF Control */ }; /* codec private data */ -- cgit v1.2.1 From 9b4156cbe9c18605d42ecf80bb99364d0c5b884a Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 17 Nov 2011 12:01:29 +0200 Subject: ASoC: alc5632: Added support of two undocumented registers There are two undocumented registers in use in alc5632_i2c_probe function. It must be added to support future rewrite of this function to use regmap API completely. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 96605423437d..be102281c0a7 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -104,6 +104,8 @@ static bool alc5632_volatile_register(struct device *dev, case ALC5632_OVER_CURR_STATUS: case ALC5632_HID_CTRL_DATA: case ALC5632_EQ_CTRL: + case ALC5632_VENDOR_ID1: + case ALC5632_VENDOR_ID2: return true; default: -- cgit v1.2.1 From 277c01bb45a4924b1741fd41c353860e8d530f6f Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 17 Nov 2011 12:01:30 +0200 Subject: ASoC: alc5632: Update of i2c_probe function to use regmap API only Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 67 +++++++++++++++++++++------------------------- 1 file changed, 30 insertions(+), 37 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index be102281c0a7..c32eadebb762 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1053,48 +1053,14 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5632_priv *alc5632; - int ret, vid1, vid2; - - vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); - if (vid1 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; - } else { - dev_info(&client->dev, "got vid1: %x\n", vid1); - } - vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); - - vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; - } else { - dev_info(&client->dev, "got vid2: %x\n", vid2); - } - vid2 = (vid2 & 0xff); - - if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { - dev_err(&client->dev, "unknown or wrong codec\n"); - dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", - 0x10ec, id->driver_data, - vid1, vid2); - return -ENODEV; - } + int ret, ret1, ret2; + unsigned int vid1, vid2; alc5632 = devm_kzalloc(&client->dev, sizeof(struct alc5632_priv), GFP_KERNEL); if (alc5632 == NULL) return -ENOMEM; - alc5632->id = vid2; - switch (alc5632->id) { - case 0x5c: - alc5632_dai.name = "alc5632-hifi"; - break; - default: - return -EINVAL; - } - i2c_set_clientdata(client, alc5632); alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); @@ -1104,6 +1070,24 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, return ret; } + ret1 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID1, &vid1); + ret2 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID2, &vid2); + if (ret1 != 0 || ret2 != 0) { + dev_err(&client->dev, + "Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2); + regmap_exit(alc5632->regmap); + return -EIO; + } + + vid2 >>= 8; + + if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) { + dev_err(&client->dev, + "Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2); + regmap_exit(alc5632->regmap); + return -EINVAL; + } + ret = alc5632_reset(alc5632->regmap); if (ret < 0) { dev_err(&client->dev, "Failed to issue reset\n"); @@ -1111,7 +1095,16 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, return ret; } - ret = snd_soc_register_codec(&client->dev, + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5632, &alc5632_dai, 1); if (ret < 0) { -- cgit v1.2.1 From 2f534edc1505ab7c6abd4b3389ba3842bf643235 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 17 Nov 2011 18:48:42 +0200 Subject: ASoC: alc5632: Remove volatile registers from regmap defaults There is no need to provide defaults for the volatile registers and doing so might cause confusion. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index c32eadebb762..2d77665eb854 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,7 +36,6 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, /* R0 - Reset */ { 2, 0x8080 }, /* R2 - Speaker Output Volume */ { 4, 0x8080 }, /* R4 - Headphone Output Volume */ { 6, 0x8080 }, /* R6 - AUXOUT Volume */ @@ -52,7 +51,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 34, 0x0000 }, /* R34 - Microphone Control */ { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost Control */ - { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC Function Select */ { 52, 0x8000 }, /* R52 - Main Serial Data Port Control @@ -70,9 +68,7 @@ static struct reg_default alc5632_reg_defaults[] = { { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ - { 84, 0x803A }, /* R84 - GPIO Pin Status */ { 86, 0x0000 }, /* R86 - Pin Sharing */ - { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ { 94, 0x3000 }, /* R94 - MISC Control */ @@ -82,9 +78,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect Block Control */ { 106, 0x0000 }, /* R106 - Private Register Address */ - { 108, 0x0000 }, /* R108 - Private Register Data */ - { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC - HPF Control */ }; /* codec private data */ -- cgit v1.2.1 From cb555318ca5dd5c1426c7a639aa1e90a88c8f024 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 12:59:52 +0000 Subject: ASoC: Use table based init for wm8731_snd_controls Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03f6f8d..f5161f383fc4 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -553,9 +553,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) /* Disable bypass path by default */ snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); - snd_soc_add_controls(codec, wm8731_snd_controls, - ARRAY_SIZE(wm8731_snd_controls)); - /* Regulators will have been enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -595,6 +592,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = wm8731_intercon, .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), + .controls = wm8731_snd_controls, + .num_controls = ARRAY_SIZE(wm8731_snd_controls), }; static const struct of_device_id wm8731_of_match[] = { -- cgit v1.2.1 From ea0756158110fef07b2f2975e38890cecde6a1ce Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 19 Nov 2011 10:15:53 +0800 Subject: ASoC: cs42l73: Return proper error code if device id mismatch Return -ENODEV instead of 0 if device id mismatch. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6fe259aceb7b..fdd8aa207730 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1369,6 +1369,7 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, if (devid != CS42L73_DEVID) { + ret = -ENODEV; dev_err(&i2c_client->dev, "CS42L73 Device ID (%X). Expected %X\n", devid, CS42L73_DEVID); -- cgit v1.2.1 From 8421f620da9717dade941d0dc9570ad731b4a9ca Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 19 Nov 2011 10:17:36 +0800 Subject: ASoC: cs42l73: Show correct revision id Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index fdd8aa207730..1b773bf4b52d 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1383,7 +1383,7 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, - "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l73->regmap, true); -- cgit v1.2.1 From afe713089a5cce680ff76fab554c42d5cbb577d0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 19 Nov 2011 13:45:34 +0800 Subject: ASoC: Remove redundant regcache_sync call in cs42l73_resume It's done in cs42l73_set_bias_level when the dapm.bias_level is switching from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 1b773bf4b52d..5544f1417a25 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1270,10 +1270,6 @@ static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) static int cs42l73_resume(struct snd_soc_codec *codec) { - - struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - regcache_sync(cs42l73->regmap); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } -- cgit v1.2.1 From 56a926dd72bd836f71216ba5b034adb7f48e80e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 15:46:51 +0000 Subject: ASoC: Convert WM8753 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 3a629d0d690e..13156c836c9a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -486,7 +486,7 @@ SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("VREF"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8753_dapm_routes[] = { /* left mixer */ {"Left Mixer", "Left Playback Switch", "Left DAC"}, {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -640,17 +640,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ACOP", NULL, "ALC Mixer"}, }; -static int wm8753_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct _pll_div { u32 div2:1; @@ -1467,10 +1456,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8753_LINVOL, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_RINVOL, 0x0100, 0x0100); - snd_soc_add_controls(codec, wm8753_snd_controls, - ARRAY_SIZE(wm8753_snd_controls)); - wm8753_add_widgets(codec); - return 0; } @@ -1492,6 +1477,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_size = ARRAY_SIZE(wm8753_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8753_reg, + + .controls = wm8753_snd_controls, + .num_controls = ARRAY_SIZE(wm8753_snd_controls), + .dapm_widgets = wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8753_dapm_widgets), + .dapm_routes = wm8753_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8753_dapm_routes), }; static const struct of_device_id wm8753_of_match[] = { -- cgit v1.2.1 From f733547aa30b9e85cc5f2739f3c236408157d2ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Nov 2011 14:47:24 +0000 Subject: ASoC: Remove WM5100 DSP memory windows from register default data They're all volatile so shouldn't have defaults and as we've got pages into the DSP memory the registers themselves aren't that useful - a further patch adding support for the DSPs will provide direct diagnostic access to the DSP memories. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 168 --------------------------------------- 1 file changed, 168 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 3e90dea4e267..9a18fae68204 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -697,90 +697,6 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_HPLPF3_2: case WM5100_HPLPF4_1: case WM5100_HPLPF4_2: - case WM5100_DSP1_DM_0: - case WM5100_DSP1_DM_1: - case WM5100_DSP1_DM_2: - case WM5100_DSP1_DM_3: - case WM5100_DSP1_DM_508: - case WM5100_DSP1_DM_509: - case WM5100_DSP1_DM_510: - case WM5100_DSP1_DM_511: - case WM5100_DSP1_PM_0: - case WM5100_DSP1_PM_1: - case WM5100_DSP1_PM_2: - case WM5100_DSP1_PM_3: - case WM5100_DSP1_PM_4: - case WM5100_DSP1_PM_5: - case WM5100_DSP1_PM_1530: - case WM5100_DSP1_PM_1531: - case WM5100_DSP1_PM_1532: - case WM5100_DSP1_PM_1533: - case WM5100_DSP1_PM_1534: - case WM5100_DSP1_PM_1535: - case WM5100_DSP1_ZM_0: - case WM5100_DSP1_ZM_1: - case WM5100_DSP1_ZM_2: - case WM5100_DSP1_ZM_3: - case WM5100_DSP1_ZM_2044: - case WM5100_DSP1_ZM_2045: - case WM5100_DSP1_ZM_2046: - case WM5100_DSP1_ZM_2047: - case WM5100_DSP2_DM_0: - case WM5100_DSP2_DM_1: - case WM5100_DSP2_DM_2: - case WM5100_DSP2_DM_3: - case WM5100_DSP2_DM_508: - case WM5100_DSP2_DM_509: - case WM5100_DSP2_DM_510: - case WM5100_DSP2_DM_511: - case WM5100_DSP2_PM_0: - case WM5100_DSP2_PM_1: - case WM5100_DSP2_PM_2: - case WM5100_DSP2_PM_3: - case WM5100_DSP2_PM_4: - case WM5100_DSP2_PM_5: - case WM5100_DSP2_PM_1530: - case WM5100_DSP2_PM_1531: - case WM5100_DSP2_PM_1532: - case WM5100_DSP2_PM_1533: - case WM5100_DSP2_PM_1534: - case WM5100_DSP2_PM_1535: - case WM5100_DSP2_ZM_0: - case WM5100_DSP2_ZM_1: - case WM5100_DSP2_ZM_2: - case WM5100_DSP2_ZM_3: - case WM5100_DSP2_ZM_2044: - case WM5100_DSP2_ZM_2045: - case WM5100_DSP2_ZM_2046: - case WM5100_DSP2_ZM_2047: - case WM5100_DSP3_DM_0: - case WM5100_DSP3_DM_1: - case WM5100_DSP3_DM_2: - case WM5100_DSP3_DM_3: - case WM5100_DSP3_DM_508: - case WM5100_DSP3_DM_509: - case WM5100_DSP3_DM_510: - case WM5100_DSP3_DM_511: - case WM5100_DSP3_PM_0: - case WM5100_DSP3_PM_1: - case WM5100_DSP3_PM_2: - case WM5100_DSP3_PM_3: - case WM5100_DSP3_PM_4: - case WM5100_DSP3_PM_5: - case WM5100_DSP3_PM_1530: - case WM5100_DSP3_PM_1531: - case WM5100_DSP3_PM_1532: - case WM5100_DSP3_PM_1533: - case WM5100_DSP3_PM_1534: - case WM5100_DSP3_PM_1535: - case WM5100_DSP3_ZM_0: - case WM5100_DSP3_ZM_1: - case WM5100_DSP3_ZM_2: - case WM5100_DSP3_ZM_3: - case WM5100_DSP3_ZM_2044: - case WM5100_DSP3_ZM_2045: - case WM5100_DSP3_ZM_2046: - case WM5100_DSP3_ZM_2047: return 1; default: return 0; @@ -1445,88 +1361,4 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ - { 0x4000, 0x0000 }, /* R16384 - DSP1 DM 0 */ - { 0x4001, 0x0000 }, /* R16385 - DSP1 DM 1 */ - { 0x4002, 0x0000 }, /* R16386 - DSP1 DM 2 */ - { 0x4003, 0x0000 }, /* R16387 - DSP1 DM 3 */ - { 0x41FC, 0x0000 }, /* R16892 - DSP1 DM 508 */ - { 0x41FD, 0x0000 }, /* R16893 - DSP1 DM 509 */ - { 0x41FE, 0x0000 }, /* R16894 - DSP1 DM 510 */ - { 0x41FF, 0x0000 }, /* R16895 - DSP1 DM 511 */ - { 0x4800, 0x0000 }, /* R18432 - DSP1 PM 0 */ - { 0x4801, 0x0000 }, /* R18433 - DSP1 PM 1 */ - { 0x4802, 0x0000 }, /* R18434 - DSP1 PM 2 */ - { 0x4803, 0x0000 }, /* R18435 - DSP1 PM 3 */ - { 0x4804, 0x0000 }, /* R18436 - DSP1 PM 4 */ - { 0x4805, 0x0000 }, /* R18437 - DSP1 PM 5 */ - { 0x4DFA, 0x0000 }, /* R19962 - DSP1 PM 1530 */ - { 0x4DFB, 0x0000 }, /* R19963 - DSP1 PM 1531 */ - { 0x4DFC, 0x0000 }, /* R19964 - DSP1 PM 1532 */ - { 0x4DFD, 0x0000 }, /* R19965 - DSP1 PM 1533 */ - { 0x4DFE, 0x0000 }, /* R19966 - DSP1 PM 1534 */ - { 0x4DFF, 0x0000 }, /* R19967 - DSP1 PM 1535 */ - { 0x5000, 0x0000 }, /* R20480 - DSP1 ZM 0 */ - { 0x5001, 0x0000 }, /* R20481 - DSP1 ZM 1 */ - { 0x5002, 0x0000 }, /* R20482 - DSP1 ZM 2 */ - { 0x5003, 0x0000 }, /* R20483 - DSP1 ZM 3 */ - { 0x57FC, 0x0000 }, /* R22524 - DSP1 ZM 2044 */ - { 0x57FD, 0x0000 }, /* R22525 - DSP1 ZM 2045 */ - { 0x57FE, 0x0000 }, /* R22526 - DSP1 ZM 2046 */ - { 0x57FF, 0x0000 }, /* R22527 - DSP1 ZM 2047 */ - { 0x6000, 0x0000 }, /* R24576 - DSP2 DM 0 */ - { 0x6001, 0x0000 }, /* R24577 - DSP2 DM 1 */ - { 0x6002, 0x0000 }, /* R24578 - DSP2 DM 2 */ - { 0x6003, 0x0000 }, /* R24579 - DSP2 DM 3 */ - { 0x61FC, 0x0000 }, /* R25084 - DSP2 DM 508 */ - { 0x61FD, 0x0000 }, /* R25085 - DSP2 DM 509 */ - { 0x61FE, 0x0000 }, /* R25086 - DSP2 DM 510 */ - { 0x61FF, 0x0000 }, /* R25087 - DSP2 DM 511 */ - { 0x6800, 0x0000 }, /* R26624 - DSP2 PM 0 */ - { 0x6801, 0x0000 }, /* R26625 - DSP2 PM 1 */ - { 0x6802, 0x0000 }, /* R26626 - DSP2 PM 2 */ - { 0x6803, 0x0000 }, /* R26627 - DSP2 PM 3 */ - { 0x6804, 0x0000 }, /* R26628 - DSP2 PM 4 */ - { 0x6805, 0x0000 }, /* R26629 - DSP2 PM 5 */ - { 0x6DFA, 0x0000 }, /* R28154 - DSP2 PM 1530 */ - { 0x6DFB, 0x0000 }, /* R28155 - DSP2 PM 1531 */ - { 0x6DFC, 0x0000 }, /* R28156 - DSP2 PM 1532 */ - { 0x6DFD, 0x0000 }, /* R28157 - DSP2 PM 1533 */ - { 0x6DFE, 0x0000 }, /* R28158 - DSP2 PM 1534 */ - { 0x6DFF, 0x0000 }, /* R28159 - DSP2 PM 1535 */ - { 0x7000, 0x0000 }, /* R28672 - DSP2 ZM 0 */ - { 0x7001, 0x0000 }, /* R28673 - DSP2 ZM 1 */ - { 0x7002, 0x0000 }, /* R28674 - DSP2 ZM 2 */ - { 0x7003, 0x0000 }, /* R28675 - DSP2 ZM 3 */ - { 0x77FC, 0x0000 }, /* R30716 - DSP2 ZM 2044 */ - { 0x77FD, 0x0000 }, /* R30717 - DSP2 ZM 2045 */ - { 0x77FE, 0x0000 }, /* R30718 - DSP2 ZM 2046 */ - { 0x77FF, 0x0000 }, /* R30719 - DSP2 ZM 2047 */ - { 0x8000, 0x0000 }, /* R32768 - DSP3 DM 0 */ - { 0x8001, 0x0000 }, /* R32769 - DSP3 DM 1 */ - { 0x8002, 0x0000 }, /* R32770 - DSP3 DM 2 */ - { 0x8003, 0x0000 }, /* R32771 - DSP3 DM 3 */ - { 0x81FC, 0x0000 }, /* R33276 - DSP3 DM 508 */ - { 0x81FD, 0x0000 }, /* R33277 - DSP3 DM 509 */ - { 0x81FE, 0x0000 }, /* R33278 - DSP3 DM 510 */ - { 0x81FF, 0x0000 }, /* R33279 - DSP3 DM 511 */ - { 0x8800, 0x0000 }, /* R34816 - DSP3 PM 0 */ - { 0x8801, 0x0000 }, /* R34817 - DSP3 PM 1 */ - { 0x8802, 0x0000 }, /* R34818 - DSP3 PM 2 */ - { 0x8803, 0x0000 }, /* R34819 - DSP3 PM 3 */ - { 0x8804, 0x0000 }, /* R34820 - DSP3 PM 4 */ - { 0x8805, 0x0000 }, /* R34821 - DSP3 PM 5 */ - { 0x8DFA, 0x0000 }, /* R36346 - DSP3 PM 1530 */ - { 0x8DFB, 0x0000 }, /* R36347 - DSP3 PM 1531 */ - { 0x8DFC, 0x0000 }, /* R36348 - DSP3 PM 1532 */ - { 0x8DFD, 0x0000 }, /* R36349 - DSP3 PM 1533 */ - { 0x8DFE, 0x0000 }, /* R36350 - DSP3 PM 1534 */ - { 0x8DFF, 0x0000 }, /* R36351 - DSP3 PM 1535 */ - { 0x9000, 0x0000 }, /* R36864 - DSP3 ZM 0 */ - { 0x9001, 0x0000 }, /* R36865 - DSP3 ZM 1 */ - { 0x9002, 0x0000 }, /* R36866 - DSP3 ZM 2 */ - { 0x9003, 0x0000 }, /* R36867 - DSP3 ZM 3 */ - { 0x97FC, 0x0000 }, /* R38908 - DSP3 ZM 2044 */ - { 0x97FD, 0x0000 }, /* R38909 - DSP3 ZM 2045 */ - { 0x97FE, 0x0000 }, /* R38910 - DSP3 ZM 2046 */ - { 0x97FF, 0x0000 }, /* R38911 - DSP3 ZM 2047 */ }; -- cgit v1.2.1 From 12a7a709a09aac117b630264cdd526e20d4d0ce2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 12:11:37 +0000 Subject: ASoC: Remove conditional I2C usage from tlv320aic3x driver The driver only supports I2C so doesn't need to do things conditionally. Signed-off-by: Mark Brown Acked-by: Jarkko Nikula --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 14cb5534ce8b..2e2bf18253c8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1481,7 +1481,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .resume = aic3x_resume, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B @@ -1548,27 +1547,22 @@ static struct i2c_driver aic3x_i2c_driver = { .remove = aic3x_i2c_remove, .id_table = aic3x_i2c_id, }; -#endif static int __init aic3x_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&aic3x_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n", ret); } -#endif return ret; } module_init(aic3x_modinit); static void __exit aic3x_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&aic3x_i2c_driver); -#endif } module_exit(aic3x_exit); -- cgit v1.2.1 From 717b8fae3873b4c83dda2274e8190f538c442000 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 18 Nov 2011 16:05:13 +0800 Subject: ASoC: cs42l73: Unify the way to define bits of register Current code defines some bits with left shift to the proper bit defined in datasheet, but some don't. Unify the definition with proper left shift and adjust the code accordingly. Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 6 +++--- sound/soc/codecs/cs42l73.h | 18 +++++++++--------- 2 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 5544f1417a25..672da66dc662 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1028,13 +1028,13 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= (PCM_MODE0 << 4); + spc |= PCM_MODE0; if (inv == SND_SOC_DAIFMT_IB_NF) - spc |= (PCM_MODE1 << 4); + spc |= PCM_MODE1; break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= (PCM_MODE1 << 4); + spc |= PCM_MODE1; break; default: return -EINVAL; diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index 7c3bf7fd2f99..f30a4c4d62e6 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -162,16 +162,16 @@ /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ #define SP_3ST (1 << 7) -#define SPDIF_I2S 0 +#define SPDIF_I2S (0 << 6) #define SPDIF_PCM (1 << 6) -#define PCM_MODE0 0 -#define PCM_MODE1 1 -#define PCM_MODE2 2 -#define PCM_BO_MSBLSB 0 -#define PCM_BO_LSBMSB 1 -#define MCK_SCLK_64FS 0 -#define MCK_SCLK_MCLK 2 -#define MCK_SCLK_PREMCLK 3 +#define PCM_MODE0 (0 << 4) +#define PCM_MODE1 (1 << 4) +#define PCM_MODE2 (2 << 4) +#define PCM_MODE_MASK (3 << 4) +#define PCM_BIT_ORDER (1 << 3) +#define MCK_SCLK_64FS (0 << 0) +#define MCK_SCLK_MCLK (2 << 0) +#define MCK_SCLK_PREMCLK (3 << 0) /* CS42L73_xSPMMCC */ #define MS_MASTER (1 << 7) -- cgit v1.2.1 From dbb1f516375b3019373f2177b46e334b47a6d8bf Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 18 Nov 2011 17:16:22 +0800 Subject: ASoC: cs42l73: Make inv and format to be unsigned int Fix below smatch warning: sound/soc/codecs/cs42l73.c +1030 cs42l73_set_dai_fmt(53) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1032 cs42l73_set_dai_fmt(55) error: inv is never equal to 768 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1036 cs42l73_set_dai_fmt(59) error: inv is never equal to 1024 (wrong type 0 - 255). Reported-by: Dan Carpenter Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 672da66dc662..9f52a940bcad 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -979,7 +979,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_codec *codec = codec_dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); u8 id = codec_dai->id; - u8 inv, format; + unsigned int inv, format; u8 spc, mmcc; spc = snd_soc_read(codec, CS42L73_SPC(id)); -- cgit v1.2.1 From 404417e6b49694931241aada4209e1ec0b4eefee Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Nov 2011 15:13:30 +0000 Subject: ASoC: Staticise and constify cs42l73_reg_defaults It's not exported and doesn't need to change. Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9f52a940bcad..d09578f397da 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -42,7 +42,7 @@ struct cs42l73_private { u32 mclk; }; -struct reg_default cs42l73_reg_defaults[] = { +static const struct reg_default cs42l73_reg_defaults[] = { { 1, 0x42 }, /* r01 - Device ID A&B */ { 2, 0xA7 }, /* r02 - Device ID C&D */ { 3, 0x30 }, /* r03 - Device ID E */ -- cgit v1.2.1 From 1db3c98e18962557ce9d9fd0b895c8a6e41c96fd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Nov 2011 23:19:41 +0000 Subject: ASoC: Convert wm8776 to table based control and DAPM init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index bfdc52370ad0..f967c59dbbef 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -414,12 +414,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); - snd_soc_add_controls(codec, wm8776_snd_controls, - ARRAY_SIZE(wm8776_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8776_dapm_widgets, - ARRAY_SIZE(wm8776_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes)); - return ret; } @@ -439,6 +433,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .reg_cache_size = ARRAY_SIZE(wm8776_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8776_reg, + + .controls = wm8776_snd_controls, + .num_controls = ARRAY_SIZE(wm8776_snd_controls), + .dapm_widgets = wm8776_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8776_dapm_widgets), + .dapm_routes = routes, + .num_dapm_routes = ARRAY_SIZE(routes), }; static const struct of_device_id wm8776_of_match[] = { -- cgit v1.2.1 From 99c92ae4ffca81f4dfba3b7648734c56d0b32d4c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:14 -0700 Subject: ASoC: Tegra PCM: Use module_platform_driver This saves some boiler-plate code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 436def1dfa39..90345ee138f3 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -392,18 +392,7 @@ static struct platform_driver tegra_pcm_driver = { .probe = tegra_pcm_platform_probe, .remove = __devexit_p(tegra_pcm_platform_remove), }; - -static int __init snd_tegra_pcm_init(void) -{ - return platform_driver_register(&tegra_pcm_driver); -} -module_init(snd_tegra_pcm_init); - -static void __exit snd_tegra_pcm_exit(void) -{ - platform_driver_unregister(&tegra_pcm_driver); -} -module_exit(snd_tegra_pcm_exit); +module_platform_driver(tegra_pcm_driver); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra PCM ASoC driver"); -- cgit v1.2.1 From f2296d7bf19a210a462a57bb90b1c9263d18a4ee Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:15 -0700 Subject: ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver module_platform_drive saves some boiler-plate code. The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.c | 45 ++++++++++----------------------------------- 1 file changed, 10 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 3b55a44146af..fa3a4426cbdd 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -172,11 +172,11 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) if (das) return -ENODEV; - das = kzalloc(sizeof(struct tegra_das), GFP_KERNEL); + das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); if (!das) { dev_err(&pdev->dev, "Can't allocate tegra_das\n"); ret = -ENOMEM; - goto exit; + goto err; } das->dev = &pdev->dev; @@ -184,22 +184,22 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "No memory resource\n"); ret = -ENODEV; - goto err_free; + goto err; } - region = request_mem_region(res->start, resource_size(res), - pdev->name); + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); if (!region) { dev_err(&pdev->dev, "Memory region already claimed\n"); ret = -EBUSY; - goto err_free; + goto err; } - das->regs = ioremap(res->start, resource_size(res)); + das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (!das->regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; - goto err_release; + goto err; } tegra_das_debug_add(das); @@ -208,32 +208,18 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) return 0; -err_release: - release_mem_region(res->start, resource_size(res)); -err_free: - kfree(das); +err: das = NULL; -exit: return ret; } static int __devexit tegra_das_remove(struct platform_device *pdev) { - struct resource *res; - if (!das) return -ENODEV; - platform_set_drvdata(pdev, NULL); - tegra_das_debug_remove(das); - iounmap(das->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - kfree(das); das = NULL; return 0; @@ -246,18 +232,7 @@ static struct platform_driver tegra_das_driver = { .name = DRV_NAME, }, }; - -static int __init tegra_das_modinit(void) -{ - return platform_driver_register(&tegra_das_driver); -} -module_init(tegra_das_modinit); - -static void __exit tegra_das_modexit(void) -{ - platform_driver_unregister(&tegra_das_driver); -} -module_exit(tegra_das_modexit); +module_platform_driver(tegra_das_driver); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra DAS driver"); -- cgit v1.2.1 From 65713ce8442b42c6f688bd8b0950a49d8f4dcf5f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:13 -0700 Subject: ASoC: Tegra: Move DAS configuration into machine drivers This removes potentially machine-specific routing knowledge from the I2S driverinto the machine drivers, which is better equipped to know what the appropriate routing configuration is. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 18 ------------------ sound/soc/tegra/tegra_wm8903.c | 13 +++++++++++++ sound/soc/tegra/trimslice.c | 15 +++++++++++++++ 3 files changed, 28 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 6728fab8c411..33e62fcdfce3 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -42,7 +42,6 @@ #include #include -#include "tegra_das.h" #include "tegra_i2s.h" #define DRV_NAME "tegra-i2s" @@ -363,23 +362,6 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) return -EINVAL; } - /* - * FIXME: Until a codec driver exists for the tegra DAS, hard-code a - * 1:1 mapping between audio controllers and audio ports. - */ - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1 + pdev->id, - TEGRA_DAS_DAP_SEL_DAC1 + pdev->id); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAP connection\n"); - return ret; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1 + pdev->id, - TEGRA_DAS_DAC_SEL_DAP1 + pdev->id); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAC connection\n"); - return ret; - } - i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL); if (!i2s) { dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index a81cf39257bf..9b0ee1510935 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -249,6 +249,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct tegra_wm8903_platform_data *pdata = machine->pdata; int ret; + ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, + TEGRA_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(card->dev, "Can't set up DAS DAP connection\n"); + return ret; + } + ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, + TEGRA_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(card->dev, "Can't set up DAS DAC connection\n"); + return ret; + } + if (gpio_is_valid(pdata->gpio_spkr_en)) { ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); if (ret) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index b3a7efa6d960..2699a6fa45f9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -118,7 +118,22 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = { static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, + TEGRA_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(card->dev, "Can't set up DAS DAP connection\n"); + return ret; + } + ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, + TEGRA_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(card->dev, "Can't set up DAS DAC connection\n"); + return ret; + } snd_soc_dapm_nc_pin(dapm, "LHPOUT"); snd_soc_dapm_nc_pin(dapm, "RHPOUT"); -- cgit v1.2.1 From bea0ed0825be288f9fc98696fc476066776b26be Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:16 -0700 Subject: ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver module_platform_drive saves some boiler-plate code. The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 45 ++++++++++----------------------------------- 1 file changed, 10 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33e62fcdfce3..76014f0d8a29 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -362,11 +362,11 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) return -EINVAL; } - i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL); + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); if (!i2s) { dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); ret = -ENOMEM; - goto exit; + goto err; } dev_set_drvdata(&pdev->dev, i2s); @@ -374,7 +374,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) if (IS_ERR(i2s->clk_i2s)) { dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); - goto err_free; + goto err; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -391,19 +391,19 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) goto err_clk_put; } - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); ret = -EBUSY; goto err_clk_put; } - i2s->regs = ioremap(mem->start, resource_size(mem)); + i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!i2s->regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; - goto err_release; + goto err_clk_put; } i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; @@ -422,43 +422,29 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_unmap; + goto err_clk_put; } tegra_i2s_debug_add(i2s, pdev->id); return 0; -err_unmap: - iounmap(i2s->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); err_clk_put: clk_put(i2s->clk_i2s); -err_free: - kfree(i2s); -exit: +err: return ret; } static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) { struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); - struct resource *res; snd_soc_unregister_dai(&pdev->dev); tegra_i2s_debug_remove(i2s); - iounmap(i2s->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - clk_put(i2s->clk_i2s); - kfree(i2s); - return 0; } @@ -470,18 +456,7 @@ static struct platform_driver tegra_i2s_driver = { .probe = tegra_i2s_platform_probe, .remove = __devexit_p(tegra_i2s_platform_remove), }; - -static int __init snd_tegra_i2s_init(void) -{ - return platform_driver_register(&tegra_i2s_driver); -} -module_init(snd_tegra_i2s_init); - -static void __exit snd_tegra_i2s_exit(void) -{ - platform_driver_unregister(&tegra_i2s_driver); -} -module_exit(snd_tegra_i2s_exit); +module_platform_driver(tegra_i2s_driver); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); -- cgit v1.2.1 From 85e7652d89293a6dab42bfd31f276f8bc072d4c5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 23 Nov 2011 11:40:40 +0100 Subject: ASoC: Constify snd_soc_dai_ops structs Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure") introduced the possibility to have constant DAI ops structures, yet this is barley used in both existing drivers and also new drivers being submitted, although none of them modifies its DAI ops structure. The later is not surprising since existing drivers are often used as templates for new drivers. So this patch just constifies all existing snd_soc_dai_ops structs to eliminate the issue altogether. The patch was generated with the following coccinelle semantic patch: // @@ identifier ops; @@ -struct snd_soc_dai_ops ops = +const struct snd_soc_dai_ops ops = { ... }; // Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 2 +- sound/soc/blackfin/bf5xx-tdm.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 4 ++-- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/max98088.c | 4 ++-- sound/soc/codecs/max98095.c | 6 +++--- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sn95031.c | 8 ++++---- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/stac9766.c | 4 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 6 +++--- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 4 ++-- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 4 ++-- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 6 +++--- sound/soc/codecs/wm8995.c | 6 +++--- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 6 +++--- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/davinci/davinci-vcif.c | 2 +- sound/soc/ep93xx/ep93xx-ac97.c | 2 +- sound/soc/ep93xx/ep93xx-i2s.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 4 ++-- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/imx/imx-ssi.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-hdmi.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +++--- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/s6000/s6000-i2s.c | 2 +- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_i2s.c | 2 +- sound/soc/tegra/tegra_spdif.c | 2 +- 121 files changed, 147 insertions(+), 147 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 71225090c49f..a67fc9b7dbe7 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -719,7 +719,7 @@ static int atmel_ssc_remove(struct snd_soc_dai *dai) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops atmel_ssc_dai_ops = { +static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 726bd651a105..7771934b93e2 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -195,7 +195,7 @@ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops alchemy_ac97c_ops = { +static const struct snd_soc_dai_ops alchemy_ac97c_ops = { .startup = alchemy_ac97c_startup, }; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 6bcf48f5884c..2d5f755ac99c 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -201,7 +201,7 @@ static int au1xi2s_startup(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops au1xi2s_dai_ops = { +static const const struct snd_soc_dai_ops au1xi2s_dai_ops = { .startup = au1xi2s_startup, .trigger = au1xi2s_trigger, .hw_params = au1xi2s_hw_params, diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 0c6acd547141..87daf456b1c9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -337,7 +337,7 @@ static int au1xpsc_ac97_probe(struct snd_soc_dai *dai) return au1xpsc_ac97_workdata ? 0 : -ENODEV; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { +static const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { .startup = au1xpsc_ac97_startup, .trigger = au1xpsc_ac97_trigger, .hw_params = au1xpsc_ac97_hw_params, diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index e03c5ce01b30..f7714d50bdaf 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -265,7 +265,7 @@ static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { +static const struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 00cc3e00b2fe..b31662e3a428 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -223,7 +223,7 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { +static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index a822d1ee1380..7876b5090fda 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -226,7 +226,7 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) #define bf5xx_tdm_resume NULL #endif -static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { +static const struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 5ca122e51183..ea305b88cb55 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1198,14 +1198,14 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_dai_ops pm860x_pcm_dai_ops = { +static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = { .digital_mute = pm860x_digital_mute, .hw_params = pm860x_pcm_hw_params, .set_fmt = pm860x_pcm_set_dai_fmt, .set_sysclk = pm860x_set_dai_sysclk, }; -static struct snd_soc_dai_ops pm860x_i2s_dai_ops = { +static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = { .digital_mute = pm860x_digital_mute, .hw_params = pm860x_i2s_hw_params, .set_fmt = pm860x_i2s_set_dai_fmt, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e715186b4300..8f3216793eb5 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -39,7 +39,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops ac97_dai_ops = { +static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 4e5c5726366b..fab0948f7a54 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -189,7 +189,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops ad1836_dai_ops = { +static const struct snd_soc_dai_ops ad1836_dai_ops = { .hw_params = ad1836_hw_params, .set_fmt = ad1836_set_dai_fmt, }; diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 120602130b5c..1901cd222233 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -312,7 +312,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops ad193x_dai_ops = { +static const struct snd_soc_dai_ops ad193x_dai_ops = { .hw_params = ad193x_hw_params, .digital_mute = ad193x_mute, .set_tdm_slot = ad193x_set_tdm_slot, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 45c63028b40d..2e040af9ad57 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1042,7 +1042,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static const struct snd_soc_dai_ops adau1373_dai_ops = { +static const const struct snd_soc_dai_ops adau1373_dai_ops = { .hw_params = adau1373_hw_params, .set_sysclk = adau1373_set_dai_sysclk, .set_fmt = adau1373_set_dai_fmt, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8b7e1c50d6e9..c69bdfe745bb 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -427,7 +427,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, #define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const struct snd_soc_dai_ops adau1701_dai_ops = { +static const const struct snd_soc_dai_ops adau1701_dai_ops = { .set_fmt = adau1701_set_dai_fmt, .hw_params = adau1701_hw_params, .digital_mute = adau1701_digital_mute, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f9f08948e5e8..d927febd02cc 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -718,7 +718,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, adav80x->rate = 0; } -static const struct snd_soc_dai_ops adav80x_dai_ops = { +static const const struct snd_soc_dai_ops adav80x_dai_ops = { .set_fmt = adav80x_set_dai_fmt, .hw_params = adav80x_hw_params, .startup = adav80x_dai_startup, diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index d3b29dce6ed7..152420ca78b8 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -170,7 +170,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); } -static struct snd_soc_dai_ops ak4101_dai_ops = { +static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, .set_fmt = ak4104_set_dai_fmt, }; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 95d782d86e7d..f6c47345bcc8 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -331,7 +331,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops ak4535_dai_ops = { +static const struct snd_soc_dai_ops ak4535_dai_ops = { .hw_params = ak4535_hw_params, .set_fmt = ak4535_set_dai_fmt, .digital_mute = ak4535_mute, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 77838586f358..3657c76cc127 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -442,14 +442,14 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000) #define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) -static struct snd_soc_dai_ops ak4641_i2s_dai_ops = { +static const struct snd_soc_dai_ops ak4641_i2s_dai_ops = { .hw_params = ak4641_i2s_hw_params, .set_fmt = ak4641_i2s_set_dai_fmt, .digital_mute = ak4641_mute, .set_sysclk = ak4641_set_dai_sysclk, }; -static struct snd_soc_dai_ops ak4641_pcm_dai_ops = { +static const struct snd_soc_dai_ops ak4641_pcm_dai_ops = { .hw_params = NULL, /* rates are controlled by BT chip */ .set_fmt = ak4641_pcm_set_dai_fmt, .digital_mute = ak4641_mute, diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 859e0155e18d..c887ddf1061e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -435,7 +435,7 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_dai_ops ak4642_dai_ops = { +static const struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, .set_sysclk = ak4642_dai_set_sysclk, diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index de9ff66d3721..4f5c69f735a9 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -594,7 +594,7 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec, #define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE -static struct snd_soc_dai_ops ak4671_dai_ops = { +static const struct snd_soc_dai_ops ak4671_dai_ops = { .hw_params = ak4671_hw_params, .set_sysclk = ak4671_set_dai_sysclk, .set_fmt = ak4671_set_dai_fmt, diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 984b14bcb605..88647d3ab24b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -839,7 +839,7 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec, | SNDRV_PCM_FMTBIT_S24_LE \ | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops alc5623_dai_ops = { +static const struct snd_soc_dai_ops alc5623_dai_ops = { .hw_params = alc5623_pcm_hw_params, .digital_mute = alc5623_mute, .set_fmt = alc5623_set_dai_fmt, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 2d77665eb854..3f750def8967 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -924,7 +924,7 @@ static int alc5632_set_bias_level(struct snd_soc_codec *codec, | SNDRV_PCM_FMTBIT_S24_LE \ | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops alc5632_dai_ops = { +static const struct snd_soc_dai_ops alc5632_dai_ops = { .hw_params = alc5632_pcm_hw_params, .digital_mute = alc5632_mute, .set_fmt = alc5632_set_dai_fmt, diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd067f79..cbb3028e2008 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -122,7 +122,7 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec, #define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) -static struct snd_soc_dai_ops cq93vc_dai_ops = { +static const struct snd_soc_dai_ops cq93vc_dai_ops = { .digital_mute = cq93vc_mute, .set_sysclk = cq93vc_set_dai_sysclk, }; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 73f46eb459f1..5396b91fa5f1 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -447,7 +447,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { snd_soc_get_volsw, cs4270_soc_put_mute), }; -static struct snd_soc_dai_ops cs4270_dai_ops = { +static const struct snd_soc_dai_ops cs4270_dai_ops = { .hw_params = cs4270_hw_params, .set_sysclk = cs4270_set_dai_sysclk, .set_fmt = cs4270_set_dai_fmt, diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 69fde1506fe1..a6f77a855f45 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -402,7 +402,7 @@ static const struct snd_kcontrol_new cs4271_snd_controls[] = { 7, 1, 1), }; -static struct snd_soc_dai_ops cs4271_dai_ops = { +static const struct snd_soc_dai_ops cs4271_dai_ops = { .hw_params = cs4271_hw_params, .set_sysclk = cs4271_set_dai_sysclk, .set_fmt = cs4271_set_dai_fmt, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 00718b5e747b..e378c4d52027 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -483,7 +483,7 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg); } -static struct snd_soc_dai_ops cs42l51_dai_ops = { +static const struct snd_soc_dai_ops cs42l51_dai_ops = { .hw_params = cs42l51_hw_params, .set_sysclk = cs42l51_set_dai_sysclk, .set_fmt = cs42l51_set_dai_fmt, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d09578f397da..75d80b2e1ec4 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1190,7 +1190,7 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const struct snd_soc_dai_ops cs42l73_ops = { +static const const struct snd_soc_dai_ops cs42l73_ops = { .startup = cs42l73_pcm_startup, .hw_params = cs42l73_pcm_hw_params, .set_fmt = cs42l73_set_dai_fmt, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8b5848a6374c..8ef820fd68c7 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -761,7 +761,7 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ -static struct snd_soc_dai_ops da7210_dai_ops = { +static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, .digital_mute = da7210_mute, diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index e373f8f06907..64a479c3429a 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -206,7 +206,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops jz4740_codec_dai_ops = { +static const struct snd_soc_dai_ops jz4740_codec_dai_ops = { .hw_params = jz4740_codec_hw_params, }; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ebbf63c79c34..48a52a1aaaaa 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1650,14 +1650,14 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec, #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 #define MAX98088_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max98088_dai1_ops = { +static const struct snd_soc_dai_ops max98088_dai1_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai1_set_fmt, .hw_params = max98088_dai1_hw_params, .digital_mute = max98088_dai1_digital_mute, }; -static struct snd_soc_dai_ops max98088_dai2_ops = { +static const struct snd_soc_dai_ops max98088_dai2_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai2_set_fmt, .hw_params = max98088_dai2_hw_params, diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 26d7b089fb9c..cc712d59ab64 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1782,19 +1782,19 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, #define MAX98095_RATES SNDRV_PCM_RATE_8000_96000 #define MAX98095_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max98095_dai1_ops = { +static const struct snd_soc_dai_ops max98095_dai1_ops = { .set_sysclk = max98095_dai_set_sysclk, .set_fmt = max98095_dai1_set_fmt, .hw_params = max98095_dai1_hw_params, }; -static struct snd_soc_dai_ops max98095_dai2_ops = { +static const struct snd_soc_dai_ops max98095_dai2_ops = { .set_sysclk = max98095_dai_set_sysclk, .set_fmt = max98095_dai2_set_fmt, .hw_params = max98095_dai2_hw_params, }; -static struct snd_soc_dai_ops max98095_dai3_ops = { +static const struct snd_soc_dai_ops max98095_dai3_ops = { .set_sysclk = max98095_dai_set_sysclk, .set_fmt = max98095_dai3_set_fmt, .hw_params = max98095_dai3_hw_params, diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 208d2ee61855..94c2b586ed5d 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -254,7 +254,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, #define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max9850_dai_ops = { +static const struct snd_soc_dai_ops max9850_dai_ops = { .hw_params = max9850_hw_params, .set_sysclk = max9850_set_dai_sysclk, .set_fmt = max9850_set_dai_fmt, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 4646e808b90a..dac4d05f512d 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1664,7 +1664,7 @@ static int rt5631_resume(struct snd_soc_codec *codec) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5631_ops = { +static const struct snd_soc_dai_ops rt5631_ops = { .hw_params = rt5631_hifi_pcm_params, .set_fmt = rt5631_hifi_codec_set_dai_fmt, .set_sysclk = rt5631_hifi_codec_set_dai_sysclk, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index bbcf921166f7..1a6564b3684e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -923,7 +923,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops sgtl5000_ops = { +static const struct snd_soc_dai_ops sgtl5000_ops = { .hw_params = sgtl5000_pcm_hw_params, .digital_mute = sgtl5000_digital_mute, .set_fmt = sgtl5000_set_dai_fmt, diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 887d618f4a63..65f2ef986c4f 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -698,21 +698,21 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, } /* Codec DAI section */ -static struct snd_soc_dai_ops sn95031_headset_dai_ops = { +static const struct snd_soc_dai_ops sn95031_headset_dai_ops = { .digital_mute = sn95031_pcm_hs_mute, .hw_params = sn95031_pcm_hw_params, }; -static struct snd_soc_dai_ops sn95031_speaker_dai_ops = { +static const struct snd_soc_dai_ops sn95031_speaker_dai_ops = { .digital_mute = sn95031_pcm_spkr_mute, .hw_params = sn95031_pcm_hw_params, }; -static struct snd_soc_dai_ops sn95031_vib1_dai_ops = { +static const struct snd_soc_dai_ops sn95031_vib1_dai_ops = { .hw_params = sn95031_pcm_hw_params, }; -static struct snd_soc_dai_ops sn95031_vib2_dai_ops = { +static const struct snd_soc_dai_ops sn95031_vib2_dai_ops = { .hw_params = sn95031_pcm_hw_params, }; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 3cb3271c5fe2..620411c384e5 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -498,7 +498,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops ssm2602_dai_ops = { +static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 3b0deafd766b..e2b1cdedb982 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -783,7 +783,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_dai_ops sta32x_dai_ops = { +static const struct snd_soc_dai_ops sta32x_dai_ops = { .hw_params = sta32x_hw_params, .set_sysclk = sta32x_set_dai_sysclk, .set_fmt = sta32x_set_dai_fmt, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 78b2b50271e2..e4783a4f71fd 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -286,11 +286,11 @@ reset: return 0; } -static struct snd_soc_dai_ops stac9766_dai_ops_analog = { +static const struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; -static struct snd_soc_dai_ops stac9766_dai_ops_digital = { +static const struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, }; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 336de8f69a02..9782631df93b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -503,7 +503,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops tlv320aic23_dai_ops = { +static const struct snd_soc_dai_ops tlv320aic23_dai_ops = { .prepare = tlv320aic23_pcm_prepare, .hw_params = tlv320aic23_hw_params, .shutdown = tlv320aic23_shutdown, diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7859bdcc93db..86d1fa38ed2e 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -275,7 +275,7 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) -static struct snd_soc_dai_ops aic26_dai_ops = { +static const struct snd_soc_dai_ops aic26_dai_ops = { .hw_params = aic26_hw_params, .digital_mute = aic26_mute, .set_sysclk = aic26_set_sysclk, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index b21c610051c0..d2e38af46aa1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -597,7 +597,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops aic32x4_ops = { +static const struct snd_soc_dai_ops aic32x4_ops = { .hw_params = aic32x4_hw_params, .digital_mute = aic32x4_mute, .set_fmt = aic32x4_set_dai_fmt, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 2e2bf18253c8..7d665ea3ac62 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1244,7 +1244,7 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops aic3x_dai_ops = { +static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index dc8a2b2bdc1c..abcb97e03405 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1499,7 +1499,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { SNDRV_PCM_RATE_48000) #define DAC33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops dac33_dai_ops = { +static const struct snd_soc_dai_ops dac33_dai_ops = { .startup = dac33_startup, .shutdown = dac33_shutdown, .hw_params = dac33_hw_params, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f798247ac1b2..2a3a52838e9c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2149,7 +2149,7 @@ static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops twl4030_dai_hifi_ops = { +static const struct snd_soc_dai_ops twl4030_dai_hifi_ops = { .startup = twl4030_startup, .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, @@ -2158,7 +2158,7 @@ static struct snd_soc_dai_ops twl4030_dai_hifi_ops = { .set_tristate = twl4030_set_tristate, }; -static struct snd_soc_dai_ops twl4030_dai_voice_ops = { +static const struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 73e11f022ded..17930edd3a2c 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1397,7 +1397,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static struct snd_soc_dai_ops twl6040_dai_ops = { +static const struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, .prepare = twl6040_prepare, diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a7b8f301bad3..486aef637eed 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -452,7 +452,7 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; -static struct snd_soc_dai_ops uda134x_dai_ops = { +static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, .hw_params = uda134x_hw_params, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index c5ca8cfea60f..6b933efc7ed3 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -643,21 +643,21 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops uda1380_dai_ops = { +static const struct snd_soc_dai_ops uda1380_dai_ops = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .trigger = uda1380_trigger, .set_fmt = uda1380_set_dai_fmt_both, }; -static struct snd_soc_dai_ops uda1380_dai_ops_playback = { +static const struct snd_soc_dai_ops uda1380_dai_ops_playback = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .trigger = uda1380_trigger, .set_fmt = uda1380_set_dai_fmt_playback, }; -static struct snd_soc_dai_ops uda1380_dai_ops_capture = { +static const struct snd_soc_dai_ops uda1380_dai_ops_capture = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .trigger = uda1380_trigger, diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index a85498982991..9531c35dccad 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -386,7 +386,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops wl1273_dai_ops = { +static const struct snd_soc_dai_ops wl1273_dai_ops = { .startup = wl1273_startup, .hw_params = wl1273_hw_params, }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f37d67f4058b..6c79d97ba181 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1661,7 +1661,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops wm5100_dai_ops = { +static const struct snd_soc_dai_ops wm5100_dai_ops = { .set_fmt = wm5100_set_fmt, .hw_params = wm5100_hw_params, }; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 35f3ad83dfb6..3b846c95f07f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1511,7 +1511,7 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect); SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8350_dai_ops = { +static const struct snd_soc_dai_ops wm8350_dai_ops = { .hw_params = wm8350_pcm_hw_params, .digital_mute = wm8350_mute, .trigger = wm8350_pcm_trigger, diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 585def1ffca6..07d84a86e14e 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1316,7 +1316,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, #define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8400_dai_ops = { +static const struct snd_soc_dai_ops wm8400_dai_ops = { .hw_params = wm8400_hw_params, .digital_mute = wm8400_mute, .set_fmt = wm8400_set_dai_fmt, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 07c9cc759e97..26571b25e440 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -509,7 +509,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8510_dai_ops = { +static const struct snd_soc_dai_ops wm8510_dai_ops = { .hw_params = wm8510_pcm_hw_params, .digital_mute = wm8510_mute, .set_fmt = wm8510_set_dai_fmt, diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index db7a6819499f..d0ae82d2b24f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -365,7 +365,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec, #define WM8523_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8523_dai_ops = { +static const struct snd_soc_dai_ops wm8523_dai_ops = { .startup = wm8523_startup, .hw_params = wm8523_hw_params, .set_sysclk = wm8523_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 8212b3c8bfdd..0aa3e4d138f4 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -776,7 +776,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8580_dai_ops_playback = { +static const struct snd_soc_dai_ops wm8580_dai_ops_playback = { .set_sysclk = wm8580_set_sysclk, .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, @@ -785,7 +785,7 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .digital_mute = wm8580_digital_mute, }; -static struct snd_soc_dai_ops wm8580_dai_ops_capture = { +static const struct snd_soc_dai_ops wm8580_dai_ops_capture = { .set_sysclk = wm8580_set_sysclk, .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 076bdb9930a1..a6f1e391314d 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -318,7 +318,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, #define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8711_ops = { +static const struct snd_soc_dai_ops wm8711_ops = { .prepare = wm8711_pcm_prepare, .hw_params = wm8711_hw_params, .shutdown = wm8711_shutdown, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 04b027efd5c0..085c2f81d8c2 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -196,7 +196,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8728_dai_ops = { +static const struct snd_soc_dai_ops wm8728_dai_ops = { .hw_params = wm8728_hw_params, .digital_mute = wm8728_mute, .set_fmt = wm8728_set_dai_fmt, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index ca59622e41d2..28972d875f7c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -465,7 +465,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8731_dai_ops = { +static const struct snd_soc_dai_ops wm8731_dai_ops = { .hw_params = wm8731_hw_params, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index f6aef58845c2..b7d661581ebf 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -521,7 +521,7 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec, #define WM8737_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8737_dai_ops = { +static const struct snd_soc_dai_ops wm8737_dai_ops = { .hw_params = wm8737_hw_params, .set_sysclk = wm8737_set_dai_sysclk, .set_fmt = wm8737_set_dai_fmt, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 57ad22aacc51..e51f4f0a93f4 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -382,7 +382,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, #define WM8741_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8741_dai_ops = { +static const struct snd_soc_dai_ops wm8741_dai_ops = { .startup = wm8741_startup, .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index ca75a8180708..dfb41ad902e1 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -643,7 +643,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8750_dai_ops = { +static const struct snd_soc_dai_ops wm8750_dai_ops = { .hw_params = wm8750_pcm_hw_params, .digital_mute = wm8750_mute, .set_fmt = wm8750_set_dai_fmt, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 13156c836c9a..fb013b152fa6 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1315,7 +1315,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { +static const struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_hifi_set_dai_fmt, @@ -1324,7 +1324,7 @@ static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = { +static const struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = { .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_voice_set_dai_fmt, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index aa05e6507f84..87957e862b9c 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -528,7 +528,7 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec, #define WM8770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8770_dai_ops = { +static const struct snd_soc_dai_ops wm8770_dai_ops = { .digital_mute = wm8770_mute, .hw_params = wm8770_hw_params, .set_fmt = wm8770_set_fmt, diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index f967c59dbbef..223fc5a5c1b0 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -327,14 +327,14 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8776_dac_ops = { +static const struct snd_soc_dai_ops wm8776_dac_ops = { .digital_mute = wm8776_mute, .hw_params = wm8776_hw_params, .set_fmt = wm8776_set_fmt, .set_sysclk = wm8776_set_sysclk, }; -static struct snd_soc_dai_ops wm8776_adc_ops = { +static const struct snd_soc_dai_ops wm8776_adc_ops = { .hw_params = wm8776_hw_params, .set_fmt = wm8776_set_fmt, .set_sysclk = wm8776_set_sysclk, diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9ee072b85975..d99c6a0a0a2d 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -670,7 +670,7 @@ err_reg_get: return ret; } -static struct snd_soc_dai_ops wm8804_dai_ops = { +static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, .set_sysclk = wm8804_set_sysclk, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 17a12c2df8da..a430930cc09f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -987,7 +987,7 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -static struct snd_soc_dai_ops wm8900_dai_ops = { +static const struct snd_soc_dai_ops wm8900_dai_ops = { .hw_params = wm8900_hw_params, .set_clkdiv = wm8900_set_dai_clkdiv, .set_pll = wm8900_set_dai_pll, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 4ad8ebd290e3..812dce95f131 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1732,7 +1732,7 @@ static irqreturn_t wm8903_irq(int irq, void *data) SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8903_dai_ops = { +static const struct snd_soc_dai_ops wm8903_dai_ops = { .hw_params = wm8903_hw_params, .digital_mute = wm8903_digital_mute, .set_fmt = wm8903_set_dai_fmt, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index bb070f835257..f0b0c7a487b3 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2205,7 +2205,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, #define WM8904_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8904_dai_ops = { +static const struct snd_soc_dai_ops wm8904_dai_ops = { .set_sysclk = wm8904_set_sysclk, .set_fmt = wm8904_set_fmt, .set_tdm_slot = wm8904_set_tdm_slot, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 1b5856b4ea7c..0dd1e0c0fc1b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -644,7 +644,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8940_dai_ops = { +static const struct snd_soc_dai_ops wm8940_dai_ops = { .hw_params = wm8940_i2s_hw_params, .set_sysclk = wm8940_set_dai_sysclk, .digital_mute = wm8940_mute, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 3c7198779c31..dbf2a8328a8e 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -859,7 +859,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, #define WM8955_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8955_dai_ops = { +static const struct snd_soc_dai_ops wm8955_dai_ops = { .set_sysclk = wm8955_set_sysclk, .set_fmt = wm8955_set_fmt, .hw_params = wm8955_hw_params, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6e22f9b3d967..06dca88a7332 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -869,7 +869,7 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8960_dai_ops = { +static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 7f2df7ba27f6..783a3d1daf51 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -929,7 +929,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8961_dai_ops = { +static const struct snd_soc_dai_ops wm8961_dai_ops = { .hw_params = wm8961_hw_params, .set_sysclk = wm8961_set_sysclk, .set_fmt = wm8961_set_fmt, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 48b5c95a0648..555311d1ce37 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3503,7 +3503,7 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8962_dai_ops = { +static const struct snd_soc_dai_ops wm8962_dai_ops = { .hw_params = wm8962_hw_params, .set_sysclk = wm8962_set_dai_sysclk, .set_fmt = wm8962_set_dai_fmt, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 3a06a95dd96f..98bfbdd62c60 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -567,7 +567,7 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8971_dai_ops = { +static const struct snd_soc_dai_ops wm8971_dai_ops = { .hw_params = wm8971_pcm_hw_params, .digital_mute = wm8971_mute, .set_fmt = wm8971_set_dai_fmt, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 7bd35b8fdcd2..16569c7a03c1 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -557,7 +557,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, #define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8974_ops = { +static const struct snd_soc_dai_ops wm8974_ops = { .hw_params = wm8974_pcm_hw_params, .digital_mute = wm8974_mute, .set_fmt = wm8974_set_dai_fmt, diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 41ca4d9ac20c..517bb2238d46 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -865,7 +865,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec, #define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8978_dai_ops = { +static const struct snd_soc_dai_ops wm8978_dai_ops = { .hw_params = wm8978_hw_params, .digital_mute = wm8978_mute, .set_fmt = wm8978_set_dai_fmt, diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 58e067b5a6a3..362298cce92c 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1035,7 +1035,7 @@ static int wm8983_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_dai_ops wm8983_dai_ops = { +static const struct snd_soc_dai_ops wm8983_dai_ops = { .digital_mute = wm8983_dac_mute, .hw_params = wm8983_hw_params, .set_fmt = wm8983_set_fmt, diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 36c4ee08e159..9e4481bb1223 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1031,7 +1031,7 @@ err_reg_get: return ret; } -static struct snd_soc_dai_ops wm8985_dai_ops = { +static const struct snd_soc_dai_ops wm8985_dai_ops = { .digital_mute = wm8985_dac_mute, .hw_params = wm8985_hw_params, .set_fmt = wm8985_set_fmt, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 514189d1923e..9d83bed5c210 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -701,7 +701,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, #define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8988_ops = { +static const struct snd_soc_dai_ops wm8988_ops = { .startup = wm8988_pcm_startup, .hw_params = wm8988_pcm_hw_params, .set_fmt = wm8988_set_dai_fmt, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index d4cbec6372db..61c620e5fe4f 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1287,7 +1287,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ -static struct snd_soc_dai_ops wm8990_dai_ops = { +static const struct snd_soc_dai_ops wm8990_dai_ops = { .hw_params = wm8990_hw_params, .digital_mute = wm8990_mute, .set_fmt = wm8990_set_dai_fmt, diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 1d46d59c82a3..ac957ece6785 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1311,7 +1311,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) #define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops wm8991_ops = { +static const struct snd_soc_dai_ops wm8991_ops = { .hw_params = wm8991_hw_params, .digital_mute = wm8991_mute, .set_fmt = wm8991_set_dai_fmt, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d1a142f48b09..780c24cdab6d 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1394,7 +1394,7 @@ out: return 0; } -static struct snd_soc_dai_ops wm8993_ops = { +static const struct snd_soc_dai_ops wm8993_ops = { .set_sysclk = wm8993_set_sysclk, .set_fmt = wm8993_set_dai_fmt, .hw_params = wm8993_hw_params, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9c982e47eb99..73db9806c475 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2531,7 +2531,7 @@ static int wm8994_aif2_probe(struct snd_soc_dai *dai) #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { +static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, @@ -2541,7 +2541,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_tristate = wm8994_set_tristate, }; -static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { +static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, @@ -2551,7 +2551,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_tristate = wm8994_set_tristate, }; -static struct snd_soc_dai_ops wm8994_aif3_dai_ops = { +static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = { .hw_params = wm8994_aif3_hw_params, .set_tristate = wm8994_set_tristate, }; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 3774acb69ddd..8f6a36d7c75b 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2155,7 +2155,7 @@ err_reg_get: #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8995_aif1_dai_ops = { +static const struct snd_soc_dai_ops wm8995_aif1_dai_ops = { .set_sysclk = wm8995_set_dai_sysclk, .set_fmt = wm8995_set_dai_fmt, .hw_params = wm8995_hw_params, @@ -2164,7 +2164,7 @@ static struct snd_soc_dai_ops wm8995_aif1_dai_ops = { .set_tristate = wm8995_set_tristate, }; -static struct snd_soc_dai_ops wm8995_aif2_dai_ops = { +static const struct snd_soc_dai_ops wm8995_aif2_dai_ops = { .set_sysclk = wm8995_set_dai_sysclk, .set_fmt = wm8995_set_dai_fmt, .hw_params = wm8995_hw_params, @@ -2173,7 +2173,7 @@ static struct snd_soc_dai_ops wm8995_aif2_dai_ops = { .set_tristate = wm8995_set_tristate, }; -static struct snd_soc_dai_ops wm8995_aif3_dai_ops = { +static const struct snd_soc_dai_ops wm8995_aif3_dai_ops = { .set_tristate = wm8995_set_tristate, }; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index fd5bb1ad6912..304a0e570cb4 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3052,7 +3052,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8996_dai_ops = { +static const struct snd_soc_dai_ops wm8996_dai_ops = { .set_fmt = wm8996_set_fmt, .hw_params = wm8996_hw_params, .set_sysclk = wm8996_set_sysclk, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index f7c0738a9da6..48bf80baf1d4 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1234,7 +1234,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm9081_dai_ops = { +static const struct snd_soc_dai_ops wm9081_dai_ops = { .hw_params = wm9081_hw_params, .set_fmt = wm9081_set_dai_fmt, .digital_mute = wm9081_digital_mute, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 646b58dda849..edf603281ce7 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -258,7 +258,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops wm9705_dai_ops = { +static const struct snd_soc_dai_ops wm9705_dai_ops = { .prepare = ac97_prepare, }; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 90117f8156e8..fd1812704af8 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -505,11 +505,11 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops wm9712_dai_ops_hifi = { +static const struct snd_soc_dai_ops wm9712_dai_ops_hifi = { .prepare = ac97_prepare, }; -static struct snd_soc_dai_ops wm9712_dai_ops_aux = { +static const struct snd_soc_dai_ops wm9712_dai_ops_aux = { .prepare = ac97_aux_prepare, }; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7167cb6787db..09360b60037c 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1026,19 +1026,19 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -static struct snd_soc_dai_ops wm9713_dai_ops_hifi = { +static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = { .prepare = ac97_hifi_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, }; -static struct snd_soc_dai_ops wm9713_dai_ops_aux = { +static const struct snd_soc_dai_ops wm9713_dai_ops_aux = { .prepare = ac97_aux_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, }; -static struct snd_soc_dai_ops wm9713_dai_ops_voice = { +static const struct snd_soc_dai_ops wm9713_dai_ops_voice = { .hw_params = wm9713_pcm_hw_params, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 300e12118c00..f3d5ae1078be 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -620,7 +620,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { +static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7173df254a91..03cea9d39c4b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -813,7 +813,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { +static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 1f11525d97e8..dae96b85fd6d 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -183,7 +183,7 @@ static int davinci_vcif_startup(struct snd_pcm_substream *substream, #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 -static struct snd_soc_dai_ops davinci_vcif_dai_ops = { +static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 3cd6158d83e1..c423d12a26cf 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -330,7 +330,7 @@ static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { +static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { .startup = ep93xx_ac97_startup, .trigger = ep93xx_ac97_trigger, }; diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 099614e16651..3dba128cc6f1 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -338,7 +338,7 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) #define ep93xx_i2s_resume NULL #endif -static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { +static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .startup = ep93xx_i2s_startup, .shutdown = ep93xx_i2s_shutdown, .hw_params = ep93xx_i2s_hw_params, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 83c4bd5b2dd7..17d857e55efe 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -514,7 +514,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, } } -static struct snd_soc_dai_ops fsl_ssi_dai_ops = { +static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index ad36b095bb79..2fb388f0150b 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -226,12 +226,12 @@ static int psc_ac97_probe(struct snd_soc_dai *cpu_dai) /** * psc_ac97_dai_template: template CPU Digital Audio Interface */ -static struct snd_soc_dai_ops psc_ac97_analog_ops = { +static const struct snd_soc_dai_ops psc_ac97_analog_ops = { .hw_params = psc_ac97_hw_analog_params, .trigger = psc_ac97_trigger, }; -static struct snd_soc_dai_ops psc_ac97_digital_ops = { +static const struct snd_soc_dai_ops psc_ac97_digital_ops = { .hw_params = psc_ac97_hw_digital_params, }; diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 87cf2a5c2b2c..e77a1f20d4d2 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -123,7 +123,7 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * psc_i2s_dai_template: template CPU Digital Audio Interface */ -static struct snd_soc_dai_ops psc_i2s_dai_ops = { +static const struct snd_soc_dai_ops psc_i2s_dai_ops = { .hw_params = psc_i2s_hw_params, .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 4c05e2b8f4d2..eed7041364e6 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -342,7 +342,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { +static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { .hw_params = imx_ssi_hw_params, .set_fmt = imx_ssi_set_dai_fmt, .set_clkdiv = imx_ssi_set_dai_clkdiv, diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index cd22a54b2f14..91255c6e1ee7 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -392,7 +392,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops jz4740_i2s_dai_ops = { +static const struct snd_soc_dai_ops jz4740_i2s_dai_ops = { .startup = jz4740_i2s_startup, .shutdown = jz4740_i2s_shutdown, .trigger = jz4740_i2s_trigger, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 715e841c0507..2b212dcb9ac7 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -373,7 +373,7 @@ static int kirkwood_i2s_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { +static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, .hw_params = kirkwood_i2s_hw_params, diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d24fc2..46d76b52529b 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -550,7 +550,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops mxs_saif_dai_ops = { +static const struct snd_soc_dai_ops mxs_saif_dai_ops = { .startup = mxs_saif_startup, .trigger = mxs_saif_trigger, .prepare = mxs_saif_prepare, diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 9c0edad90d8b..7544d249807e 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -291,7 +291,7 @@ static int nuc900_ac97_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops nuc900_ac97_dai_ops = { +static const struct snd_soc_dai_ops nuc900_ac97_dai_ops = { .trigger = nuc900_ac97_trigger, }; diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index ccb8a6aa1817..a04a4338fdac 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -474,7 +474,7 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) } /* Our codec DAI probably doesn't have its own .ops structure */ -static struct snd_soc_dai_ops ams_delta_dai_ops = { +static const struct snd_soc_dai_ops ams_delta_dai_ops = { .digital_mute = ams_delta_digital_mute, }; diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 36c6eaeffb02..9bb1cf89b4a4 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -83,7 +83,7 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, return err; } -static struct snd_soc_dai_ops omap_hdmi_dai_ops = { +static const struct snd_soc_dai_ops omap_hdmi_dai_ops = { .startup = omap_hdmi_dai_startup, .hw_params = omap_hdmi_dai_hw_params, }; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 4314647e735e..d91e6efd2600 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -599,7 +599,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } -static struct snd_soc_dai_ops mcbsp_dai_ops = { +static const struct snd_soc_dai_ops mcbsp_dai_ops = { .startup = omap_mcbsp_dai_startup, .shutdown = omap_mcbsp_dai_shutdown, .trigger = omap_mcbsp_dai_trigger, diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 41d17067cc73..cc8ceff25dbd 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -367,7 +367,7 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { +static const struct snd_soc_dai_ops omap_mcpdm_dai_ops = { .startup = omap_mcpdm_dai_startup, .shutdown = omap_mcpdm_dai_shutdown, .hw_params = omap_mcpdm_dai_hw_params, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 8ad93ee2e92b..9c9a51ef67c3 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -771,7 +771,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops pxa_ssp_dai_ops = { +static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index ac51c6d25c42..3fec2f35b8f8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -163,15 +163,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { +static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { .hw_params = pxa2xx_ac97_hw_params, }; -static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { +static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { .hw_params = pxa2xx_ac97_hw_aux_params, }; -static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { +static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { .hw_params = pxa2xx_ac97_hw_mic_params, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 11be5952a506..609abd51e55f 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -331,7 +331,7 @@ static int pxa2xx_i2s_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -static struct snd_soc_dai_ops pxa_i2s_dai_ops = { +static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 3052f64b2403..13716a9317fb 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -409,7 +409,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai) SNDRV_PCM_RATE_8000_192000) #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops s6000_i2s_dai_ops = { +static const struct snd_soc_dai_ops s6000_i2s_dai_ops = { .set_fmt = s6000_i2s_set_dai_fmt, .set_clkdiv = s6000_i2s_set_clkdiv, .hw_params = s6000_i2s_hw_params, diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 16521e3ffc0c..09035afdeb74 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -329,12 +329,12 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops s3c_ac97_dai_ops = { +static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; -static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { +static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index bff42bf370b9..03ee8ce46a29 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -923,7 +923,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops samsung_i2s_dai_ops = { +static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { .trigger = i2s_trigger, .hw_params = i2s_hw_params, .set_fmt = i2s_set_fmt, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 05a47cf7f06e..2df2762f3000 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -452,7 +452,7 @@ static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -static struct snd_soc_dai_ops s3c_pcm_dai_ops = { +static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_sysclk = s3c_pcm_set_sysclk, .set_clkdiv = s3c_pcm_set_clkdiv, .trigger = s3c_pcm_trigger, diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7bbec25e6e15..545773d0641c 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -142,7 +142,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { +static const struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { .hw_params = s3c2412_i2s_hw_params, }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 558c64bbed2e..2a98bed2db02 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -444,7 +444,7 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { +static const struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { .trigger = s3c24xx_i2s_trigger, .hw_params = s3c24xx_i2s_hw_params, .set_fmt = s3c24xx_i2s_set_fmt, diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 468cff1bb1af..a1fee1a414c9 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -334,7 +334,7 @@ static int spdif_resume(struct snd_soc_dai *cpu_dai) #define spdif_resume NULL #endif -static struct snd_soc_dai_ops spdif_dai_ops = { +static const struct snd_soc_dai_ops spdif_dai_ops = { .set_sysclk = spdif_set_sysclk, .trigger = spdif_trigger, .hw_params = spdif_hw_params, diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 99ed61024166..aa3033075a0d 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1096,7 +1096,7 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops fsi_dai_ops = { +static const struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index c87e3ff28a0a..a1f307b9a82d 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -static struct snd_soc_dai_ops hac_dai_ops = { +static const struct snd_soc_dai_ops hac_dai_ops = { .hw_params = hac_hw_params, }; diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index edacfeb13b94..93dea49ff1a7 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -707,7 +707,7 @@ epclkget: return ret; } -static struct snd_soc_dai_ops siu_dai_ops = { +static const struct snd_soc_dai_ops siu_dai_ops = { .startup = siu_dai_startup, .shutdown = siu_dai_shutdown, .prepare = siu_dai_prepare, diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index e0c621c0553b..1fda16a00e6a 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -static struct snd_soc_dai_ops ssi_dai_ops = { +static const struct snd_soc_dai_ops ssi_dai_ops = { .startup = ssi_startup, .shutdown = ssi_shutdown, .trigger = ssi_trigger, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685a5d38..bf41d9071f1e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -735,7 +735,7 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); #define snd_soc_resume NULL #endif -static struct snd_soc_dai_ops null_dai_ops = { +static const struct snd_soc_dai_ops null_dai_ops = { }; static int soc_bind_dai_link(struct snd_soc_card *card, int num) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 76014f0d8a29..1acbb5541772 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -305,7 +305,7 @@ static int tegra_i2s_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops tegra_i2s_dai_ops = { +static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { .set_fmt = tegra_i2s_set_fmt, .hw_params = tegra_i2s_hw_params, .trigger = tegra_i2s_trigger, diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index dd11d0c63474..ea9c92036aa1 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -226,7 +226,7 @@ static int tegra_spdif_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops tegra_spdif_dai_ops = { +static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { .hw_params = tegra_spdif_hw_params, .trigger = tegra_spdif_trigger, }; -- cgit v1.2.1 From 186bcda6f6217dc4b5353c3474121bc1194847f6 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:18 -0700 Subject: ASoC: Tegra DAS: Add device tree binding Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index fa3a4426cbdd..5b82b4e79231 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -225,11 +225,18 @@ static int __devexit tegra_das_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id tegra_das_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-das", }, + {}, +}; + static struct platform_driver tegra_das_driver = { .probe = tegra_das_probe, .remove = __devexit_p(tegra_das_remove), .driver = { .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra_das_of_match, }, }; module_platform_driver(tegra_das_driver); @@ -238,3 +245,4 @@ MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra DAS driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_das_of_match); -- cgit v1.2.1 From e4e4c18a930ff11940ba2c525676566bd631706f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:20 -0700 Subject: ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver module_platform_driver saves some boiler-plate code. The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 9b0ee1510935..33feee81668c 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -390,17 +390,19 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) return -EINVAL; } - machine = kzalloc(sizeof(struct tegra_wm8903), GFP_KERNEL); + machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8903), + GFP_KERNEL); if (!machine) { dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err; } machine->pdata = pdata; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err_free_machine; + goto err; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -431,8 +433,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); -err_free_machine: - kfree(machine); +err: return ret; } @@ -460,8 +461,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); - kfree(machine); - return 0; } @@ -474,18 +473,7 @@ static struct platform_driver tegra_wm8903_driver = { .probe = tegra_wm8903_driver_probe, .remove = __devexit_p(tegra_wm8903_driver_remove), }; - -static int __init tegra_wm8903_modinit(void) -{ - return platform_driver_register(&tegra_wm8903_driver); -} -module_init(tegra_wm8903_modinit); - -static void __exit tegra_wm8903_modexit(void) -{ - platform_driver_unregister(&tegra_wm8903_driver); -} -module_exit(tegra_wm8903_modexit); +module_platform_driver(tegra_wm8903_driver); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra+WM8903 machine ASoC driver"); -- cgit v1.2.1 From 45c26091205eb6ad737329c5973f46fd7c122595 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Nov 2011 18:21:21 -0700 Subject: ASoC: Tegra TrimSlice machine: Use devm_ APIs and module_platform_driver module_platform_driver saves some boiler-plate code. The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/trimslice.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 2699a6fa45f9..d564b40756a9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -170,15 +170,17 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) struct tegra_trimslice *trimslice; int ret; - trimslice = kzalloc(sizeof(struct tegra_trimslice), GFP_KERNEL); + trimslice = devm_kzalloc(&pdev->dev, sizeof(struct tegra_trimslice), + GFP_KERNEL); if (!trimslice) { dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err; } ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); if (ret) - goto err_free_trimslice; + goto err; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -195,8 +197,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&trimslice->util_data); -err_free_trimslice: - kfree(trimslice); +err: return ret; } @@ -209,8 +210,6 @@ static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&trimslice->util_data); - kfree(trimslice); - return 0; } @@ -222,18 +221,7 @@ static struct platform_driver tegra_snd_trimslice_driver = { .probe = tegra_snd_trimslice_probe, .remove = __devexit_p(tegra_snd_trimslice_remove), }; - -static int __init snd_tegra_trimslice_init(void) -{ - return platform_driver_register(&tegra_snd_trimslice_driver); -} -module_init(snd_tegra_trimslice_init); - -static void __exit snd_tegra_trimslice_exit(void) -{ - platform_driver_unregister(&tegra_snd_trimslice_driver); -} -module_exit(snd_tegra_trimslice_exit); +module_platform_driver(tegra_snd_trimslice_driver); MODULE_AUTHOR("Mike Rapoport "); MODULE_DESCRIPTION("Trimslice machine ASoC driver"); -- cgit v1.2.1 From 890754a878c887de50bc0c9f9041b8b73bd09937 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 23 Nov 2011 14:11:21 +0100 Subject: ASoC: Cleanup duplicated const Commit 85e7652("ASoC: Constify snd_soc_dai_ops structs") accidentally introduced a few duplicated consts. This patch cleans it up. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/au1x/i2sc.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 2d5f755ac99c..6bcf48f5884c 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -201,7 +201,7 @@ static int au1xi2s_startup(struct snd_pcm_substream *substream, return 0; } -static const const struct snd_soc_dai_ops au1xi2s_dai_ops = { +static const struct snd_soc_dai_ops au1xi2s_dai_ops = { .startup = au1xi2s_startup, .trigger = au1xi2s_trigger, .hw_params = au1xi2s_hw_params, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2e040af9ad57..45c63028b40d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1042,7 +1042,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static const const struct snd_soc_dai_ops adau1373_dai_ops = { +static const struct snd_soc_dai_ops adau1373_dai_ops = { .hw_params = adau1373_hw_params, .set_sysclk = adau1373_set_dai_sysclk, .set_fmt = adau1373_set_dai_fmt, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index c69bdfe745bb..8b7e1c50d6e9 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -427,7 +427,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, #define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const const struct snd_soc_dai_ops adau1701_dai_ops = { +static const struct snd_soc_dai_ops adau1701_dai_ops = { .set_fmt = adau1701_set_dai_fmt, .hw_params = adau1701_hw_params, .digital_mute = adau1701_digital_mute, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index d927febd02cc..f9f08948e5e8 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -718,7 +718,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, adav80x->rate = 0; } -static const const struct snd_soc_dai_ops adav80x_dai_ops = { +static const struct snd_soc_dai_ops adav80x_dai_ops = { .set_fmt = adav80x_set_dai_fmt, .hw_params = adav80x_hw_params, .startup = adav80x_dai_startup, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 75d80b2e1ec4..d09578f397da 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1190,7 +1190,7 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const const struct snd_soc_dai_ops cs42l73_ops = { +static const struct snd_soc_dai_ops cs42l73_ops = { .startup = cs42l73_pcm_startup, .hw_params = cs42l73_pcm_hw_params, .set_fmt = cs42l73_set_dai_fmt, -- cgit v1.2.1 From 16c88583dca05034f284ad5c52f007a47673cf35 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Nov 2011 14:59:54 +0000 Subject: ASoC: Remove unused variable in wm8776 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 223fc5a5c1b0..359319cbc784 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -392,7 +392,6 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); -- cgit v1.2.1 From 2c043bcbf287dc69848054d5c02c55c20f7a7bc5 Mon Sep 17 00:00:00 2001 From: Rajendra Nayak Date: Fri, 18 Nov 2011 16:47:19 +0530 Subject: regulator: pass additional of_node to regulator_register() With device tree support for regulators, its needed that the regulator_dev->dev device has the right of_node attached. To be able to do this add an additional parameter to the regulator_register() api, wherein the dt-adapted driver can then pass this additional info onto the regulator core. Signed-off-by: Rajendra Nayak Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d15695d1c273..fc7ab30572d0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -833,7 +833,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->voltage = voltage; ldo->dev = regulator_register(&ldo->desc, codec->dev, - init_data, ldo); + init_data, ldo, NULL); if (IS_ERR(ldo->dev)) { int ret = PTR_ERR(ldo->dev); -- cgit v1.2.1 From d4a2eca781bfd7323bfd98dbc7fd63c7d613fef2 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 23 Nov 2011 13:33:25 -0700 Subject: ASoC: Tegra I2S: Remove dependency on pdev->id When devices are instantiated from device-tree, pdev->id is set to -1. Rework the driver so it doesn't depend on the ID. Tegra I2S instantiated from board files are configured with pdev name "tegra-i2s" and ID 0 or 1. The driver core then names the device "tegra-i2s.0" or "tegra-i2s.1". This is not changing. When a device is instantiated from device-tree, it will have pdev->name="" and pdev->id=-1. For this reason, the pdev->id value is not something we can rely on. This patch doesn't actually change any names though: When a device is instantiated from device-tree, the overall device name will be "${unit_address}.${node_name}". This causes issues such as clk_get() failures due to lack of a device-name match. To solve that, AUXDATA was invented, to force a specific device name, thus allowing dev_name() to return the same as the non-device-tree case. Tegra currently uses AUXDATA for the I2S controllers. Eventually, AUXDATA will go away, most likely replaced by phandle-based references within the device tree. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 72 +++++++++++++++------------------------------ sound/soc/tegra/tegra_i2s.h | 1 + 2 files changed, 24 insertions(+), 49 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 1acbb5541772..ca4d0c0a913e 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -98,13 +98,11 @@ static const struct file_operations tegra_i2s_debug_fops = { .release = single_release, }; -static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static void tegra_i2s_debug_add(struct tegra_i2s *i2s) { - char name[] = DRV_NAME ".0"; - - snprintf(name, sizeof(name), DRV_NAME".%1d", id); - i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root, - i2s, &tegra_i2s_debug_fops); + i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, + snd_soc_debugfs_root, i2s, + &tegra_i2s_debug_fops); } static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) @@ -311,43 +309,22 @@ static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { .trigger = tegra_i2s_trigger, }; -static struct snd_soc_dai_driver tegra_i2s_dai[] = { - { - .name = DRV_NAME ".0", - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, +static const struct snd_soc_dai_driver tegra_i2s_dai_template = { + .probe = tegra_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - { - .name = DRV_NAME ".1", - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &tegra_i2s_dai_ops, + .symmetric_rates = 1, }; static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) @@ -356,12 +333,6 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) struct resource *mem, *memregion, *dmareq; int ret; - if ((pdev->id < 0) || - (pdev->id >= ARRAY_SIZE(tegra_i2s_dai))) { - dev_err(&pdev->dev, "ID %d out of range\n", pdev->id); - return -EINVAL; - } - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); if (!i2s) { dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); @@ -370,6 +341,9 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); + i2s->dai = tegra_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + i2s->clk_i2s = clk_get(&pdev->dev, NULL); if (IS_ERR(i2s->clk_i2s)) { dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); @@ -418,14 +392,14 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; - ret = snd_soc_register_dai(&pdev->dev, &tegra_i2s_dai[pdev->id]); + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; goto err_clk_put; } - tegra_i2s_debug_add(i2s, pdev->id); + tegra_i2s_debug_add(i2s); return 0; diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h index 2b38a096f46c..15ce1e2e8bde 100644 --- a/sound/soc/tegra/tegra_i2s.h +++ b/sound/soc/tegra/tegra_i2s.h @@ -153,6 +153,7 @@ #define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) struct tegra_i2s { + struct snd_soc_dai_driver dai; struct clk *clk_i2s; int clk_refs; struct tegra_pcm_dma_params capture_dma_data; -- cgit v1.2.1 From 1633281b79fd276f1c7c2fb37c3b97da74e42ae5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 23 Nov 2011 12:42:04 -0700 Subject: ASoC: Implement fully_routed card property A card is fully routed if the DAPM route table describes all connections on the board. When a card is fully routed, some operations can be automated by the ASoC core. The first, and currently only, such operation is described below, and implemented by this patch. Codecs often have a large number of external pins, and not all of these pins will be connected on all board designs. Some machine drivers therefore call snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core never to activate them. However, when a card is fully routed, the information needed to derive the set of unused pins is present in card->dapm_routes. In this case, have the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused codec pin. This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 +++ sound/soc/soc-dapm.c | 73 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b194842c4cc3..2abaf6dcdb0a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1488,6 +1488,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_widgets(&card->dapm); + if (card->fully_routed) + list_for_each_entry(codec, &codec_list, list) + snd_soc_dapm_auto_nc_codec_pins(codec); + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f42e8b9fb17d..1ecd1b4927f9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2947,6 +2947,79 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); +static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, &card->paths, list) { + if ((p->source == w) || (p->sink == w)) { + dev_dbg(card->dev, + "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n", + p->source->name, p->source->id, p->source->dapm, + p->sink->name, p->sink->id, p->sink->dapm); + + /* Connected to something other than the codec */ + if (p->source->dapm != p->sink->dapm) + return true; + /* + * Loopback connection from codec external pin to + * codec external pin + */ + if (p->sink->id == snd_soc_dapm_input) { + switch (p->source->id) { + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + return true; + default: + break; + } + } + } + } + + return false; +} + +/** + * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins + * @codec: The codec whose pins should be processed + * + * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec + * which are unused. Pins are used if they are connected externally to the + * codec, whether that be to some other device, or a loop-back connection to + * the codec itself. + */ +void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_widget *w; + + dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n", + &card->dapm, &codec->dapm); + + list_for_each_entry(w, &card->widgets, list) { + if (w->dapm != dapm) + continue; + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + dev_dbg(card->dev, "Auto NC: Checking widget %s\n", + w->name); + if (!snd_soc_dapm_widget_in_card_paths(card, w)) { + dev_dbg(card->dev, + "... Not in map; disabling\n"); + snd_soc_dapm_nc_pin(dapm, w->name); + } + break; + default: + break; + } + } +} + /** * snd_soc_dapm_free - free dapm resources * @dapm: DAPM context -- cgit v1.2.1 From 6e5fdba9c9d4e2fdb19bf19633cb7b9bb72dccb1 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 23 Nov 2011 12:42:05 -0700 Subject: ASoC: Tegra+WM903 machine: Set the new fully_routed flag Set card.fully_routed to request the ASoC core calculated unused codec pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded calls. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 33feee81668c..b260f54a4462 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -331,27 +331,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); - /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony()) - snd_soc_dapm_nc_pin(dapm, "IN1L"); - if (!machine_is_seaboard() && !machine_is_aebl()) - snd_soc_dapm_nc_pin(dapm, "IN1R"); - snd_soc_dapm_nc_pin(dapm, "IN2L"); - if (!machine_is_kaen()) - snd_soc_dapm_nc_pin(dapm, "IN2R"); - snd_soc_dapm_nc_pin(dapm, "IN3L"); - snd_soc_dapm_nc_pin(dapm, "IN3R"); - - if (machine_is_aebl()) { - snd_soc_dapm_nc_pin(dapm, "LON"); - snd_soc_dapm_nc_pin(dapm, "RON"); - snd_soc_dapm_nc_pin(dapm, "ROP"); - snd_soc_dapm_nc_pin(dapm, "LOP"); - } else { - snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); - snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); - } - return 0; } @@ -375,6 +354,7 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tegra_wm8903_dapm_widgets), + .fully_routed = true, }; static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) -- cgit v1.2.1 From 504855d171f4183ac231a5ecdf0273ac249cda2b Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 23 Nov 2011 12:42:06 -0700 Subject: ASoC: TrimSlice machine: Set the new fully_routed flag Set card.fully_routed to request the ASoC core calculated unused codec pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded calls. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/trimslice.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index d564b40756a9..043eb7c7eb73 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -119,7 +119,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, @@ -135,10 +134,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_nc_pin(dapm, "LHPOUT"); - snd_soc_dapm_nc_pin(dapm, "RHPOUT"); - snd_soc_dapm_nc_pin(dapm, "MICIN"); - return 0; } @@ -162,6 +157,7 @@ static struct snd_soc_card snd_soc_trimslice = { .num_dapm_widgets = ARRAY_SIZE(trimslice_dapm_widgets), .dapm_routes = trimslice_audio_map, .num_dapm_routes = ARRAY_SIZE(trimslice_audio_map), + .fully_routed = true, }; static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) -- cgit v1.2.1 From 39afd66cead742e99c051d6f3b07f89d09eebbbb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Nov 2011 21:33:07 +0000 Subject: ASoC: Add fully_routed flag to Speyside machines Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 3 +-- sound/soc/samsung/speyside_wm8962.c | 1 + 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 85bf541a771d..efa5187f6197 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -222,8 +222,6 @@ static struct snd_soc_dai_link speyside_dai[] = { static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_nc_pin(dapm, "LINEOUT"); - /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); @@ -308,6 +306,7 @@ static struct snd_soc_card speyside = { .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, .late_probe = speyside_late_probe, }; diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index e3e27166cc50..a681c8d74118 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -208,6 +208,7 @@ static struct snd_soc_card speyside_wm8962 = { .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, .late_probe = speyside_wm8962_late_probe, }; -- cgit v1.2.1 From 45f3121615b2b354f7d95d30f795bc5fe0043e92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 23 Nov 2011 16:55:34 -0800 Subject: ASoC: fsi-ak4642: modify specification method of FSI / ak464x Current fsi-ak4642 was using id_entry name in order to specify FSI port and ak464x codec. But it was no sense, no flexibility. Platform can specify FSI/ak464x pair by this patch. Acked-by: Paul Mundt Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 114 +++++----------------------------------------- 1 file changed, 11 insertions(+), 103 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index dff64b95f5dc..11d2d7ff29d9 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -58,27 +58,23 @@ static struct platform_device *fsi_snd_device; static int fsi_ak4642_probe(struct platform_device *pdev) { int ret = -ENOMEM; - const struct platform_device_id *id_entry; - struct fsi_ak4642_data *pdata; + struct fsi_ak4642_info *pinfo = pdev->dev.platform_data; - id_entry = pdev->id_entry; - if (!id_entry) { - dev_err(&pdev->dev, "unknown fsi ak4642\n"); - return -ENODEV; + if (!pinfo) { + dev_err(&pdev->dev, "no info for fsi ak4642\n"); + goto out; } - pdata = (struct fsi_ak4642_data *)id_entry->driver_data; - - fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); + fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id); if (!fsi_snd_device) goto out; - fsi_dai_link.name = pdata->name; - fsi_dai_link.stream_name = pdata->name; - fsi_dai_link.cpu_dai_name = pdata->cpu_dai; - fsi_dai_link.platform_name = pdata->platform; - fsi_dai_link.codec_name = pdata->codec; - fsi_soc_card.name = pdata->card; + fsi_dai_link.name = pinfo->name; + fsi_dai_link.stream_name = pinfo->name; + fsi_dai_link.cpu_dai_name = pinfo->cpu_dai; + fsi_dai_link.platform_name = pinfo->platform; + fsi_dai_link.codec_name = pinfo->codec; + fsi_soc_card.name = pinfo->card; platform_set_drvdata(fsi_snd_device, &fsi_soc_card); ret = platform_device_add(fsi_snd_device); @@ -96,100 +92,12 @@ static int fsi_ak4642_remove(struct platform_device *pdev) return 0; } -static struct fsi_ak4642_data fsi_a_ak4642 = { - .name = "AK4642", - .card = "FSIA-AK4642", - .cpu_dai = "fsia-dai", - .codec = "ak4642-codec.0-0012", - .platform = "sh_fsi.0", - .id = FSI_PORT_A, -}; - -static struct fsi_ak4642_data fsi_b_ak4642 = { - .name = "AK4642", - .card = "FSIB-AK4642", - .cpu_dai = "fsib-dai", - .codec = "ak4642-codec.0-0012", - .platform = "sh_fsi.0", - .id = FSI_PORT_B, -}; - -static struct fsi_ak4642_data fsi_a_ak4643 = { - .name = "AK4643", - .card = "FSIA-AK4643", - .cpu_dai = "fsia-dai", - .codec = "ak4642-codec.0-0013", - .platform = "sh_fsi.0", - .id = FSI_PORT_A, -}; - -static struct fsi_ak4642_data fsi_b_ak4643 = { - .name = "AK4643", - .card = "FSIB-AK4643", - .cpu_dai = "fsib-dai", - .codec = "ak4642-codec.0-0013", - .platform = "sh_fsi.0", - .id = FSI_PORT_B, -}; - -static struct fsi_ak4642_data fsi2_a_ak4642 = { - .name = "AK4642", - .card = "FSI2A-AK4642", - .cpu_dai = "fsia-dai", - .codec = "ak4642-codec.0-0012", - .platform = "sh_fsi2", - .id = FSI_PORT_A, -}; - -static struct fsi_ak4642_data fsi2_b_ak4642 = { - .name = "AK4642", - .card = "FSI2B-AK4642", - .cpu_dai = "fsib-dai", - .codec = "ak4642-codec.0-0012", - .platform = "sh_fsi2", - .id = FSI_PORT_B, -}; - -static struct fsi_ak4642_data fsi2_a_ak4643 = { - .name = "AK4643", - .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", - .codec = "ak4642-codec.0-0013", - .platform = "sh_fsi2", - .id = FSI_PORT_A, -}; - -static struct fsi_ak4642_data fsi2_b_ak4643 = { - .name = "AK4643", - .card = "FSI2B-AK4643", - .cpu_dai = "fsib-dai", - .codec = "ak4642-codec.0-0013", - .platform = "sh_fsi2", - .id = FSI_PORT_B, -}; - -static struct platform_device_id fsi_id_table[] = { - /* FSI */ - { "sh_fsi_a_ak4642", (kernel_ulong_t)&fsi_a_ak4642 }, - { "sh_fsi_b_ak4642", (kernel_ulong_t)&fsi_b_ak4642 }, - { "sh_fsi_a_ak4643", (kernel_ulong_t)&fsi_a_ak4643 }, - { "sh_fsi_b_ak4643", (kernel_ulong_t)&fsi_b_ak4643 }, - - /* FSI 2 */ - { "sh_fsi2_a_ak4642", (kernel_ulong_t)&fsi2_a_ak4642 }, - { "sh_fsi2_b_ak4642", (kernel_ulong_t)&fsi2_b_ak4642 }, - { "sh_fsi2_a_ak4643", (kernel_ulong_t)&fsi2_a_ak4643 }, - { "sh_fsi2_b_ak4643", (kernel_ulong_t)&fsi2_b_ak4643 }, - {}, -}; - static struct platform_driver fsi_ak4642 = { .driver = { .name = "fsi-ak4642-audio", }, .probe = fsi_ak4642_probe, .remove = fsi_ak4642_remove, - .id_table = fsi_id_table, }; static int __init fsi_ak4642_init(void) -- cgit v1.2.1 From ee18f6314fa16376d53c29ecf9704011f2ce8180 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 12:07:55 +0800 Subject: ASoC: Convert ep93xx directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Mika Westerberg Acked-by: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 12 +----------- sound/soc/ep93xx/ep93xx-ac97.c | 12 +----------- sound/soc/ep93xx/ep93xx-i2s.c | 13 +------------ sound/soc/ep93xx/ep93xx-pcm.c | 13 +------------ sound/soc/ep93xx/simone.c | 12 +----------- sound/soc/ep93xx/snappercl15.c | 13 +------------ 6 files changed, 6 insertions(+), 69 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 51930b6a83af..6b90c757cf4c 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -131,17 +131,7 @@ static struct platform_driver edb93xx_driver = { .remove = __devexit_p(edb93xx_remove), }; -static int __init edb93xx_init(void) -{ - return platform_driver_register(&edb93xx_driver); -} -module_init(edb93xx_init); - -static void __exit edb93xx_exit(void) -{ - platform_driver_unregister(&edb93xx_driver); -} -module_exit(edb93xx_exit); +module_platform_driver(edb93xx_driver); MODULE_AUTHOR("Alexander Sverdlin "); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index c423d12a26cf..0678637abd66 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -449,17 +449,7 @@ static struct platform_driver ep93xx_ac97_driver = { }, }; -static int __init ep93xx_ac97_init(void) -{ - return platform_driver_register(&ep93xx_ac97_driver); -} -module_init(ep93xx_ac97_init); - -static void __exit ep93xx_ac97_exit(void) -{ - platform_driver_unregister(&ep93xx_ac97_driver); -} -module_exit(ep93xx_ac97_exit); +module_platform_driver(ep93xx_ac97_driver); MODULE_DESCRIPTION("EP93xx AC97 ASoC Driver"); MODULE_AUTHOR("Mika Westerberg "); diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 3dba128cc6f1..f7a62348e3fe 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -464,18 +464,7 @@ static struct platform_driver ep93xx_i2s_driver = { }, }; -static int __init ep93xx_i2s_init(void) -{ - return platform_driver_register(&ep93xx_i2s_driver); -} - -static void __exit ep93xx_i2s_exit(void) -{ - platform_driver_unregister(&ep93xx_i2s_driver); -} - -module_init(ep93xx_i2s_init); -module_exit(ep93xx_i2s_exit); +module_platform_driver(ep93xx_i2s_driver); MODULE_ALIAS("platform:ep93xx-i2s"); MODULE_AUTHOR("Ryan Mallon"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index d00230a591b1..a2de9c42b702 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -339,18 +339,7 @@ static struct platform_driver ep93xx_pcm_driver = { .remove = __devexit_p(ep93xx_soc_platform_remove), }; -static int __init ep93xx_soc_platform_init(void) -{ - return platform_driver_register(&ep93xx_pcm_driver); -} - -static void __exit ep93xx_soc_platform_exit(void) -{ - platform_driver_unregister(&ep93xx_pcm_driver); -} - -module_init(ep93xx_soc_platform_init); -module_exit(ep93xx_soc_platform_exit); +module_platform_driver(ep93xx_pcm_driver); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 968cb316d511..1e00b33cc508 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -81,17 +81,7 @@ static struct platform_driver simone_driver = { .remove = __devexit_p(simone_remove), }; -static int __init simone_init(void) -{ - return platform_driver_register(&simone_driver); -} -module_init(simone_init); - -static void __exit simone_exit(void) -{ - platform_driver_unregister(&simone_driver); -} -module_exit(simone_exit); +module_platform_driver(simone_driver); MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); MODULE_AUTHOR("Mika Westerberg "); diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index 2cde43321eec..33901d647b72 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -147,18 +147,7 @@ static struct platform_driver snappercl15_driver = { .remove = __devexit_p(snappercl15_remove), }; -static int __init snappercl15_init(void) -{ - return platform_driver_register(&snappercl15_driver); -} - -static void __exit snappercl15_exit(void) -{ - platform_driver_unregister(&snappercl15_driver); -} - -module_init(snappercl15_init); -module_exit(snappercl15_exit); +module_platform_driver(snappercl15_driver); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); -- cgit v1.2.1 From 880dd7210cd04205c6584922ad16b2d5731ab2c0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 12:14:56 +0800 Subject: ASoC: Convert s6000 directory to module_platform_driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 12 +----------- sound/soc/s6000/s6000-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 13716a9317fb..aaabdbaec19c 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -604,17 +604,7 @@ static struct platform_driver s6000_i2s_driver = { }, }; -static int __init s6000_i2s_init(void) -{ - return platform_driver_register(&s6000_i2s_driver); -} -module_init(s6000_i2s_init); - -static void __exit s6000_i2s_exit(void) -{ - platform_driver_unregister(&s6000_i2s_driver); -} -module_exit(s6000_i2s_exit); +module_platform_driver(s6000_i2s_driver); MODULE_AUTHOR("Daniel Gloeckner"); MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 55efc2bdf0bd..43c014f362f6 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -520,17 +520,7 @@ static struct platform_driver s6000_pcm_driver = { .remove = __devexit_p(s6000_soc_platform_remove), }; -static int __init snd_s6000_pcm_init(void) -{ - return platform_driver_register(&s6000_pcm_driver); -} -module_init(snd_s6000_pcm_init); - -static void __exit snd_s6000_pcm_exit(void) -{ - platform_driver_unregister(&s6000_pcm_driver); -} -module_exit(snd_s6000_pcm_exit); +module_platform_driver(s6000_pcm_driver); MODULE_AUTHOR("Daniel Gloeckner"); MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); -- cgit v1.2.1 From 85aa0960d8ef22edbb092446559b3b700a5512ef Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 14:21:29 +0800 Subject: ASoC: Convert mxs directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Wolfram Sang Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 12 +----------- sound/soc/mxs/mxs-saif.c | 12 +----------- sound/soc/mxs/mxs-sgtl5000.c | 12 +----------- 3 files changed, 3 insertions(+), 33 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index dea5aa4aa647..612ad3d9052d 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -346,14 +346,4 @@ static struct platform_driver mxs_pcm_driver = { .remove = __devexit_p(mxs_soc_platform_remove), }; -static int __init snd_mxs_pcm_init(void) -{ - return platform_driver_register(&mxs_pcm_driver); -} -module_init(snd_mxs_pcm_init); - -static void __exit snd_mxs_pcm_exit(void) -{ - platform_driver_unregister(&mxs_pcm_driver); -} -module_exit(snd_mxs_pcm_exit); +module_platform_driver(mxs_pcm_driver); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 46d76b52529b..1a13ab8b8e0d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -781,18 +781,8 @@ static struct platform_driver mxs_saif_driver = { }, }; -static int __init mxs_saif_init(void) -{ - return platform_driver_register(&mxs_saif_driver); -} - -static void __exit mxs_saif_exit(void) -{ - platform_driver_unregister(&mxs_saif_driver); -} +module_platform_driver(mxs_saif_driver); -module_init(mxs_saif_init); -module_exit(mxs_saif_exit); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ASoC SAIF driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 7fbeaec06eb4..200a9282b7cc 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -156,17 +156,7 @@ static struct platform_driver mxs_sgtl5000_audio_driver = { .remove = __devexit_p(mxs_sgtl5000_remove), }; -static int __init mxs_sgtl5000_init(void) -{ - return platform_driver_register(&mxs_sgtl5000_audio_driver); -} -module_init(mxs_sgtl5000_init); - -static void __exit mxs_sgtl5000_exit(void) -{ - platform_driver_unregister(&mxs_sgtl5000_audio_driver); -} -module_exit(mxs_sgtl5000_exit); +module_platform_driver(mxs_sgtl5000_audio_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ALSA SoC Machine driver"); -- cgit v1.2.1 From fb80297e4379640653b525e897b65b0b05a5b845 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 14:44:52 +0800 Subject: ASoC: Convert blackfin directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-ac97.c | 13 +------------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-i2s.c | 13 +------------ sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-tdm.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +----------- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +----------- 9 files changed, 9 insertions(+), 101 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 56815c1d47b3..fcff58390848 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -475,17 +475,7 @@ static struct platform_driver bf5xx_pcm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bf5xx_pcm_init(void) -{ - return platform_driver_register(&bf5xx_pcm_driver); -} -module_init(snd_bf5xx_pcm_init); - -static void __exit snd_bf5xx_pcm_exit(void) -{ - platform_driver_unregister(&bf5xx_pcm_driver); -} -module_exit(snd_bf5xx_pcm_exit); +module_platform_driver(bf5xx_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 6d2162590889..f4e9dc4e262e 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -375,18 +375,7 @@ static struct platform_driver asoc_bfin_ac97_driver = { .remove = __devexit_p(asoc_bfin_ac97_remove), }; -static int __init bfin_ac97_init(void) -{ - return platform_driver_register(&asoc_bfin_ac97_driver); -} -module_init(bfin_ac97_init); - -static void __exit bfin_ac97_exit(void) -{ - platform_driver_unregister(&asoc_bfin_ac97_driver); -} -module_exit(bfin_ac97_exit); - +module_platform_driver(asoc_bfin_ac97_driver); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 7565e1576ffa..6ec3d41b9b6d 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_i2s_pcm_driver = { .remove = __devexit_p(bfin_i2s_soc_platform_remove), }; -static int __init snd_bfin_i2s_pcm_init(void) -{ - return platform_driver_register(&bfin_i2s_pcm_driver); -} -module_init(snd_bfin_i2s_pcm_init); - -static void __exit snd_bfin_i2s_pcm_exit(void) -{ - platform_driver_unregister(&bfin_i2s_pcm_driver); -} -module_exit(snd_bfin_i2s_pcm_exit); +module_platform_driver(bfin_i2s_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index b31662e3a428..4dccf0374fe7 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -288,18 +288,7 @@ static struct platform_driver bfin_i2s_driver = { }, }; -static int __init bfin_i2s_init(void) -{ - return platform_driver_register(&bfin_i2s_driver); -} - -static void __exit bfin_i2s_exit(void) -{ - platform_driver_unregister(&bfin_i2s_driver); -} - -module_init(bfin_i2s_init); -module_exit(bfin_i2s_exit); +module_platform_driver(bfin_i2s_driver); /* Module information */ MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index c95cc03d583d..4406f9a865ae 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -339,17 +339,7 @@ static struct platform_driver bfin_tdm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(snd_bfin_tdm_init); - -static void __exit snd_bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(snd_bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); MODULE_AUTHOR("Barry Song"); MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 7876b5090fda..594f88217c74 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_tdm_driver = { }, }; -static int __init bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(bfin_tdm_init); - -static void __exit bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 8df2a3b0cb36..85ed39abe10e 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -184,17 +184,7 @@ static struct platform_driver bfin_eval_adau1373_driver = { .remove = __devexit_p(bfin_eval_adau1373_remove), }; -static int __init bfin_eval_adau1373_init(void) -{ - return platform_driver_register(&bfin_eval_adau1373_driver); -} -module_init(bfin_eval_adau1373_init); - -static void __exit bfin_eval_adau1373_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1373_driver); -} -module_exit(bfin_eval_adau1373_exit); +module_platform_driver(bfin_eval_adau1373_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index e5550acba2c2..1a88fe9ce34c 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -121,17 +121,7 @@ static struct platform_driver bfin_eval_adau1701_driver = { .remove = __devexit_p(bfin_eval_adau1701_remove), }; -static int __init bfin_eval_adau1701_init(void) -{ - return platform_driver_register(&bfin_eval_adau1701_driver); -} -module_init(bfin_eval_adau1701_init); - -static void __exit bfin_eval_adau1701_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1701_driver); -} -module_exit(bfin_eval_adau1701_exit); +module_platform_driver(bfin_eval_adau1701_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 897cfa68a2a6..0bc995fb6283 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -157,17 +157,7 @@ static struct platform_driver bfin_eval_adav80x_driver = { .id_table = bfin_eval_adav80x_ids, }; -static int __init bfin_eval_adav80x_init(void) -{ - return platform_driver_register(&bfin_eval_adav80x_driver); -} -module_init(bfin_eval_adav80x_init); - -static void __exit bfin_eval_adav80x_exit(void) -{ - platform_driver_unregister(&bfin_eval_adav80x_driver); -} -module_exit(bfin_eval_adav80x_exit); +module_platform_driver(bfin_eval_adav80x_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -- cgit v1.2.1 From 7a24b2ba59fda5e6d1367d5d3cb0d4d0f811713b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 15:03:50 +0800 Subject: ASoC: Convert imx directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +----------- sound/soc/imx/imx-pcm-fiq.c | 12 +----------- sound/soc/imx/imx-ssi.c | 13 +------------ 3 files changed, 3 insertions(+), 34 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 43fdc24f7e8d..1cf2fe889f6a 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -326,16 +326,6 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 8df0fae21943..d7ea0b354124 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -331,14 +331,4 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index eed7041364e6..01d1f749cf02 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -757,18 +757,7 @@ static struct platform_driver imx_ssi_driver = { }, }; -static int __init imx_ssi_init(void) -{ - return platform_driver_register(&imx_ssi_driver); -} - -static void __exit imx_ssi_exit(void) -{ - platform_driver_unregister(&imx_ssi_driver); -} - -module_init(imx_ssi_init); -module_exit(imx_ssi_exit); +module_platform_driver(imx_ssi_driver); /* Module information */ MODULE_AUTHOR("Sascha Hauer, "); -- cgit v1.2.1 From c32986e66bd72c02f9ecef490769248c7fcb5145 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 10:13:03 +0800 Subject: ASoC: Convert jz4740 directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 12 +----------- sound/soc/jz4740/jz4740-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 91255c6e1ee7..a5af7c42e62b 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -519,17 +519,7 @@ static struct platform_driver jz4740_i2s_driver = { }, }; -static int __init jz4740_i2s_init(void) -{ - return platform_driver_register(&jz4740_i2s_driver); -} -module_init(jz4740_i2s_init); - -static void __exit jz4740_i2s_exit(void) -{ - platform_driver_unregister(&jz4740_i2s_driver); -} -module_exit(jz4740_i2s_exit); +module_platform_driver(jz4740_i2s_driver); MODULE_AUTHOR("Lars-Peter Clausen, "); MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver"); diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index d1989cde9f14..50cda9ea9156 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -356,17 +356,7 @@ static struct platform_driver jz4740_pcm_driver = { }, }; -static int __init jz4740_soc_platform_init(void) -{ - return platform_driver_register(&jz4740_pcm_driver); -} -module_init(jz4740_soc_platform_init); - -static void __exit jz4740_soc_platform_exit(void) -{ - return platform_driver_unregister(&jz4740_pcm_driver); -} -module_exit(jz4740_soc_platform_exit); +module_platform_driver(jz4740_pcm_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); -- cgit v1.2.1 From d0efa6a279e53df0f695382a5a6958e8a9863bff Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 10:45:32 +0800 Subject: ASoC: Convert nuc900 directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 13 +------------ sound/soc/nuc900/nuc900-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 7544d249807e..f0c790451bd6 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -405,18 +405,7 @@ static struct platform_driver nuc900_ac97_driver = { .remove = __devexit_p(nuc900_ac97_drvremove), }; -static int __init nuc900_ac97_init(void) -{ - return platform_driver_register(&nuc900_ac97_driver); -} - -static void __exit nuc900_ac97_exit(void) -{ - platform_driver_unregister(&nuc900_ac97_driver); -} - -module_init(nuc900_ac97_init); -module_exit(nuc900_ac97_exit); +module_platform_driver(nuc900_ac97_driver); MODULE_AUTHOR("Wan ZongShun "); MODULE_DESCRIPTION("NUC900 AC97 SoC driver!"); diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index ae8d6806966b..37585b47f4e3 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -358,17 +358,7 @@ static struct platform_driver nuc900_pcm_driver = { .remove = __devexit_p(nuc900_soc_platform_remove), }; -static int __init nuc900_pcm_init(void) -{ - return platform_driver_register(&nuc900_pcm_driver); -} -module_init(nuc900_pcm_init); - -static void __exit nuc900_pcm_exit(void) -{ - platform_driver_unregister(&nuc900_pcm_driver); -} -module_exit(nuc900_pcm_exit); +module_platform_driver(nuc900_pcm_driver); MODULE_AUTHOR("Wan ZongShun, "); MODULE_DESCRIPTION("nuc900 Audio DMA module"); -- cgit v1.2.1 From 41b1022509733bb3347b15d670f3c1609ddf928f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 11:43:09 +0800 Subject: ASoC: Convert kirkwood directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 12 +----------- sound/soc/kirkwood/kirkwood-i2s.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cd33de1c5b7a..210438261a49 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -388,17 +388,7 @@ static struct platform_driver kirkwood_pcm_driver = { .remove = __devexit_p(kirkwood_soc_platform_remove), }; -static int __init kirkwood_pcm_init(void) -{ - return platform_driver_register(&kirkwood_pcm_driver); -} -module_init(kirkwood_pcm_init); - -static void __exit kirkwood_pcm_exit(void) -{ - platform_driver_unregister(&kirkwood_pcm_driver); -} -module_exit(kirkwood_pcm_exit); +module_platform_driver(kirkwood_pcm_driver); MODULE_AUTHOR("Arnaud Patard "); MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 2b212dcb9ac7..f6bb21156876 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -483,17 +483,7 @@ static struct platform_driver kirkwood_i2s_driver = { }, }; -static int __init kirkwood_i2s_init(void) -{ - return platform_driver_register(&kirkwood_i2s_driver); -} -module_init(kirkwood_i2s_init); - -static void __exit kirkwood_i2s_exit(void) -{ - platform_driver_unregister(&kirkwood_i2s_driver); -} -module_exit(kirkwood_i2s_exit); +module_platform_driver(kirkwood_i2s_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, "); -- cgit v1.2.1 From 29515d62db425796d82e2e2d9209a44b9e324ff4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 24 Nov 2011 11:51:56 +0800 Subject: ASoC: Convert mid-x86 directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 14 +------------- sound/soc/mid-x86/sst_platform.c | 14 +------------- 2 files changed, 2 insertions(+), 26 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index cca693ae1bd4..e53f8e473a78 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -428,19 +428,7 @@ static struct platform_driver snd_mfld_mc_driver = { .remove = __devexit_p(snd_mfld_mc_remove), }; -static int __init snd_mfld_driver_init(void) -{ - pr_debug("snd_mfld_driver_init called\n"); - return platform_driver_register(&snd_mfld_mc_driver); -} -module_init(snd_mfld_driver_init); - -static void __exit snd_mfld_driver_exit(void) -{ - pr_debug("snd_mfld_driver_exit called\n"); - platform_driver_unregister(&snd_mfld_mc_driver); -} -module_exit(snd_mfld_driver_exit); +module_platform_driver(snd_mfld_mc_driver); MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); MODULE_AUTHOR("Vinod Koul "); diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 23057020aa0f..94f70b3f94e6 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -472,19 +472,7 @@ static struct platform_driver sst_platform_driver = { .remove = sst_platform_remove, }; -static int __init sst_soc_platform_init(void) -{ - pr_debug("sst_soc_platform_init called\n"); - return platform_driver_register(&sst_platform_driver); -} -module_init(sst_soc_platform_init); - -static void __exit sst_soc_platform_exit(void) -{ - platform_driver_unregister(&sst_platform_driver); - pr_debug("sst_soc_platform_exit success\n"); -} -module_exit(sst_soc_platform_exit); +module_platform_driver(sst_platform_driver); MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul "); -- cgit v1.2.1 From e00c3f555f1f404b38d44bcfe19db674a92c809a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Nov 2011 15:20:13 +0000 Subject: ASoC: Convert Samsung directory to module_platform_driver Saves some boilerplate code. Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/ac97.c | 12 +----------- sound/soc/samsung/dma.c | 12 +----------- sound/soc/samsung/i2s.c | 12 +----------- sound/soc/samsung/idma.c | 12 +----------- sound/soc/samsung/lowland.c | 12 +----------- sound/soc/samsung/pcm.c | 12 +----------- sound/soc/samsung/s3c2412-i2s.c | 12 +----------- sound/soc/samsung/s3c24xx-i2s.c | 12 +----------- sound/soc/samsung/s3c24xx_simtec_hermes.c | 16 ++-------------- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 16 ++-------------- sound/soc/samsung/s3c24xx_uda134x.c | 14 +------------- sound/soc/samsung/smdk_wm8580pcm.c | 14 +------------- sound/soc/samsung/spdif.c | 12 +----------- sound/soc/samsung/speyside.c | 12 +----------- sound/soc/samsung/speyside_wm8962.c | 12 +----------- 15 files changed, 17 insertions(+), 175 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 09035afdeb74..7b9bf93e3701 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -509,17 +509,7 @@ static struct platform_driver s3c_ac97_driver = { }, }; -static int __init s3c_ac97_init(void) -{ - return platform_driver_register(&s3c_ac97_driver); -} -module_init(s3c_ac97_init); - -static void __exit s3c_ac97_exit(void) -{ - platform_driver_unregister(&s3c_ac97_driver); -} -module_exit(s3c_ac97_exit); +module_platform_driver(s3c_ac97_driver); MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index a68b26441784..797c3d5e79e5 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -458,17 +458,7 @@ static struct platform_driver asoc_dma_driver = { .remove = __devexit_p(samsung_asoc_platform_remove), }; -static int __init samsung_asoc_init(void) -{ - return platform_driver_register(&asoc_dma_driver); -} -module_init(samsung_asoc_init); - -static void __exit samsung_asoc_exit(void) -{ - platform_driver_unregister(&asoc_dma_driver); -} -module_exit(samsung_asoc_exit); +module_platform_driver(asoc_dma_driver); MODULE_AUTHOR("Ben Dooks, "); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03ee8ce46a29..fb80f2886c70 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1144,17 +1144,7 @@ static struct platform_driver samsung_i2s_driver = { }, }; -static int __init samsung_i2s_init(void) -{ - return platform_driver_register(&samsung_i2s_driver); -} -module_init(samsung_i2s_init); - -static void __exit samsung_i2s_exit(void) -{ - platform_driver_unregister(&samsung_i2s_driver); -} -module_exit(samsung_i2s_exit); +module_platform_driver(samsung_i2s_driver); /* Module information */ MODULE_AUTHOR("Jaswinder Singh, "); diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index c41178efc908..6ca3d8c221a0 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -437,17 +437,7 @@ static struct platform_driver asoc_idma_driver = { .remove = __devexit_p(asoc_idma_platform_remove), }; -static int __init asoc_idma_init(void) -{ - return platform_driver_register(&asoc_idma_driver); -} -module_init(asoc_idma_init); - -static void __exit asoc_idma_exit(void) -{ - platform_driver_unregister(&asoc_idma_driver); -} -module_exit(asoc_idma_exit); +module_platform_driver(asoc_idma_driver); MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index eff1b4b65df4..4216a06b45f5 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -228,17 +228,7 @@ static struct platform_driver lowland_driver = { .remove = __devexit_p(lowland_remove), }; -static int __init lowland_audio_init(void) -{ - return platform_driver_register(&lowland_driver); -} -module_init(lowland_audio_init); - -static void __exit lowland_audio_exit(void) -{ - platform_driver_unregister(&lowland_driver); -} -module_exit(lowland_audio_exit); +module_platform_driver(lowland_driver); MODULE_DESCRIPTION("Lowland audio support"); MODULE_AUTHOR("Mark Brown "); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 2df2762f3000..beef63fca052 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -632,17 +632,7 @@ static struct platform_driver s3c_pcm_driver = { }, }; -static int __init s3c_pcm_init(void) -{ - return platform_driver_register(&s3c_pcm_driver); -} -module_init(s3c_pcm_init); - -static void __exit s3c_pcm_exit(void) -{ - platform_driver_unregister(&s3c_pcm_driver); -} -module_exit(s3c_pcm_exit); +module_platform_driver(s3c_pcm_driver); /* Module information */ MODULE_AUTHOR("Jaswinder Singh, "); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 545773d0641c..72185078ddf8 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -184,17 +184,7 @@ static struct platform_driver s3c2412_iis_driver = { }, }; -static int __init s3c2412_i2s_init(void) -{ - return platform_driver_register(&s3c2412_iis_driver); -} -module_init(s3c2412_i2s_init); - -static void __exit s3c2412_i2s_exit(void) -{ - platform_driver_unregister(&s3c2412_iis_driver); -} -module_exit(s3c2412_i2s_exit); +module_platform_driver(s3c2412_iis_driver); /* Module information */ MODULE_AUTHOR("Ben Dooks, "); diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 2a98bed2db02..c4aa4d412fbf 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -489,17 +489,7 @@ static struct platform_driver s3c24xx_iis_driver = { }, }; -static int __init s3c24xx_i2s_init(void) -{ - return platform_driver_register(&s3c24xx_iis_driver); -} -module_init(s3c24xx_i2s_init); - -static void __exit s3c24xx_i2s_exit(void) -{ - platform_driver_unregister(&s3c24xx_iis_driver); -} -module_exit(s3c24xx_i2s_exit); +module_platform_driver(s3c24xx_iis_driver); /* Module information */ MODULE_AUTHOR("Ben Dooks, "); diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index d125e79baf7f..502798100f21 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -114,21 +114,9 @@ static struct platform_driver simtec_audio_hermes_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); - -static int __init simtec_hermes_modinit(void) -{ - return platform_driver_register(&simtec_audio_hermes_platdrv); -} - -static void __exit simtec_hermes_modexit(void) -{ - platform_driver_unregister(&simtec_audio_hermes_platdrv); -} - -module_init(simtec_hermes_modinit); -module_exit(simtec_hermes_modexit); +module_platform_driver(simtec_audio_hermes_platdrv); +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); MODULE_AUTHOR("Ben Dooks "); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 5e4fd46b7200..7324609833d8 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -102,21 +102,9 @@ static struct platform_driver simtec_audio_tlv320aic23_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); - -static int __init simtec_tlv320aic23_modinit(void) -{ - return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); -} - -static void __exit simtec_tlv320aic23_modexit(void) -{ - platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); -} - -module_init(simtec_tlv320aic23_modinit); -module_exit(simtec_tlv320aic23_modexit); +module_platform_driver(simtec_audio_tlv320aic32_driver); +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); MODULE_AUTHOR("Ben Dooks "); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 548c6ac6e7b0..62b69fb6a085 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -343,19 +343,7 @@ static struct platform_driver s3c24xx_uda134x_driver = { }, }; -static int __init s3c24xx_uda134x_init(void) -{ - return platform_driver_register(&s3c24xx_uda134x_driver); -} - -static void __exit s3c24xx_uda134x_exit(void) -{ - platform_driver_unregister(&s3c24xx_uda134x_driver); -} - - -module_init(s3c24xx_uda134x_init); -module_exit(s3c24xx_uda134x_exit); +module_platform_driver(s3c24xx_uda134x_driver); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin "); MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0677473e6b60..49dfafbf3df6 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -188,19 +188,7 @@ static struct platform_driver snd_smdk_driver = { .remove = __devexit_p(snd_smdk_remove), }; -static int __init smdk_audio_init(void) -{ - return platform_driver_register(&snd_smdk_driver); -} - -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_driver_unregister(&snd_smdk_driver); -} - -module_exit(smdk_audio_exit); +module_platform_driver(snd_smdk_driver); MODULE_AUTHOR("Sangbeom Kim, "); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index a1fee1a414c9..a5a56a120345 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -483,17 +483,7 @@ static struct platform_driver samsung_spdif_driver = { }, }; -static int __init spdif_init(void) -{ - return platform_driver_register(&samsung_spdif_driver); -} -module_init(spdif_init); - -static void __exit spdif_exit(void) -{ - platform_driver_unregister(&samsung_spdif_driver); -} -module_exit(spdif_exit); +module_platform_driver(samsung_spdif_driver); MODULE_AUTHOR("Seungwhan Youn, "); MODULE_DESCRIPTION("Samsung S/PDIF Controller Driver"); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index efa5187f6197..11196b9b319e 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -347,17 +347,7 @@ static struct platform_driver speyside_driver = { .remove = __devexit_p(speyside_remove), }; -static int __init speyside_audio_init(void) -{ - return platform_driver_register(&speyside_driver); -} -module_init(speyside_audio_init); - -static void __exit speyside_audio_exit(void) -{ - platform_driver_unregister(&speyside_driver); -} -module_exit(speyside_audio_exit); +module_platform_driver(speyside_driver); MODULE_DESCRIPTION("Speyside audio support"); MODULE_AUTHOR("Mark Brown "); diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index a681c8d74118..c09648efab61 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -249,17 +249,7 @@ static struct platform_driver speyside_wm8962_driver = { .remove = __devexit_p(speyside_wm8962_remove), }; -static int __init speyside_wm8962_audio_init(void) -{ - return platform_driver_register(&speyside_wm8962_driver); -} -module_init(speyside_wm8962_audio_init); - -static void __exit speyside_wm8962_audio_exit(void) -{ - platform_driver_unregister(&speyside_wm8962_driver); -} -module_exit(speyside_wm8962_audio_exit); +module_platform_driver(speyside_wm8962_driver); MODULE_DESCRIPTION("Speyside WM8962 audio support"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.1 From 878042d19c760178ba08ed24025d08ba750e38c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Nov 2011 17:31:12 +0000 Subject: ASoC: Staticise non-exported symbols in sta32x Signed-off-by: Mark Brown Acked-by: Johannes Stezenbach --- sound/soc/codecs/sta32x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e2b1cdedb982..edcbeef43735 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -265,7 +265,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } -int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) +static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int cfud; @@ -290,7 +290,7 @@ int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) return 0; } -int sta32x_cache_sync(struct snd_soc_codec *codec) +static int sta32x_cache_sync(struct snd_soc_codec *codec) { unsigned int mute; int rc; -- cgit v1.2.1 From a81b82c09e70db853cb270ed9ac166b6c50d7b8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Nov 2011 18:28:51 +0000 Subject: ASoC: Use devm_kzalloc() in wm5100 Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 6c79d97ba181..844d5d287976 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2607,7 +2607,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i; - wm5100 = kzalloc(sizeof(struct wm5100_priv), GFP_KERNEL); + wm5100 = devm_kzalloc(&i2c->dev, sizeof(struct wm5100_priv), + GFP_KERNEL); if (wm5100 == NULL) return -ENOMEM; @@ -2616,7 +2617,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err_alloc; + goto err; } for (i = 0; i < ARRAY_SIZE(wm5100->fll); i++) @@ -2774,8 +2775,7 @@ err_core: wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); -err_alloc: - kfree(wm5100); +err: return ret; } @@ -2799,7 +2799,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); regmap_exit(wm5100->regmap); - kfree(wm5100); return 0; } -- cgit v1.2.1 From b31c9056e400ddf10ec9691c6fada2fba1709330 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:05:45 +0800 Subject: ASoC: Convert atmel directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 12 +----------- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index f81d4c3f8956..60de05525c06 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -495,17 +495,7 @@ static struct platform_driver atmel_pcm_driver = { .remove = __devexit_p(atmel_soc_platform_remove), }; -static int __init snd_atmel_pcm_init(void) -{ - return platform_driver_register(&atmel_pcm_driver); -} -module_init(snd_atmel_pcm_init); - -static void __exit snd_atmel_pcm_exit(void) -{ - platform_driver_unregister(&atmel_pcm_driver); -} -module_exit(snd_atmel_pcm_exit); +module_platform_driver(atmel_pcm_driver); MODULE_AUTHOR("Sedji Gaouaou "); MODULE_DESCRIPTION("Atmel PCM module"); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index a67fc9b7dbe7..354341ec0f42 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -859,17 +859,7 @@ int atmel_ssc_set_audio(int ssc_id) } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -static int __init snd_atmel_ssc_init(void) -{ - return platform_driver_register(&asoc_ssc_driver); -} -module_init(snd_atmel_ssc_init); - -static void __exit snd_atmel_ssc_exit(void) -{ - platform_driver_unregister(&asoc_ssc_driver); -} -module_exit(snd_atmel_ssc_exit); +module_platform_driver(asoc_ssc_driver); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com"); -- cgit v1.2.1 From 8a124f9cc9bafc40f5650e63a84ba1ff98a36ea0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:06:59 +0800 Subject: ASoC: Convert au1x directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/au1x/db1000.c | 13 +------------ sound/soc/au1x/db1200.c | 13 +------------ sound/soc/au1x/dbdma2.c | 13 +------------ sound/soc/au1x/dma.c | 13 +------------ sound/soc/au1x/i2sc.c | 13 +------------ sound/soc/au1x/psc-i2s.c | 13 +------------ 6 files changed, 6 insertions(+), 72 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 127477a5e0c7..094a20723bc6 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -57,18 +57,7 @@ static struct platform_driver db1000_audio_driver = { .remove = __devexit_p(db1000_audio_remove), }; -static int __init db1000_audio_load(void) -{ - return platform_driver_register(&db1000_audio_driver); -} - -static void __exit db1000_audio_unload(void) -{ - platform_driver_unregister(&db1000_audio_driver); -} - -module_init(db1000_audio_load); -module_exit(db1000_audio_unload); +module_platform_driver(db1000_audio_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio"); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 289312c14b99..80733331733f 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -133,18 +133,7 @@ static struct platform_driver db1200_audio_driver = { .remove = __devexit_p(db1200_audio_remove), }; -static int __init db1200_audio_load(void) -{ - return platform_driver_register(&db1200_audio_driver); -} - -static void __exit db1200_audio_unload(void) -{ - platform_driver_unregister(&db1200_audio_driver); -} - -module_init(db1200_audio_load); -module_exit(db1200_audio_unload); +module_platform_driver(db1200_audio_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("DB1200 ASoC audio support"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index d7d04e26eee5..09699de9b337 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -384,18 +384,7 @@ static struct platform_driver au1xpsc_pcm_driver = { .remove = __devexit_p(au1xpsc_pcm_drvremove), }; -static int __init au1xpsc_audio_dbdma_load(void) -{ - return platform_driver_register(&au1xpsc_pcm_driver); -} - -static void __exit au1xpsc_audio_dbdma_unload(void) -{ - platform_driver_unregister(&au1xpsc_pcm_driver); -} - -module_init(au1xpsc_audio_dbdma_load); -module_exit(au1xpsc_audio_dbdma_unload); +module_platform_driver(au1xpsc_pcm_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 177f7137a9c8..dc4dae48aed9 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -359,18 +359,7 @@ static struct platform_driver alchemy_pcmdma_driver = { .remove = __devexit_p(alchemy_pcm_drvremove), }; -static int __init alchemy_pcmdma_load(void) -{ - return platform_driver_register(&alchemy_pcmdma_driver); -} - -static void __exit alchemy_pcmdma_unload(void) -{ - platform_driver_unregister(&alchemy_pcmdma_driver); -} - -module_init(alchemy_pcmdma_load); -module_exit(alchemy_pcmdma_unload); +module_platform_driver(alchemy_pcmdma_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver"); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 6bcf48f5884c..cb53ad87d0a9 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -331,18 +331,7 @@ static struct platform_driver au1xi2s_driver = { .remove = __devexit_p(au1xi2s_drvremove), }; -static int __init au1xi2s_load(void) -{ - return platform_driver_register(&au1xi2s_driver); -} - -static void __exit au1xi2s_unload(void) -{ - platform_driver_unregister(&au1xi2s_driver); -} - -module_init(au1xi2s_load); -module_exit(au1xi2s_unload); +module_platform_driver(au1xi2s_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver"); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f7714d50bdaf..5c1dc8a141ab 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -435,18 +435,7 @@ static struct platform_driver au1xpsc_i2s_driver = { .remove = __devexit_p(au1xpsc_i2s_drvremove), }; -static int __init au1xpsc_i2s_load(void) -{ - return platform_driver_register(&au1xpsc_i2s_driver); -} - -static void __exit au1xpsc_i2s_unload(void) -{ - platform_driver_unregister(&au1xpsc_i2s_driver); -} - -module_init(au1xpsc_i2s_load); -module_exit(au1xpsc_i2s_unload); +module_platform_driver(au1xpsc_i2s_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -- cgit v1.2.1 From 2f702a19154ddbd294825c0588593e1eef10b1e2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:13:37 +0800 Subject: ASoC: Convert pxa directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Marek Vasut Acked-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/soc/pxa/hx4700.c | 13 +------------ sound/soc/pxa/mioa701_wm9713.c | 13 +------------ sound/soc/pxa/palm27x.c | 13 +------------ sound/soc/pxa/pxa-ssp.c | 12 +----------- sound/soc/pxa/pxa2xx-ac97.c | 12 +----------- sound/soc/pxa/pxa2xx-pcm.c | 12 +----------- 6 files changed, 6 insertions(+), 69 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 65c124831a00..e32afaf1ebbb 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -236,18 +236,7 @@ static struct platform_driver hx4700_audio_driver = { .remove = __devexit_p(hx4700_audio_remove), }; -static int __init hx4700_modinit(void) -{ - return platform_driver_register(&hx4700_audio_driver); -} -module_init(hx4700_modinit); - -static void __exit hx4700_modexit(void) -{ - platform_driver_unregister(&hx4700_audio_driver); -} - -module_exit(hx4700_modexit); +module_platform_driver(hx4700_audio_driver); MODULE_AUTHOR("Philipp Zabel"); MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0b8d1ee738a4..0e73a7f718e4 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -227,18 +227,7 @@ static struct platform_driver mioa701_wm9713_driver = { }, }; -static int __init mioa701_asoc_init(void) -{ - return platform_driver_register(&mioa701_wm9713_driver); -} - -static void __exit mioa701_asoc_exit(void) -{ - platform_driver_unregister(&mioa701_wm9713_driver); -} - -module_init(mioa701_asoc_init); -module_exit(mioa701_asoc_exit); +module_platform_driver(mioa701_wm9713_driver); /* Module information */ MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 7edc1fb71fae..f313eca40fdc 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -201,18 +201,7 @@ static struct platform_driver palm27x_wm9712_driver = { }, }; -static int __init palm27x_asoc_init(void) -{ - return platform_driver_register(&palm27x_wm9712_driver); -} - -static void __exit palm27x_asoc_exit(void) -{ - platform_driver_unregister(&palm27x_wm9712_driver); -} - -module_init(palm27x_asoc_init); -module_exit(palm27x_asoc_exit); +module_platform_driver(palm27x_wm9712_driver); /* Module information */ MODULE_AUTHOR("Marek Vasut "); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9c9a51ef67c3..a57cfbc038e3 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -825,17 +825,7 @@ static struct platform_driver asoc_ssp_driver = { .remove = __devexit_p(asoc_ssp_remove), }; -static int __init pxa_ssp_init(void) -{ - return platform_driver_register(&asoc_ssp_driver); -} -module_init(pxa_ssp_init); - -static void __exit pxa_ssp_exit(void) -{ - platform_driver_unregister(&asoc_ssp_driver); -} -module_exit(pxa_ssp_exit); +module_platform_driver(asoc_ssp_driver); /* Module information */ MODULE_AUTHOR("Mark Brown "); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 3fec2f35b8f8..837ff341fd6d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -263,17 +263,7 @@ static struct platform_driver pxa2xx_ac97_driver = { }, }; -static int __init pxa_ac97_init(void) -{ - return platform_driver_register(&pxa2xx_ac97_driver); -} -module_init(pxa_ac97_init); - -static void __exit pxa_ac97_exit(void) -{ - platform_driver_unregister(&pxa2xx_ac97_driver); -} -module_exit(pxa_ac97_exit); +module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 600676f709a9..fdd6bedef9bd 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -141,17 +141,7 @@ static struct platform_driver pxa_pcm_driver = { .remove = __devexit_p(pxa2xx_soc_platform_remove), }; -static int __init snd_pxa_pcm_init(void) -{ - return platform_driver_register(&pxa_pcm_driver); -} -module_init(snd_pxa_pcm_init); - -static void __exit snd_pxa_pcm_exit(void) -{ - platform_driver_unregister(&pxa_pcm_driver); -} -module_exit(snd_pxa_pcm_exit); +module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -- cgit v1.2.1 From cb5e87387cfa8172faca36682e2df069b006efdf Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:15:07 +0800 Subject: ASoC: Convert sh directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 12 +----------- sound/soc/sh/fsi-ak4642.c | 13 +------------ sound/soc/sh/fsi-hdmi.c | 13 +------------ sound/soc/sh/fsi.c | 13 +------------ sound/soc/sh/hac.c | 12 +----------- sound/soc/sh/siu_dai.c | 13 +------------ sound/soc/sh/ssi.c | 12 +----------- 7 files changed, 7 insertions(+), 81 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index db74005f37ce..7da20186b19e 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -369,17 +369,7 @@ static struct platform_driver sh7760_pcm_driver = { .remove = __devexit_p(sh7760_soc_platform_remove), }; -static int __init snd_sh7760_pcm_init(void) -{ - return platform_driver_register(&sh7760_pcm_driver); -} -module_init(snd_sh7760_pcm_init); - -static void __exit snd_sh7760_pcm_exit(void) -{ - platform_driver_unregister(&sh7760_pcm_driver); -} -module_exit(snd_sh7760_pcm_exit); +module_platform_driver(sh7760_pcm_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index 11d2d7ff29d9..eb52778d0f90 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -100,18 +100,7 @@ static struct platform_driver fsi_ak4642 = { .remove = fsi_ak4642_remove, }; -static int __init fsi_ak4642_init(void) -{ - return platform_driver_register(&fsi_ak4642); -} - -static void __exit fsi_ak4642_exit(void) -{ - platform_driver_unregister(&fsi_ak4642); -} - -module_init(fsi_ak4642_init); -module_exit(fsi_ak4642_exit); +module_platform_driver(fsi_ak4642); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index 3ebebe706ad3..621aea155ac1 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -110,18 +110,7 @@ static struct platform_driver fsi_hdmi = { .id_table = fsi_id_table, }; -static int __init fsi_hdmi_init(void) -{ - return platform_driver_register(&fsi_hdmi); -} - -static void __exit fsi_hdmi_exit(void) -{ - platform_driver_unregister(&fsi_hdmi); -} - -module_init(fsi_hdmi_init); -module_exit(fsi_hdmi_exit); +module_platform_driver(fsi_hdmi); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Generic SH4 FSI-HDMI sound card"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index aa3033075a0d..a27c30636b82 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1468,18 +1468,7 @@ static struct platform_driver fsi_driver = { .id_table = fsi_id_table, }; -static int __init fsi_mobile_init(void) -{ - return platform_driver_register(&fsi_driver); -} - -static void __exit fsi_mobile_exit(void) -{ - platform_driver_unregister(&fsi_driver); -} - -module_init(fsi_mobile_init); -module_exit(fsi_mobile_exit); +module_platform_driver(fsi_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index a1f307b9a82d..3474d7befe5a 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -332,17 +332,7 @@ static struct platform_driver hac_pcm_driver = { .remove = __devexit_p(hac_soc_platform_remove), }; -static int __init sh4_hac_pcm_init(void) -{ - return platform_driver_register(&hac_pcm_driver); -} -module_init(sh4_hac_pcm_init); - -static void __exit sh4_hac_pcm_exit(void) -{ - platform_driver_unregister(&hac_pcm_driver); -} -module_exit(sh4_hac_pcm_exit); +module_platform_driver(hac_pcm_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 93dea49ff1a7..11c608570820 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -852,18 +852,7 @@ static struct platform_driver siu_driver = { .remove = __devexit_p(siu_remove), }; -static int __init siu_init(void) -{ - return platform_driver_register(&siu_driver); -} - -static void __exit siu_exit(void) -{ - platform_driver_unregister(&siu_driver); -} - -module_init(siu_init) -module_exit(siu_exit) +module_platform_driver(siu_driver); MODULE_AUTHOR("Carlos Munoz "); MODULE_DESCRIPTION("ALSA SoC SH7722 SIU driver"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 1fda16a00e6a..ff82b56a8860 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -401,17 +401,7 @@ static struct platform_driver sh4_ssi_driver = { .remove = __devexit_p(sh4_soc_dai_remove), }; -static int __init snd_sh4_ssi_init(void) -{ - return platform_driver_register(&sh4_ssi_driver); -} -module_init(snd_sh4_ssi_init); - -static void __exit snd_sh4_ssi_exit(void) -{ - platform_driver_unregister(&sh4_ssi_driver); -} -module_exit(snd_sh4_ssi_exit); +module_platform_driver(sh4_ssi_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); -- cgit v1.2.1 From 33d316cd8b39fda7106332e5554f5959dc04b4dc Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:16:10 +0800 Subject: ASoC: Convert txx9 directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-ac97.c | 13 +------------ sound/soc/txx9/txx9aclc.c | 12 +----------- 2 files changed, 2 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index a4e3f5501847..28db4ca997ca 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -223,18 +223,7 @@ static struct platform_driver txx9aclc_ac97_driver = { }, }; -static int __init txx9aclc_ac97_init(void) -{ - return platform_driver_register(&txx9aclc_ac97_driver); -} - -static void __exit txx9aclc_ac97_exit(void) -{ - platform_driver_unregister(&txx9aclc_ac97_driver); -} - -module_init(txx9aclc_ac97_init); -module_exit(txx9aclc_ac97_exit); +module_platform_driver(txx9aclc_ac97_driver); MODULE_AUTHOR("Atsushi Nemoto "); MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 3de99af8cb82..93931def0dce 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -438,17 +438,7 @@ static struct platform_driver txx9aclc_pcm_driver = { .remove = __devexit_p(txx9aclc_soc_platform_remove), }; -static int __init snd_txx9aclc_pcm_init(void) -{ - return platform_driver_register(&txx9aclc_pcm_driver); -} -module_init(snd_txx9aclc_pcm_init); - -static void __exit snd_txx9aclc_pcm_exit(void) -{ - platform_driver_unregister(&txx9aclc_pcm_driver); -} -module_exit(snd_txx9aclc_pcm_exit); +module_platform_driver(txx9aclc_pcm_driver); MODULE_AUTHOR("Atsushi Nemoto "); MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver"); -- cgit v1.2.1 From f9b8a51493d69841bab3c5e85f335b6af0c8e5c2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:09:27 +0800 Subject: ASoC: Convert davinci directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 12 +----------- sound/soc/davinci/davinci-mcasp.c | 12 +----------- sound/soc/davinci/davinci-pcm.c | 12 +----------- sound/soc/davinci/davinci-vcif.c | 12 +----------- 4 files changed, 4 insertions(+), 44 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index f3d5ae1078be..ec187100367e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -774,17 +774,7 @@ static struct platform_driver davinci_mcbsp_driver = { }, }; -static int __init davinci_i2s_init(void) -{ - return platform_driver_register(&davinci_mcbsp_driver); -} -module_init(davinci_i2s_init); - -static void __exit davinci_i2s_exit(void) -{ - platform_driver_unregister(&davinci_mcbsp_driver); -} -module_exit(davinci_i2s_exit); +module_platform_driver(davinci_mcbsp_driver); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 03cea9d39c4b..2152ff5c04f6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -991,17 +991,7 @@ static struct platform_driver davinci_mcasp_driver = { }, }; -static int __init davinci_mcasp_init(void) -{ - return platform_driver_register(&davinci_mcasp_driver); -} -module_init(davinci_mcasp_init); - -static void __exit davinci_mcasp_exit(void) -{ - platform_driver_unregister(&davinci_mcasp_driver); -} -module_exit(davinci_mcasp_exit); +module_platform_driver(davinci_mcasp_driver); MODULE_AUTHOR("Steve Chen"); MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index d5fe08cc5db7..65bff3d30dd7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -886,17 +886,7 @@ static struct platform_driver davinci_pcm_driver = { .remove = __devexit_p(davinci_soc_platform_remove), }; -static int __init snd_davinci_pcm_init(void) -{ - return platform_driver_register(&davinci_pcm_driver); -} -module_init(snd_davinci_pcm_init); - -static void __exit snd_davinci_pcm_exit(void) -{ - platform_driver_unregister(&davinci_pcm_driver); -} -module_exit(snd_davinci_pcm_exit); +module_platform_driver(davinci_pcm_driver); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index dae96b85fd6d..70ce10c5d998 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -265,17 +265,7 @@ static struct platform_driver davinci_vcif_driver = { }, }; -static int __init davinci_vcif_init(void) -{ - return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); -} -module_init(davinci_vcif_init); - -static void __exit davinci_vcif_exit(void) -{ - platform_driver_unregister(&davinci_vcif_driver); -} -module_exit(davinci_vcif_exit); +module_platform_driver(davinci_vcif_driver); MODULE_AUTHOR("Miguel Aguilar"); MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); -- cgit v1.2.1 From beda5bf575a93823289fbeb868b42e75e9f08d96 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:12:16 +0800 Subject: ASoC: Convert omap directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi.c | 12 +----------- sound/soc/omap/omap-mcbsp.c | 12 +----------- sound/soc/omap/omap-mcpdm.c | 12 +----------- sound/soc/omap/omap-pcm.c | 12 +----------- sound/soc/omap/omap4-hdmi-card.c | 12 +----------- 5 files changed, 5 insertions(+), 55 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 9bb1cf89b4a4..38e0defa7078 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -139,17 +139,7 @@ static struct platform_driver hdmi_dai_driver = { .remove = __devexit_p(omap_hdmi_remove), }; -static int __init hdmi_dai_init(void) -{ - return platform_driver_register(&hdmi_dai_driver); -} -module_init(hdmi_dai_init); - -static void __exit hdmi_dai_exit(void) -{ - platform_driver_unregister(&hdmi_dai_driver); -} -module_exit(hdmi_dai_exit); +module_platform_driver(hdmi_dai_driver); MODULE_AUTHOR("Jorge Candelaria "); MODULE_AUTHOR("Ricardo Neri "); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d91e6efd2600..bd11d2568584 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -785,17 +785,7 @@ static struct platform_driver asoc_mcbsp_driver = { .remove = __devexit_p(asoc_mcbsp_remove), }; -static int __init snd_omap_mcbsp_init(void) -{ - return platform_driver_register(&asoc_mcbsp_driver); -} -module_init(snd_omap_mcbsp_init); - -static void __exit snd_omap_mcbsp_exit(void) -{ - platform_driver_unregister(&asoc_mcbsp_driver); -} -module_exit(snd_omap_mcbsp_exit); +module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index cc8ceff25dbd..b50ac60be7db 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -520,17 +520,7 @@ static struct platform_driver asoc_mcpdm_driver = { .remove = __devexit_p(asoc_mcpdm_remove), }; -static int __init snd_omap_mcpdm_init(void) -{ - return platform_driver_register(&asoc_mcpdm_driver); -} -module_init(snd_omap_mcpdm_init); - -static void __exit snd_omap_mcpdm_exit(void) -{ - platform_driver_unregister(&asoc_mcpdm_driver); -} -module_exit(snd_omap_mcpdm_exit); +module_platform_driver(asoc_mcpdm_driver); MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6ede7dc6c10a..52a0f634948e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -433,17 +433,7 @@ static struct platform_driver omap_pcm_driver = { .remove = __devexit_p(omap_pcm_remove), }; -static int __init snd_omap_pcm_init(void) -{ - return platform_driver_register(&omap_pcm_driver); -} -module_init(snd_omap_pcm_init); - -static void __exit snd_omap_pcm_exit(void) -{ - platform_driver_unregister(&omap_pcm_driver); -} -module_exit(snd_omap_pcm_exit); +module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP PCM DMA module"); diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 8671261ba16d..52d471c1eeed 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -112,17 +112,7 @@ static struct platform_driver omap4_hdmi_driver = { .remove = __devexit_p(omap4_hdmi_remove), }; -static int __init omap4_hdmi_init(void) -{ - return platform_driver_register(&omap4_hdmi_driver); -} -module_init(omap4_hdmi_init); - -static void __exit omap4_hdmi_exit(void) -{ - platform_driver_unregister(&omap4_hdmi_driver); -} -module_exit(omap4_hdmi_exit); +module_platform_driver(omap4_hdmi_driver); MODULE_AUTHOR("Ricardo Neri "); MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver"); -- cgit v1.2.1 From 679acec1f240b433dc3879714655b6c6452385ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 12:11:46 +0000 Subject: ASoC: Remove driver versioning from ak4642 It's never been updated so it can't be that useful and it makes the driver needlessly chatty. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c887ddf1061e..30ce3d660d9e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -32,8 +32,6 @@ #include #include -#define AK4642_VERSION "0.0.1" - #define PW_MGMT1 0x00 #define PW_MGMT2 0x01 #define SG_SL1 0x02 @@ -473,8 +471,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); int ret; - dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- cgit v1.2.1 From 997c2ea916edb516f23d6e1848cd1f4a10e62740 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 12:14:37 +0000 Subject: ASoC: Remove unneeded platform_device.h inclusions from CODECs They've not been needed for a long time if they were ever required. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm_hubs.c | 1 - 46 files changed, 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index f6c47345bcc8..e1f531085453 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 3657c76cc127..f53f31480565 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 30ce3d660d9e..9b4ee6c63d28 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -25,7 +25,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 88647d3ab24b..6a5c001e8ba8 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5396b91fa5f1..dc77ff7ba339 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -22,7 +22,6 @@ */ #include -#include #include #include #include diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 2f268f20268a..528510b8e5de 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -22,7 +22,6 @@ */ #include -#include #include #include #include diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8ef820fd68c7..e4ca61c18605 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,7 +17,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 48a52a1aaaaa..9b6036e5738a 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -15,7 +15,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index cc712d59ab64..01f4ad725149 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -15,7 +15,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index dac4d05f512d..9fd50bd77c49 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1a6564b3684e..ff0a1079efec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 620411c384e5..0d43e4b4a586 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,7 +33,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index edcbeef43735..b3d1c78e361f 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9782631df93b..cba798e1a07e 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index d2e38af46aa1..f55337567379 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7d665ea3ac62..21625dddde23 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,7 +40,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index abcb97e03405..6b0f0e220f85 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a3b9cbb20ee9..01b1abe7a36b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 844d5d287976..8be5dae83cae 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 26571b25e440..3a655719ba2c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index d0ae82d2b24f..0c89f8e2daaf 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 0aa3e4d138f4..764b2bf80a71 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a6f1e391314d..760080e43015 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 28972d875f7c..c18dee06f29c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index b7d661581ebf..c13e4f7809cf 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index e51f4f0a93f4..bf471dc57114 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index dfb41ad902e1..b312fccbf67a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fb013b152fa6..dc3153852d8a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -39,7 +39,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 87957e862b9c..391c385ec43e 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 359319cbc784..af542a2f5941 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index a430930cc09f..6ac80cf80b31 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 812dce95f131..5957a8b52eda 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f0b0c7a487b3..babca49c8766 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 0dd1e0c0fc1b..9f1cce8d105d 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -28,7 +28,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index dbf2a8328a8e..ca38722bc3fe 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -16,7 +16,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 06dca88a7332..ed2773f623ca 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 783a3d1daf51..c0587013fdfa 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 555311d1ce37..018257c69bca 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -20,7 +20,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 98bfbdd62c60..b01df56b824a 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 16569c7a03c1..e41f9993c652 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 517bb2238d46..649a2e3c02ae 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 9d83bed5c210..608c6721e4f1 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 61c620e5fe4f..58d7f0bff990 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index ac957ece6785..35c5389e5ef7 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 48bf80baf1d4..ba126906f82c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -18,7 +18,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index bde0e84e8214..d1debfb20c60 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.1 From 5fe803f56ad41cf008399f71ee48280f0cf9732b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 15:56:55 +0000 Subject: ASoC: Convert wm1250-ev1 driver to use devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index cd0ec0fd1dba..aefb4f89be0e 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -116,7 +116,7 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) if (!pdata) return 0; - wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); if (!wm1250) { dev_err(&i2c->dev, "Unable to allocate private data\n"); ret = -ENOMEM; @@ -134,15 +134,13 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); if (ret != 0) { dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); - goto err_alloc; + goto err; } dev_set_drvdata(&i2c->dev, wm1250); return ret; -err_alloc: - kfree(wm1250); err: return ret; } @@ -151,10 +149,8 @@ static void wm1250_ev1_free(struct i2c_client *i2c) { struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); - if (wm1250) { + if (wm1250) gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); - kfree(wm1250); - } } static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, -- cgit v1.2.1 From 897f7847e6fec6f24efef4268993afcfc36dca23 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 15:57:44 +0000 Subject: ASoC: Convert wm9081 driver to use devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ba126906f82c..8a4b97060444 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1361,7 +1361,8 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret; - wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); + wm9081 = devm_kzalloc(&i2c->dev, sizeof(struct wm9081_priv), + GFP_KERNEL); if (wm9081 == NULL) return -ENOMEM; @@ -1405,7 +1406,6 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, err_regmap: regmap_exit(wm9081->regmap); err: - kfree(wm9081); return ret; } @@ -1416,7 +1416,6 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm9081->regmap); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From a290986b2a184941da60921ada71bcb47a0d4af2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 15:59:23 +0000 Subject: ASoC: Convert wm8996 to use devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 304a0e570cb4..41cc9d2d5ae9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3104,7 +3104,8 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, int ret, i; unsigned int reg; - wm8996 = kzalloc(sizeof(struct wm8996_priv), GFP_KERNEL); + wm8996 = devm_kzalloc(&i2c->dev, sizeof(struct wm8996_priv), + GFP_KERNEL); if (wm8996 == NULL) return -ENOMEM; @@ -3216,7 +3217,6 @@ err_gpio: if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); err: - kfree(wm8996); return ret; } @@ -3234,7 +3234,6 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } - kfree(wm8996); return 0; } -- cgit v1.2.1 From b05d8dc15f346224306bda4b4ae39fc5ace74ee6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 19:38:34 +0000 Subject: ASoC: Fix CODEC enumeration for auto_nc_codec_pins We need to enumerate all the CODECs that are part of the card we're instantiating, not all the CODECs that are in the system as the system may have multiple cards. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2abaf6dcdb0a..ec783f0a27e9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1489,7 +1489,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_widgets(&card->dapm); if (card->fully_routed) - list_for_each_entry(codec, &codec_list, list) + list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); ret = snd_card_register(card->snd_card); -- cgit v1.2.1 From a094b80bb603d602bef5d8c02faedab8d06ed484 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 19:42:20 +0000 Subject: ASoC: Log automatic pin disconnection per CODEC rather than per card This makes the output a bit less confusing on multi-CODEC systems as the same pin may appear in multiple CODECs. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ecd1b4927f9..da5c1ae7cc30 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2996,7 +2996,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n", + dev_dbg(codec->dev, "Auto NC: DAPMs: card:%p codec:%p\n", &card->dapm, &codec->dapm); list_for_each_entry(w, &card->widgets, list) { @@ -3006,10 +3006,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: - dev_dbg(card->dev, "Auto NC: Checking widget %s\n", + dev_dbg(codec->dev, "Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(card->dev, + dev_dbg(codec->dev, "... Not in map; disabling\n"); snd_soc_dapm_nc_pin(dapm, w->name); } -- cgit v1.2.1 From be086aa8ca7aac8292db9f1a6a17756fb1cfda81 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 19:56:52 +0000 Subject: ASoC: Convert WM8962 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 018257c69bca..8810988522eb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4150,7 +4150,8 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i; - wm8962 = kzalloc(sizeof(struct wm8962_priv), GFP_KERNEL); + wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), + GFP_KERNEL); if (wm8962 == NULL) return -ENOMEM; @@ -4167,7 +4168,7 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err_alloc; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), @@ -4241,8 +4242,7 @@ err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); err_get: regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); -err_alloc: - kfree(wm8962); +err: return ret; } @@ -4253,7 +4253,6 @@ static __devexit int wm8962_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8962->regmap); regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 5bbcc3c0d0f063318ec83146d1958acf7154c66f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Nov 2011 22:52:08 +0000 Subject: ASoC: Convert CODEC drivers to module_platform_driver Factors out a bit of boilerplate. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 12 +----------- sound/soc/codecs/ac97.c | 12 +----------- sound/soc/codecs/ad1980.c | 12 +----------- sound/soc/codecs/ad73311.c | 12 +----------- sound/soc/codecs/ads117x.c | 12 +----------- sound/soc/codecs/cq93vc.c | 12 +----------- sound/soc/codecs/cx20442.c | 12 +----------- sound/soc/codecs/dfbmcs320.c | 12 +----------- sound/soc/codecs/dmic.c | 12 +----------- sound/soc/codecs/jz4740.c | 12 +----------- sound/soc/codecs/pcm3008.c | 12 +----------- sound/soc/codecs/sn95031.c | 14 +------------- sound/soc/codecs/spdif_transciever.c | 13 +------------ sound/soc/codecs/stac9766.c | 12 +----------- sound/soc/codecs/twl4030.c | 12 +----------- sound/soc/codecs/twl6040.c | 12 +----------- sound/soc/codecs/uda134x.c | 12 +----------- sound/soc/codecs/wl1273.c | 12 +----------- sound/soc/codecs/wm8350.c | 12 +----------- sound/soc/codecs/wm8400.c | 12 +----------- sound/soc/codecs/wm8727.c | 12 +----------- sound/soc/codecs/wm8782.c | 12 +----------- sound/soc/codecs/wm8994.c | 13 +------------ sound/soc/codecs/wm9705.c | 12 +----------- sound/soc/codecs/wm9712.c | 12 +----------- sound/soc/codecs/wm9713.c | 12 +----------- 26 files changed, 26 insertions(+), 290 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index ea305b88cb55..2d39123dd21a 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1481,17 +1481,7 @@ static struct platform_driver pm860x_codec_driver = { .remove = __devexit_p(pm860x_codec_remove), }; -static __init int pm860x_init(void) -{ - return platform_driver_register(&pm860x_codec_driver); -} -module_init(pm860x_init); - -static __exit void pm860x_exit(void) -{ - platform_driver_unregister(&pm860x_codec_driver); -} -module_exit(pm860x_exit); +module_platform_driver(pm860x_codec_driver); MODULE_DESCRIPTION("ASoC 88PM860x driver"); MODULE_AUTHOR("Haojian Zhuang "); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8f3216793eb5..221ec29f68e3 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -148,17 +148,7 @@ static struct platform_driver ac97_codec_driver = { .remove = __devexit_p(ac97_remove), }; -static int __init ac97_init(void) -{ - return platform_driver_register(&ac97_codec_driver); -} -module_init(ac97_init); - -static void __exit ac97_exit(void) -{ - platform_driver_unregister(&ac97_codec_driver); -} -module_exit(ac97_exit); +module_platform_driver(ac97_codec_driver); MODULE_DESCRIPTION("Soc Generic AC97 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index e3931cc5e66c..9bba7f849464 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -277,17 +277,7 @@ static struct platform_driver ad1980_codec_driver = { .remove = __devexit_p(ad1980_remove), }; -static int __init ad1980_init(void) -{ - return platform_driver_register(&ad1980_codec_driver); -} -module_init(ad1980_init); - -static void __exit ad1980_exit(void) -{ - platform_driver_unregister(&ad1980_codec_driver); -} -module_exit(ad1980_exit); +module_platform_driver(ad1980_codec_driver); MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 8d793e993e9a..ee7a68dcefd2 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -63,17 +63,7 @@ static struct platform_driver ad73311_codec_driver = { .remove = __devexit_p(ad73311_remove), }; -static int __init ad73311_init(void) -{ - return platform_driver_register(&ad73311_codec_driver); -} -module_init(ad73311_init); - -static void __exit ad73311_exit(void) -{ - platform_driver_unregister(&ad73311_codec_driver); -} -module_exit(ad73311_exit); +module_platform_driver(ad73311_codec_driver); MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 9082e0f729f3..8103b938b8c0 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -58,17 +58,7 @@ static struct platform_driver ads117x_codec_driver = { .remove = __devexit_p(ads117x_remove), }; -static int __init ads117x_init(void) -{ - return platform_driver_register(&ads117x_codec_driver); -} -module_init(ads117x_init); - -static void __exit ads117x_exit(void) -{ - platform_driver_unregister(&ads117x_codec_driver); -} -module_exit(ads117x_exit); +module_platform_driver(ads117x_codec_driver); MODULE_DESCRIPTION("ASoC ads117x driver"); MODULE_AUTHOR("Graeme Gregory"); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index cbb3028e2008..4854b472d5fd 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -206,17 +206,7 @@ static struct platform_driver cq93vc_codec_driver = { .remove = __devexit_p(cq93vc_platform_remove), }; -static int __init cq93vc_init(void) -{ - return platform_driver_register(&cq93vc_codec_driver); -} -module_init(cq93vc_init); - -static void __exit cq93vc_exit(void) -{ - platform_driver_unregister(&cq93vc_codec_driver); -} -module_exit(cq93vc_exit); +module_platform_driver(cq93vc_codec_driver); MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); MODULE_AUTHOR("Miguel Aguilar"); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067db8ae4..ae55e31bfc72 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -391,17 +391,7 @@ static struct platform_driver cx20442_platform_driver = { .remove = __exit_p(cx20442_platform_remove), }; -static int __init cx20442_init(void) -{ - return platform_driver_register(&cx20442_platform_driver); -} -module_init(cx20442_init); - -static void __exit cx20442_exit(void) -{ - platform_driver_unregister(&cx20442_platform_driver); -} -module_exit(cx20442_exit); +module_platform_driver(cx20442_platform_driver); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c index 704bbde65737..bfe46aa90362 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/dfbmcs320.c @@ -55,17 +55,7 @@ static struct platform_driver dfmcs320_driver = { .remove = __devexit_p(dfbmcs320_remove), }; -static int __init dfbmcs320_init(void) -{ - return platform_driver_register(&dfmcs320_driver); -} -module_init(dfbmcs320_init); - -static void __exit dfbmcs320_exit(void) -{ - platform_driver_unregister(&dfmcs320_driver); -} -module_exit(dfbmcs320_exit); +module_platform_driver(dfmcs320_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 6fae765e3ad8..3e929f079a1f 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -89,17 +89,7 @@ static struct platform_driver dmic_driver = { .remove = __devexit_p(dmic_dev_remove), }; -static int __init dmic_init(void) -{ - return platform_driver_register(&dmic_driver); -} -module_init(dmic_init); - -static void __exit dmic_exit(void) -{ - platform_driver_unregister(&dmic_driver); -} -module_exit(dmic_exit); +module_platform_driver(dmic_driver); MODULE_DESCRIPTION("Generic DMIC driver"); MODULE_AUTHOR("Liam Girdwood "); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 64a479c3429a..4fca8bccd535 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -424,17 +424,7 @@ static struct platform_driver jz4740_codec_driver = { }, }; -static int __init jz4740_codec_init(void) -{ - return platform_driver_register(&jz4740_codec_driver); -} -module_init(jz4740_codec_init); - -static void __exit jz4740_codec_exit(void) -{ - platform_driver_unregister(&jz4740_codec_driver); -} -module_exit(jz4740_codec_exit); +module_platform_driver(jz4740_codec_driver); MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); MODULE_AUTHOR("Lars-Peter Clausen "); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f7316519432c..b12d01f67990 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -172,17 +172,7 @@ static struct platform_driver pcm3008_codec_driver = { }, }; -static int __init pcm3008_modinit(void) -{ - return platform_driver_register(&pcm3008_codec_driver); -} -module_init(pcm3008_modinit); - -static void __exit pcm3008_exit(void) -{ - platform_driver_unregister(&pcm3008_codec_driver); -} -module_exit(pcm3008_exit); +module_platform_driver(pcm3008_codec_driver); MODULE_DESCRIPTION("Soc PCM3008 driver"); MODULE_AUTHOR("Hugo Villeneuve"); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 65f2ef986c4f..f99baa0b8c39 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -920,19 +920,7 @@ static struct platform_driver sn95031_codec_driver = { .remove = __devexit_p(sn95031_device_remove), }; -static int __init sn95031_init(void) -{ - pr_debug("driver init called\n"); - return platform_driver_register(&sn95031_codec_driver); -} -module_init(sn95031_init); - -static void __exit sn95031_exit(void) -{ - pr_debug("driver exit called\n"); - platform_driver_unregister(&sn95031_codec_driver); -} -module_exit(sn95031_exit); +module_platform_driver(sn95031_codec_driver); MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul "); diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 6a1a7e705cd7..112a49d66e39 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -61,18 +61,7 @@ static struct platform_driver spdif_dit_driver = { }, }; -static int __init dit_modinit(void) -{ - return platform_driver_register(&spdif_dit_driver); -} - -static void __exit dit_exit(void) -{ - platform_driver_unregister(&spdif_dit_driver); -} - -module_init(dit_modinit); -module_exit(dit_exit); +module_platform_driver(spdif_dit_driver); MODULE_AUTHOR("Steve Chen "); MODULE_DESCRIPTION("SPDIF dummy codec driver"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e4783a4f71fd..55819537b677 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -408,17 +408,7 @@ static struct platform_driver stac9766_codec_driver = { .remove = __devexit_p(stac9766_remove), }; -static int __init stac9766_init(void) -{ - return platform_driver_register(&stac9766_codec_driver); -} -module_init(stac9766_init); - -static void __exit stac9766_exit(void) -{ - platform_driver_unregister(&stac9766_codec_driver); -} -module_exit(stac9766_exit); +module_platform_driver(stac9766_codec_driver); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl "); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a3a52838e9c..61d8a9065ff3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2294,17 +2294,7 @@ static struct platform_driver twl4030_codec_driver = { }, }; -static int __init twl4030_modinit(void) -{ - return platform_driver_register(&twl4030_codec_driver); -} -module_init(twl4030_modinit); - -static void __exit twl4030_exit(void) -{ - platform_driver_unregister(&twl4030_codec_driver); -} -module_exit(twl4030_exit); +module_platform_driver(twl4030_codec_driver); MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 17930edd3a2c..a4a65dc9e33a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1620,17 +1620,7 @@ static struct platform_driver twl6040_codec_driver = { .remove = __devexit_p(twl6040_codec_remove), }; -static int __init twl6040_codec_init(void) -{ - return platform_driver_register(&twl6040_codec_driver); -} -module_init(twl6040_codec_init); - -static void __exit twl6040_codec_exit(void) -{ - platform_driver_unregister(&twl6040_codec_driver); -} -module_exit(twl6040_codec_exit); +module_platform_driver(twl6040_codec_driver); MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); MODULE_AUTHOR("Misael Lopez Cruz"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 486aef637eed..d0f9d904ce8f 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -625,17 +625,7 @@ static struct platform_driver uda134x_codec_driver = { .remove = __devexit_p(uda134x_codec_remove), }; -static int __init uda134x_codec_init(void) -{ - return platform_driver_register(&uda134x_codec_driver); -} -module_init(uda134x_codec_init); - -static void __exit uda134x_codec_exit(void) -{ - platform_driver_unregister(&uda134x_codec_driver); -} -module_exit(uda134x_codec_exit); +module_platform_driver(uda134x_codec_driver); MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin "); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 9531c35dccad..44aacf927ba9 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -510,17 +510,7 @@ static struct platform_driver wl1273_platform_driver = { .remove = __devexit_p(wl1273_platform_remove), }; -static int __init wl1273_init(void) -{ - return platform_driver_register(&wl1273_platform_driver); -} -module_init(wl1273_init); - -static void __exit wl1273_exit(void) -{ - platform_driver_unregister(&wl1273_platform_driver); -} -module_exit(wl1273_exit); +module_platform_driver(wl1273_platform_driver); MODULE_AUTHOR("Matti Aaltonen "); MODULE_DESCRIPTION("ASoC WL1273 codec driver"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b846c95f07f..3f1ed5f5ccf4 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1711,17 +1711,7 @@ static struct platform_driver wm8350_codec_driver = { .remove = __devexit_p(wm8350_remove), }; -static __init int wm8350_init(void) -{ - return platform_driver_register(&wm8350_codec_driver); -} -module_init(wm8350_init); - -static __exit void wm8350_exit(void) -{ - platform_driver_unregister(&wm8350_codec_driver); -} -module_exit(wm8350_exit); +module_platform_driver(wm8350_codec_driver); MODULE_DESCRIPTION("ASoC WM8350 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 07d84a86e14e..a1173eb7936d 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1477,17 +1477,7 @@ static struct platform_driver wm8400_codec_driver = { .remove = __devexit_p(wm8400_remove), }; -static __init int wm8400_init(void) -{ - return platform_driver_register(&wm8400_codec_driver); -} -module_init(wm8400_init); - -static __exit void wm8400_exit(void) -{ - platform_driver_unregister(&wm8400_codec_driver); -} -module_exit(wm8400_exit); +module_platform_driver(wm8400_codec_driver); MODULE_DESCRIPTION("ASoC WM8400 driver"); MODULE_AUTHOR("Mark Brown"); diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 748808285119..fad90a35f399 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -67,17 +67,7 @@ static struct platform_driver wm8727_codec_driver = { .remove = __devexit_p(wm8727_remove), }; -static int __init wm8727_init(void) -{ - return platform_driver_register(&wm8727_codec_driver); -} -module_init(wm8727_init); - -static void __exit wm8727_exit(void) -{ - platform_driver_unregister(&wm8727_codec_driver); -} -module_exit(wm8727_exit); +module_platform_driver(wm8727_codec_driver); MODULE_DESCRIPTION("ASoC wm8727 driver"); MODULE_AUTHOR("Neil Jones"); diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f2ced71328b0..3fdea98f732e 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -63,17 +63,7 @@ static struct platform_driver wm8782_codec_driver = { .remove = __devexit_p(wm8782_remove), }; -static int __init wm8782_init(void) -{ - return platform_driver_register(&wm8782_codec_driver); -} -module_init(wm8782_init); - -static void __exit wm8782_exit(void) -{ - platform_driver_unregister(&wm8782_codec_driver); -} -module_exit(wm8782_exit); +module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach "); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 73db9806c475..380e3f2f3190 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3579,18 +3579,7 @@ static struct platform_driver wm8994_codec_driver = { .remove = __devexit_p(wm8994_remove), }; -static __init int wm8994_init(void) -{ - return platform_driver_register(&wm8994_codec_driver); -} -module_init(wm8994_init); - -static __exit void wm8994_exit(void) -{ - platform_driver_unregister(&wm8994_codec_driver); -} -module_exit(wm8994_exit); - +module_platform_driver(wm8994_codec_driver); MODULE_DESCRIPTION("ASoC WM8994 driver"); MODULE_AUTHOR("Mark Brown "); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index edf603281ce7..b720a43c422c 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,17 +406,7 @@ static struct platform_driver wm9705_codec_driver = { .remove = __devexit_p(wm9705_remove), }; -static int __init wm9705_init(void) -{ - return platform_driver_register(&wm9705_codec_driver); -} -module_init(wm9705_init); - -static void __exit wm9705_exit(void) -{ - platform_driver_unregister(&wm9705_codec_driver); -} -module_exit(wm9705_exit); +module_platform_driver(wm9705_codec_driver); MODULE_DESCRIPTION("ASoC WM9705 driver"); MODULE_AUTHOR("Ian Molton"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd1812704af8..4ce73f59df20 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -694,17 +694,7 @@ static struct platform_driver wm9712_codec_driver = { .remove = __devexit_p(wm9712_remove), }; -static int __init wm9712_init(void) -{ - return platform_driver_register(&wm9712_codec_driver); -} -module_init(wm9712_init); - -static void __exit wm9712_exit(void) -{ - platform_driver_unregister(&wm9712_codec_driver); -} -module_exit(wm9712_exit); +module_platform_driver(wm9712_codec_driver); MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 09360b60037c..edb598182c69 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1277,17 +1277,7 @@ static struct platform_driver wm9713_codec_driver = { .remove = __devexit_p(wm9713_remove), }; -static int __init wm9713_init(void) -{ - return platform_driver_register(&wm9713_codec_driver); -} -module_init(wm9713_init); - -static void __exit wm9713_exit(void) -{ - platform_driver_unregister(&wm9713_codec_driver); -} -module_exit(wm9713_exit); +module_platform_driver(wm9713_codec_driver); MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); MODULE_AUTHOR("Liam Girdwood"); -- cgit v1.2.1 From 5032dc34294d1084b7367877dadb6edb2d45ad7c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 12:20:08 +0000 Subject: ASoC: Convert WM8903 MICBIAS to a supply widget Also rename it to MICBIAS to reflect the pin name and help any out of tree users notice the change. Signed-off-by: Mark Brown Acked-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/tegra/tegra_wm8903.c | 18 +++++++++--------- 2 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 5957a8b52eda..70a2268c5498 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -838,7 +838,7 @@ SND_SOC_DAPM_OUTPUT("LON"), SND_SOC_DAPM_OUTPUT("ROP"), SND_SOC_DAPM_OUTPUT("RON"), -SND_SOC_DAPM_MICBIAS("Mic Bias", WM8903_MIC_BIAS_CONTROL_0, 0, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8903_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Left Input Mux", SND_SOC_NOPM, 0, 0, &linput_mux), SND_SOC_DAPM_MUX("Left Input Inverting Mux", SND_SOC_NOPM, 0, 0, @@ -947,7 +947,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0), static const struct snd_soc_dapm_route wm8903_intercon[] = { { "CLK_DSP", NULL, "CLK_SYS" }, - { "Mic Bias", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "CLK_SYS" }, { "HPL_DCS", NULL, "CLK_SYS" }, { "HPR_DCS", NULL, "CLK_SYS" }, { "LINEOUTL_DCS", NULL, "CLK_SYS" }, diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index b260f54a4462..2f5b1074a8d9 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -201,8 +201,8 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1L", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1L", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route seaboard_audio_map[] = { @@ -212,8 +212,8 @@ static const struct snd_soc_dapm_route seaboard_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1R", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route kaen_audio_map[] = { @@ -223,8 +223,8 @@ static const struct snd_soc_dapm_route kaen_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN2R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN2R", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route aebl_audio_map[] = { @@ -232,8 +232,8 @@ static const struct snd_soc_dapm_route aebl_audio_map[] = { {"Headphone Jack", NULL, "HPOUTL"}, {"Int Spk", NULL, "LINEOUTR"}, {"Int Spk", NULL, "LINEOUTL"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1R", NULL, "Mic Jack"}, }; static const struct snd_kcontrol_new tegra_wm8903_controls[] = { @@ -329,7 +329,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); return 0; } -- cgit v1.2.1 From fd26f9474676bb2232ba9dded148edc41fd02ef4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 28 Nov 2011 15:45:40 +0200 Subject: ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC Add support for OMAP4 Digital Microphone interface. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 549 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 ++++++ 4 files changed, 623 insertions(+) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fe83d0d176be..052254ac333f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -2,6 +2,9 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP +config SND_OMAP_SOC_DMIC + tristate + config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 052fd758722e..1fd723fb559d 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,10 +1,12 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o +snd-soc-omap-dmic-objs := omap-dmic.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o +obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c new file mode 100644 index 000000000000..9c73c0c70d39 --- /dev/null +++ b/sound/soc/omap/omap-dmic.c @@ -0,0 +1,549 @@ +/* + * omap-dmic.c -- OMAP ASoC DMIC DAI driver + * + * Copyright (C) 2010 - 2011 Texas Instruments + * + * Author: David Lambert + * Misael Lopez Cruz + * Liam Girdwood + * Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "omap-pcm.h" +#include "omap-dmic.h" + +struct omap_dmic { + struct device *dev; + void __iomem *io_base; + struct clk *fclk; + int fclk_freq; + int out_freq; + int clk_div; + int sysclk; + int threshold; + u32 ch_enabled; + bool active; + struct mutex mutex; +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_dmic_dai_dma_params = { + .name = "DMIC capture", + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, +}; + +static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) +{ + __raw_writel(val, dmic->io_base + reg); +} + +static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg) +{ + return __raw_readl(dmic->io_base + reg); +} + +static inline void omap_dmic_start(struct omap_dmic *dmic) +{ + u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG); + + /* Configure DMA controller */ + omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_SET_REG, + OMAP_DMIC_DMA_ENABLE); + + omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl | dmic->ch_enabled); +} + +static inline void omap_dmic_stop(struct omap_dmic *dmic) +{ + u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG); + omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, + ctrl & ~OMAP_DMIC_UP_ENABLE_MASK); + + /* Disable DMA request generation */ + omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_CLR_REG, + OMAP_DMIC_DMA_ENABLE); + +} + +static inline int dmic_is_enabled(struct omap_dmic *dmic) +{ + return omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG) & + OMAP_DMIC_UP_ENABLE_MASK; +} + +static int omap_dmic_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + mutex_lock(&dmic->mutex); + + if (!dai->active) { + pm_runtime_get_sync(dmic->dev); + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + dmic->active = 1; + } else { + ret = -EBUSY; + } + + mutex_unlock(&dmic->mutex); + + return ret; +} + +static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + + mutex_lock(&dmic->mutex); + + if (!dai->active) { + pm_runtime_put_sync(dmic->dev); + dmic->active = 0; + } + + mutex_unlock(&dmic->mutex); +} + +static int omap_dmic_select_divider(struct omap_dmic *dmic, int sample_rate) +{ + int divider = -EINVAL; + + /* + * 192KHz rate is only supported with 19.2MHz/3.84MHz clock + * configuration. + */ + if (sample_rate == 192000) { + if (dmic->fclk_freq == 19200000 && dmic->out_freq == 3840000) + divider = 0x6; /* Divider: 5 (192KHz sampling rate) */ + else + dev_err(dmic->dev, + "invalid clock configuration for 192KHz\n"); + + return divider; + } + + switch (dmic->out_freq) { + case 1536000: + if (dmic->fclk_freq != 24576000) + goto div_err; + divider = 0x4; /* Divider: 16 */ + break; + case 2400000: + switch (dmic->fclk_freq) { + case 12000000: + divider = 0x5; /* Divider: 5 */ + break; + case 19200000: + divider = 0x0; /* Divider: 8 */ + break; + case 24000000: + divider = 0x2; /* Divider: 10 */ + break; + default: + goto div_err; + } + break; + case 3072000: + if (dmic->fclk_freq != 24576000) + goto div_err; + divider = 0x3; /* Divider: 8 */ + break; + case 3840000: + if (dmic->fclk_freq != 19200000) + goto div_err; + divider = 0x1; /* Divider: 5 (96KHz sampling rate) */ + break; + default: + dev_err(dmic->dev, "invalid out frequency: %dHz\n", + dmic->out_freq); + break; + } + + return divider; + +div_err: + dev_err(dmic->dev, "invalid out frequency %dHz for %dHz input\n", + dmic->out_freq, dmic->fclk_freq); + return -EINVAL; +} + +static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + int channels; + + dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params)); + if (dmic->clk_div < 0) { + dev_err(dmic->dev, "no valid divider for %dHz from %dHz\n", + dmic->out_freq, dmic->fclk_freq); + return -EINVAL; + } + + dmic->ch_enabled = 0; + channels = params_channels(params); + switch (channels) { + case 6: + dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; + case 4: + dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; + case 2: + dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; + break; + default: + dev_err(dmic->dev, "invalid number of legacy channels\n"); + return -EINVAL; + } + + /* packet size is threshold * channels */ + omap_dmic_dai_dma_params.packet_size = dmic->threshold * channels; + snd_soc_dai_set_dma_data(dai, substream, &omap_dmic_dai_dma_params); + + return 0; +} + +static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + + /* Configure uplink threshold */ + omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); + + ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG); + + /* Set dmic out format */ + ctrl &= ~(OMAP_DMIC_FORMAT | OMAP_DMIC_POLAR_MASK); + ctrl |= (OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 | + OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3); + + /* Configure dmic clock divider */ + ctrl &= ~OMAP_DMIC_CLK_DIV_MASK; + ctrl |= OMAP_DMIC_CLK_DIV(dmic->clk_div); + + omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl); + + omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, + ctrl | OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 | + OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3); + + return 0; +} + +static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + omap_dmic_start(dmic); + break; + case SNDRV_PCM_TRIGGER_STOP: + omap_dmic_stop(dmic); + break; + default: + break; + } + + return 0; +} + +static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, + unsigned int freq) +{ + struct clk *parent_clk; + char *parent_clk_name; + int ret = 0; + + switch (freq) { + case 12000000: + case 19200000: + case 24000000: + case 24576000: + break; + default: + dev_err(dmic->dev, "invalid input frequency: %dHz\n", freq); + dmic->fclk_freq = 0; + return -EINVAL; + } + + if (dmic->sysclk == clk_id) { + dmic->fclk_freq = freq; + return 0; + } + + /* re-parent not allowed if a stream is ongoing */ + if (dmic->active && dmic_is_enabled(dmic)) { + dev_err(dmic->dev, "can't re-parent when DMIC active\n"); + return -EBUSY; + } + + switch (clk_id) { + case OMAP_DMIC_SYSCLK_PAD_CLKS: + parent_clk_name = "pad_clks_ck"; + break; + case OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS: + parent_clk_name = "slimbus_clk"; + break; + case OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS: + parent_clk_name = "dmic_sync_mux_ck"; + break; + default: + dev_err(dmic->dev, "fclk clk_id (%d) not supported\n", clk_id); + return -EINVAL; + } + + parent_clk = clk_get(dmic->dev, parent_clk_name); + if (IS_ERR(parent_clk)) { + dev_err(dmic->dev, "can't get %s\n", parent_clk_name); + return -ENODEV; + } + + mutex_lock(&dmic->mutex); + if (dmic->active) { + /* disable clock while reparenting */ + pm_runtime_put_sync(dmic->dev); + ret = clk_set_parent(dmic->fclk, parent_clk); + pm_runtime_get_sync(dmic->dev); + } else { + ret = clk_set_parent(dmic->fclk, parent_clk); + } + mutex_unlock(&dmic->mutex); + + if (ret < 0) { + dev_err(dmic->dev, "re-parent failed\n"); + goto err_busy; + } + + dmic->sysclk = clk_id; + dmic->fclk_freq = freq; + +err_busy: + clk_put(parent_clk); + + return ret; +} + +static int omap_dmic_select_outclk(struct omap_dmic *dmic, int clk_id, + unsigned int freq) +{ + int ret = 0; + + if (clk_id != OMAP_DMIC_ABE_DMIC_CLK) { + dev_err(dmic->dev, "output clk_id (%d) not supported\n", + clk_id); + return -EINVAL; + } + + switch (freq) { + case 1536000: + case 2400000: + case 3072000: + case 3840000: + dmic->out_freq = freq; + break; + default: + dev_err(dmic->dev, "invalid out frequency: %dHz\n", freq); + dmic->out_freq = 0; + ret = -EINVAL; + } + + return ret; +} + +static int omap_dmic_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN) + return omap_dmic_select_fclk(dmic, clk_id, freq); + else if (dir == SND_SOC_CLOCK_OUT) + return omap_dmic_select_outclk(dmic, clk_id, freq); + + dev_err(dmic->dev, "invalid clock direction (%d)\n", dir); + return -EINVAL; +} + +static const struct snd_soc_dai_ops omap_dmic_dai_ops = { + .startup = omap_dmic_dai_startup, + .shutdown = omap_dmic_dai_shutdown, + .hw_params = omap_dmic_dai_hw_params, + .prepare = omap_dmic_dai_prepare, + .trigger = omap_dmic_dai_trigger, + .set_sysclk = omap_dmic_set_dai_sysclk, +}; + +static int omap_dmic_probe(struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + + pm_runtime_enable(dmic->dev); + + /* Disable lines while request is ongoing */ + pm_runtime_get_sync(dmic->dev); + omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, 0x00); + pm_runtime_put_sync(dmic->dev); + + /* Configure DMIC threshold value */ + dmic->threshold = OMAP_DMIC_THRES_MAX - 3; + return 0; +} + +static int omap_dmic_remove(struct snd_soc_dai *dai) +{ + struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); + + pm_runtime_disable(dmic->dev); + + return 0; +} + +static struct snd_soc_dai_driver omap_dmic_dai = { + .name = "omap-dmic", + .probe = omap_dmic_probe, + .remove = omap_dmic_remove, + .capture = { + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &omap_dmic_dai_ops, +}; + +static __devinit int asoc_dmic_probe(struct platform_device *pdev) +{ + struct omap_dmic *dmic; + struct resource *res; + int ret; + + dmic = devm_kzalloc(&pdev->dev, sizeof(struct omap_dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + platform_set_drvdata(pdev, dmic); + dmic->dev = &pdev->dev; + dmic->sysclk = OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS; + + mutex_init(&dmic->mutex); + + dmic->fclk = clk_get(dmic->dev, "dmic_fck"); + if (IS_ERR(dmic->fclk)) { + dev_err(dmic->dev, "cant get dmic_fck\n"); + return -ENODEV; + } + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (!res) { + dev_err(dmic->dev, "invalid dma memory resource\n"); + ret = -ENODEV; + goto err_put_clk; + } + omap_dmic_dai_dma_params.port_addr = res->start + OMAP_DMIC_DATA_REG; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dmic->dev, "invalid dma resource\n"); + ret = -ENODEV; + goto err_put_clk; + } + omap_dmic_dai_dma_params.dma_req = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!res) { + dev_err(dmic->dev, "invalid memory resource\n"); + ret = -ENODEV; + goto err_put_clk; + } + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_err(dmic->dev, "memory region already claimed\n"); + ret = -ENODEV; + goto err_put_clk; + } + + dmic->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!dmic->io_base) { + ret = -ENOMEM; + goto err_put_clk; + } + + ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai); + if (ret) + goto err_put_clk; + + return 0; + +err_put_clk: + clk_put(dmic->fclk); + return ret; +} + +static int __devexit asoc_dmic_remove(struct platform_device *pdev) +{ + struct omap_dmic *dmic = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + clk_put(dmic->fclk); + + return 0; +} + +static struct platform_driver asoc_dmic_driver = { + .driver = { + .name = "omap-dmic", + .owner = THIS_MODULE, + }, + .probe = asoc_dmic_probe, + .remove = __devexit_p(asoc_dmic_remove), +}; + +module_platform_driver(asoc_dmic_driver); + +MODULE_ALIAS("platform:omap-dmic"); +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_DESCRIPTION("OMAP DMIC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-dmic.h b/sound/soc/omap/omap-dmic.h new file mode 100644 index 000000000000..231e728bff0e --- /dev/null +++ b/sound/soc/omap/omap-dmic.h @@ -0,0 +1,69 @@ +/* + * omap-dmic.h -- OMAP Digital Microphone Controller + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _OMAP_DMIC_H +#define _OMAP_DMIC_H + +#define OMAP_DMIC_REVISION_REG 0x00 +#define OMAP_DMIC_SYSCONFIG_REG 0x10 +#define OMAP_DMIC_IRQSTATUS_RAW_REG 0x24 +#define OMAP_DMIC_IRQSTATUS_REG 0x28 +#define OMAP_DMIC_IRQENABLE_SET_REG 0x2C +#define OMAP_DMIC_IRQENABLE_CLR_REG 0x30 +#define OMAP_DMIC_IRQWAKE_EN_REG 0x34 +#define OMAP_DMIC_DMAENABLE_SET_REG 0x38 +#define OMAP_DMIC_DMAENABLE_CLR_REG 0x3C +#define OMAP_DMIC_DMAWAKEEN_REG 0x40 +#define OMAP_DMIC_CTRL_REG 0x44 +#define OMAP_DMIC_DATA_REG 0x48 +#define OMAP_DMIC_FIFO_CTRL_REG 0x4C +#define OMAP_DMIC_FIFO_DMIC1R_DATA_REG 0x50 +#define OMAP_DMIC_FIFO_DMIC1L_DATA_REG 0x54 +#define OMAP_DMIC_FIFO_DMIC2R_DATA_REG 0x58 +#define OMAP_DMIC_FIFO_DMIC2L_DATA_REG 0x5C +#define OMAP_DMIC_FIFO_DMIC3R_DATA_REG 0x60 +#define OMAP_DMIC_FIFO_DMIC3L_DATA_REG 0x64 + +/* IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR bit fields */ +#define OMAP_DMIC_IRQ (1 << 0) +#define OMAP_DMIC_IRQ_FULL (1 << 1) +#define OMAP_DMIC_IRQ_ALMST_EMPTY (1 << 2) +#define OMAP_DMIC_IRQ_EMPTY (1 << 3) +#define OMAP_DMIC_IRQ_MASK 0x07 + +/* DMIC_DMAENABLE bit fields */ +#define OMAP_DMIC_DMA_ENABLE 0x1 + +/* DMIC_CTRL bit fields */ +#define OMAP_DMIC_UP1_ENABLE (1 << 0) +#define OMAP_DMIC_UP2_ENABLE (1 << 1) +#define OMAP_DMIC_UP3_ENABLE (1 << 2) +#define OMAP_DMIC_UP_ENABLE_MASK 0x7 +#define OMAP_DMIC_FORMAT (1 << 3) +#define OMAP_DMIC_POLAR1 (1 << 4) +#define OMAP_DMIC_POLAR2 (1 << 5) +#define OMAP_DMIC_POLAR3 (1 << 6) +#define OMAP_DMIC_POLAR_MASK (0x7 << 4) +#define OMAP_DMIC_CLK_DIV(x) (((x) & 0x7) << 7) +#define OMAP_DMIC_CLK_DIV_MASK (0x7 << 7) +#define OMAP_DMIC_RESET (1 << 10) + +#define OMAP_DMICOUTFORMAT_LJUST (0 << 3) +#define OMAP_DMICOUTFORMAT_RJUST (1 << 3) + +/* DMIC_FIFO_CTRL bit fields */ +#define OMAP_DMIC_THRES_MAX 0xF + +enum omap_dmic_clk { + OMAP_DMIC_SYSCLK_PAD_CLKS, /* PAD_CLKS */ + OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS, /* SLIMBUS_CLK */ + OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS, /* DMIC_SYNC_MUX_CLK */ + OMAP_DMIC_ABE_DMIC_CLK, /* abe_dmic_clk */ +}; + +#endif -- cgit v1.2.1 From 6524c8e3e6525891d6085c7fb0f7fe5ce18e5b50 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 28 Nov 2011 15:45:43 +0200 Subject: ASoC: sdp4430: Add support for digital microphones OMAP4 SDP/Blaze boards have digital microphones. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 2 ++ sound/soc/omap/sdp4430.c | 85 +++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 76 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 052254ac333f..fb1bf2581efb 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,8 +100,10 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_SDP4430 tristate "SoC Audio support for Texas Instruments SDP4430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP + select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 + select SND_SOC_DMIC help Say Y if you want to add support for SoC audio on Texas Instruments SDP4430. diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 03d9fa4192fe..2735fa03b74b 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -33,6 +33,7 @@ #include #include +#include "omap-dmic.h" #include "omap-mcpdm.h" #include "omap-pcm.h" #include "../codecs/twl6040.h" @@ -67,6 +68,32 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set DMIC cpu system clock\n"); + return ret; + } + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + printk(KERN_ERR "can't set DMIC output clock\n"); + return ret; + } + return 0; +} + +static struct snd_soc_ops sdp4430_dmic_ops = { + .hw_params = sdp4430_dmic_hw_params, +}; + /* Headset jack */ static struct snd_soc_jack hs_jack; @@ -148,23 +175,59 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Digital Mic", NULL), +}; + +static const struct snd_soc_dapm_route dmic_audio_map[] = { + {"DMic", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic"}, +}; + +static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, + ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); + if (ret) + return ret; + + return snd_soc_dapm_add_routes(dapm, dmic_audio_map, + ARRAY_SIZE(dmic_audio_map)); +} + /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp4430_dai = { - .name = "TWL6040", - .stream_name = "TWL6040", - .cpu_dai_name = "omap-mcpdm", - .codec_dai_name = "twl6040-legacy", - .platform_name = "omap-pcm-audio", - .codec_name = "twl6040-codec", - .init = sdp4430_twl6040_init, - .ops = &sdp4430_ops, +static struct snd_soc_dai_link sdp4430_dai[] = { + { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", + .platform_name = "omap-pcm-audio", + .codec_name = "twl6040-codec", + .init = sdp4430_twl6040_init, + .ops = &sdp4430_ops, + }, + { + .name = "DMIC", + .stream_name = "DMIC Capture", + .cpu_dai_name = "omap-dmic", + .codec_dai_name = "dmic-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "dmic-codec", + .init = sdp4430_dmic_init, + .ops = &sdp4430_dmic_ops, + }, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", - .dai_link = &sdp4430_dai, - .num_links = 1, + .dai_link = sdp4430_dai, + .num_links = ARRAY_SIZE(sdp4430_dai), .dapm_widgets = sdp4430_twl6040_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), -- cgit v1.2.1 From ba0a7e024d2a0ccdb887cda149f3e11f1ce27101 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 10:10:55 +0800 Subject: ASoC: Convert fsl directory to module_platform_driver Factor out some boilerplate code. Signed-off-by: Axel Lin Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 15 +-------------- sound/soc/fsl/fsl_ssi.c | 15 +-------------- sound/soc/fsl/mpc5200_dma.c | 12 +----------- sound/soc/fsl/mpc5200_psc_ac97.c | 16 +--------------- sound/soc/fsl/mpc5200_psc_i2s.c | 16 +--------------- 5 files changed, 5 insertions(+), 69 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ef15402a3bc4..4f59bbaba48f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -992,20 +992,7 @@ static struct platform_driver fsl_soc_dma_driver = { .remove = __devexit_p(fsl_soc_dma_remove), }; -static int __init fsl_soc_dma_init(void) -{ - pr_info("Freescale Elo DMA ASoC PCM Driver\n"); - - return platform_driver_register(&fsl_soc_dma_driver); -} - -static void __exit fsl_soc_dma_exit(void) -{ - platform_driver_unregister(&fsl_soc_dma_driver); -} - -module_init(fsl_soc_dma_init); -module_exit(fsl_soc_dma_exit); +module_platform_driver(fsl_soc_dma_driver); MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM Driver"); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 17d857e55efe..3e066966d878 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -793,20 +793,7 @@ static struct platform_driver fsl_ssi_driver = { .remove = fsl_ssi_remove, }; -static int __init fsl_ssi_init(void) -{ - printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - - return platform_driver_register(&fsl_ssi_driver); -} - -static void __exit fsl_ssi_exit(void) -{ - platform_driver_unregister(&fsl_ssi_driver); -} - -module_init(fsl_ssi_init); -module_exit(fsl_ssi_exit); +module_platform_driver(fsl_ssi_driver); MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 5c6c2457386e..e7803d34c425 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -526,17 +526,7 @@ static struct platform_driver mpc5200_hpcd_of_driver = { } }; -static int __init mpc5200_hpcd_init(void) -{ - return platform_driver_register(&mpc5200_hpcd_of_driver); -} -module_init(mpc5200_hpcd_init); - -static void __exit mpc5200_hpcd_exit(void) -{ - platform_driver_unregister(&mpc5200_hpcd_of_driver); -} -module_exit(mpc5200_hpcd_exit); +module_platform_driver(mpc5200_hpcd_of_driver); MODULE_AUTHOR("Grant Likely "); MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 2fb388f0150b..ffa00a2eb770 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -325,21 +325,7 @@ static struct platform_driver psc_ac97_driver = { }, }; -/* --------------------------------------------------------------------- - * Module setup and teardown; simply register the of_platform driver - * for the PSC in AC97 mode. - */ -static int __init psc_ac97_init(void) -{ - return platform_driver_register(&psc_ac97_driver); -} -module_init(psc_ac97_init); - -static void __exit psc_ac97_exit(void) -{ - platform_driver_unregister(&psc_ac97_driver); -} -module_exit(psc_ac97_exit); +module_platform_driver(psc_ac97_driver); MODULE_AUTHOR("Jon Smirl "); MODULE_DESCRIPTION("mpc5200 AC97 module"); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index e77a1f20d4d2..7b530327553a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -222,21 +222,7 @@ static struct platform_driver psc_i2s_driver = { }, }; -/* --------------------------------------------------------------------- - * Module setup and teardown; simply register the of_platform driver - * for the PSC in I2S mode. - */ -static int __init psc_i2s_init(void) -{ - return platform_driver_register(&psc_i2s_driver); -} -module_init(psc_i2s_init); - -static void __exit psc_i2s_exit(void) -{ - platform_driver_unregister(&psc_i2s_driver); -} -module_exit(psc_i2s_exit); +module_platform_driver(psc_i2s_driver); MODULE_AUTHOR("Grant Likely "); MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); -- cgit v1.2.1 From b90d4183f70e8a922db781b7ecfc823d37a3202a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:06 +0100 Subject: ASoC: ad193x: Use table based DAPM and controls setup Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 1901cd222233..1dfda5ca2789 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -345,7 +345,6 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; codec->control_data = ad193x->regmap; @@ -371,17 +370,17 @@ static int ad193x_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); - snd_soc_add_controls(codec, ad193x_snd_controls, - ARRAY_SIZE(ad193x_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets, - ARRAY_SIZE(ad193x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return ret; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, + .controls = ad193x_snd_controls, + .num_controls = ARRAY_SIZE(ad193x_snd_controls), + .dapm_widgets = ad193x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad193x_dapm_widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.1 From 591c034a32a8e3034c447308ad7a4ef19e7ca617 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:07 +0100 Subject: ASoC: ad193x: Provide dB ranges for the volume controls Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 1dfda5ca2789..7da7e29753b6 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -35,16 +35,18 @@ static const char *ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; static const struct soc_enum ad193x_deemp_enum = SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp); +static const DECLARE_TLV_DB_MINMAX(adau193x_tlv, -9563, 0); + static const struct snd_kcontrol_new ad193x_snd_controls[] = { /* DAC volume control */ - SOC_DOUBLE_R("DAC1 Volume", AD193X_DAC_L1_VOL, - AD193X_DAC_R1_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC2 Volume", AD193X_DAC_L2_VOL, - AD193X_DAC_R2_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC3 Volume", AD193X_DAC_L3_VOL, - AD193X_DAC_R3_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC4 Volume", AD193X_DAC_L4_VOL, - AD193X_DAC_R4_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R_TLV("DAC1 Volume", AD193X_DAC_L1_VOL, + AD193X_DAC_R1_VOL, 0, 0xFF, 1, adau193x_tlv), + SOC_DOUBLE_R_TLV("DAC2 Volume", AD193X_DAC_L2_VOL, + AD193X_DAC_R2_VOL, 0, 0xFF, 1, adau193x_tlv), + SOC_DOUBLE_R_TLV("DAC3 Volume", AD193X_DAC_L3_VOL, + AD193X_DAC_R3_VOL, 0, 0xFF, 1, adau193x_tlv), + SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, + AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), /* ADC switch control */ SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, -- cgit v1.2.1 From c4e7a4a2768aad0bb83988922a164b4a96393713 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:08 +0100 Subject: ASoC: ad193x: Make enum items const char * const Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 7da7e29753b6..665af5cb257d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -30,7 +30,7 @@ struct ad193x_priv { /* * AD193X volume/mute/de-emphasis etc. controls */ -static const char *ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; +static const char * const ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; static const struct soc_enum ad193x_deemp_enum = SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp); -- cgit v1.2.1 From b21990b47d799152f5039c2873c38622fa7ae0f2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:09 +0100 Subject: ASoC: ad193x: Remove non-functional DAPM route controls DAPM route controls only take effect on paths where the sink is a mixer or a mux, furthermore the control must be a control assigned to the mixer or mux. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 665af5cb257d..c52ebd389c3d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -90,12 +90,12 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "ADC", NULL, "PLL_PWR" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", "DAC1 Switch", "DAC" }, - { "DAC2OUT", "DAC2 Switch", "DAC" }, - { "DAC3OUT", "DAC3 Switch", "DAC" }, - { "DAC4OUT", "DAC4 Switch", "DAC" }, - { "ADC", "ADC1 Switch", "ADC1IN" }, - { "ADC", "ADC2 Switch", "ADC2IN" }, + { "DAC1OUT", NULL, "DAC" }, + { "DAC2OUT", NULL, "DAC" }, + { "DAC3OUT", NULL, "DAC" }, + { "DAC4OUT", NULL, "DAC" }, + { "ADC", NULL, "ADC1IN" }, + { "ADC", NULL, "ADC2IN" }, }; /* -- cgit v1.2.1 From 0718fd27775fcc335c728cfa4965ce78c0662b67 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:10 +0100 Subject: ASoC: ad193x: Add sysclk DAPM supply Add a DAPM supply widget for the internal sysclk, so it can be disabled automatically when not needed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c52ebd389c3d..c19e2232f10d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -77,6 +77,7 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), @@ -86,8 +87,8 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { }; static const struct snd_soc_dapm_route audio_paths[] = { - { "DAC", NULL, "PLL_PWR" }, - { "ADC", NULL, "PLL_PWR" }, + { "DAC", NULL, "SYSCLK" }, + { "ADC", NULL, "SYSCLK" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC" }, @@ -96,6 +97,7 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC4OUT", NULL, "DAC" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, + { "SYSCLK", NULL, "PLL_PWR" }, }; /* -- cgit v1.2.1 From b82ca578fd8b28d9600a077f4e24e22a71383fe8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:11 +0100 Subject: ASoC: ad193x: Use snd_soc_update_bits where appropriate We can reduce the code size here a bit by using snd_soc_update_bits instead of open-coding the read-modify-write cycle. The conversion done in this patch is not completely straightforward and some minor code restructuring has been incorporated to further reduce the code size. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 96 ++++++++++++++++++----------------------------- sound/soc/codecs/ad193x.h | 17 +++++---- 2 files changed, 45 insertions(+), 68 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c19e2232f10d..7d64f2021b06 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -123,35 +123,29 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { struct snd_soc_codec *codec = dai->codec; - int dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1); - int adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2); - - dac_reg &= ~AD193X_DAC_CHAN_MASK; - adc_reg &= ~AD193X_ADC_CHAN_MASK; + unsigned int channels; switch (slots) { case 2: - dac_reg |= AD193X_DAC_2_CHANNELS << AD193X_DAC_CHAN_SHFT; - adc_reg |= AD193X_ADC_2_CHANNELS << AD193X_ADC_CHAN_SHFT; + channels = AD193X_2_CHANNELS; break; case 4: - dac_reg |= AD193X_DAC_4_CHANNELS << AD193X_DAC_CHAN_SHFT; - adc_reg |= AD193X_ADC_4_CHANNELS << AD193X_ADC_CHAN_SHFT; + channels = AD193X_4_CHANNELS; break; case 8: - dac_reg |= AD193X_DAC_8_CHANNELS << AD193X_DAC_CHAN_SHFT; - adc_reg |= AD193X_ADC_8_CHANNELS << AD193X_ADC_CHAN_SHFT; + channels = AD193X_8_CHANNELS; break; case 16: - dac_reg |= AD193X_DAC_16_CHANNELS << AD193X_DAC_CHAN_SHFT; - adc_reg |= AD193X_ADC_16_CHANNELS << AD193X_ADC_CHAN_SHFT; + channels = AD193X_16_CHANNELS; break; default: return -EINVAL; } - snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg); - snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg); + snd_soc_update_bits(codec, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, + channels << AD193X_DAC_CHAN_SHFT); + snd_soc_update_bits(codec, AD193X_ADC_CTRL2, AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -160,23 +154,19 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - int adc_reg1, adc_reg2, dac_reg; - - adc_reg1 = snd_soc_read(codec, AD193X_ADC_CTRL1); - adc_reg2 = snd_soc_read(codec, AD193X_ADC_CTRL2); - dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1); + unsigned int adc_serfmt = 0; + unsigned int adc_fmt = 0; + unsigned int dac_fmt = 0; /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - adc_reg1 &= ~AD193X_ADC_SERFMT_MASK; - adc_reg1 |= AD193X_ADC_SERFMT_TDM; + adc_serfmt |= AD193X_ADC_SERFMT_TDM; break; case SND_SOC_DAIFMT_DSP_A: - adc_reg1 &= ~AD193X_ADC_SERFMT_MASK; - adc_reg1 |= AD193X_ADC_SERFMT_AUX; + adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: return -EINVAL; @@ -184,29 +174,20 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ - adc_reg2 &= ~AD193X_ADC_LEFT_HIGH; - adc_reg2 &= ~AD193X_ADC_BCLK_INV; - dac_reg &= ~AD193X_DAC_LEFT_HIGH; - dac_reg &= ~AD193X_DAC_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */ - adc_reg2 |= AD193X_ADC_LEFT_HIGH; - adc_reg2 &= ~AD193X_ADC_BCLK_INV; - dac_reg |= AD193X_DAC_LEFT_HIGH; - dac_reg &= ~AD193X_DAC_BCLK_INV; + adc_fmt |= AD193X_ADC_LEFT_HIGH; + dac_fmt |= AD193X_DAC_LEFT_HIGH; break; case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */ - adc_reg2 &= ~AD193X_ADC_LEFT_HIGH; - adc_reg2 |= AD193X_ADC_BCLK_INV; - dac_reg &= ~AD193X_DAC_LEFT_HIGH; - dac_reg |= AD193X_DAC_BCLK_INV; + adc_fmt |= AD193X_ADC_BCLK_INV; + dac_fmt |= AD193X_DAC_BCLK_INV; break; - case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */ - adc_reg2 |= AD193X_ADC_LEFT_HIGH; - adc_reg2 |= AD193X_ADC_BCLK_INV; - dac_reg |= AD193X_DAC_LEFT_HIGH; - dac_reg |= AD193X_DAC_BCLK_INV; + adc_fmt |= AD193X_ADC_LEFT_HIGH; + adc_fmt |= AD193X_ADC_BCLK_INV; + dac_fmt |= AD193X_DAC_LEFT_HIGH; + dac_fmt |= AD193X_DAC_BCLK_INV; break; default: return -EINVAL; @@ -214,36 +195,31 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ - adc_reg2 |= AD193X_ADC_LCR_MASTER; - adc_reg2 |= AD193X_ADC_BCLK_MASTER; - dac_reg |= AD193X_DAC_LCR_MASTER; - dac_reg |= AD193X_DAC_BCLK_MASTER; + adc_fmt |= AD193X_ADC_LCR_MASTER; + adc_fmt |= AD193X_ADC_BCLK_MASTER; + dac_fmt |= AD193X_DAC_LCR_MASTER; + dac_fmt |= AD193X_DAC_BCLK_MASTER; break; case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ - adc_reg2 |= AD193X_ADC_LCR_MASTER; - adc_reg2 &= ~AD193X_ADC_BCLK_MASTER; - dac_reg |= AD193X_DAC_LCR_MASTER; - dac_reg &= ~AD193X_DAC_BCLK_MASTER; + adc_fmt |= AD193X_ADC_LCR_MASTER; + dac_fmt |= AD193X_DAC_LCR_MASTER; break; case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ - adc_reg2 &= ~AD193X_ADC_LCR_MASTER; - adc_reg2 |= AD193X_ADC_BCLK_MASTER; - dac_reg &= ~AD193X_DAC_LCR_MASTER; - dac_reg |= AD193X_DAC_BCLK_MASTER; + adc_fmt |= AD193X_ADC_BCLK_MASTER; + dac_fmt |= AD193X_DAC_BCLK_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ - adc_reg2 &= ~AD193X_ADC_LCR_MASTER; - adc_reg2 &= ~AD193X_ADC_BCLK_MASTER; - dac_reg &= ~AD193X_DAC_LCR_MASTER; - dac_reg &= ~AD193X_DAC_BCLK_MASTER; break; default: return -EINVAL; } - snd_soc_write(codec, AD193X_ADC_CTRL1, adc_reg1); - snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg2); - snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg); + snd_soc_update_bits(codec, AD193X_ADC_CTRL1, AD193X_ADC_SERFMT_MASK, + adc_serfmt); + snd_soc_update_bits(codec, AD193X_ADC_CTRL2, AD193X_ADC_FMT_MASK, + adc_fmt); + snd_soc_update_bits(codec, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, + dac_fmt); return 0; } diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 1507eaa425a3..473388049992 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -23,16 +23,14 @@ #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) #define AD193X_DAC_CTRL1 0x03 -#define AD193X_DAC_2_CHANNELS 0 -#define AD193X_DAC_4_CHANNELS 1 -#define AD193X_DAC_8_CHANNELS 2 -#define AD193X_DAC_16_CHANNELS 3 #define AD193X_DAC_CHAN_SHFT 1 #define AD193X_DAC_CHAN_MASK (3 << AD193X_DAC_CHAN_SHFT) #define AD193X_DAC_LCR_MASTER (1 << 4) #define AD193X_DAC_BCLK_MASTER (1 << 5) #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) +#define AD193X_DAC_FMT_MASK (AD193X_DAC_LCR_MASTER | \ + AD193X_DAC_BCLK_MASTER | AD193X_DAC_LEFT_HIGH | AD193X_DAC_BCLK_INV) #define AD193X_DAC_CTRL2 0x04 #define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 @@ -68,16 +66,19 @@ #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x10 -#define AD193X_ADC_2_CHANNELS 0 -#define AD193X_ADC_4_CHANNELS 1 -#define AD193X_ADC_8_CHANNELS 2 -#define AD193X_ADC_16_CHANNELS 3 #define AD193X_ADC_CHAN_SHFT 4 #define AD193X_ADC_CHAN_MASK (3 << AD193X_ADC_CHAN_SHFT) #define AD193X_ADC_LCR_MASTER (1 << 3) #define AD193X_ADC_BCLK_MASTER (1 << 6) #define AD193X_ADC_LEFT_HIGH (1 << 2) #define AD193X_ADC_BCLK_INV (1 << 1) +#define AD193X_ADC_FMT_MASK (AD193X_ADC_LCR_MASTER | \ + AD193X_ADC_BCLK_MASTER | AD193X_ADC_LEFT_HIGH | AD193X_ADC_BCLK_INV) + +#define AD193X_2_CHANNELS 0 +#define AD193X_4_CHANNELS 1 +#define AD193X_8_CHANNELS 2 +#define AD193X_16_CHANNELS 3 #define AD193X_NUM_REGS 17 -- cgit v1.2.1 From 34cbe16833a1840d6cde592123335fb3ad75b5d4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 17:28:12 +0100 Subject: ASoC: ad193x: Convert to direct regmap API usage Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 61 ++++++++++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 7d64f2021b06..c1b7d928c347 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -106,14 +106,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { static int ad193x_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(dai->codec); if (mute) - snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL2, AD193X_DAC_MASTER_MUTE, AD193X_DAC_MASTER_MUTE); else - snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL2, AD193X_DAC_MASTER_MUTE, 0); return 0; @@ -122,7 +122,7 @@ static int ad193x_mute(struct snd_soc_dai *dai, int mute) static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { - struct snd_soc_codec *codec = dai->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(dai->codec); unsigned int channels; switch (slots) { @@ -142,10 +142,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return -EINVAL; } - snd_soc_update_bits(codec, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, - channels << AD193X_DAC_CHAN_SHFT); - snd_soc_update_bits(codec, AD193X_ADC_CTRL2, AD193X_ADC_CHAN_MASK, - channels << AD193X_ADC_CHAN_SHFT); + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, + AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -153,7 +153,7 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec_dai->codec); unsigned int adc_serfmt = 0; unsigned int adc_fmt = 0; unsigned int dac_fmt = 0; @@ -214,12 +214,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - snd_soc_update_bits(codec, AD193X_ADC_CTRL1, AD193X_ADC_SERFMT_MASK, - adc_serfmt); - snd_soc_update_bits(codec, AD193X_ADC_CTRL2, AD193X_ADC_FMT_MASK, - adc_fmt); - snd_soc_update_bits(codec, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, - dac_fmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, + AD193X_DAC_FMT_MASK, dac_fmt); return 0; } @@ -279,14 +279,14 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } - snd_soc_update_bits(codec, AD193X_PLL_CLK_CTRL0, + regmap_update_bits(ad193x->regmap, AD193X_PLL_CLK_CTRL0, AD193X_PLL_INPUT_MASK, master_rate); - snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL2, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - snd_soc_update_bits(codec, AD193X_ADC_CTRL1, + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, AD193X_ADC_WORD_LEN_MASK, word_len); return 0; @@ -337,18 +337,18 @@ static int ad193x_probe(struct snd_soc_codec *codec) /* default setting for ad193x */ /* unmute dac channels */ - snd_soc_write(codec, AD193X_DAC_CHNL_MUTE, 0x0); + regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ - snd_soc_write(codec, AD193X_DAC_CTRL2, 0x1A); + regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* powerdown dac, dac in tdm mode */ - snd_soc_write(codec, AD193X_DAC_CTRL0, 0x41); + regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x41); /* high-pass filter enable */ - snd_soc_write(codec, AD193X_ADC_CTRL0, 0x3); + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ - snd_soc_write(codec, AD193X_ADC_CTRL1, 0x43); + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); /* pll input: mclki/xi */ - snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ - snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); + regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ + regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); return ret; } @@ -363,6 +363,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .num_dapm_routes = ARRAY_SIZE(audio_paths), }; +static bool adau193x_reg_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + #if defined(CONFIG_SPI_MASTER) static const struct regmap_config ad193x_spi_regmap_config = { @@ -370,6 +375,9 @@ static const struct regmap_config ad193x_spi_regmap_config = { .reg_bits = 16, .read_flag_mask = 0x09, .write_flag_mask = 0x08, + + .max_register = AD193X_NUM_REGS - 1, + .volatile_reg = adau193x_reg_volatile, }; static int __devinit ad193x_spi_probe(struct spi_device *spi) @@ -429,6 +437,9 @@ static struct spi_driver ad193x_spi_driver = { static const struct regmap_config ad193x_i2c_regmap_config = { .val_bits = 8, .reg_bits = 8, + + .max_register = AD193X_NUM_REGS - 1, + .volatile_reg = adau193x_reg_volatile, }; static const struct i2c_device_id ad193x_id[] = { -- cgit v1.2.1 From 0a590b1de28813c81effa2c291f24ef1f47444e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Nov 2011 22:05:41 +0000 Subject: ASoC: Add basic 1277-EV1 Littlemill audio driver The Littlemill audio card supports a number of pluggable miniboards, normally for the WM8994 family of devices. As all these devices look mostly the same from an external configuration point of view and are runtime enumerable we can write a standard machine driver which will work out of the box with any of them. Start doing that with the bare bones of a driver, only supporting AIF1. Future patches will flesh this out to be more fully featured. Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 6 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/littlemill.c | 227 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 235 insertions(+) create mode 100644 sound/soc/samsung/littlemill.c (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 71f38de18222..7aaaf8e8056f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -205,3 +205,9 @@ config SND_SOC_LOWLAND select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 + +config SND_SOC_LITTLEMILL + tristate "Audio support for Wolfson Littlemill" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM8994 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 7802c25db775..c9564e3547bb 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -41,6 +41,7 @@ snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-speyside-wm8962-objs := speyside_wm8962.o snd-soc-lowland-objs := lowland.o +snd-soc-littlemill-objs := littlemill.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -63,3 +64,4 @@ obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o +obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c new file mode 100644 index 000000000000..d2a44ab3c207 --- /dev/null +++ b/sound/soc/samsung/littlemill.c @@ -0,0 +1,227 @@ +/* + * Littlemill audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include + +#include "../codecs/wm8994.h" + +static int sample_rate = 44100; + +static int littlemill_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + /* + * If we've not already clocked things via hw_params() + * then do so now, otherwise these are noops. + */ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + WM8994_FLL_SRC_MCLK2, 32768, + sample_rate * 512); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL1, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int littlemill_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int littlemill_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + sample_rate = params_rate(params); + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + WM8994_FLL_SRC_MCLK2, 32768, + sample_rate * 512); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL1, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_ops littlemill_ops = { + .hw_params = littlemill_hw_params, +}; + +static struct snd_soc_dai_link littlemill_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &littlemill_ops, + }, +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUT1L" }, + { "Headphone", NULL, "HPOUT1R" }, +}; + +static int littlemill_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_card littlemill = { + .name = "Littlemill", + .dai_link = littlemill_dai, + .num_links = ARRAY_SIZE(littlemill_dai), + + .set_bias_level = littlemill_set_bias_level, + .set_bias_level_post = littlemill_set_bias_level_post, + + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + + .late_probe = littlemill_late_probe, +}; + +static __devinit int littlemill_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &littlemill; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit littlemill_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver littlemill_driver = { + .driver = { + .name = "littlemill", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = littlemill_probe, + .remove = __devexit_p(littlemill_remove), +}; + +module_platform_driver(littlemill_driver); + +MODULE_DESCRIPTION("Littlemill audio support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:littlemill"); -- cgit v1.2.1 From af3c2621a9b4d22b8927b91bc9cc02a13087e12b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 28 Nov 2011 18:55:03 +0800 Subject: ASoC: Convert tegra_spdif to use module_platform_driver() Use the module_platform_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_spdif.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index ea9c92036aa1..475428cf270e 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -352,17 +352,7 @@ static struct platform_driver tegra_spdif_driver = { .remove = __devexit_p(tegra_spdif_platform_remove), }; -static int __init snd_tegra_spdif_init(void) -{ - return platform_driver_register(&tegra_spdif_driver); -} -module_init(snd_tegra_spdif_init); - -static void __exit snd_tegra_spdif_exit(void) -{ - platform_driver_unregister(&tegra_spdif_driver); -} -module_exit(snd_tegra_spdif_exit); +module_platform_driver(tegra_spdif_driver); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); -- cgit v1.2.1 From cc0b401ad87e830843d3034f892c4017f9837fae Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Mon, 28 Nov 2011 15:49:31 -0600 Subject: ASoC: Convert CS42L73 to devm_kzalloc() Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d09578f397da..9fd5de77cafb 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1339,7 +1339,8 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, unsigned int devid = 0; unsigned int reg; - cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), + GFP_KERNEL); if (!cs42l73) { dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; @@ -1394,8 +1395,6 @@ err_regmap: regmap_exit(cs42l73->regmap); err: - kfree(cs42l73); - return ret; } @@ -1406,7 +1405,6 @@ static __devexit int cs42l73_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(cs42l73->regmap); - kfree(cs42l73); return 0; } -- cgit v1.2.1 From 1175f71197140dfdb8ad31767030175d88cbea2b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 28 Nov 2011 18:53:57 +0800 Subject: ASoC: Convert smdk_wm8994pcm to use module_platform_driver() Use the module_platform_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994pcm.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index da9c2a264d93..23c7fb71ddfa 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -158,19 +158,7 @@ static struct platform_driver snd_smdk_driver = { .remove = __devexit_p(snd_smdk_remove), }; -static int __init smdk_audio_init(void) -{ - return platform_driver_register(&snd_smdk_driver); -} - -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_driver_unregister(&snd_smdk_driver); -} - -module_exit(smdk_audio_exit); +module_platform_driver(snd_smdk_driver); MODULE_AUTHOR("Sangbeom Kim, "); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); -- cgit v1.2.1 From 7b282cbbf3c7bbad20505761a9eadd6d9a7280c7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 29 Nov 2011 19:47:38 +0800 Subject: ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt What we want is to clear BIT[5:4](PCM_MODE_MASK) and BIT[3](PCM_BIT_ORDER) bits, but current code clears BIT[2:0]. Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9fd5de77cafb..da3125aa55f9 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1024,7 +1024,8 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } if (spc & SPDIF_PCM) { - spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ + spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) -- cgit v1.2.1 From 40216ce7aa88c2e70869723a0f5929fdbd4a91c5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 09:44:17 +0100 Subject: ASoC: Move SigmaDSP firmware loader to ASoC It has been pointed out previously, that the firmware subsystem is not the right place for the SigmaDSP firmware loader. Furthermore the SigmaDSP is currently only used in audio products and we are aiming for better integration into the ASoC framework in the future, with support for ALSA controls for firmware parameters and support dynamic power management as well. So the natural choice for the SigmaDSP firmware loader is the ASoC subsystem. Signed-off-by: Lars-Peter Clausen Acked-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/sigmadsp.c | 154 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sigmadsp.h | 55 ++++++++++++++++ 5 files changed, 217 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp.c create mode 100644 sound/soc/codecs/sigmadsp.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 686f45a07f34..593174c78d7b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,7 @@ config SND_SOC_AD73311 tristate config SND_SOC_ADAU1701 - select SIGMA + select SND_SOC_SIGMADSP tristate config SND_SOC_ADAU1373 @@ -234,6 +234,10 @@ config SND_SOC_RT5631 config SND_SOC_SGTL5000 tristate +config SND_SOC_SIGMADSP + tristate + select CRC32 + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 62b01e4e7983..fa15006fcac5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -33,6 +33,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o +snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -134,6 +135,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o +obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8b7e1c50d6e9..6a6af567f02a 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -12,13 +12,13 @@ #include #include #include -#include #include #include #include #include #include +#include "sigmadsp.h" #include "adau1701.h" #define ADAU1701_DSPCTRL 0x1c diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c new file mode 100644 index 000000000000..acb97a9834aa --- /dev/null +++ b/sound/soc/codecs/sigmadsp.c @@ -0,0 +1,154 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include +#include +#include +#include + +#include "sigmadsp.h" + +static size_t sigma_action_size(struct sigma_action *sa) +{ + size_t payload = 0; + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + payload = sigma_action_len(sa); + break; + default: + break; + } + + payload = ALIGN(payload, 2); + + return payload + sizeof(struct sigma_action); +} + +/* + * Returns a negative error value in case of an error, 0 if processing of + * the firmware should be stopped after this action, 1 otherwise. + */ +static int +process_sigma_action(struct i2c_client *client, struct sigma_action *sa) +{ + size_t len = sigma_action_len(sa); + int ret; + + pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, + sa->instr, sa->addr, len); + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + ret = i2c_master_send(client, (void *)&sa->addr, len); + if (ret < 0) + return -EINVAL; + break; + case SIGMA_ACTION_DELAY: + udelay(len); + len = 0; + break; + case SIGMA_ACTION_END: + return 0; + default: + return -EINVAL; + } + + return 1; +} + +static int +process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) +{ + struct sigma_action *sa; + size_t size; + int ret; + + while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { + sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + + size = sigma_action_size(sa); + ssfw->pos += size; + if (ssfw->pos > ssfw->fw->size || size == 0) + break; + + ret = process_sigma_action(client, sa); + + pr_debug("%s: action returned %i\n", __func__, ret); + + if (ret <= 0) + return ret; + } + + if (ssfw->pos != ssfw->fw->size) + return -EINVAL; + + return 0; +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + int ret; + struct sigma_firmware_header *ssfw_head; + struct sigma_firmware ssfw; + const struct firmware *fw; + u32 crc; + + pr_debug("%s: loading firmware %s\n", __func__, name); + + /* first load the blob */ + ret = request_firmware(&fw, name, &client->dev); + if (ret) { + pr_debug("%s: request_firmware() failed with %i\n", __func__, ret); + return ret; + } + ssfw.fw = fw; + + /* then verify the header */ + ret = -EINVAL; + + /* + * Reject too small or unreasonable large files. The upper limit has been + * chosen a bit arbitrarily, but it should be enough for all practical + * purposes and having the limit makes it easier to avoid integer + * overflows later in the loading process. + */ + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) + goto done; + + ssfw_head = (void *)fw->data; + if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) + goto done; + + crc = crc32(0, fw->data + sizeof(*ssfw_head), + fw->size - sizeof(*ssfw_head)); + pr_debug("%s: crc=%x\n", __func__, crc); + if (crc != le32_to_cpu(ssfw_head->crc)) + goto done; + + ssfw.pos = sizeof(*ssfw_head); + + /* finally process all of the actions */ + ret = process_sigma_actions(client, &ssfw); + + done: + release_firmware(fw); + + pr_debug("%s: loaded %s\n", __func__, name); + + return ret; +} +EXPORT_SYMBOL(process_sigma_firmware); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h new file mode 100644 index 000000000000..d0de882c0d96 --- /dev/null +++ b/sound/soc/codecs/sigmadsp.h @@ -0,0 +1,55 @@ +/* + * Load firmware files from Analog Devices SigmaStudio + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __SIGMA_FIRMWARE_H__ +#define __SIGMA_FIRMWARE_H__ + +#include +#include + +struct i2c_client; + +#define SIGMA_MAGIC "ADISIGM" + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; +}; + +struct sigma_firmware_header { + unsigned char magic[7]; + u8 version; + __le32 crc; +}; + +enum { + SIGMA_ACTION_WRITEXBYTES = 0, + SIGMA_ACTION_WRITESINGLE, + SIGMA_ACTION_WRITESAFELOAD, + SIGMA_ACTION_DELAY, + SIGMA_ACTION_PLLWAIT, + SIGMA_ACTION_NOOP, + SIGMA_ACTION_END, +}; + +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +}; + +static inline u32 sigma_action_len(struct sigma_action *sa) +{ + return (sa->len_hi << 16) | le16_to_cpu(sa->len); +} + +extern int process_sigma_firmware(struct i2c_client *client, const char *name); + +#endif -- cgit v1.2.1 From 48afc5272eec2e1a7cf17aee0d2949810a45994a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 09:44:18 +0100 Subject: ASoC: SigmaDSP: Provide diagnostic error messages Provide some error messages when loading the firmware fails, so it is possible to diagnose the reason for the failure. Signed-off-by: Lars-Peter Clausen Acked-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index acb97a9834aa..c0ad88516f30 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -124,18 +124,25 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) * purposes and having the limit makes it easier to avoid integer * overflows later in the loading process. */ - if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) { + dev_err(&client->dev, "Failed to load firmware: Invalid size\n"); goto done; + } ssfw_head = (void *)fw->data; - if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) + if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) { + dev_err(&client->dev, "Failed to load firmware: Invalid magic\n"); goto done; + } crc = crc32(0, fw->data + sizeof(*ssfw_head), fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); - if (crc != le32_to_cpu(ssfw_head->crc)) + if (crc != le32_to_cpu(ssfw_head->crc)) { + dev_err(&client->dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n", + le32_to_cpu(ssfw_head->crc), crc); goto done; + } ssfw.pos = sizeof(*ssfw_head); -- cgit v1.2.1 From a4c1d7e66719b326431c6e617da07cab0caedbca Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 09:44:19 +0100 Subject: ASoC: SigmaDSP: Move private structs and functions to C file Move the structs and functions only used by SigmaDSP firmware loader itself from the header to the C file. Signed-off-by: Lars-Peter Clausen Acked-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sigmadsp.h | 39 --------------------------------------- 2 files changed, 36 insertions(+), 39 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index c0ad88516f30..aa223c56b2b6 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -15,6 +15,42 @@ #include "sigmadsp.h" +#define SIGMA_MAGIC "ADISIGM" + +struct sigma_firmware_header { + unsigned char magic[7]; + u8 version; + __le32 crc; +} __packed; + +enum { + SIGMA_ACTION_WRITEXBYTES = 0, + SIGMA_ACTION_WRITESINGLE, + SIGMA_ACTION_WRITESAFELOAD, + SIGMA_ACTION_DELAY, + SIGMA_ACTION_PLLWAIT, + SIGMA_ACTION_NOOP, + SIGMA_ACTION_END, +}; + +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; +}; + +static inline u32 sigma_action_len(struct sigma_action *sa) +{ + return (sa->len_hi << 16) | le16_to_cpu(sa->len); +} + static size_t sigma_action_size(struct sigma_action *sa) { size_t payload = 0; diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index d0de882c0d96..99a609157b2e 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -9,47 +9,8 @@ #ifndef __SIGMA_FIRMWARE_H__ #define __SIGMA_FIRMWARE_H__ -#include -#include - struct i2c_client; -#define SIGMA_MAGIC "ADISIGM" - -struct sigma_firmware { - const struct firmware *fw; - size_t pos; -}; - -struct sigma_firmware_header { - unsigned char magic[7]; - u8 version; - __le32 crc; -}; - -enum { - SIGMA_ACTION_WRITEXBYTES = 0, - SIGMA_ACTION_WRITESINGLE, - SIGMA_ACTION_WRITESAFELOAD, - SIGMA_ACTION_DELAY, - SIGMA_ACTION_PLLWAIT, - SIGMA_ACTION_NOOP, - SIGMA_ACTION_END, -}; - -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -}; - -static inline u32 sigma_action_len(struct sigma_action *sa) -{ - return (sa->len_hi << 16) | le16_to_cpu(sa->len); -} - extern int process_sigma_firmware(struct i2c_client *client, const char *name); #endif -- cgit v1.2.1 From 38fd54ee38624a52c28d65fadfd452c9c49fb152 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 28 Nov 2011 09:44:20 +0100 Subject: ASoC: SigmaDSP: Add regmap support Add support for loading the SigmaDSP firmware using regmap. This allows us to transparently use SPI or I2C as the transport protocol on devices which support them. For now we keep the old I2C support since we have one user of this which is not straight forward to convert to regmap, due to variable length registers. Signed-off-by: Lars-Peter Clausen Acked-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 75 +++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/sigmadsp.h | 5 +++ 2 files changed, 67 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index aa223c56b2b6..5be42bf56996 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -11,6 +11,7 @@ #include #include #include +#include #include #include "sigmadsp.h" @@ -44,6 +45,10 @@ struct sigma_action { struct sigma_firmware { const struct firmware *fw; size_t pos; + + void *control_data; + int (*write)(void *control_data, const struct sigma_action *sa, + size_t len); }; static inline u32 sigma_action_len(struct sigma_action *sa) @@ -75,7 +80,7 @@ static size_t sigma_action_size(struct sigma_action *sa) * the firmware should be stopped after this action, 1 otherwise. */ static int -process_sigma_action(struct i2c_client *client, struct sigma_action *sa) +process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa) { size_t len = sigma_action_len(sa); int ret; @@ -87,7 +92,7 @@ process_sigma_action(struct i2c_client *client, struct sigma_action *sa) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - ret = i2c_master_send(client, (void *)&sa->addr, len); + ret = ssfw->write(ssfw->control_data, sa, len); if (ret < 0) return -EINVAL; break; @@ -105,7 +110,7 @@ process_sigma_action(struct i2c_client *client, struct sigma_action *sa) } static int -process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) +process_sigma_actions(struct sigma_firmware *ssfw) { struct sigma_action *sa; size_t size; @@ -119,7 +124,7 @@ process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) if (ssfw->pos > ssfw->fw->size || size == 0) break; - ret = process_sigma_action(client, sa); + ret = process_sigma_action(ssfw, sa); pr_debug("%s: action returned %i\n", __func__, ret); @@ -133,23 +138,23 @@ process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) return 0; } -int process_sigma_firmware(struct i2c_client *client, const char *name) +static int _process_sigma_firmware(struct device *dev, + struct sigma_firmware *ssfw, const char *name) { int ret; struct sigma_firmware_header *ssfw_head; - struct sigma_firmware ssfw; const struct firmware *fw; u32 crc; pr_debug("%s: loading firmware %s\n", __func__, name); /* first load the blob */ - ret = request_firmware(&fw, name, &client->dev); + ret = request_firmware(&fw, name, dev); if (ret) { pr_debug("%s: request_firmware() failed with %i\n", __func__, ret); return ret; } - ssfw.fw = fw; + ssfw->fw = fw; /* then verify the header */ ret = -EINVAL; @@ -161,13 +166,13 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) * overflows later in the loading process. */ if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) { - dev_err(&client->dev, "Failed to load firmware: Invalid size\n"); + dev_err(dev, "Failed to load firmware: Invalid size\n"); goto done; } ssfw_head = (void *)fw->data; if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) { - dev_err(&client->dev, "Failed to load firmware: Invalid magic\n"); + dev_err(dev, "Failed to load firmware: Invalid magic\n"); goto done; } @@ -175,15 +180,15 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); if (crc != le32_to_cpu(ssfw_head->crc)) { - dev_err(&client->dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n", + dev_err(dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n", le32_to_cpu(ssfw_head->crc), crc); goto done; } - ssfw.pos = sizeof(*ssfw_head); + ssfw->pos = sizeof(*ssfw_head); /* finally process all of the actions */ - ret = process_sigma_actions(client, &ssfw); + ret = process_sigma_actions(ssfw); done: release_firmware(fw); @@ -192,6 +197,50 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) return ret; } + +#if IS_ENABLED(CONFIG_I2C) + +static int sigma_action_write_i2c(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return i2c_master_send(control_data, (const unsigned char *)&sa->addr, + len); +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = client; + ssfw.write = sigma_action_write_i2c; + + return _process_sigma_firmware(&client->dev, &ssfw, name); +} EXPORT_SYMBOL(process_sigma_firmware); +#endif + +#if IS_ENABLED(CONFIG_REGMAP) + +static int sigma_action_write_regmap(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + sa->payload, len - 2); +} + +int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, + const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = regmap; + ssfw.write = sigma_action_write_regmap; + + return _process_sigma_firmware(dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware_regmap); + +#endif + MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index 99a609157b2e..e439cbd7af7d 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -9,8 +9,13 @@ #ifndef __SIGMA_FIRMWARE_H__ #define __SIGMA_FIRMWARE_H__ +#include +#include + struct i2c_client; extern int process_sigma_firmware(struct i2c_client *client, const char *name); +extern int process_sigma_firmware_regmap(struct device *dev, + struct regmap *regmap, const char *name); #endif -- cgit v1.2.1 From 4cdf5e49ce8ff79038ee5388cc5f97097238bb29 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Nov 2011 14:36:17 +0000 Subject: ASoC: Ensure SYSCLK is enabled for WM8958 accessory detection Ensure SYSCLK is enabled while running accessory detection on WM8958. It is always required so there is no sense in requiring machine drivers to individually do this. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5ea0c3c15254..0a16de743dd8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3016,6 +3016,8 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, cb_data = codec; } + snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS"); + wm8994->micdet[0].jack = jack; wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; @@ -3025,6 +3027,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } else { snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); + snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS"); } return 0; -- cgit v1.2.1 From 9b8f5695a155308a4e0355a29747961bec9757c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 21:35:40 +0000 Subject: ASoC: Fix __iomem annotation for IDMA registers We always store the register address as __iomem but pass it around as a plain void * which upsets sparse. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/idma.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fb80f2886c70..5de500ce5dd4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -881,7 +881,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) writel(CON_RSTCLR, i2s->addr + I2SCON); if (i2s->quirks & QUIRK_SEC_DAI) - idma_reg_addr_init((void *)i2s->addr, + idma_reg_addr_init(i2s->addr, i2s->sec_dai->idma_playback.dma_addr); probe_exit: diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 6ca3d8c221a0..baf97ebadd48 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -403,7 +403,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) return ret; } -void idma_reg_addr_init(void *regs, dma_addr_t addr) +void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) { spin_lock_init(&idma.lock); idma.regs = regs; diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h index 48273216166e..8644946973e5 100644 --- a/sound/soc/samsung/idma.h +++ b/sound/soc/samsung/idma.h @@ -14,7 +14,7 @@ #ifndef __SND_SOC_SAMSUNG_IDMA_H_ #define __SND_SOC_SAMSUNG_IDMA_H_ -extern void idma_reg_addr_init(void *regs, dma_addr_t addr); +extern void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr); /* dma_state */ #define LPAM_DMA_STOP 0 -- cgit v1.2.1 From 500fa30ed5795a1d8e8539d0cd81f73b34f831a3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Nov 2011 19:58:19 +0000 Subject: ASoC: Put WM8958 and WM1811 MICBIAS into bypass mode when no audio When we don't have any active audio we can put the microphone biases into bypass mode to save power at the expense of performance. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0a16de743dd8..207bccd156f1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2025,6 +2025,18 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: + /* MICBIAS into regulating mode */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, 0); + break; + default: + break; + } break; case SND_SOC_BIAS_STANDBY: @@ -2077,7 +2089,20 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } - + /* MICBIAS into bypass mode on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, + WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, + WM8958_MICB2_MODE); + break; + default: + break; + } break; case SND_SOC_BIAS_OFF: @@ -3371,6 +3396,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + /* Put MICBIAS into bypass mode by default on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, WM8958_MICB2_MODE); + break; + default: + break; + } + wm8994_update_class_w(codec); wm8994_handle_pdata(wm8994); -- cgit v1.2.1 From b00adf76a6fa492c39f8225fc42debc01bbbdc1d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 13 Aug 2011 11:57:18 +0900 Subject: ASoC: Enhance default WM8958 microphone detection Actively manage the detection rate for microphones with WM8958, providing improved power consumption and maximising the benefit from the hardware debounce. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 120 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm8994.h | 2 + 2 files changed, 111 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 207bccd156f1..027bf683efce 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -53,6 +53,56 @@ static int wm8994_retune_mobile_base[] = { WM8994_AIF2_EQ_GAINS_1, }; +static void wm8958_default_micdet(u16 status, void *data); + +static const struct { + int sysclk; + bool idle; + int start; + int rate; +} wm8958_micd_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 6 }, + { 44100 * 256, false, 7, 6 }, +}; + +static void wm8958_micd_set_rate(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int best, i, sysclk, val; + bool idle; + + if (wm8994->jack_cb != wm8958_default_micdet) + return; + + idle = !wm8994->jack_mic; + + sysclk = snd_soc_read(codec, WM8994_CLOCKING_1); + if (sysclk & WM8994_SYSCLK_SRC) + sysclk = wm8994->aifclk[1]; + else + sysclk = wm8994->aifclk[0]; + + best = 0; + for (i = 0; i < ARRAY_SIZE(wm8958_micd_rates); i++) { + if (wm8958_micd_rates[i].idle != idle) + continue; + if (abs(wm8958_micd_rates[i].sysclk - sysclk) < + abs(wm8958_micd_rates[best].sysclk - sysclk)) + best = i; + else if (wm8958_micd_rates[best].idle != idle) + best = i; + } + + val = wm8958_micd_rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT + | wm8958_micd_rates[best].rate << WM8958_MICD_RATE_SHIFT; + + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_BIAS_STARTTIME_MASK | + WM8958_MICD_RATE_MASK, val); +} + static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -221,8 +271,10 @@ static int configure_clock(struct snd_soc_codec *codec) */ /* If they're equal it doesn't matter which is used */ - if (wm8994->aifclk[0] == wm8994->aifclk[1]) + if (wm8994->aifclk[0] == wm8994->aifclk[1]) { + wm8958_micd_set_rate(codec); return 0; + } if (wm8994->aifclk[0] < wm8994->aifclk[1]) new = WM8994_SYSCLK_SRC; @@ -236,6 +288,8 @@ static int configure_clock(struct snd_soc_codec *codec) snd_soc_dapm_sync(&codec->dapm); + wm8958_micd_set_rate(codec); + return 0; } @@ -2987,21 +3041,56 @@ static void wm8958_default_micdet(u16 status, void *data) { struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int report = 0; /* If nothing present then clear our statuses */ - if (!(status & WM8958_MICD_STS)) - goto done; + if (!(status & WM8958_MICD_STS)) { + dev_dbg(codec->dev, "Detected open circuit\n"); + wm8994->jack_mic = false; + wm8994->detecting = true; + + wm8958_micd_set_rate(codec); - report = SND_JACK_MICROPHONE; + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_BTN_0 | SND_JACK_HEADSET); + + return; + } - /* Everything else is buttons; just assign slots */ - if (status & 0x1c) - report |= SND_JACK_BTN_0; + /* If the measurement is showing a high impedence we've got a + * microphone. + */ + if (wm8994->detecting && (status & 0x600)) { + dev_dbg(codec->dev, "Detected microphone\n"); + + wm8994->detecting = false; + wm8994->jack_mic = true; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADSET, + SND_JACK_HEADSET); + } -done: - snd_soc_jack_report(wm8994->micdet[0].jack, report, - SND_JACK_BTN_0 | SND_JACK_MICROPHONE); + + if (wm8994->detecting && status & 0x4) { + dev_dbg(codec->dev, "Detected headphone\n"); + wm8994->detecting = false; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + } + + /* Report short circuit as a button */ + if (wm8994->jack_mic) { + if (status & 0x4) + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_BTN_0, SND_JACK_BTN_0); + else + snd_soc_jack_report(wm8994->micdet[0].jack, + 0, SND_JACK_BTN_0); + } } /** @@ -3047,6 +3136,15 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; + wm8994->detecting = true; + wm8994->jack_mic = false; + + wm8958_micd_set_rate(codec); + + /* Detect microphones and short circuits */ + snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, + WM8958_MICD_LVL_SEL_MASK, 0x41); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); } else { diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f4f1355efc82..1087425cbac0 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -126,6 +126,8 @@ struct wm8994_priv { struct soc_enum enh_eq_enum; struct wm8994_micdet micdet[2]; + bool detecting; + bool jack_mic; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- cgit v1.2.1 From bf55499e6ee927e047feed85349365481289bd75 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 29 Nov 2011 18:36:48 -0700 Subject: ASoC: Tegra I2S: Add device tree binding Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index ca4d0c0a913e..33509de52540 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -36,6 +36,7 @@ #include #include #include +#include #include #include #include @@ -331,6 +332,8 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { struct tegra_i2s * i2s; struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); @@ -360,9 +363,16 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; } memregion = devm_request_mem_region(&pdev->dev, mem->start, @@ -383,12 +393,12 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dmareq->start; + i2s->capture_dma_data.req_sel = dma_ch; i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; i2s->playback_dma_data.wrap = 4; i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dmareq->start; + i2s->playback_dma_data.req_sel = dma_ch; i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; @@ -422,10 +432,16 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + static struct platform_driver tegra_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = tegra_i2s_of_match, }, .probe = tegra_i2s_platform_probe, .remove = __devexit_p(tegra_i2s_platform_remove), @@ -436,3 +452,4 @@ MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); -- cgit v1.2.1 From 6414261f0a2af00c6ffc80f847e9202344360bb4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Nov 2011 13:30:27 +0000 Subject: ASoC: Rename Speyside WM8962 to Tobermory All the other machine drivers for non-default configurations are named after the relevant audio module so do so for Tobermory also. Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/speyside_wm8962.c | 257 ------------------------------------ sound/soc/samsung/tobermory.c | 257 ++++++++++++++++++++++++++++++++++++ 4 files changed, 261 insertions(+), 261 deletions(-) delete mode 100644 sound/soc/samsung/speyside_wm8962.c create mode 100644 sound/soc/samsung/tobermory.c (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7aaaf8e8056f..09d636cc3658 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -193,8 +193,8 @@ config SND_SOC_SPEYSIDE select SND_SOC_WM9081 select SND_SOC_WM1250_EV1 -config SND_SOC_SPEYSIDE_WM8962 - tristate "Audio support for Wolfson Speyside with WM8962" +config SND_SOC_TOBERMORY + tristate "Audio support for Wolfson Tobermory" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8962 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index c9564e3547bb..9d03beb40c86 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -39,7 +39,7 @@ snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o -snd-soc-speyside-wm8962-objs := speyside_wm8962.o +snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o @@ -62,6 +62,6 @@ obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o -obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o +obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c deleted file mode 100644 index c09648efab61..000000000000 --- a/sound/soc/samsung/speyside_wm8962.c +++ /dev/null @@ -1,257 +0,0 @@ -/* - * Speyside with WM8962 audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include - -#include "../codecs/wm8962.h" - -static int sample_rate = 44100; - -static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, 32768, - sample_rate * 512); - if (ret < 0) - pr_err("Failed to start FLL: %d\n", ret); - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, - sample_rate * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - -static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - sample_rate = params_rate(params); - - return 0; -} - -static struct snd_soc_ops speyside_wm8962_ops = { - .hw_params = speyside_wm8962_hw_params, -}; - -static struct snd_soc_dai_link speyside_wm8962_dai[] = { - { - .name = "CPU", - .stream_name = "CPU", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8962", - .platform_name = "samsung-audio", - .codec_name = "wm8962.1-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, - .ops = &speyside_wm8962_ops, - }, -}; - -static const struct snd_kcontrol_new controls[] = { - SOC_DAPM_PIN_SWITCH("Main Speaker"), - SOC_DAPM_PIN_SWITCH("DMIC"), -}; - -static struct snd_soc_dapm_widget widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - - SND_SOC_DAPM_MIC("DMIC", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - - SND_SOC_DAPM_SPK("Main Speaker", NULL), -}; - -static struct snd_soc_dapm_route audio_paths[] = { - { "Headphone", NULL, "HPOUTL" }, - { "Headphone", NULL, "HPOUTR" }, - - { "Main Speaker", NULL, "SPKOUTL" }, - { "Main Speaker", NULL, "SPKOUTR" }, - - { "Headset Mic", NULL, "MICBIAS" }, - { "IN4L", NULL, "Headset Mic" }, - { "IN4R", NULL, "Headset Mic" }, - - { "AMIC", NULL, "MICBIAS" }, - { "IN1L", NULL, "AMIC" }, - { "IN1R", NULL, "AMIC" }, - - { "DMIC", NULL, "MICBIAS" }, - { "DMICDAT", NULL, "DMIC" }, -}; - -static struct snd_soc_jack speyside_wm8962_headset; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static int speyside_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &speyside_wm8962_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_wm8962_headset, - ARRAY_SIZE(speyside_wm8962_headset_pins), - speyside_wm8962_headset_pins); - if (ret) - return ret; - - wm8962_mic_detect(codec, &speyside_wm8962_headset); - - return 0; -} - -static struct snd_soc_card speyside_wm8962 = { - .name = "Speyside WM8962", - .dai_link = speyside_wm8962_dai, - .num_links = ARRAY_SIZE(speyside_wm8962_dai), - - .set_bias_level = speyside_wm8962_set_bias_level, - .set_bias_level_post = speyside_wm8962_set_bias_level_post, - - .controls = controls, - .num_controls = ARRAY_SIZE(controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = audio_paths, - .num_dapm_routes = ARRAY_SIZE(audio_paths), - .fully_routed = true, - - .late_probe = speyside_wm8962_late_probe, -}; - -static __devinit int speyside_wm8962_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &speyside_wm8962; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; - } - - return 0; -} - -static int __devexit speyside_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - -static struct platform_driver speyside_wm8962_driver = { - .driver = { - .name = "speyside-wm8962", - .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, - }, - .probe = speyside_wm8962_probe, - .remove = __devexit_p(speyside_wm8962_remove), -}; - -module_platform_driver(speyside_wm8962_driver); - -MODULE_DESCRIPTION("Speyside WM8962 audio support"); -MODULE_AUTHOR("Mark Brown "); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:speyside-wm8962"); diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c new file mode 100644 index 000000000000..6f91c65c5a0e --- /dev/null +++ b/sound/soc/samsung/tobermory.c @@ -0,0 +1,257 @@ +/* + * Tobermory audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include + +#include "../codecs/wm8962.h" + +static int sample_rate = 44100; + +static int tobermory_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 32768, + sample_rate * 512); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int tobermory_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int tobermory_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + + return 0; +} + +static struct snd_soc_ops tobermory_ops = { + .hw_params = tobermory_hw_params, +}; + +static struct snd_soc_dai_link tobermory_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8962", + .platform_name = "samsung-audio", + .codec_name = "wm8962.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &tobermory_ops, + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("DMIC"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_MIC("DMIC", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUTL" }, + { "Headphone", NULL, "HPOUTR" }, + + { "Main Speaker", NULL, "SPKOUTL" }, + { "Main Speaker", NULL, "SPKOUTR" }, + + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, + + { "AMIC", NULL, "MICBIAS" }, + { "IN1L", NULL, "AMIC" }, + { "IN1R", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, +}; + +static struct snd_soc_jack tobermory_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin tobermory_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int tobermory_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &tobermory_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&tobermory_headset, + ARRAY_SIZE(tobermory_headset_pins), + tobermory_headset_pins); + if (ret) + return ret; + + wm8962_mic_detect(codec, &tobermory_headset); + + return 0; +} + +static struct snd_soc_card tobermory = { + .name = "Tobermory", + .dai_link = tobermory_dai, + .num_links = ARRAY_SIZE(tobermory_dai), + + .set_bias_level = tobermory_set_bias_level, + .set_bias_level_post = tobermory_set_bias_level_post, + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, + + .late_probe = tobermory_late_probe, +}; + +static __devinit int tobermory_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &tobermory; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit tobermory_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver tobermory_driver = { + .driver = { + .name = "tobermory", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tobermory_probe, + .remove = __devexit_p(tobermory_remove), +}; + +module_platform_driver(tobermory_driver); + +MODULE_DESCRIPTION("Tobermory audio support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:tobermory"); -- cgit v1.2.1 From a1691343a397157dd5f67bce50435f64024a462d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Nov 2011 14:56:40 +0000 Subject: ASoC: Provide debug log of accessory status on WM8958 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 027bf683efce..16e2bd7c3cea 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3042,6 +3042,8 @@ static void wm8958_default_micdet(u16 status, void *data) struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + dev_dbg(codec->dev, "MICDET %x\n", status); + /* If nothing present then clear our statuses */ if (!(status & WM8958_MICD_STS)) { dev_dbg(codec->dev, "Detected open circuit\n"); -- cgit v1.2.1 From 2a8a856d427fea68a5d538adf52edae4cdbb246b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 Jul 2011 12:20:41 +0100 Subject: ASoC: Don't use control_data to get struct wm8994 This will support refactoring to make use of the regmap API more directly in the core. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 72 +++++++++++++++++++++++++---------------------- sound/soc/codecs/wm8994.h | 5 ++-- 2 files changed, 41 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 16e2bd7c3cea..d36b62b492c1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -106,7 +106,7 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; switch (reg) { case WM8994_GPIO_1: @@ -1822,7 +1822,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, unsigned int freq_in, unsigned int freq_out) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; @@ -2071,8 +2071,8 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, static int wm8994_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->wm8994; switch (level) { case SND_SOC_BIAS_ON: @@ -2174,7 +2174,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; int ms = 0; @@ -2474,7 +2475,8 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->wm8994; int aif1_reg; int aif1 = 0; @@ -2705,7 +2707,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int i, ret; switch (control->type) { @@ -2736,7 +2738,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int i, ret; unsigned int val, mask; @@ -2958,7 +2960,7 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_micdet *micdet; - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int reg; if (control->type != WM8994) @@ -3115,7 +3117,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micdet_cb cb, void *cb_data) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; switch (control->type) { case WM1811: @@ -3247,6 +3249,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm8994); + + wm8994->wm8994 = dev_get_drvdata(codec->dev->parent); wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; @@ -3328,14 +3332,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } - wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); - ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); if (ret == 0) @@ -3355,7 +3359,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ret); } - ret = wm8994_request_irq(codec->control_data, + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, wm8994_mic_irq, "Mic 1 short", wm8994); @@ -3364,7 +3368,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) "Failed to request Mic1 short IRQ: %d\n", ret); - ret = wm8994_request_irq(codec->control_data, + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET, wm8994_mic_irq, "Mic 2 detect", wm8994); @@ -3373,7 +3377,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) "Failed to request Mic2 detect IRQ: %d\n", ret); - ret = wm8994_request_irq(codec->control_data, + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC2_SHRT, wm8994_mic_irq, "Mic 2 short", wm8994); @@ -3400,7 +3404,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->fll_locked_irq = true; for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) { - ret = wm8994_request_irq(codec->control_data, + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, wm8994_fll_locked_irq, "FLL lock", &wm8994->fll_locked[i]); @@ -3620,19 +3624,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return 0; err_irq: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_SHRT, wm8994); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET, wm8994); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, wm8994); if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) - wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); - wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, &wm8994->hubs); - wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); - wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); - wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3641,7 +3645,7 @@ err: static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int i; wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -3649,24 +3653,24 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) pm_runtime_disable(codec->dev); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) - wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); - wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, &wm8994->hubs); - wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); - wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); - wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET, wm8994); break; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1087425cbac0..c3e71d72eb6a 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -70,10 +70,11 @@ struct wm8994_fll_config { #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 +struct wm8994; + struct wm8994_priv { struct wm_hubs_data hubs; - enum snd_soc_control_type control_type; - void *control_data; + struct wm8994 *wm8994; struct snd_soc_codec *codec; int sysclk[2]; int sysclk_rate[2]; -- cgit v1.2.1 From 604533de0f60c3be6ae99fdaf44d1d79f38b307e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Dec 2011 12:51:25 +0000 Subject: ASoC: Tune down active mode detection rate for WM8958 mic detection Saves a little power. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d36b62b492c1..45bfa09f2e45 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -63,8 +63,8 @@ static const struct { } wm8958_micd_rates[] = { { 32768, true, 1, 4 }, { 32768, false, 1, 1 }, - { 44100 * 256, true, 7, 6 }, - { 44100 * 256, false, 7, 6 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, }; static void wm8958_micd_set_rate(struct snd_soc_codec *codec) -- cgit v1.2.1 From 4585790d1cde32a5719c24412e9845e031358e08 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Nov 2011 10:55:14 +0000 Subject: ASoC: Allow more WM8958/WM1811 button levels with default handler The WM8958 and WM1811 support detecting a range of buttons. Allow the user to provide platform data enabling more of these levels without having to write a custom detection handler. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 42 ++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 35 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 45bfa09f2e45..3e52d40866d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3043,6 +3043,7 @@ static void wm8958_default_micdet(u16 status, void *data) { struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int report; dev_dbg(codec->dev, "MICDET %x\n", status); @@ -3055,7 +3056,7 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); snd_soc_jack_report(wm8994->micdet[0].jack, 0, - SND_JACK_BTN_0 | SND_JACK_HEADSET); + wm8994->btn_mask | SND_JACK_HEADSET); return; } @@ -3088,12 +3089,27 @@ static void wm8958_default_micdet(u16 status, void *data) /* Report short circuit as a button */ if (wm8994->jack_mic) { + report = 0; if (status & 0x4) - snd_soc_jack_report(wm8994->micdet[0].jack, - SND_JACK_BTN_0, SND_JACK_BTN_0); - else - snd_soc_jack_report(wm8994->micdet[0].jack, - 0, SND_JACK_BTN_0); + report |= SND_JACK_BTN_0; + + if (status & 0x8) + report |= SND_JACK_BTN_1; + + if (status & 0x10) + report |= SND_JACK_BTN_2; + + if (status & 0x20) + report |= SND_JACK_BTN_3; + + if (status & 0x40) + report |= SND_JACK_BTN_4; + + if (status & 0x80) + report |= SND_JACK_BTN_5; + + snd_soc_jack_report(wm8994->micdet[0].jack, report, + wm8994->btn_mask); } } @@ -3118,6 +3134,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + u16 micd_lvl_sel; switch (control->type) { case WM1811: @@ -3145,9 +3162,18 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micd_set_rate(codec); - /* Detect microphones and short circuits */ + /* Detect microphones and short circuits by default */ + if (wm8994->pdata->micd_lvl_sel) + micd_lvl_sel = wm8994->pdata->micd_lvl_sel; + else + micd_lvl_sel = 0x41; + + wm8994->btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5; + snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, - WM8958_MICD_LVL_SEL_MASK, 0x41); + WM8958_MICD_LVL_SEL_MASK, micd_lvl_sel); snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c3e71d72eb6a..77e3d8c9eeb8 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -129,6 +129,7 @@ struct wm8994_priv { struct wm8994_micdet micdet[2]; bool detecting; bool jack_mic; + int btn_mask; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- cgit v1.2.1 From 157a75e664f8c811c660de1d1b9abb16a1f72579 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Nov 2011 13:43:51 +0000 Subject: ASoC: Rename WM8994 detecting flag to mic_detecting More specific and avoids confusion with a following change. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 12 ++++++------ sound/soc/codecs/wm8994.h | 2 +- 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3e52d40866d2..e65745bc1003 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3051,7 +3051,7 @@ static void wm8958_default_micdet(u16 status, void *data) if (!(status & WM8958_MICD_STS)) { dev_dbg(codec->dev, "Detected open circuit\n"); wm8994->jack_mic = false; - wm8994->detecting = true; + wm8994->mic_detecting = true; wm8958_micd_set_rate(codec); @@ -3064,10 +3064,10 @@ static void wm8958_default_micdet(u16 status, void *data) /* If the measurement is showing a high impedence we've got a * microphone. */ - if (wm8994->detecting && (status & 0x600)) { + if (wm8994->mic_detecting && (status & 0x600)) { dev_dbg(codec->dev, "Detected microphone\n"); - wm8994->detecting = false; + wm8994->mic_detecting = false; wm8994->jack_mic = true; wm8958_micd_set_rate(codec); @@ -3077,9 +3077,9 @@ static void wm8958_default_micdet(u16 status, void *data) } - if (wm8994->detecting && status & 0x4) { + if (wm8994->mic_detecting && status & 0x4) { dev_dbg(codec->dev, "Detected headphone\n"); - wm8994->detecting = false; + wm8994->mic_detecting = false; wm8958_micd_set_rate(codec); @@ -3157,7 +3157,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; - wm8994->detecting = true; + wm8994->mic_detecting = true; wm8994->jack_mic = false; wm8958_micd_set_rate(codec); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 77e3d8c9eeb8..8622bc4db2fe 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -127,7 +127,7 @@ struct wm8994_priv { struct soc_enum enh_eq_enum; struct wm8994_micdet micdet[2]; - bool detecting; + bool mic_detecting; bool jack_mic; int btn_mask; -- cgit v1.2.1 From af6b6fe41c4bc9e7933d66bbbf5106e0e7e6e484 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Nov 2011 20:32:05 +0000 Subject: ASoC: Implement support for WM1811A jack detection The WM1811A features an advanced low power accessory detection subsystem which allows the device to be maintained in a very low power state while the system is idle without sacrificing any accessory detection features. Implement software support for this, automatically managing the power configuration of the device depending on the detected accessory. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 264 ++++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm8994.h | 3 + 2 files changed, 248 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e65745bc1003..2e28f472b963 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -38,6 +38,11 @@ #include "wm8994.h" #include "wm_hubs.h" +#define WM1811_JACKDET_MODE_NONE 0x0000 +#define WM1811_JACKDET_MODE_JACK 0x0100 +#define WM1811_JACKDET_MODE_MIC 0x0080 +#define WM1811_JACKDET_MODE_AUDIO 0x0180 + #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 @@ -55,23 +60,34 @@ static int wm8994_retune_mobile_base[] = { static void wm8958_default_micdet(u16 status, void *data); -static const struct { +struct wm8958_micd_rate { int sysclk; bool idle; int start; int rate; -} wm8958_micd_rates[] = { +}; + +static const struct wm8958_micd_rate micdet_rates[] = { { 32768, true, 1, 4 }, { 32768, false, 1, 1 }, { 44100 * 256, true, 7, 10 }, { 44100 * 256, false, 7, 10 }, }; +static const struct wm8958_micd_rate jackdet_rates[] = { + { 32768, true, 0, 1 }, + { 32768, false, 0, 1 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, +}; + static void wm8958_micd_set_rate(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int best, i, sysclk, val; bool idle; + const struct wm8958_micd_rate *rates; + int num_rates; if (wm8994->jack_cb != wm8958_default_micdet) return; @@ -84,19 +100,27 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) else sysclk = wm8994->aifclk[0]; + if (wm8994->jackdet) { + rates = jackdet_rates; + num_rates = ARRAY_SIZE(jackdet_rates); + } else { + rates = micdet_rates; + num_rates = ARRAY_SIZE(micdet_rates); + } + best = 0; - for (i = 0; i < ARRAY_SIZE(wm8958_micd_rates); i++) { - if (wm8958_micd_rates[i].idle != idle) + for (i = 0; i < num_rates; i++) { + if (rates[i].idle != idle) continue; - if (abs(wm8958_micd_rates[i].sysclk - sysclk) < - abs(wm8958_micd_rates[best].sysclk - sysclk)) + if (abs(rates[i].sysclk - sysclk) < + abs(rates[best].sysclk - sysclk)) best = i; - else if (wm8958_micd_rates[best].idle != idle) + else if (rates[best].idle != idle) best = i; } - val = wm8958_micd_rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT - | wm8958_micd_rates[best].rate << WM8958_MICD_RATE_SHIFT; + val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT + | rates[best].rate << WM8958_MICD_RATE_SHIFT; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_BIAS_STARTTIME_MASK | @@ -762,6 +786,74 @@ SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, mixin_boost_tlv), }; +/* We run all mode setting through a function to enforce audio mode */ +static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (wm8994->active_refcount) + mode = WM1811_JACKDET_MODE_AUDIO; + + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, mode); + + if (mode == WM1811_JACKDET_MODE_MIC) + msleep(2); +} + +static void active_reference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + mutex_lock(&wm8994->accdet_lock); + + wm8994->active_refcount++; + + dev_dbg(codec->dev, "Active refcount incremented, now %d\n", + wm8994->active_refcount); + + if (wm8994->active_refcount == 1) { + /* If we're using jack detection go into audio mode */ + if (wm8994->jackdet && wm8994->jack_cb) { + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + WM1811_JACKDET_MODE_AUDIO); + msleep(2); + } + } + + mutex_unlock(&wm8994->accdet_lock); +} + +static void active_dereference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + u16 mode; + + mutex_lock(&wm8994->accdet_lock); + + wm8994->active_refcount--; + + dev_dbg(codec->dev, "Active refcount decremented, now %d\n", + wm8994->active_refcount); + + if (wm8994->active_refcount == 0) { + /* Go into appropriate detection only mode */ + if (wm8994->jackdet && wm8994->jack_cb) { + if (wm8994->jack_mic || wm8994->mic_detecting) + mode = WM1811_JACKDET_MODE_MIC; + else + mode = WM1811_JACKDET_MODE_JACK; + + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + mode); + } + } + + mutex_unlock(&wm8994->accdet_lock); +} + static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1919,6 +2011,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (freq_out) { /* Enable VMID if we need it */ if (!was_enabled) { + active_reference(codec); + switch (control->type) { case WM8994: vmid_reference(codec); @@ -1962,6 +2056,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, default: break; } + + active_dereference(codec); } } @@ -2091,6 +2187,9 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, default: break; } + + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + active_reference(codec); break; case SND_SOC_BIAS_STANDBY: @@ -2143,6 +2242,9 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) + active_dereference(codec); + /* MICBIAS into bypass mode on newer devices */ switch (control->type) { case WM8958: @@ -2168,6 +2270,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } codec->dapm.bias_level = level; + return 0; } @@ -2715,6 +2818,9 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); break; case WM1811: + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, 0); + /* Fall through */ case WM8958: snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -2784,6 +2890,13 @@ static int wm8994_resume(struct snd_soc_codec *codec) WM8994_MICD_ENA, WM8994_MICD_ENA); break; case WM1811: + if (wm8994->jackdet && wm8994->jack_cb) { + /* Restart from idle */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + WM1811_JACKDET_MODE_JACK); + break; + } case WM8958: if (wm8994->jack_cb) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, @@ -3047,17 +3160,20 @@ static void wm8958_default_micdet(u16 status, void *data) dev_dbg(codec->dev, "MICDET %x\n", status); - /* If nothing present then clear our statuses */ + /* Either nothing present or just starting detection */ if (!(status & WM8958_MICD_STS)) { - dev_dbg(codec->dev, "Detected open circuit\n"); - wm8994->jack_mic = false; - wm8994->mic_detecting = true; + if (!wm8994->jackdet) { + /* If nothing present then clear our statuses */ + dev_dbg(codec->dev, "Detected open circuit\n"); + wm8994->jack_mic = false; + wm8994->mic_detecting = true; - wm8958_micd_set_rate(codec); - - snd_soc_jack_report(wm8994->micdet[0].jack, 0, - wm8994->btn_mask | SND_JACK_HEADSET); + wm8958_micd_set_rate(codec); + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + wm8994->btn_mask | + SND_JACK_HEADSET); + } return; } @@ -3085,6 +3201,15 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + + /* If we have jackdet that will detect removal */ + if (wm8994->jackdet) { + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); + } } /* Report short circuit as a button */ @@ -3113,6 +3238,56 @@ static void wm8958_default_micdet(u16 status, void *data) } } +static irqreturn_t wm1811_jackdet_irq(int irq, void *data) +{ + struct wm8994_priv *wm8994 = data; + struct snd_soc_codec *codec = wm8994->codec; + int reg; + + mutex_lock(&wm8994->accdet_lock); + + reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); + if (reg < 0) { + dev_err(codec->dev, "Failed to read jack status: %d\n", reg); + mutex_unlock(&wm8994->accdet_lock); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "JACKDET %x\n", reg); + + if (reg & WM1811_JACKDET_LVL) { + dev_dbg(codec->dev, "Jack detected\n"); + + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + + /* + * Start off measument of microphone impedence to find + * out what's actually there. + */ + wm8994->mic_detecting = true; + wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_MIC); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + } else { + dev_dbg(codec->dev, "Jack not detected\n"); + + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + + wm8994->mic_detecting = false; + wm8994->jack_mic = false; + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); + } + + mutex_unlock(&wm8994->accdet_lock); + + return IRQ_HANDLED; +} + /** * wm8958_mic_detect - Enable microphone detection via the WM8958 IRQ * @@ -3175,8 +3350,22 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, WM8958_MICD_LVL_SEL_MASK, micd_lvl_sel); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); + WARN_ON(codec->dapm.bias_level > SND_SOC_BIAS_STANDBY); + + /* + * If we can use jack detection start off with that, + * otherwise jump straight to microphone detection. + */ + if (wm8994->jackdet) { + snd_soc_update_bits(codec, WM8994_LDO_1, + WM8994_LDO1_DISCH, 0); + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); + } else { + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + } + } else { snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -3193,6 +3382,18 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) struct snd_soc_codec *codec = wm8994->codec; int reg, count; + mutex_lock(&wm8994->accdet_lock); + + /* + * Jack detection may have detected a removal simulataneously + * with an update of the MICDET status; if so it will have + * stopped detection and we can ignore this interrupt. + */ + if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) { + mutex_unlock(&wm8994->accdet_lock); + return IRQ_HANDLED; + } + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3200,6 +3401,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) do { reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); if (reg < 0) { + mutex_unlock(&wm8994->accdet_lock); dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); @@ -3230,6 +3432,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + mutex_unlock(&wm8994->accdet_lock); + return IRQ_HANDLED; } @@ -3280,6 +3484,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + mutex_init(&wm8994->accdet_lock); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); @@ -3428,6 +3634,21 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } } + switch (control->type) { + case WM1811: + if (wm8994->revision > 1) { + ret = wm8994_request_irq(wm8994->wm8994, + WM8994_IRQ_GPIO(6), + wm1811_jackdet_irq, "JACKDET", + wm8994); + if (ret == 0) + wm8994->jackdet = true; + } + break; + default: + break; + } + wm8994->fll_locked_irq = true; for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) { ret = wm8994_request_irq(wm8994->wm8994, @@ -3650,6 +3871,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return 0; err_irq: + if (wm8994->jackdet) + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm8994); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_SHRT, wm8994); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, wm8994); @@ -3688,6 +3911,9 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); + if (wm8994->jackdet) + wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm8994); + switch (control->type) { case WM8994: if (wm8994->micdet_irq) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 8622bc4db2fe..6ef3f11878c6 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -85,6 +85,7 @@ struct wm8994_priv { bool fll_locked_irq; int vmid_refcount; + int active_refcount; int dac_rates[2]; int lrclk_shared[2]; @@ -126,10 +127,12 @@ struct wm8994_priv { const char **enh_eq_texts; struct soc_enum enh_eq_enum; + struct mutex accdet_lock; struct wm8994_micdet micdet[2]; bool mic_detecting; bool jack_mic; int btn_mask; + bool jackdet; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- cgit v1.2.1 From 52ac7ab2475da2b577e4a4616c98b5d1fa3a3cfd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Dec 2011 12:43:26 +0000 Subject: ASoC: Ensure we reconfigure WM8958 microphone detection on rate changes We don't need to rerun DAPM if the clock source is the same but we do need to adjust the microphone detection rate in case we are moving from an audio to a non-audio rate. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2e28f472b963..91f3638ab33f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -307,10 +307,8 @@ static int configure_clock(struct snd_soc_codec *codec) change = snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); - if (!change) - return 0; - - snd_soc_dapm_sync(&codec->dapm); + if (change) + snd_soc_dapm_sync(&codec->dapm); wm8958_micd_set_rate(codec); -- cgit v1.2.1 From cd1707a99a2cb43cd8ab0c1952b455b218f15884 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Dec 2011 13:44:25 +0000 Subject: ASoC: Add platform data for WM8958/WM1811 microphone detection rates Allow systems to override the default microphone detection rates using platform data in case the settings are not suitable (eg, due to an unusually noisy jack). Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 91f3638ab33f..6bdf8137c7e8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -60,13 +60,6 @@ static int wm8994_retune_mobile_base[] = { static void wm8958_default_micdet(u16 status, void *data); -struct wm8958_micd_rate { - int sysclk; - bool idle; - int start; - int rate; -}; - static const struct wm8958_micd_rate micdet_rates[] = { { 32768, true, 1, 4 }, { 32768, false, 1, 1 }, @@ -100,7 +93,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) else sysclk = wm8994->aifclk[0]; - if (wm8994->jackdet) { + if (wm8994->pdata && wm8994->pdata->micd_rates) { + rates = wm8994->pdata->micd_rates; + num_rates = wm8994->pdata->num_micd_rates; + } else if (wm8994->jackdet) { rates = jackdet_rates; num_rates = ARRAY_SIZE(jackdet_rates); } else { -- cgit v1.2.1 From 7270cebef293c7af3f91afdbe7514797ca53a5dd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Dec 2011 14:00:19 +0000 Subject: ASoC: Convert WM8994 to devm_kzalloc() Still have a manual free in there for some realloc()ed memory as there's no devm version of that. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6bdf8137c7e8..0699ed2fb793 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3003,8 +3003,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) }; /* We need an array of texts for the enum API */ - wm8994->drc_texts = kmalloc(sizeof(char *) - * pdata->num_drc_cfgs, GFP_KERNEL); + wm8994->drc_texts = devm_kzalloc(wm8994->codec->dev, + sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); if (!wm8994->drc_texts) { dev_err(wm8994->codec->dev, "Failed to allocate %d DRC config texts\n", @@ -3468,7 +3468,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) codec->control_data = dev_get_drvdata(codec->dev->parent); control = codec->control_data; - wm8994 = kzalloc(sizeof(struct wm8994_priv), GFP_KERNEL); + wm8994 = devm_kzalloc(codec->dev, sizeof(struct wm8994_priv), + GFP_KERNEL); if (wm8994 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm8994); @@ -3880,8 +3881,6 @@ err_irq: wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); -err: - kfree(wm8994); return ret; } @@ -3933,8 +3932,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) if (wm8994->enh_eq) release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - kfree(wm8994->drc_texts); - kfree(wm8994); return 0; } -- cgit v1.2.1 From 778825801d9dc3745417d295344b5b1e27de0d86 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Tue, 22 Nov 2011 23:52:21 +0800 Subject: ASoC: mxs-saif: remove function in platform_data Add master_mode and master_id in platfrom_data since it's board specific and board knows it. Then we can remove the function pointer in platfrom_data to make the driver more devicetree friendly. Signed-off-by: Dong Aisheng Acked-by: Mark Brown Signed-off-by: Shawn Guo --- sound/soc/mxs/mxs-saif.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d24fc2..1ef697fe1731 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -625,13 +625,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - pdata = pdev->dev.platform_data; - if (pdata && pdata->init) { - ret = pdata->init(); - if (ret) - return ret; - } - saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -639,12 +632,17 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[pdev->id] = saif; saif->id = pdev->id; - saif->master_id = saif->id; - if (pdata && pdata->get_master_id) { - saif->master_id = pdata->get_master_id(saif->id); + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) { + saif->master_id = pdata->master_id; if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif)) + saif->master_id >= ARRAY_SIZE(mxs_saif) || + saif->master_id == saif->id) { + dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; + } + } else { + saif->master_id = saif->id; } saif->clk = clk_get(&pdev->dev, NULL); -- cgit v1.2.1 From 1ab97c8cad98de016cb36a870e118feaf0a0caaf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 16:21:51 +0000 Subject: ASoC: Add signal generator widget type A signal generator behaves as an input would but is not considered for any of the special behaviour associated with external input pins. This is especially useful when automatically working out not connected widgets. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index da5c1ae7cc30..6bb327e431a5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -339,6 +339,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_output: case snd_soc_dapm_adc: case snd_soc_dapm_input: + case snd_soc_dapm_siggen: case snd_soc_dapm_dac: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: @@ -772,6 +773,11 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return widget->inputs; } + /* signal generator */ + if (widget->id == snd_soc_dapm_siggen) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } } list_for_each_entry(path, &widget->sources, list_sink) { @@ -1982,6 +1988,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_out_drv: case snd_soc_dapm_input: case snd_soc_dapm_output: + case snd_soc_dapm_siggen: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_pre: -- cgit v1.2.1 From dea8e237415f1992694b3a8625570f8920927f28 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 16:24:05 +0000 Subject: ASoC: Make WM5100 tone generator widgets signal generators Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm5100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8be5dae83cae..a234b70377fc 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -952,7 +952,7 @@ SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), -SND_SOC_DAPM_INPUT("TONE"), +SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_PGA_E("IN1L PGA", WM5100_INPUT_ENABLES, WM5100_IN1L_ENA_SHIFT, 0, NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), -- cgit v1.2.1 From 36c6b54cb0ec1908bc98c4d2d3b8584219f4d532 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 16:24:18 +0000 Subject: ASoC: Make WM8962 beep a signal generator Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8810988522eb..be35b6468cb1 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2675,7 +2675,7 @@ SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), -SND_SOC_DAPM_INPUT("Beep"), +SND_SOC_DAPM_SIGGEN("Beep"), SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), -- cgit v1.2.1 From 84b315ee893676e9a9ce8ac42ab5ef44e2af3ee1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 2 Dec 2011 10:18:28 +0100 Subject: ASoC: Drop unused state parameter from CODEC suspend callback The existence of this parameter is purely historical. None of the CODEC drivers uses it and we always pass in the same value anyway, so it should be safe to remove it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/stac9766.c | 3 +-- sound/soc/codecs/tlv320aic23.c | 3 +-- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 3 +-- sound/soc/codecs/wm9713.c | 3 +-- sound/soc/soc-core.c | 2 +- 66 files changed, 66 insertions(+), 71 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 221ec29f68e3..1bbad4c16d28 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -99,7 +99,7 @@ static int ac97_soc_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int ac97_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int ac97_soc_suspend(struct snd_soc_codec *codec) { snd_ac97_suspend(codec->ac97); diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index fab0948f7a54..919322daf6dd 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_driver ad183x_dais[] = { }; #ifdef CONFIG_PM -static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ad1836_suspend(struct snd_soc_codec *codec) { /* reset clock control mode */ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 45c63028b40d..637b114bea7f 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1321,7 +1321,7 @@ static int adau1373_remove(struct snd_soc_codec *codec) return 0; } -static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int adau1373_suspend(struct snd_soc_codec *codec) { return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f9f08948e5e8..ebd7b37b902b 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -798,7 +798,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } -static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int adav80x_suspend(struct snd_soc_codec *codec) { return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1f531085453..96296fd172f9 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -354,7 +354,7 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ak4535_suspend(struct snd_soc_codec *codec) { ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index f53f31480565..90184701480d 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -498,7 +498,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = { }, }; -static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ak4641_suspend(struct snd_soc_codec *codec) { ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a5c001e8ba8..da97f024ec74 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -868,7 +868,7 @@ static struct snd_soc_dai_driver alc5623_dai = { .ops = &alc5623_dai_ops, }; -static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int alc5623_suspend(struct snd_soc_codec *codec) { alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 3f750def8967..08613c7e1091 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -956,7 +956,7 @@ static struct snd_soc_dai_driver alc5632_dai = { }; #ifdef CONFIG_PM -static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int alc5632_suspend(struct snd_soc_codec *codec) { alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index dc77ff7ba339..fef0f48330e4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -578,7 +578,7 @@ static int cs4270_remove(struct snd_soc_codec *codec) * and all registers are written back to the hardware when resuming. */ -static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int cs4270_soc_suspend(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int reg, ret; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a6f77a855f45..f6fe846b6a6c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -430,7 +430,7 @@ static struct snd_soc_dai_driver cs4271_dai = { }; #ifdef CONFIG_PM -static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; /* Set power-down bit */ diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index da3125aa55f9..9d38db8f1919 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1262,7 +1262,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { } }; -static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int cs42l73_suspend(struct snd_soc_codec *codec) { cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4fca8bccd535..517e2a516daf 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -311,7 +311,7 @@ static int jz4740_codec_dev_remove(struct snd_soc_codec *codec) #ifdef CONFIG_PM_SLEEP -static int jz4740_codec_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int jz4740_codec_suspend(struct snd_soc_codec *codec) { return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 9b6036e5738a..ba4f6f167a13 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1946,7 +1946,7 @@ static void max98088_handle_pdata(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int max98088_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max98088_suspend(struct snd_soc_codec *codec) { max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 01f4ad725149..c69dd022bea8 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2174,7 +2174,7 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int max98095_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max98095_suspend(struct snd_soc_codec *codec) { max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 94c2b586ed5d..7dfd6e84796d 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_driver max9850_dai = { }; #ifdef CONFIG_PM -static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max9850_suspend(struct snd_soc_codec *codec) { max9850_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index b12d01f67990..edcaa7ea5487 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -118,7 +118,7 @@ static int pcm3008_soc_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 9fd50bd77c49..f6e4f5ed9286 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1641,7 +1641,7 @@ static int rt5631_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int rt5631_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int rt5631_suspend(struct snd_soc_codec *codec) { rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ff0a1079efec..250175755eb2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -967,7 +967,7 @@ static int sgtl5000_volatile_register(struct snd_soc_codec *codec, } #ifdef CONFIG_SUSPEND -static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int sgtl5000_suspend(struct snd_soc_codec *codec) { sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0d43e4b4a586..7dfc7b08114c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -523,7 +523,7 @@ static struct snd_soc_dai_driver ssm2602_dai = { .ops = &ssm2602_dai_ops, }; -static int ssm2602_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ssm2602_suspend(struct snd_soc_codec *codec) { ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index b3d1c78e361f..6648af6656c8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -801,7 +801,7 @@ static struct snd_soc_dai_driver sta32x_dai = { }; #ifdef CONFIG_PM -static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int sta32x_suspend(struct snd_soc_codec *codec) { sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 55819537b677..e34969cdc0e8 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -256,8 +256,7 @@ static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) return 0; } -static int stac9766_codec_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int stac9766_codec_suspend(struct snd_soc_codec *codec) { stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cba798e1a07e..60d08aeac22a 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -528,8 +528,7 @@ static struct snd_soc_dai_driver tlv320aic23_dai = { .ops = &tlv320aic23_dai_ops, }; -static int tlv320aic23_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int tlv320aic23_suspend(struct snd_soc_codec *codec) { tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f55337567379..81a26e1090b3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -621,7 +621,7 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int aic32x4_suspend(struct snd_soc_codec *codec) { aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 21625dddde23..6f963c50e76e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1268,7 +1268,7 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int aic3x_suspend(struct snd_soc_codec *codec) { aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 6b0f0e220f85..c7a61fbdae4b 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1460,7 +1460,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int dac33_soc_suspend(struct snd_soc_codec *codec) { dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 61d8a9065ff3..18e71014cc2e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2202,7 +2202,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int twl4030_soc_suspend(struct snd_soc_codec *codec) { twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a4a65dc9e33a..3376e6fad2a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1470,7 +1470,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }; #ifdef CONFIG_PM -static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int twl6040_suspend(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index d0f9d904ce8f..8f4f469d6411 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -571,8 +571,7 @@ static int uda134x_soc_remove(struct snd_soc_codec *codec) } #if defined(CONFIG_PM) -static int uda134x_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int uda134x_soc_suspend(struct snd_soc_codec *codec) { uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6b933efc7ed3..d08b91dde53d 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -705,7 +705,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { }, }; -static int uda1380_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int uda1380_suspend(struct snd_soc_codec *codec) { uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3f1ed5f5ccf4..f39497fc13e0 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1315,7 +1315,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8350_suspend(struct snd_soc_codec *codec) { wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index a1173eb7936d..56a7b7256efa 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1352,7 +1352,7 @@ static struct snd_soc_dai_driver wm8400_dai = { .ops = &wm8400_dai_ops, }; -static int wm8400_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8400_suspend(struct snd_soc_codec *codec) { wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 3a655719ba2c..5e847506138e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -534,7 +534,7 @@ static struct snd_soc_dai_driver wm8510_dai = { .symmetric_rates = 1, }; -static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8510_suspend(struct snd_soc_codec *codec) { wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 0c89f8e2daaf..7fea2c3bf7e7 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -384,7 +384,7 @@ static struct snd_soc_dai_driver wm8523_dai = { }; #ifdef CONFIG_PM -static int wm8523_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8523_suspend(struct snd_soc_codec *codec) { wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 760080e43015..b9b1a2f8360f 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -338,7 +338,7 @@ static struct snd_soc_dai_driver wm8711_dai = { .ops = &wm8711_ops, }; -static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8711_suspend(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8711_ACTIVE, 0x0); wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 085c2f81d8c2..b1f01d9273be 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -214,7 +214,7 @@ static struct snd_soc_dai_driver wm8728_dai = { .ops = &wm8728_dai_ops, }; -static int wm8728_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8728_suspend(struct snd_soc_codec *codec) { wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c18dee06f29c..8821af70e660 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -490,7 +490,7 @@ static struct snd_soc_dai_driver wm8731_dai = { }; #ifdef CONFIG_PM -static int wm8731_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8731_suspend(struct snd_soc_codec *codec) { wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index c13e4f7809cf..ff95e62c56b9 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -539,7 +539,7 @@ static struct snd_soc_dai_driver wm8737_dai = { }; #ifdef CONFIG_PM -static int wm8737_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8737_suspend(struct snd_soc_codec *codec) { wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b312fccbf67a..48cb78fd0103 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -666,7 +666,7 @@ static struct snd_soc_dai_driver wm8750_dai = { .ops = &wm8750_dai_ops, }; -static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8750_suspend(struct snd_soc_codec *codec) { wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index dc3153852d8a..b114c19f530a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1380,7 +1380,7 @@ static void wm8753_work(struct work_struct *work) wm8753_set_bias_level(codec, dapm->bias_level); } -static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 391c385ec43e..8976eb5796d3 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -555,7 +555,7 @@ static struct snd_soc_dai_driver wm8770_dai = { }; #ifdef CONFIG_PM -static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8770_suspend(struct snd_soc_codec *codec) { wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index af542a2f5941..fbf80c5220d0 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -371,7 +371,7 @@ static struct snd_soc_dai_driver wm8776_dai[] = { }; #ifdef CONFIG_PM -static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8776_suspend(struct snd_soc_codec *codec) { wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d99c6a0a0a2d..ae4b8fb3c3e5 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -542,7 +542,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8804_suspend(struct snd_soc_codec *codec) { wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6ac80cf80b31..85632ffcb872 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1106,7 +1106,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8900_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8900_suspend(struct snd_soc_codec *codec) { struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); int fll_out = wm8900->fll_out; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 70a2268c5498..d663c97785d7 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1758,7 +1758,7 @@ static struct snd_soc_dai_driver wm8903_dai = { .symmetric_rates = 1, }; -static int wm8903_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8903_suspend(struct snd_soc_codec *codec) { wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index babca49c8766..f0ae01bbaa94 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2234,7 +2234,7 @@ static struct snd_soc_dai_driver wm8904_dai = { }; #ifdef CONFIG_PM -static int wm8904_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8904_suspend(struct snd_soc_codec *codec) { wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 9f1cce8d105d..0fe4545eef89 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -672,7 +672,7 @@ static struct snd_soc_dai_driver wm8940_dai = { .symmetric_rates = 1, }; -static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8940_suspend(struct snd_soc_codec *codec) { return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ca38722bc3fe..cdd51398e1f4 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -878,7 +878,7 @@ static struct snd_soc_dai_driver wm8955_dai = { }; #ifdef CONFIG_PM -static int wm8955_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8955_suspend(struct snd_soc_codec *codec) { wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f623ca..55b9a25cd1b3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -894,7 +894,7 @@ static struct snd_soc_dai_driver wm8960_dai = { .symmetric_rates = 1, }; -static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8960_suspend(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index c0587013fdfa..9bcf846e93b0 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1038,7 +1038,7 @@ static int wm8961_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm8961_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8961_suspend(struct snd_soc_codec *codec) { wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b01df56b824a..aadd14a14661 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -599,7 +599,7 @@ static void wm8971_work(struct work_struct *work) wm8971_set_bias_level(codec, codec->dapm.bias_level); } -static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8971_suspend(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index e41f9993c652..a5fd017c4332 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -582,7 +582,7 @@ static struct snd_soc_dai_driver wm8974_dai = { .symmetric_rates = 1, }; -static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8974_suspend(struct snd_soc_codec *codec) { wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 649a2e3c02ae..85d514d63a4c 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -892,7 +892,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .ops = &wm8978_dai_ops, }; -static int wm8978_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8978_suspend(struct snd_soc_codec *codec) { wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 362298cce92c..cebde568d191 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -974,7 +974,7 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8983_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8983_suspend(struct snd_soc_codec *codec) { wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 9e4481bb1223..c0c86b3c6adf 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -945,7 +945,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8985_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8985_suspend(struct snd_soc_codec *codec) { wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 608c6721e4f1..093884705b01 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -728,7 +728,7 @@ static struct snd_soc_dai_driver wm8988_dai = { .symmetric_rates = 1, }; -static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8988_suspend(struct snd_soc_codec *codec) { wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 58d7f0bff990..b417d2e0cdfd 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1313,7 +1313,7 @@ static struct snd_soc_dai_driver wm8990_dai = { .ops = &wm8990_dai_ops, }; -static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8990_suspend(struct snd_soc_codec *codec) { wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 35c5389e5ef7..7ee40da8dbb5 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1240,7 +1240,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8991_suspend(struct snd_soc_codec *codec) { wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 780c24cdab6d..0f8278b4f0ad 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1544,7 +1544,7 @@ static int wm8993_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm8993_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8993_suspend(struct snd_soc_codec *codec) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int fll_fout = wm8993->fll_fout; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0699ed2fb793..d9faa39d826d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2801,7 +2801,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { }; #ifdef CONFIG_PM -static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8994_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 8f6a36d7c75b..5863406b459d 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2009,7 +2009,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8995_suspend(struct snd_soc_codec *codec) { wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8a4b97060444..1f2672b1e03e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1302,7 +1302,7 @@ static int wm9081_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm9081_suspend(struct snd_soc_codec *codec) { wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f94c06057c64..5cb8759868df 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -604,7 +604,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9090_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm9090_suspend(struct snd_soc_codec *codec) { wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index b720a43c422c..40c92ead85a3 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9705_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int wm9705_soc_suspend(struct snd_soc_codec *codec) { soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 4ce73f59df20..b7b31f84c10b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -583,8 +583,7 @@ err: return -EIO; } -static int wm9712_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int wm9712_soc_suspend(struct snd_soc_codec *codec) { wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index edb598182c69..2b8479bfcd93 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1140,8 +1140,7 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm9713_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int wm9713_soc_suspend(struct snd_soc_codec *codec) { u16 reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ec783f0a27e9..5195f0653b35 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -572,7 +572,7 @@ int snd_soc_suspend(struct device *dev) switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec, PMSG_SUSPEND); + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; break; -- cgit v1.2.1 From 6f526f0a86dbb22fd2fc5a873f55c9e2341a79c0 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 1 Dec 2011 13:49:19 -0700 Subject: ASoC: WM8903: Disallow all invalid gpio_cfg pdata values The GPIO registers are 15 bits wide. Hence values, higher than 0x7fff are not legal GPIO register values. Modify the pdata.gpio_cfg handling code to reject all illegal values, not just WM8903_GPIO_NO_CONFIG (0x8000). This will allow the later use of 0xffffffff as an invalid value in future device tree bindings, meaning "don't touch this GPIO's configuration". Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d663c97785d7..60ad8cdc046c 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1936,11 +1936,11 @@ static int wm8903_probe(struct snd_soc_codec *codec) bool mic_gpio = false; for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) + if (pdata->gpio_cfg[i] > 0x7fff) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0xffff); + pdata->gpio_cfg[i] & 0x7fff); val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) >> WM8903_GP1_FN_SHIFT; -- cgit v1.2.1 From a806aa9207ad59933464efbe6009394723713c0d Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 1 Dec 2011 19:52:46 -0600 Subject: ASoC: p1022ds: add support for fsl,P1022 and fsl,P1022DS model names Commit ab827d97 ("powerpc/85xx: Rework P1022DS device tree") renamed the the /model property of the P1022DS device tree from "fsl,P1022" to "fsl,P1022DS". To support both old and new device trees, the ASoC machine driver for the P1022DS needs to query the /model property and update the platform driver object dynamically. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_ds.c | 36 ++++++++++++++++++++++++++++-------- 1 file changed, 28 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 2c064a9824ad..309162652287 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -540,12 +540,6 @@ static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, .remove = __devexit_p(p1022_ds_remove), .driver = { - /* The name must match the 'model' property in the device tree, - * in lowercase letters, but only the part after that last - * comma. This is because some model properties have a "fsl," - * prefix. - */ - .name = "snd-soc-p1022", .owner = THIS_MODULE, }, }; @@ -559,13 +553,39 @@ static int __init p1022_ds_init(void) { struct device_node *guts_np; struct resource res; + const char *sprop; + + /* + * Check if we're actually running on a P1022DS. Older device trees + * have a model of "fsl,P1022" and newer ones use "fsl,P1022DS", so we + * need to support both. The SSI driver uses that property to link to + * the machine driver, so have to match it. + */ + sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); + if (!sprop) { + pr_err("snd-soc-p1022ds: missing /model node"); + return -ENODEV; + } + + pr_debug("snd-soc-p1022ds: board model name is %s\n", sprop); - pr_info("Freescale P1022 DS ALSA SoC machine driver\n"); + /* + * The name of this board, taken from the device tree. Normally, this is a* + * fixed string, but some P1022DS device trees have a /model property of + * "fsl,P1022", and others have "fsl,P1022DS". + */ + if (strcasecmp(sprop, "fsl,p1022ds") == 0) + p1022_ds_driver.driver.name = "snd-soc-p1022ds"; + else if (strcasecmp(sprop, "fsl,p1022") == 0) + p1022_ds_driver.driver.name = "snd-soc-p1022"; + else + return -ENODEV; /* Get the physical address of the global utilities registers */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("p1022-ds: missing/invalid global utilities node\n"); + pr_err("snd-soc-p1022ds: missing/invalid global utils node\n"); + of_node_put(guts_np); return -EINVAL; } guts_phys = res.start; -- cgit v1.2.1 From 6132725eac521b89dee3d58df3c6d04a1e50844c Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 1 Dec 2011 19:52:47 -0600 Subject: ASoC: fsl/powerpc: don't rely on the cell-index property Instead of using the 'cell-index' property in the I2C adapter node to determine the adapter number, just query the i2c_adapter object directly. Previously, the I2C nodes always appeared in cell-index order, so the dynamic numbering coincided with the cell-index property. With commit ab827d97 ("powerpc/85xx: Rework P1022DS device tree"), the I2C nodes are unintentionally reversed in the device tree, and so the machine driver guesses the wrong I2C adapter number. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 13 ++++++++----- sound/soc/fsl/p1022_ds.c | 13 ++++++++----- 2 files changed, 16 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index ae49f1c78c6d..0ea4a5a96e06 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include @@ -249,8 +250,9 @@ static int get_parent_cell_index(struct device_node *np) static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) { const u32 *iprop; - int bus, addr; + int addr; char temp[DAI_NAME_SIZE]; + struct i2c_client *i2c; of_modalias_node(np, temp, DAI_NAME_SIZE); @@ -260,11 +262,12 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) addr = be32_to_cpup(iprop); - bus = get_parent_cell_index(np); - if (bus < 0) - return bus; + /* We need the adapter number */ + i2c = of_find_i2c_device_by_node(np); + if (!i2c) + return -ENODEV; - snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s-codec.%u-%04x", temp, i2c->adapter->nr, addr); return 0; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 309162652287..a5d4e80a9cf4 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include @@ -252,8 +253,9 @@ static int get_parent_cell_index(struct device_node *np) static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) { const u32 *iprop; - int bus, addr; + int addr; char temp[DAI_NAME_SIZE]; + struct i2c_client *i2c; of_modalias_node(np, temp, DAI_NAME_SIZE); @@ -263,11 +265,12 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) addr = be32_to_cpup(iprop); - bus = get_parent_cell_index(np); - if (bus < 0) - return bus; + /* We need the adapter number */ + i2c = of_find_i2c_device_by_node(np); + if (!i2c) + return -ENODEV; - snprintf(buf, len, "%s.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); return 0; } -- cgit v1.2.1 From 3631e8d43e385e851f88637244a287433246c097 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 10:55:12 +0000 Subject: ASoC: Add missing err label Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d9faa39d826d..83e8033f5719 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3881,6 +3881,7 @@ err_irq: wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); +err: return ret; } -- cgit v1.2.1 From f2e2026c98b74028b55901711c5df98e6d2ad8c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 15:55:52 +0000 Subject: ASoC: Add WM8958 based headset detection on Littlemill The board supports CODECs that won't work with this but the CODEC driver will check to see if it's running on the right chip for us. Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index d2a44ab3c207..5d7680f4b7c1 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -154,8 +154,11 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1R" }, }; +static struct snd_soc_jack littlemill_headset; + static int littlemill_late_probe(struct snd_soc_card *card) { + struct snd_soc_codec *codec = card->rtd[0].codec; struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; @@ -164,6 +167,18 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset); + if (ret) + return ret; + + /* This will check device compatibility itself */ + wm8958_mic_detect(codec, &littlemill_headset, NULL, NULL); + return 0; } -- cgit v1.2.1 From 91e20854e58075231c72a9c63d4454616787557e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 16:01:41 +0000 Subject: ASoC: Convert WM8994 MICBIASes to supply widgets There are some in tree systems using the driver but none use the MICBIAS widgets. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index d1debfb20c60..2a61094075f8 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -610,8 +610,8 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), -SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), -- cgit v1.2.1 From 8f103167fecb1f4b5888fbcfc81b67e3c810dee0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 17:36:06 +0000 Subject: ASoC: Map microphones on Littlemill Littlemill has one analogue microphone on the board (connected to IN1LN) and an array of four DMICs connected to both DMICDAT lines. The biases can be selected by jumpers but pick the default jumper fit. Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5d7680f4b7c1..5cea59beec9f 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -147,11 +147,21 @@ static struct snd_soc_dai_link littlemill_dai[] = { static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), + + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), }; static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1L" }, { "Headphone", NULL, "HPOUT1R" }, + + { "AMIC", NULL, "MICBIAS1" }, /* Default for AMICBIAS jumper */ + { "IN1LN", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, }; static struct snd_soc_jack littlemill_headset; -- cgit v1.2.1 From 2950cd2208174af9be430f6b6f1507d429c366ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 10:59:32 +0000 Subject: ASoC: Convert WM8903 to devm_kzalloc() Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 60ad8cdc046c..12eedaf938d7 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2079,7 +2079,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, struct wm8903_priv *wm8903; int ret; - wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); + wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), + GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; @@ -2088,15 +2089,13 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); - if (ret < 0) - kfree(wm8903); + return ret; } static __devexit int wm8903_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From f4a10837c9dd473cd615766cf38f33a3c1f745cf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:21:28 +0000 Subject: ASoC: Use table based control init for WM8903 Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 12eedaf938d7..76c7c2bd3cd1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2034,9 +2034,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); - snd_soc_add_controls(codec, wm8903_snd_controls, - ARRAY_SIZE(wm8903_snd_controls)); - wm8903_init_gpio(codec); return ret; @@ -2066,6 +2063,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, .seq_notifier = wm8903_seq_notifier, + .controls = wm8903_snd_controls, + .num_controls = ARRAY_SIZE(wm8903_snd_controls), .dapm_widgets = wm8903_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8903_dapm_widgets), .dapm_routes = wm8903_intercon, -- cgit v1.2.1 From 88a1b12b9c70d1b2ea4d11bdfa6ae65c9570909b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:21:52 +0000 Subject: ASoC: WM8903 only supports I2C so don't ifdef it Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 76c7c2bd3cd1..745681258eda 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2071,7 +2071,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .num_dapm_routes = ARRAY_SIZE(wm8903_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2113,27 +2112,22 @@ static struct i2c_driver wm8903_i2c_driver = { .remove = __devexit_p(wm8903_i2c_remove), .id_table = wm8903_i2c_id, }; -#endif static int __init wm8903_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8903_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8903 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8903_modinit); static void __exit wm8903_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8903_i2c_driver); -#endif } module_exit(wm8903_exit); -- cgit v1.2.1 From 45e967553f3466f773ecd418c09fe92b753f18b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:23:37 +0000 Subject: ASoC: Use a normal cache sync for WM8903 The driver used to use a complicated method to sync the register cache after having brought the bias level up to standby in resume due to the use of the write sequencer to manage the initial power up. Now that we don't use the write sequencer there is no need for this and we can just use snd_soc_cache_sync() directly. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 18 +++--------------- 1 file changed, 3 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 745681258eda..fdc3ff053f98 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1767,23 +1767,11 @@ static int wm8903_suspend(struct snd_soc_codec *codec) static int wm8903_resume(struct snd_soc_codec *codec) { - int i; - u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), - GFP_KERNEL); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - /* Bring the codec back up to standby first to minimise pop/clicks */ - wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_cache_sync(codec); - /* Sync back everything else */ - if (tmp_cache) { - for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) - if (tmp_cache[i] != reg_cache[i]) - snd_soc_write(codec, i, tmp_cache[i]); - kfree(tmp_cache); - } else { - dev_err(codec->dev, "Failed to allocate temporary cache\n"); - } + wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } -- cgit v1.2.1 From 82ae55dbcc4a37a4288346795755da5e07c09d33 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:24:40 +0000 Subject: ASoC: Don't resync WM8903 register cache on reset We only do this on initial power on so it's at best a waste of time as the core will have already defaulted to the same values. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fdc3ff053f98..d840cbfc34ac 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -260,8 +260,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re static void wm8903_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); - memcpy(codec->reg_cache, wm8903_reg_defaults, - sizeof(wm8903_reg_defaults)); } static int wm8903_cp_event(struct snd_soc_dapm_widget *w, -- cgit v1.2.1 From ee244ce4ea5651989229d7f287f777f68104a59a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:33:32 +0000 Subject: ASoC: Convert WM8903 to direct regmap API usage Converting to an rbtree cache as regcache doesn't have a flat cache. Since the top of the register map is fairly sparse this should be an overall win. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 378 ++++++++++++++++++++++++---------------------- 1 file changed, 196 insertions(+), 182 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d840cbfc34ac..0b12a5525c15 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -37,184 +38,84 @@ #include "wm8903.h" /* Register defaults at reset */ -static u16 wm8903_reg_defaults[] = { - 0x8903, /* R0 - SW Reset and ID */ - 0x0000, /* R1 - Revision Number */ - 0x0000, /* R2 */ - 0x0000, /* R3 */ - 0x0018, /* R4 - Bias Control 0 */ - 0x0000, /* R5 - VMID Control 0 */ - 0x0000, /* R6 - Mic Bias Control 0 */ - 0x0000, /* R7 */ - 0x0001, /* R8 - Analogue DAC 0 */ - 0x0000, /* R9 */ - 0x0001, /* R10 - Analogue ADC 0 */ - 0x0000, /* R11 */ - 0x0000, /* R12 - Power Management 0 */ - 0x0000, /* R13 - Power Management 1 */ - 0x0000, /* R14 - Power Management 2 */ - 0x0000, /* R15 - Power Management 3 */ - 0x0000, /* R16 - Power Management 4 */ - 0x0000, /* R17 - Power Management 5 */ - 0x0000, /* R18 - Power Management 6 */ - 0x0000, /* R19 */ - 0x0400, /* R20 - Clock Rates 0 */ - 0x0D07, /* R21 - Clock Rates 1 */ - 0x0000, /* R22 - Clock Rates 2 */ - 0x0000, /* R23 */ - 0x0050, /* R24 - Audio Interface 0 */ - 0x0242, /* R25 - Audio Interface 1 */ - 0x0008, /* R26 - Audio Interface 2 */ - 0x0022, /* R27 - Audio Interface 3 */ - 0x0000, /* R28 */ - 0x0000, /* R29 */ - 0x00C0, /* R30 - DAC Digital Volume Left */ - 0x00C0, /* R31 - DAC Digital Volume Right */ - 0x0000, /* R32 - DAC Digital 0 */ - 0x0000, /* R33 - DAC Digital 1 */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x00C0, /* R36 - ADC Digital Volume Left */ - 0x00C0, /* R37 - ADC Digital Volume Right */ - 0x0000, /* R38 - ADC Digital 0 */ - 0x0073, /* R39 - Digital Microphone 0 */ - 0x09BF, /* R40 - DRC 0 */ - 0x3241, /* R41 - DRC 1 */ - 0x0020, /* R42 - DRC 2 */ - 0x0000, /* R43 - DRC 3 */ - 0x0085, /* R44 - Analogue Left Input 0 */ - 0x0085, /* R45 - Analogue Right Input 0 */ - 0x0044, /* R46 - Analogue Left Input 1 */ - 0x0044, /* R47 - Analogue Right Input 1 */ - 0x0000, /* R48 */ - 0x0000, /* R49 */ - 0x0008, /* R50 - Analogue Left Mix 0 */ - 0x0004, /* R51 - Analogue Right Mix 0 */ - 0x0000, /* R52 - Analogue Spk Mix Left 0 */ - 0x0000, /* R53 - Analogue Spk Mix Left 1 */ - 0x0000, /* R54 - Analogue Spk Mix Right 0 */ - 0x0000, /* R55 - Analogue Spk Mix Right 1 */ - 0x0000, /* R56 */ - 0x002D, /* R57 - Analogue OUT1 Left */ - 0x002D, /* R58 - Analogue OUT1 Right */ - 0x0039, /* R59 - Analogue OUT2 Left */ - 0x0039, /* R60 - Analogue OUT2 Right */ - 0x0100, /* R61 */ - 0x0139, /* R62 - Analogue OUT3 Left */ - 0x0139, /* R63 - Analogue OUT3 Right */ - 0x0000, /* R64 */ - 0x0000, /* R65 - Analogue SPK Output Control 0 */ - 0x0000, /* R66 */ - 0x0010, /* R67 - DC Servo 0 */ - 0x0100, /* R68 */ - 0x00A4, /* R69 - DC Servo 2 */ - 0x0807, /* R70 */ - 0x0000, /* R71 */ - 0x0000, /* R72 */ - 0x0000, /* R73 */ - 0x0000, /* R74 */ - 0x0000, /* R75 */ - 0x0000, /* R76 */ - 0x0000, /* R77 */ - 0x0000, /* R78 */ - 0x000E, /* R79 */ - 0x0000, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 */ - 0x0000, /* R83 */ - 0x0000, /* R84 */ - 0x0000, /* R85 */ - 0x0000, /* R86 */ - 0x0006, /* R87 */ - 0x0000, /* R88 */ - 0x0000, /* R89 */ - 0x0000, /* R90 - Analogue HP 0 */ - 0x0060, /* R91 */ - 0x0000, /* R92 */ - 0x0000, /* R93 */ - 0x0000, /* R94 - Analogue Lineout 0 */ - 0x0060, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 - Charge Pump 0 */ - 0x1F25, /* R99 */ - 0x2B19, /* R100 */ - 0x01C0, /* R101 */ - 0x01EF, /* R102 */ - 0x2B00, /* R103 */ - 0x0000, /* R104 - Class W 0 */ - 0x01C0, /* R105 */ - 0x1C10, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 - Write Sequencer 0 */ - 0x0000, /* R109 - Write Sequencer 1 */ - 0x0000, /* R110 - Write Sequencer 2 */ - 0x0000, /* R111 - Write Sequencer 3 */ - 0x0000, /* R112 - Write Sequencer 4 */ - 0x0000, /* R113 */ - 0x0000, /* R114 - Control Interface */ - 0x0000, /* R115 */ - 0x00A8, /* R116 - GPIO Control 1 */ - 0x00A8, /* R117 - GPIO Control 2 */ - 0x00A8, /* R118 - GPIO Control 3 */ - 0x0220, /* R119 - GPIO Control 4 */ - 0x01A0, /* R120 - GPIO Control 5 */ - 0x0000, /* R121 - Interrupt Status 1 */ - 0xFFFF, /* R122 - Interrupt Status 1 Mask */ - 0x0000, /* R123 - Interrupt Polarity 1 */ - 0x0000, /* R124 */ - 0x0003, /* R125 */ - 0x0000, /* R126 - Interrupt Control */ - 0x0000, /* R127 */ - 0x0005, /* R128 */ - 0x0000, /* R129 - Control Interface Test 1 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x03FF, /* R135 */ - 0x0007, /* R136 */ - 0x0040, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x4000, /* R148 */ - 0x6810, /* R149 - Charge Pump Test 1 */ - 0x0004, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0028, /* R164 - Clock Rate Test 4 */ - 0x0004, /* R165 */ - 0x0000, /* R166 */ - 0x0060, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 - Analogue Output Bias 0 */ +static const struct reg_default wm8903_reg_defaults[] = { + { 4, 0x0018 }, /* R4 - Bias Control 0 */ + { 5, 0x0000 }, /* R5 - VMID Control 0 */ + { 6, 0x0000 }, /* R6 - Mic Bias Control 0 */ + { 8, 0x0001 }, /* R8 - Analogue DAC 0 */ + { 10, 0x0001 }, /* R10 - Analogue ADC 0 */ + { 12, 0x0000 }, /* R12 - Power Management 0 */ + { 13, 0x0000 }, /* R13 - Power Management 1 */ + { 14, 0x0000 }, /* R14 - Power Management 2 */ + { 15, 0x0000 }, /* R15 - Power Management 3 */ + { 16, 0x0000 }, /* R16 - Power Management 4 */ + { 17, 0x0000 }, /* R17 - Power Management 5 */ + { 18, 0x0000 }, /* R18 - Power Management 6 */ + { 20, 0x0400 }, /* R20 - Clock Rates 0 */ + { 21, 0x0D07 }, /* R21 - Clock Rates 1 */ + { 22, 0x0000 }, /* R22 - Clock Rates 2 */ + { 24, 0x0050 }, /* R24 - Audio Interface 0 */ + { 25, 0x0242 }, /* R25 - Audio Interface 1 */ + { 26, 0x0008 }, /* R26 - Audio Interface 2 */ + { 27, 0x0022 }, /* R27 - Audio Interface 3 */ + { 30, 0x00C0 }, /* R30 - DAC Digital Volume Left */ + { 31, 0x00C0 }, /* R31 - DAC Digital Volume Right */ + { 32, 0x0000 }, /* R32 - DAC Digital 0 */ + { 33, 0x0000 }, /* R33 - DAC Digital 1 */ + { 36, 0x00C0 }, /* R36 - ADC Digital Volume Left */ + { 37, 0x00C0 }, /* R37 - ADC Digital Volume Right */ + { 38, 0x0000 }, /* R38 - ADC Digital 0 */ + { 39, 0x0073 }, /* R39 - Digital Microphone 0 */ + { 40, 0x09BF }, /* R40 - DRC 0 */ + { 41, 0x3241 }, /* R41 - DRC 1 */ + { 42, 0x0020 }, /* R42 - DRC 2 */ + { 43, 0x0000 }, /* R43 - DRC 3 */ + { 44, 0x0085 }, /* R44 - Analogue Left Input 0 */ + { 45, 0x0085 }, /* R45 - Analogue Right Input 0 */ + { 46, 0x0044 }, /* R46 - Analogue Left Input 1 */ + { 47, 0x0044 }, /* R47 - Analogue Right Input 1 */ + { 50, 0x0008 }, /* R50 - Analogue Left Mix 0 */ + { 51, 0x0004 }, /* R51 - Analogue Right Mix 0 */ + { 52, 0x0000 }, /* R52 - Analogue Spk Mix Left 0 */ + { 53, 0x0000 }, /* R53 - Analogue Spk Mix Left 1 */ + { 54, 0x0000 }, /* R54 - Analogue Spk Mix Right 0 */ + { 55, 0x0000 }, /* R55 - Analogue Spk Mix Right 1 */ + { 57, 0x002D }, /* R57 - Analogue OUT1 Left */ + { 58, 0x002D }, /* R58 - Analogue OUT1 Right */ + { 59, 0x0039 }, /* R59 - Analogue OUT2 Left */ + { 60, 0x0039 }, /* R60 - Analogue OUT2 Right */ + { 62, 0x0139 }, /* R62 - Analogue OUT3 Left */ + { 63, 0x0139 }, /* R63 - Analogue OUT3 Right */ + { 64, 0x0000 }, /* R65 - Analogue SPK Output Control 0 */ + { 67, 0x0010 }, /* R67 - DC Servo 0 */ + { 69, 0x00A4 }, /* R69 - DC Servo 2 */ + { 90, 0x0000 }, /* R90 - Analogue HP 0 */ + { 94, 0x0000 }, /* R94 - Analogue Lineout 0 */ + { 98, 0x0000 }, /* R98 - Charge Pump 0 */ + { 104, 0x0000 }, /* R104 - Class W 0 */ + { 108, 0x0000 }, /* R108 - Write Sequencer 0 */ + { 109, 0x0000 }, /* R109 - Write Sequencer 1 */ + { 110, 0x0000 }, /* R110 - Write Sequencer 2 */ + { 111, 0x0000 }, /* R111 - Write Sequencer 3 */ + { 112, 0x0000 }, /* R112 - Write Sequencer 4 */ + { 114, 0x0000 }, /* R114 - Control Interface */ + { 116, 0x00A8 }, /* R116 - GPIO Control 1 */ + { 117, 0x00A8 }, /* R117 - GPIO Control 2 */ + { 118, 0x00A8 }, /* R118 - GPIO Control 3 */ + { 119, 0x0220 }, /* R119 - GPIO Control 4 */ + { 120, 0x01A0 }, /* R120 - GPIO Control 5 */ + { 122, 0xFFFF }, /* R122 - Interrupt Status 1 Mask */ + { 123, 0x0000 }, /* R123 - Interrupt Polarity 1 */ + { 126, 0x0000 }, /* R126 - Interrupt Control */ + { 129, 0x0000 }, /* R129 - Control Interface Test 1 */ + { 149, 0x6810 }, /* R149 - Charge Pump Test 1 */ + { 164, 0x0028 }, /* R164 - Clock Rate Test 4 */ + { 172, 0x0000 }, /* R172 - Analogue Output Bias 0 */ }; struct wm8903_priv { struct snd_soc_codec *codec; + struct regmap *regmap; int sysclk; int irq; @@ -239,7 +140,93 @@ struct wm8903_priv { #endif }; -static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8903_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8903_SW_RESET_AND_ID: + case WM8903_REVISION_NUMBER: + case WM8903_BIAS_CONTROL_0: + case WM8903_VMID_CONTROL_0: + case WM8903_MIC_BIAS_CONTROL_0: + case WM8903_ANALOGUE_DAC_0: + case WM8903_ANALOGUE_ADC_0: + case WM8903_POWER_MANAGEMENT_0: + case WM8903_POWER_MANAGEMENT_1: + case WM8903_POWER_MANAGEMENT_2: + case WM8903_POWER_MANAGEMENT_3: + case WM8903_POWER_MANAGEMENT_4: + case WM8903_POWER_MANAGEMENT_5: + case WM8903_POWER_MANAGEMENT_6: + case WM8903_CLOCK_RATES_0: + case WM8903_CLOCK_RATES_1: + case WM8903_CLOCK_RATES_2: + case WM8903_AUDIO_INTERFACE_0: + case WM8903_AUDIO_INTERFACE_1: + case WM8903_AUDIO_INTERFACE_2: + case WM8903_AUDIO_INTERFACE_3: + case WM8903_DAC_DIGITAL_VOLUME_LEFT: + case WM8903_DAC_DIGITAL_VOLUME_RIGHT: + case WM8903_DAC_DIGITAL_0: + case WM8903_DAC_DIGITAL_1: + case WM8903_ADC_DIGITAL_VOLUME_LEFT: + case WM8903_ADC_DIGITAL_VOLUME_RIGHT: + case WM8903_ADC_DIGITAL_0: + case WM8903_DIGITAL_MICROPHONE_0: + case WM8903_DRC_0: + case WM8903_DRC_1: + case WM8903_DRC_2: + case WM8903_DRC_3: + case WM8903_ANALOGUE_LEFT_INPUT_0: + case WM8903_ANALOGUE_RIGHT_INPUT_0: + case WM8903_ANALOGUE_LEFT_INPUT_1: + case WM8903_ANALOGUE_RIGHT_INPUT_1: + case WM8903_ANALOGUE_LEFT_MIX_0: + case WM8903_ANALOGUE_RIGHT_MIX_0: + case WM8903_ANALOGUE_SPK_MIX_LEFT_0: + case WM8903_ANALOGUE_SPK_MIX_LEFT_1: + case WM8903_ANALOGUE_SPK_MIX_RIGHT_0: + case WM8903_ANALOGUE_SPK_MIX_RIGHT_1: + case WM8903_ANALOGUE_OUT1_LEFT: + case WM8903_ANALOGUE_OUT1_RIGHT: + case WM8903_ANALOGUE_OUT2_LEFT: + case WM8903_ANALOGUE_OUT2_RIGHT: + case WM8903_ANALOGUE_OUT3_LEFT: + case WM8903_ANALOGUE_OUT3_RIGHT: + case WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0: + case WM8903_DC_SERVO_0: + case WM8903_DC_SERVO_2: + case WM8903_DC_SERVO_READBACK_1: + case WM8903_DC_SERVO_READBACK_2: + case WM8903_DC_SERVO_READBACK_3: + case WM8903_DC_SERVO_READBACK_4: + case WM8903_ANALOGUE_HP_0: + case WM8903_ANALOGUE_LINEOUT_0: + case WM8903_CHARGE_PUMP_0: + case WM8903_CLASS_W_0: + case WM8903_WRITE_SEQUENCER_0: + case WM8903_WRITE_SEQUENCER_1: + case WM8903_WRITE_SEQUENCER_2: + case WM8903_WRITE_SEQUENCER_3: + case WM8903_WRITE_SEQUENCER_4: + case WM8903_CONTROL_INTERFACE: + case WM8903_GPIO_CONTROL_1: + case WM8903_GPIO_CONTROL_2: + case WM8903_GPIO_CONTROL_3: + case WM8903_GPIO_CONTROL_4: + case WM8903_GPIO_CONTROL_5: + case WM8903_INTERRUPT_STATUS_1: + case WM8903_INTERRUPT_STATUS_1_MASK: + case WM8903_INTERRUPT_POLARITY_1: + case WM8903_INTERRUPT_CONTROL: + case WM8903_CLOCK_RATE_TEST_4: + case WM8903_ANALOGUE_OUTPUT_BIAS_0: + return true; + default: + return false; + } +} + +static bool wm8903_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8903_SW_RESET_AND_ID: @@ -1767,7 +1754,7 @@ static int wm8903_resume(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - snd_soc_cache_sync(codec); + regcache_sync(wm8903->regmap); wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1897,15 +1884,16 @@ static int wm8903_probe(struct snd_soc_codec *codec) u16 val; wm8903->codec = codec; + codec->control_data = wm8903->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } val = snd_soc_read(codec, WM8903_SW_RESET_AND_ID); - if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { + if (val != 0x8903) { dev_err(codec->dev, "Device with ID register %x is not a WM8903\n", val); return -ENODEV; @@ -2044,10 +2032,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .suspend = wm8903_suspend, .resume = wm8903_resume, .set_bias_level = wm8903_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8903_reg_defaults, - .volatile_register = wm8903_volatile_register, .seq_notifier = wm8903_seq_notifier, .controls = wm8903_snd_controls, .num_controls = ARRAY_SIZE(wm8903_snd_controls), @@ -2057,6 +2041,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .num_dapm_routes = ARRAY_SIZE(wm8903_intercon), }; +static const struct regmap_config wm8903_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8903_MAX_REGISTER, + .volatile_reg = wm8903_volatile_register, + .readable_reg = wm8903_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8903_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8903_reg_defaults), +}; + static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2068,18 +2065,35 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, if (wm8903 == NULL) return -ENOMEM; + wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap); + if (IS_ERR(wm8903->regmap)) { + ret = PTR_ERR(wm8903->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + i2c_set_clientdata(i2c, wm8903); wm8903->irq = i2c->irq; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); + if (ret != 0) + goto err; + return 0; +err: + regmap_exit(wm8903->regmap); return ret; } static __devexit int wm8903_i2c_remove(struct i2c_client *client) { + struct wm8903_priv *wm8903 = i2c_get_clientdata(client); + + regmap_exit(wm8903->regmap); snd_soc_unregister_codec(&client->dev); + return 0; } -- cgit v1.2.1 From 7d46a528c609418e0a61121aac75edaf4992b622 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 18:39:17 +0000 Subject: ASoC: Move initial WM8903 identification and reset to I2C probe Get control of the device earlier and avoid trying to do an ASoC probe on a card that won't work. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/wm8903.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0b12a5525c15..a75688b5a568 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -244,11 +244,6 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg) } } -static void wm8903_reset(struct snd_soc_codec *codec) -{ - snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); -} - static int wm8903_cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1892,19 +1887,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - val = snd_soc_read(codec, WM8903_SW_RESET_AND_ID); - if (val != 0x8903) { - dev_err(codec->dev, - "Device with ID register %x is not a WM8903\n", val); - return -ENODEV; - } - - val = snd_soc_read(codec, WM8903_REVISION_NUMBER); - dev_info(codec->dev, "WM8903 revision %c\n", - (val & WM8903_CHIP_REV_MASK) + 'A'); - - wm8903_reset(codec); - /* Set up GPIOs and microphone detection */ if (pdata) { bool mic_gpio = false; @@ -2058,6 +2040,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8903_priv *wm8903; + unsigned int val; int ret; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), @@ -2076,6 +2059,28 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8903); wm8903->irq = i2c->irq; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + goto err; + } + if (val != 0x8903) { + dev_err(&i2c->dev, "Device with ID %x is not a WM8903\n", val); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(wm8903->regmap, WM8903_REVISION_NUMBER, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip revision: %d\n", ret); + goto err; + } + dev_info(&i2c->dev, "WM8903 revision %c\n", + (val & WM8903_CHIP_REV_MASK) + 'A'); + + /* Reset the device */ + regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) -- cgit v1.2.1 From c0eb27cf84ffd79347907f07ae33061ba0034c41 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 2 Dec 2011 15:08:38 -0700 Subject: ASoC: WM8903: Create default platform data structure When no platform data is supplied, point pdata at a default platform structure. This enables two future changes: a) Defines the default platform data values in a single place. b) There is always a valid pdata pointer, so some conditional code can be simplified by a later patch. Based on work by John Bonesio, but significantly reworked since then. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a75688b5a568..e6ecede576dc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -114,6 +114,7 @@ static const struct reg_default wm8903_reg_defaults[] = { }; struct wm8903_priv { + struct wm8903_platform_data *pdata; struct snd_soc_codec *codec; struct regmap *regmap; @@ -1834,7 +1835,7 @@ static struct gpio_chip wm8903_template_chip = { static void wm8903_init_gpio(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); + struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; @@ -1872,8 +1873,8 @@ static void wm8903_free_gpio(struct snd_soc_codec *codec) static int wm8903_probe(struct snd_soc_codec *codec) { - struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + struct wm8903_platform_data *pdata = wm8903->pdata; int ret, i; int trigger, irq_pol; u16 val; @@ -2039,6 +2040,7 @@ static const struct regmap_config wm8903_regmap = { static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; unsigned int val; int ret; @@ -2059,6 +2061,19 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8903); wm8903->irq = i2c->irq; + /* If no platform data was supplied, create storage for defaults */ + if (pdata) { + wm8903->pdata = pdata; + } else { + wm8903->pdata = devm_kzalloc(&i2c->dev, + sizeof(struct wm8903_platform_data), + GFP_KERNEL); + if (wm8903->pdata == NULL) { + dev_err(&i2c->dev, "Failed to allocate pdata\n"); + return -ENOMEM; + } + } + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); -- cgit v1.2.1 From 091edccf7f500837f2b3942be0d40362d25234c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 22:08:49 +0000 Subject: ASoC: Remove unused -codec from Wolfson device driver names Devices that aren't MFDs don't need to distinguish this. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8727.c | 2 +- sound/soc/codecs/wm8900.c | 4 ++-- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 5e847506138e..00f8dfa14b1c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -666,7 +666,7 @@ MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); static struct i2c_driver wm8510_i2c_driver = { .driver = { - .name = "wm8510-codec", + .name = "wm8510", .owner = THIS_MODULE, .of_match_table = wm8510_of_match, }, diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index fad90a35f399..e81705620718 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -59,7 +59,7 @@ static int __devexit wm8727_remove(struct platform_device *pdev) static struct platform_driver wm8727_codec_driver = { .driver = { - .name = "wm8727-codec", + .name = "wm8727", .owner = THIS_MODULE, }, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 85632ffcb872..e427a38032cc 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1258,7 +1258,7 @@ static int __devexit wm8900_spi_remove(struct spi_device *spi) static struct spi_driver wm8900_spi_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_spi_probe, @@ -1302,7 +1302,7 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f0ae01bbaa94..f31c754c8865 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2564,7 +2564,7 @@ MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); static struct i2c_driver wm8904_i2c_driver = { .driver = { - .name = "wm8904-codec", + .name = "wm8904", .owner = THIS_MODULE, }, .probe = wm8904_i2c_probe, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 0fe4545eef89..14039ea2f3e4 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -779,7 +779,7 @@ MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); static struct i2c_driver wm8940_i2c_driver = { .driver = { - .name = "wm8940-codec", + .name = "wm8940", .owner = THIS_MODULE, }, .probe = wm8940_i2c_probe, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index cdd51398e1f4..924548182d58 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1037,7 +1037,7 @@ MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); static struct i2c_driver wm8955_i2c_driver = { .driver = { - .name = "wm8955-codec", + .name = "wm8955", .owner = THIS_MODULE, }, .probe = wm8955_i2c_probe, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 55b9a25cd1b3..3446f9c25b83 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1030,7 +1030,7 @@ MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); static struct i2c_driver wm8960_i2c_driver = { .driver = { - .name = "wm8960-codec", + .name = "wm8960", .owner = THIS_MODULE, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 9bcf846e93b0..dc087c155975 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1116,7 +1116,7 @@ MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); static struct i2c_driver wm8961_i2c_driver = { .driver = { - .name = "wm8961-codec", + .name = "wm8961", .owner = THIS_MODULE, }, .probe = wm8961_i2c_probe, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index aadd14a14661..4af893601f00 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -724,7 +724,7 @@ MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); static struct i2c_driver wm8971_i2c_driver = { .driver = { - .name = "wm8971-codec", + .name = "wm8971", .owner = THIS_MODULE, }, .probe = wm8971_i2c_probe, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a5fd017c4332..4a6a7b5a61ba 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -671,7 +671,7 @@ MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); static struct i2c_driver wm8974_i2c_driver = { .driver = { - .name = "wm8974-codec", + .name = "wm8974", .owner = THIS_MODULE, }, .probe = wm8974_i2c_probe, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 093884705b01..ab52963dd04c 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -822,7 +822,7 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) static struct spi_driver wm8988_spi_driver = { .driver = { - .name = "wm8988-codec", + .name = "wm8988", .owner = THIS_MODULE, }, .probe = wm8988_spi_probe, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b417d2e0cdfd..e538edaae1f0 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1417,7 +1417,7 @@ MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { - .name = "wm8990-codec", + .name = "wm8990", .owner = THIS_MODULE, }, .probe = wm8990_i2c_probe, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 0f8278b4f0ad..f472ea6ecf6b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1641,7 +1641,7 @@ MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); static struct i2c_driver wm8993_i2c_driver = { .driver = { - .name = "wm8993-codec", + .name = "wm8993", .owner = THIS_MODULE, }, .probe = wm8993_i2c_probe, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 5cb8759868df..31869afa7007 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -685,7 +685,7 @@ MODULE_DEVICE_TABLE(i2c, wm9090_id); static struct i2c_driver wm9090_i2c_driver = { .driver = { - .name = "wm9090-codec", + .name = "wm9090", .owner = THIS_MODULE, }, .probe = wm9090_i2c_probe, -- cgit v1.2.1 From 3a0d077f3d013811cc9f2208089d765ae79a2695 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:20:37 +0000 Subject: ASoC: Remove I2C ifdefs from WM8960 The driver only supports I2C as the control interface. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 3446f9c25b83..ee8d97f56bf3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -994,7 +994,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .reg_cache_default = wm8960_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1037,27 +1036,22 @@ static struct i2c_driver wm8960_i2c_driver = { .remove = __devexit_p(wm8960_i2c_remove), .id_table = wm8960_i2c_id, }; -#endif static int __init wm8960_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8960_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8960_modinit); static void __exit wm8960_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8960_i2c_driver); -#endif } module_exit(wm8960_exit); -- cgit v1.2.1 From 6cd4eb959294990cbf38051fd9ab1e9091b3e926 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:20:58 +0000 Subject: ASoC: Remove unused AUDIO_NAME define from WM8960 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ee8d97f56bf3..6a9c41d351de 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -25,8 +25,6 @@ #include "wm8960.h" -#define AUDIO_NAME "wm8960" - /* R25 - Power 1 */ #define WM8960_VMID_MASK 0x180 #define WM8960_VREF 0x40 -- cgit v1.2.1 From b03e96e4d619183cbe9aea55f2340596c1fecf64 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:28:31 +0000 Subject: ASoC: Convert WM2000 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 01b1abe7a36b..2726f6651990 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -740,7 +740,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, return -EINVAL; } - wm2000 = kzalloc(sizeof(struct wm2000_priv), GFP_KERNEL); + wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), + GFP_KERNEL); if (wm2000 == NULL) { dev_err(&i2c->dev, "Unable to allocate private data\n"); return -ENOMEM; @@ -779,7 +780,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, /* Pre-cook the concatenation of the register address onto the image */ wm2000->anc_download_size = fw->size + 2; - wm2000->anc_download = kmalloc(wm2000->anc_download_size, GFP_KERNEL); + wm2000->anc_download = devm_kzalloc(&i2c->dev, + wm2000->anc_download_size, + GFP_KERNEL); if (wm2000->anc_download == NULL) { dev_err(&i2c->dev, "Out of memory\n"); ret = -ENOMEM; @@ -810,7 +813,6 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, err_fw: release_firmware(fw); err: - kfree(wm2000); return ret; } @@ -821,8 +823,6 @@ static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) wm2000_anc_transition(wm2000, ANC_OFF); wm2000_i2c = NULL; - kfree(wm2000->anc_download); - kfree(wm2000); return 0; } -- cgit v1.2.1 From 0d1fe0d4521436d8af2111045a682c4c8aa1b55d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:29:38 +0000 Subject: ASoC: Convert WM8350 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index f39497fc13e0..7b095aeef695 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1553,7 +1553,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -EINVAL; } - priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL); + priv = devm_kzalloc(codec->dev, sizeof(struct wm8350_data), + GFP_KERNEL); if (priv == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); @@ -1564,7 +1565,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies), priv->supplies); if (ret != 0) - goto err_priv; + return ret; wm8350->codec.codec = codec; codec->control_data = wm8350; @@ -1640,10 +1641,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; - -err_priv: - kfree(priv); - return ret; } static int wm8350_codec_remove(struct snd_soc_codec *codec) @@ -1676,7 +1673,7 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies); - kfree(priv); + return 0; } -- cgit v1.2.1 From b903c0ed2e85155c3a67cfc54117223a61bb483f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:41:27 +0000 Subject: ASoC: Convert WM8400 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 56a7b7256efa..aef7e4dcefdd 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1383,7 +1383,8 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) int ret; u16 reg; - priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL); + priv = devm_kzalloc(codec->dev, sizeof(struct wm8400_priv), + GFP_KERNEL); if (priv == NULL) return -ENOMEM; @@ -1395,7 +1396,7 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(power), &power[0]); if (ret != 0) { dev_err(codec->dev, "Failed to get regulators: %d\n", ret); - goto err; + return ret; } INIT_WORK(&priv->work, wm8400_probe_deferred); @@ -1426,14 +1427,11 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) err_regulator: regulator_bulk_free(ARRAY_SIZE(power), power); -err: - kfree(priv); return ret; } static int wm8400_codec_remove(struct snd_soc_codec *codec) { - struct wm8400_priv *priv = snd_soc_codec_get_drvdata(codec); u16 reg; reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); @@ -1441,7 +1439,6 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) reg & (~WM8400_CODEC_ENA)); regulator_bulk_free(ARRAY_SIZE(power), power); - kfree(priv); return 0; } -- cgit v1.2.1 From 5aefb306e35541d35c8d5838ae97f3f9d8ad1a12 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 17:17:05 +0000 Subject: ASoC: Convert WM8741 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index bf471dc57114..24d8ec53f35b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -503,7 +503,8 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, struct wm8741_priv *wm8741; int ret; - wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + wm8741 = devm_kzalloc(&i2c->dev, sizeof(struct wm8741_priv), + GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; @@ -512,20 +513,13 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret != 0) - goto err; return ret; - -err: - kfree(wm8741); - return ret; } static int wm8741_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } @@ -553,7 +547,8 @@ static int __devinit wm8741_spi_probe(struct spi_device *spi) struct wm8741_priv *wm8741; int ret; - wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + wm8741 = devm_kzalloc(&spi->dev, sizeof(struct wm8741_priv), + GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; @@ -562,15 +557,12 @@ static int __devinit wm8741_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret < 0) - kfree(wm8741); return ret; } static int __devexit wm8741_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- cgit v1.2.1 From 2edaed82b70c22b63bb918e1ca9c34876da21320 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 17:18:37 +0000 Subject: ASoC: Convert WM8750 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 48cb78fd0103..fa5732d78225 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -744,7 +744,8 @@ static int __devinit wm8750_spi_probe(struct spi_device *spi) struct wm8750_priv *wm8750; int ret; - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + wm8750 = devm_kzalloc(&spi->dev, sizeof(struct wm8750_priv), + GFP_KERNEL); if (wm8750 == NULL) return -ENOMEM; @@ -753,15 +754,12 @@ static int __devinit wm8750_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8750, &wm8750_dai, 1); - if (ret < 0) - kfree(wm8750); return ret; } static int __devexit wm8750_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -791,7 +789,8 @@ static __devinit int wm8750_i2c_probe(struct i2c_client *i2c, struct wm8750_priv *wm8750; int ret; - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + wm8750 = devm_kzalloc(&i2c->dev, sizeof(struct wm8750_priv), + GFP_KERNEL); if (wm8750 == NULL) return -ENOMEM; @@ -800,15 +799,12 @@ static __devinit int wm8750_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8750, &wm8750_dai, 1); - if (ret < 0) - kfree(wm8750); return ret; } static __devexit int wm8750_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 455b91bfe86fd4773a15593eb7a834b9f552797d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 22:09:47 +0000 Subject: ASoC: Convert WM9090 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 31869afa7007..d1d2c703eab2 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -647,7 +647,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, struct wm9090_priv *wm9090; int ret; - wm9090 = kzalloc(sizeof(*wm9090), GFP_KERNEL); + wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL); if (wm9090 == NULL) { dev_err(&i2c->dev, "Can not allocate memory\n"); return -ENOMEM; @@ -661,8 +661,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); - if (ret < 0) - kfree(wm9090); return ret; } @@ -671,7 +669,6 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); snd_soc_unregister_codec(&i2c->dev); - kfree(wm9090); return 0; } -- cgit v1.2.1 From e6c94e9f6dd77c928419dc05af2b3d17ed9463b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:31:58 +0000 Subject: ASoC: Convert WM8350 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 36 ++++++++---------------------------- 1 file changed, 8 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 7b095aeef695..8c4c9591ec05 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -696,7 +696,7 @@ static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN3L"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8350_dapm_routes[] = { /* left playback mixer */ {"Left Playback Mixer", "Playback Switch", "Left DAC"}, @@ -777,29 +777,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Beep", NULL, "IN3R PGA"}, }; -static int wm8350_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, - wm8350_dapm_widgets, - ARRAY_SIZE(wm8350_dapm_widgets)); - if (ret != 0) { - dev_err(codec->dev, "dapm control register failed\n"); - return ret; - } - - /* set up audio paths */ - ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (ret != 0) { - dev_err(codec->dev, "DAPM route register failed\n"); - return ret; - } - - return 0; -} - static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -1634,10 +1611,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_mic_handler, 0, "Microphone detect", priv); - snd_soc_add_controls(codec, wm8350_snd_controls, - ARRAY_SIZE(wm8350_snd_controls)); - wm8350_add_widgets(codec); - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -1685,6 +1658,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .read = wm8350_codec_read, .write = wm8350_codec_write, .set_bias_level = wm8350_set_bias_level, + + .controls = wm8350_snd_controls, + .num_controls = ARRAY_SIZE(wm8350_snd_controls), + .dapm_widgets = wm8350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8350_dapm_widgets), + .dapm_routes = wm8350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8350_dapm_routes), }; static int __devinit wm8350_probe(struct platform_device *pdev) -- cgit v1.2.1 From b4505ab141a72f65bf7bb1f7c120411ab129181a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:34:34 +0000 Subject: ASoC: Convert WM8400 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 29 ++++++++--------------------- 1 file changed, 8 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index aef7e4dcefdd..898979d23010 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -353,13 +353,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, }; -/* add non dapm controls */ -static int wm8400_add_controls(struct snd_soc_codec *codec) -{ - return snd_soc_add_controls(codec, wm8400_snd_controls, - ARRAY_SIZE(wm8400_snd_controls)); -} - /* * _DAPM_ Controls */ @@ -783,7 +776,7 @@ SND_SOC_DAPM_OUTPUT("RON"), SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -909,17 +902,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RON", NULL, "RONMIX"}, }; -static int wm8400_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8400_dapm_widgets, - ARRAY_SIZE(wm8400_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* * Clock after FLL and dividers */ @@ -1421,8 +1403,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) ret = -EINVAL; goto err_regulator; } - wm8400_add_controls(codec); - wm8400_add_widgets(codec); return 0; err_regulator: @@ -1451,6 +1431,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .read = wm8400_read, .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, + + .controls = wm8400_snd_controls, + .num_controls = ARRAY_SIZE(wm8400_snd_controls), + .dapm_widgets = wm8400_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8400_dapm_widgets), + .dapm_routes = wm8400_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8400_dapm_routes), }; static int __devinit wm8400_probe(struct platform_device *pdev) -- cgit v1.2.1 From b6709f3bbd7550fd4a10943513df72e7fa41c962 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:41:45 +0000 Subject: ASoC: Convert WM8510 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 00f8dfa14b1c..9166126bd312 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -181,7 +181,7 @@ SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8510_dapm_routes[] = { /* Mono output mixer */ {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, @@ -213,17 +213,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ADC", NULL, "Boost Mixer"}, }; -static int wm8510_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8510_dapm_widgets, - ARRAY_SIZE(wm8510_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - struct pll_ { unsigned int pre_div:4; /* prescale - 1 */ unsigned int n:4; @@ -561,9 +550,6 @@ static int wm8510_probe(struct snd_soc_codec *codec) /* power on device */ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8510_snd_controls, - ARRAY_SIZE(wm8510_snd_controls)); - wm8510_add_widgets(codec); return ret; } @@ -587,6 +573,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .reg_cache_size = ARRAY_SIZE(wm8510_reg), .reg_word_size = sizeof(u16), .reg_cache_default =wm8510_reg, + + .controls = wm8510_snd_controls, + .num_controls = ARRAY_SIZE(wm8510_snd_controls), + .dapm_widgets = wm8510_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8510_dapm_widgets), + .dapm_routes = wm8510_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8510_dapm_routes), }; static const struct of_device_id wm8510_of_match[] = { -- cgit v1.2.1 From f235c649c1301ae85d5c7e51e88b13adb30ed6a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:42:01 +0000 Subject: ASoC: Convert WM8580 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 764b2bf80a71..b1c8d3de08b2 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -272,7 +272,7 @@ SND_SOC_DAPM_INPUT("AINL"), SND_SOC_DAPM_INPUT("AINR"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8580_dapm_routes[] = { { "VOUT1L", NULL, "DAC1" }, { "VOUT1R", NULL, "DAC1" }, @@ -286,17 +286,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "ADC", NULL, "AINR" }, }; -static int wm8580_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets, - ARRAY_SIZE(wm8580_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct _pll_div { u32 prescale:1; @@ -856,10 +845,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8580_snd_controls, - ARRAY_SIZE(wm8580_snd_controls)); - wm8580_add_widgets(codec); - return 0; err_regulator_enable: @@ -889,6 +874,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .reg_cache_size = ARRAY_SIZE(wm8580_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8580_reg, + + .controls = wm8580_snd_controls, + .num_controls = ARRAY_SIZE(wm8580_snd_controls), + .dapm_widgets = wm8580_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8580_dapm_widgets), + .dapm_routes = wm8580_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8580_dapm_routes), }; static const struct of_device_id wm8580_of_match[] = { -- cgit v1.2.1 From 0e62780f5f27f24a30d5a08ed731088115e1fe80 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 17:15:06 +0000 Subject: ASoC: Convert WM8741 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 24d8ec53f35b..3941f50bf187 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -85,24 +85,13 @@ SND_SOC_DAPM_OUTPUT("VOUTRP"), SND_SOC_DAPM_OUTPUT("VOUTRN"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8741_dapm_routes[] = { { "VOUTLP", NULL, "DACL" }, { "VOUTLN", NULL, "DACL" }, { "VOUTRP", NULL, "DACR" }, { "VOUTRN", NULL, "DACR" }, }; -static int wm8741_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8741_dapm_widgets, - ARRAY_SIZE(wm8741_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static struct { int value; int ratio; @@ -456,10 +445,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, WM8741_UPDATERM, WM8741_UPDATERM); - snd_soc_add_controls(codec, wm8741_snd_controls, - ARRAY_SIZE(wm8741_snd_controls)); - wm8741_add_widgets(codec); - dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -488,6 +473,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8741_reg_defaults, + + .controls = wm8741_snd_controls, + .num_controls = ARRAY_SIZE(wm8741_snd_controls), + .dapm_widgets = wm8741_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), + .dapm_routes = wm8741_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8741_dapm_routes), }; static const struct of_device_id wm8741_of_match[] = { -- cgit v1.2.1 From 0f185e3f8b06c1d0fd817af3d4c03f9c21d776e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 17:21:43 +0000 Subject: ASoC: Convert WM8750 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index fa5732d78225..e4c50ce7d9c0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -301,7 +301,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT3"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8750_dapm_routes[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, @@ -395,17 +395,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "Right ADC Mux"}, }; -static int wm8750_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - struct _coeff_div { u32 mclk; u32 rate; @@ -708,9 +697,6 @@ static int wm8750_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8750_LINVOL, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8750_RINVOL, 0x0100, 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); return ret; } @@ -729,6 +715,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .reg_cache_size = ARRAY_SIZE(wm8750_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8750_reg, + + .controls = wm8750_snd_controls, + .num_controls = ARRAY_SIZE(wm8750_snd_controls), + .dapm_widgets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = wm8750_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8750_dapm_routes), }; static const struct of_device_id wm8750_of_match[] = { -- cgit v1.2.1 From 2f5374d8cf05d8b71f593633bf20972102f591c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:47:23 +0000 Subject: ASoC: Convert WM8711 to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index b9b1a2f8360f..0b76d1dca5ea 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -374,9 +374,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); - snd_soc_add_controls(codec, wm8711_snd_controls, - ARRAY_SIZE(wm8711_snd_controls)); - return ret; } @@ -397,6 +394,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .reg_cache_size = ARRAY_SIZE(wm8711_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8711_reg, + .controls = wm8711_snd_controls, + .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8711_dapm_widgets), .dapm_routes = wm8711_intercon, -- cgit v1.2.1 From e41d5a3b7a04e9b82279293055d09cb8164ec29e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 11:49:02 +0000 Subject: ASoC: Convert WM8728 to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index b1f01d9273be..fc3d59e49084 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -243,9 +243,6 @@ static int wm8728_probe(struct snd_soc_codec *codec) /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8728_snd_controls, - ARRAY_SIZE(wm8728_snd_controls)); - return ret; } @@ -264,6 +261,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8728_reg_defaults, + .controls = wm8728_snd_controls, + .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8728_dapm_widgets), .dapm_routes = wm8728_intercon, -- cgit v1.2.1 From 012d12db0c42119356f3ff876289b191c2e730a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:29:50 +0000 Subject: ASoC: Remove unused struct wm2000_setup_data Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.h | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h index 0b6f056f73cc..28a51ed5dc41 100644 --- a/sound/soc/codecs/wm2000.h +++ b/sound/soc/codecs/wm2000.h @@ -9,11 +9,6 @@ #ifndef _WM2000_H #define _WM2000_H -struct wm2000_setup_data { - unsigned short i2c_address; - int mclk_div; /* Set to a non-zero value if MCLK_DIV_2 required */ -}; - extern int wm2000_add_controls(struct snd_soc_codec *codec); #define WM2000_REG_SYS_START 0x8000 -- cgit v1.2.1 From 8aa1fe81c56d98e484f6d8dfc7ac434dad9acd1c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:57:19 +0000 Subject: ASoC: Convert wm2000 to use regmap API The driver wasn't even using the ASoC common code. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 75 ++++++++++++++++++++--------------------------- 1 file changed, 31 insertions(+), 44 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 2726f6651990..a5f57ce44665 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -51,6 +52,7 @@ enum wm2000_anc_mode { struct wm2000_priv { struct i2c_client *i2c; + struct regmap *regmap; enum wm2000_anc_mode anc_mode; @@ -70,54 +72,21 @@ static struct i2c_client *wm2000_i2c; static int wm2000_write(struct i2c_client *i2c, unsigned int reg, unsigned int value) { - u8 data[3]; - int ret; - - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = value & 0xff; - - dev_vdbg(&i2c->dev, "write %x = %x\n", reg, value); - - ret = i2c_master_send(i2c, data, 3); - if (ret == 3) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); + return regmap_write(wm2000->regmap, reg, value); } static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) { - struct i2c_msg xfer[2]; - u8 reg[2]; - u8 data; + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); + unsigned int val; int ret; - /* Write register */ - reg[0] = (r >> 8) & 0xff; - reg[1] = r & 0xff; - xfer[0].addr = i2c->addr; - xfer[0].flags = 0; - xfer[0].len = sizeof(reg); - xfer[0].buf = ®[0]; - - /* Read data */ - xfer[1].addr = i2c->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 1; - xfer[1].buf = &data; - - ret = i2c_transfer(i2c->adapter, xfer, 2); - if (ret != 2) { - dev_err(&i2c->dev, "i2c_transfer() returned %d\n", ret); - return 0; - } - - dev_vdbg(&i2c->dev, "read %x from %x\n", data, r); + ret = regmap_read(wm2000->regmap, r, &val); + if (ret < 0) + return -1; - return data; + return val; } static void wm2000_reset(struct wm2000_priv *wm2000) @@ -725,6 +694,11 @@ int wm2000_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(wm2000_add_controls); +static const struct regmap_config wm2000_regmap = { + .reg_bits = 8, + .val_bits = 8, +}; + static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { @@ -747,6 +721,16 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } + dev_set_drvdata(&i2c->dev, wm2000); + + wm2000->regmap = regmap_init_i2c(i2c, &wm2000_regmap); + if (IS_ERR(wm2000->regmap)) { + ret = PTR_ERR(wm2000->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + /* Verify that this is a WM2000 */ reg = wm2000_read(i2c, WM2000_REG_ID1); id = reg << 8; @@ -756,7 +740,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (id != 0x2000) { dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); ret = -ENODEV; - goto err; + goto err_regmap; } reg = wm2000_read(i2c, WM2000_REG_REVISON); @@ -775,7 +759,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = request_firmware(&fw, filename, &i2c->dev); if (ret != 0) { dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); - goto err; + goto err_regmap; } /* Pre-cook the concatenation of the register address onto the image */ @@ -795,7 +779,6 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, release_firmware(fw); - dev_set_drvdata(&i2c->dev, wm2000); wm2000->anc_eng_ena = 1; wm2000->anc_active = 1; wm2000->spk_ena = 1; @@ -812,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, err_fw: release_firmware(fw); +err_regmap: + regmap_exit(wm2000->regmap); err: return ret; } @@ -822,6 +807,8 @@ static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) wm2000_anc_transition(wm2000, ANC_OFF); + regmap_exit(wm2000->regmap); + wm2000_i2c = NULL; return 0; -- cgit v1.2.1 From 4911ccdb9d052d0389353cec5cc591a3669f39cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 21:59:18 +0000 Subject: ASoC: Convert WM2000 into a standard CODEC driver We've been able to handle external amps for a while now. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/wm2000.c | 145 ++++++++++++++++++++++------------------------ sound/soc/codecs/wm2000.h | 2 - 4 files changed, 73 insertions(+), 84 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 593174c78d7b..08e9d40e533a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -289,6 +289,9 @@ config SND_SOC_WL1273 config SND_SOC_WM1250_EV1 tristate +config SND_SOC_WM2000 + tristate + config SND_SOC_WM5100 tristate @@ -425,8 +428,5 @@ config SND_SOC_MAX9877 config SND_SOC_TPA6130A2 tristate -config SND_SOC_WM2000 - tristate - config SND_SOC_WM9090 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fa15006fcac5..adfa22ea2938 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -50,6 +50,7 @@ snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o +snd-soc-wm2000-objs := wm2000.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o @@ -97,7 +98,6 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o -snd-soc-wm2000-objs := wm2000.o snd-soc-wm9090-objs := wm9090.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o @@ -152,6 +152,7 @@ obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o +obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o @@ -199,5 +200,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o -obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a5f57ce44665..c2880907fced 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -67,8 +67,6 @@ struct wm2000_priv { char *anc_download; }; -static struct i2c_client *wm2000_i2c; - static int wm2000_write(struct i2c_client *i2c, unsigned int reg, unsigned int value) { @@ -580,7 +578,8 @@ static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); ucontrol->value.enumerated.item[0] = wm2000->anc_active; @@ -590,7 +589,8 @@ static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int anc_active = ucontrol->value.enumerated.item[0]; if (anc_active > 1) @@ -604,7 +604,8 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); ucontrol->value.enumerated.item[0] = wm2000->spk_ena; @@ -614,7 +615,8 @@ static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int val = ucontrol->value.enumerated.item[0]; if (val > 1) @@ -637,7 +639,8 @@ static const struct snd_kcontrol_new wm2000_controls[] = { static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); if (SND_SOC_DAPM_EVENT_ON(event)) wm2000->anc_eng_ena = 1; @@ -650,11 +653,11 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { /* Externally visible pins */ -SND_SOC_DAPM_OUTPUT("WM2000 SPKN"), -SND_SOC_DAPM_OUTPUT("WM2000 SPKP"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), -SND_SOC_DAPM_INPUT("WM2000 LINN"), -SND_SOC_DAPM_INPUT("WM2000 LINP"), +SND_SOC_DAPM_INPUT("LINN"), +SND_SOC_DAPM_INPUT("LINP"), SND_SOC_DAPM_PGA_E("ANC Engine", SND_SOC_NOPM, 0, 0, NULL, 0, wm2000_anc_power_event, @@ -662,43 +665,68 @@ SND_SOC_DAPM_PGA_E("ANC Engine", SND_SOC_NOPM, 0, 0, NULL, 0, }; /* Target, Path, Source */ -static const struct snd_soc_dapm_route audio_map[] = { - { "WM2000 SPKN", NULL, "ANC Engine" }, - { "WM2000 SPKP", NULL, "ANC Engine" }, - { "ANC Engine", NULL, "WM2000 LINN" }, - { "ANC Engine", NULL, "WM2000 LINP" }, +static const struct snd_soc_dapm_route wm2000_audio_map[] = { + { "SPKN", NULL, "ANC Engine" }, + { "SPKP", NULL, "ANC Engine" }, + { "ANC Engine", NULL, "LINN" }, + { "ANC Engine", NULL, "LINP" }, }; -/* Called from the machine driver */ -int wm2000_add_controls(struct snd_soc_codec *codec) +#ifdef CONFIG_PM +static int wm2000_suspend(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - if (!wm2000_i2c) { - pr_err("WM2000 not yet probed\n"); - return -ENODEV; - } - - ret = snd_soc_dapm_new_controls(dapm, wm2000_dapm_widgets, - ARRAY_SIZE(wm2000_dapm_widgets)); - if (ret < 0) - return ret; + return wm2000_anc_transition(wm2000, ANC_OFF); +} - ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (ret < 0) - return ret; +static int wm2000_resume(struct snd_soc_codec *codec) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - return snd_soc_add_controls(codec, wm2000_controls, - ARRAY_SIZE(wm2000_controls)); + return wm2000_anc_set_mode(wm2000); } -EXPORT_SYMBOL_GPL(wm2000_add_controls); +#else +#define wm2000_suspend NULL +#define wm2000_resume NULL +#endif static const struct regmap_config wm2000_regmap = { .reg_bits = 8, .val_bits = 8, }; +static int wm2000_probe(struct snd_soc_codec *codec) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + + /* This will trigger a transition to standby mode by default */ + wm2000_anc_set_mode(wm2000); + + return 0; +} + +static int wm2000_remove(struct snd_soc_codec *codec) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + + return wm2000_anc_transition(wm2000, ANC_OFF); +} + +static struct snd_soc_codec_driver soc_codec_dev_wm2000 = { + .probe = wm2000_probe, + .remove = wm2000_remove, + .suspend = wm2000_suspend, + .resume = wm2000_resume, + + .dapm_widgets = wm2000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + .dapm_routes = wm2000_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), + .controls = wm2000_controls, + .num_controls = ARRAY_SIZE(wm2000_controls), +}; + static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { @@ -709,11 +737,6 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, int reg, ret; u16 id; - if (wm2000_i2c) { - dev_err(&i2c->dev, "Another WM2000 is already registered\n"); - return -EINVAL; - } - wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), GFP_KERNEL); if (wm2000 == NULL) { @@ -786,10 +809,10 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, wm2000_reset(wm2000); - /* This will trigger a transition to standby mode by default */ - wm2000_anc_set_mode(wm2000); - - wm2000_i2c = i2c; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, + NULL, 0); + if (ret != 0) + goto err_fw; return 0; @@ -805,42 +828,12 @@ static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - wm2000_anc_transition(wm2000, ANC_OFF); - + snd_soc_unregister_codec(&i2c->dev); regmap_exit(wm2000->regmap); - wm2000_i2c = NULL; - return 0; } -static void wm2000_i2c_shutdown(struct i2c_client *i2c) -{ - struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - - wm2000_anc_transition(wm2000, ANC_OFF); -} - -#ifdef CONFIG_PM -static int wm2000_i2c_suspend(struct device *dev) -{ - struct i2c_client *i2c = to_i2c_client(dev); - struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - - return wm2000_anc_transition(wm2000, ANC_OFF); -} - -static int wm2000_i2c_resume(struct device *dev) -{ - struct i2c_client *i2c = to_i2c_client(dev); - struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - - return wm2000_anc_set_mode(wm2000); -} -#endif - -static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume); - static const struct i2c_device_id wm2000_i2c_id[] = { { "wm2000", 0 }, { } @@ -851,11 +844,9 @@ static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", .owner = THIS_MODULE, - .pm = &wm2000_pm, }, .probe = wm2000_i2c_probe, .remove = __devexit_p(wm2000_i2c_remove), - .shutdown = wm2000_i2c_shutdown, .id_table = wm2000_i2c_id, }; diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h index 28a51ed5dc41..abcd82a93995 100644 --- a/sound/soc/codecs/wm2000.h +++ b/sound/soc/codecs/wm2000.h @@ -9,8 +9,6 @@ #ifndef _WM2000_H #define _WM2000_H -extern int wm2000_add_controls(struct snd_soc_codec *codec); - #define WM2000_REG_SYS_START 0x8000 #define WM2000_REG_SPEECH_CLARITY 0x8fef #define WM2000_REG_SYS_WATCHDOG 0x8ff6 -- cgit v1.2.1 From 59792aa91fa90ab89f58152afa09d6447fdfc754 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Dec 2011 22:00:39 +0000 Subject: ASoC: Sort WM9090 in with the CODEC drivers The driver itself has been a regular CODEC driver for a while now. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 +++--- sound/soc/codecs/Makefile | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 08e9d40e533a..bc2364aced1b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -409,6 +409,9 @@ config SND_SOC_WM8996 config SND_SOC_WM9081 tristate +config SND_SOC_WM9090 + tristate + config SND_SOC_WM9705 tristate @@ -427,6 +430,3 @@ config SND_SOC_MAX9877 config SND_SOC_TPA6130A2 tristate - -config SND_SOC_WM9090 - tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index adfa22ea2938..9aa6e669e6ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -90,6 +90,7 @@ snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8994-tables.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o snd-soc-wm9081-objs := wm9081.o +snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -98,7 +99,6 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o -snd-soc-wm9090-objs := wm9090.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o @@ -192,6 +192,7 @@ obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o +obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o @@ -200,4 +201,3 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o -obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o -- cgit v1.2.1 From dd85ecc269a3aa537fd045e113b4755a3cd1284f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 4 Dec 2011 08:15:17 +0800 Subject: ASoC: Make SND_SOC_LITTLEMILL select MFD_WM8994 SND_SOC_LITTLEMILL selects SND_SOC_WM8994, but SND_SOC_WM8994 needs MFD_WM8994. Thus we need to select MFD_WM8994 to fix below build error: LD .tmp_vmlinux1 sound/built-in.o: In function `wm8994_write': sound/soc/codecs/wm8994.c:201: undefined reference to `wm8994_reg_write' sound/built-in.o: In function `wm8994_read': sound/soc/codecs/wm8994.c:222: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_resume': sound/soc/codecs/wm8994.c:2847: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_codec_probe': sound/soc/codecs/wm8994.c:3501: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3660: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3672: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8958_dsp2_fw': sound/soc/codecs/wm8958-dsp2.c:154: undefined reference to `wm8994_bulk_write' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 09d636cc3658..f3417f2311b8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -210,4 +210,5 @@ config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 -- cgit v1.2.1 From aec60f51e5127fb750b66eb7905047c67372177f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 5 Dec 2011 17:09:11 +0800 Subject: ASoC: Convert e740_wm9705 to use gpio_request_one() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e740_wm9705.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 35ed7eb8cff2..818dc57b0b2f 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -146,29 +146,21 @@ static int __init e740_init(void) if (!machine_is_e740()) return -ENODEV; - ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + /* Disable audio */ + ret = gpio_request_one(GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp"); if (ret) return ret; - ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + ret = gpio_request_one(GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, + "Output amp"); if (ret) goto free_mic_amp_gpio; - ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + ret = gpio_request_one(GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, + "Audio power"); if (ret) goto free_op_amp_gpio; - /* Disable audio */ - ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); - if (ret) - goto free_apwr_gpio; - e740_snd_device = platform_device_alloc("soc-audio", -1); if (!e740_snd_device) { ret = -ENOMEM; -- cgit v1.2.1 From 68020db8ac1046e50c758545b75850eb356a0651 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 5 Dec 2011 07:58:25 +0800 Subject: ASoC: uda1380: Convert to gpio_request_one() Using gpio_request_one can make the error handling simpler. Also remove a redundant "Failed to issue reset" error message. We already show the error message in uda1380_reset() error path. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 28 ++++++++-------------------- 1 file changed, 8 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 39c228c89e46..83e45d2b3e84 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -732,27 +732,21 @@ static int uda1380_probe(struct snd_soc_codec *codec) return -EINVAL; if (gpio_is_valid(pdata->gpio_reset)) { - ret = gpio_request(pdata->gpio_reset, "uda1380 reset"); + ret = gpio_request_one(pdata->gpio_reset, GPIOF_OUT_INIT_LOW, + "uda1380 reset"); if (ret) goto err_out; - ret = gpio_direction_output(pdata->gpio_reset, 0); - if (ret) - goto err_gpio_reset_conf; } if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request(pdata->gpio_power, "uda1380 power"); - if (ret) - goto err_gpio; - ret = gpio_direction_output(pdata->gpio_power, 0); + ret = gpio_request_one(pdata->gpio_power, GPIOF_OUT_INIT_LOW, + "uda1380 power"); if (ret) - goto err_gpio_power_conf; + goto err_free_gpio; } else { ret = uda1380_reset(codec); - if (ret) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_reset; - } + if (ret) + goto err_free_gpio; } INIT_WORK(&uda1380->work, uda1380_flush_work); @@ -776,13 +770,7 @@ static int uda1380_probe(struct snd_soc_codec *codec) return 0; -err_reset: -err_gpio_power_conf: - if (gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); - -err_gpio_reset_conf: -err_gpio: +err_free_gpio: if (gpio_is_valid(pdata->gpio_reset)) gpio_free(pdata->gpio_reset); err_out: -- cgit v1.2.1 From f031efe9402e4ab6a6cd86bbda54b30ed9171237 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 5 Dec 2011 10:06:04 +0800 Subject: ASoC: Fix reg_cache_size for stac9766 reg_cache_size is supposed to be the number of elements in the register cache, not the size in bytes. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e34969cdc0e8..cc0566c22ec1 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -379,7 +379,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, - .reg_cache_size = sizeof(stac9766_reg), + .reg_cache_size = ARRAY_SIZE(stac9766_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = stac9766_reg, -- cgit v1.2.1 From 03c33042dbcd087303062c51f462c4575eb630d6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 5 Dec 2011 19:13:41 +0530 Subject: ASoC: sst_platform: fix the dsp driver interface lower level drivers typically register with upper layers. So fix by exporting symbols from sst_platform driver for dsp driver to register to sst platform driver Now this driver doesnt depend on sst driver, so remove the dependency and the header files Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/mid-x86/Kconfig | 1 - sound/soc/mid-x86/sst_platform.c | 130 +++++++++++++++++++++++++-------------- sound/soc/mid-x86/sst_platform.h | 82 +++++++++++++++++++++--- 3 files changed, 158 insertions(+), 55 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index 29350428f1c2..61c10bf503d2 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,7 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC - depends on SND_INTEL_SST select SND_SOC_SN95031 select SND_SST_PLATFORM help diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 94f70b3f94e6..24f947146947 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -32,10 +32,51 @@ #include #include #include -#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h" -#include "../../../drivers/staging/intel_sst/intel_sst.h" #include "sst_platform.h" +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + static struct snd_pcm_hardware sst_platform_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_DOUBLE | @@ -135,37 +176,34 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) } static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct snd_sst_stream_params *param) + struct sst_pcm_params *param) { - param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM; - param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; - param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; - param->uc.pcm_params.reserved = 0; - param->uc.pcm_params.sfreq = substream->runtime->rate; - param->uc.pcm_params.ring_buffer_size = - snd_pcm_lib_buffer_bytes(substream); - param->uc.pcm_params.period_count = substream->runtime->period_size; - param->uc.pcm_params.ring_buffer_addr = - virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", - param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); + param->codec = SST_CODEC_TYPE_PCM; + param->num_chan = (u8) substream->runtime->channels; + param->pcm_wd_sz = substream->runtime->sample_bits; + param->reserved = 0; + param->sfreq = substream->runtime->rate; + param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); + param->period_count = substream->runtime->period_size; + param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct snd_sst_stream_params param = {{{0,},},}; - struct snd_sst_params str_params = {0}; + struct sst_pcm_params param = {0}; + struct sst_stream_params str_params = {0}; int ret_val; /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.uc.pcm_params.codec; + str_params.codec = param.codec; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { str_params.ops = STREAM_OPS_PLAYBACK; str_params.device_type = substream->pcm->device + 1; @@ -177,7 +215,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) pr_debug("Capture stream,Device %d\n", substream->pcm->device); } - ret_val = stream->sstdrv_ops->pcm_control->open(&str_params); + ret_val = stream->ops->open(&str_params); pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); if (ret_val < 0) return ret_val; @@ -216,7 +254,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_STREAM_INIT, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); @@ -229,7 +267,6 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val = 0; pr_debug("sst_platform_open called\n"); @@ -243,27 +280,27 @@ static int sst_platform_open(struct snd_pcm_substream *substream) if (!stream) return -ENOMEM; spin_lock_init(&stream->status_lock); - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), - GFP_KERNEL); - if (!stream->sstdrv_ops) { - pr_err("sst: mem allocation for ops fail\n"); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); kfree(stream); - return -ENOMEM; + return -ENODEV; } - stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; - stream->sstdrv_ops->module_name = SST_CARD_NAMES; - /* registering with SST driver to get access to SST APIs to use */ - ret_val = register_sst_card(stream->sstdrv_ops); - if (ret_val) { - pr_err("sst: sst card registration failed\n"); - kfree(stream->sstdrv_ops); + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); kfree(stream); - return ret_val; + return -ENODEV; } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ runtime->private_data = stream; return 0; @@ -278,9 +315,8 @@ static int sst_platform_close(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->sstdrv_ops->pcm_control->close(str_id); - unregister_sst_card(stream->sstdrv_ops); - kfree(stream->sstdrv_ops); + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); kfree(stream); return ret_val; } @@ -294,8 +330,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->sstdrv_ops->pcm_control->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); return ret_val; } @@ -347,8 +383,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, default: return -EINVAL; } - ret_val = stream->sstdrv_ops->pcm_control->device_control(str_cmd, - &str_id); + ret_val = stream->ops->device_control(str_cmd, &str_id); if (!ret_val) sst_set_stream_status(stream, status); @@ -368,7 +403,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_BUFFER_POINTER, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); @@ -439,6 +474,7 @@ static int sst_platform_probe(struct platform_device *pdev) int ret; pr_debug("sst_platform_probe called\n"); + sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index df370286694f..f04f4f72daa0 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -42,14 +42,14 @@ #define SST_MIN_PERIODS 2 #define SST_MAX_PERIODS (1024*2) #define SST_FIFO_SIZE 0 -#define SST_CARD_NAMES "intel_mid_card" -#define MSIC_VENDOR_ID 3 +#define SST_CODEC_TYPE_PCM 1 -struct sst_runtime_stream { - int stream_status; - struct pcm_stream_info stream_info; - struct intel_sst_card_ops *sstdrv_ops; - spinlock_t status_lock; +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; }; enum sst_drv_status { @@ -60,4 +60,72 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); #endif -- cgit v1.2.1 From a0f203d384fadacba514748cd0095efeadeed96c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 2 Dec 2011 15:08:37 -0700 Subject: ASoC: WM8903: Fix platform data gpio_cfg confusion wm8903_platform_data.gpio_cfg[] was intended to be interpreted as follows: 0: Don't touch this GPIO's configuration register 1..7fff: Write that value to the GPIO's configuration register 8000: Write zero to the GPIO's configuration register other: Undefined (invalid) The rationale is that platform data is usually global data, and a value of zero means that the field wasn't explicitly set to anything (e.g. because the field was new to the pdata type, and existing users weren't update to initialize it) and hence the value zero should be ignored. 0x8000 is an explicit way to get 0 in the register. The code worked this way until commit 7cfe561 "ASoC: wm8903: Expose GPIOs through gpiolib", where the behaviour was changed due to my lack of awareness of the above rationale. This patch reverts to the intended behaviour, and updates all in-tree users to use the correct scheme. This also makes WM8903 consistent with other devices that use a similar scheme. WM8903_GPIO_NO_CONFIG is also renamed to WM8903_GPIO_CONFIG_ZERO so that its name accurately reflects its purpose. Signed-off-by: Stephen Warren Cc: Olof Johansson Cc: Colin Cross Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e6ecede576dc..184b67730c39 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1893,7 +1893,8 @@ static int wm8903_probe(struct snd_soc_codec *codec) bool mic_gpio = false; for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (pdata->gpio_cfg[i] > 0x7fff) + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, -- cgit v1.2.1 From db81778409227a0dc46ab95b95e1c7184ae9ef48 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 2 Dec 2011 15:08:39 -0700 Subject: ASoC: WM8903: Remove conditionals checking pdata != NULL The pdata pointer is now always valid. Remove any conditions that check its validity. This patch is mostly just removing an indentation level. One variable had to be moved due to the removal of a scope, and one comment was split into two. Viewing the patch with git show/diff -b will show that it's actually very small. Note that WM8903_MIC_BIAS_CONTROL_0 is now written unconditionally, whereas it used to be written only if pdata was supplied. Since defpdata.micdet_cfg = 0, this unconditional write simply echos the HW defaults in the case where pdata is not supplied. Based on work by John Bonesio, but significantly reworked since then. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 74 +++++++++++++++++++++++------------------------ 1 file changed, 36 insertions(+), 38 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 184b67730c39..b114468d6453 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1842,7 +1842,7 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec) wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; wm8903->gpio_chip.dev = codec->dev; - if (pdata && pdata->gpio_base) + if (pdata->gpio_base) wm8903->gpio_chip.base = pdata->gpio_base; else wm8903->gpio_chip.base = -1; @@ -1878,6 +1878,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) int ret, i; int trigger, irq_pol; u16 val; + bool mic_gpio = false; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -1888,52 +1889,49 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs and microphone detection */ - if (pdata) { - bool mic_gpio = false; - - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; + /* Set up GPIOs, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); + snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; } + } - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); + /* Set up microphone detection */ + snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; - wm8903->mic_delay = pdata->micdet_delay; - } - if (wm8903->irq) { - if (pdata && pdata->irq_active_low) { + if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8903_IRQ_POL; } else { -- cgit v1.2.1 From 9d35f3e100eb5cfb91d777c8621fb585ad0327cd Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 2 Dec 2011 15:08:40 -0700 Subject: ASoC: WM8903: Get default irq_active_low from IRQ controller If the WM8903 is hooked up to an interrupt, set the irq_active_low flag in the default platform data based on the IRQ's IRQ_TYPE. Map IRQ_TYPE_NONE (a lack of explicit configuration/restriction) to irq_active_low = false; the previous default. This code is mainly added to support device tree interrupt bindings, although will work perfectly well in a non device tree system too. Any interrupt controller that supports only a single IRQ_TYPE could set each IRQ's type based on that restriction. This applies equally with and without device tree. To cater for interrupt controllers that don't do this, for which irqd_get_trigger_type() will return IRQ_TYPE_NONE, the platform data irq_active_low field may be used in systems that don't use device tree. With device tree, every IRQ must have some IRQ_TYPE set. Controllers that support DT and multiple IRQ_TYPEs must define the interrupts property (as used in interrupt source nodes) such that it defines the IRQ_TYPE to use. When the core DT setup code initializes wm8903->irq, the interrupts property will be parsed, and as a side- effect, set the IRQ's IRQ_TYPE for the WM8903 probe() function to read. Controllers that support DT and a single IRQ_TYPE could arrange to set the IRQ_TYPE somehow during their initialization, or hard-code it during the processing of the child interrupts property. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b114468d6453..b4f2c906b2f3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -2036,6 +2037,39 @@ static const struct regmap_config wm8903_regmap = { .num_reg_defaults = ARRAY_SIZE(wm8903_reg_defaults), }; +static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, + struct wm8903_platform_data *pdata) +{ + struct irq_data *irq_data = irq_get_irq_data(i2c->irq); + if (!irq_data) { + dev_err(&i2c->dev, "Invalid IRQ: %d\n", + i2c->irq); + return -EINVAL; + } + + switch (irqd_get_trigger_type(irq_data)) { + case IRQ_TYPE_NONE: + /* + * We assume the controller imposes no restrictions, + * so we are able to select active-high + */ + /* Fall-through */ + case IRQ_TYPE_LEVEL_HIGH: + pdata->irq_active_low = false; + break; + case IRQ_TYPE_LEVEL_LOW: + pdata->irq_active_low = true; + break; + default: + dev_err(&i2c->dev, + "Unsupported IRQ_TYPE %x\n", + irqd_get_trigger_type(irq_data)); + return -EINVAL; + } + + return 0; +} + static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2071,6 +2105,12 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to allocate pdata\n"); return -ENOMEM; } + + if (i2c->irq) { + ret = wm8903_set_pdata_irq_trigger(i2c, wm8903->pdata); + if (ret != 0) + return ret; + } } ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); -- cgit v1.2.1 From 5d680b3a84b3e870fc1ea01495935e58e17de7aa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 2 Dec 2011 15:08:41 -0700 Subject: ASoC: WM8903: Add device tree binding Document the device tree binding for the WM8903 codec, and modify the driver to extract platform data from the device tree, if present. Based on work by John Bonesio, but significantly reworked since then. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b4f2c906b2f3..adfbefaaab21 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2070,6 +2070,49 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, return 0; } +static int wm8903_set_pdata_from_of(struct i2c_client *i2c, + struct wm8903_platform_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + u32 val32; + int i; + + if (of_property_read_u32(np, "micdet-cfg", &val32) >= 0) + pdata->micdet_cfg = val32; + + if (of_property_read_u32(np, "micdet-delay", &val32) >= 0) + pdata->micdet_delay = val32; + + if (of_property_read_u32_array(np, "gpio-cfg", pdata->gpio_cfg, + ARRAY_SIZE(pdata->gpio_cfg)) >= 0) { + /* + * In device tree: 0 means "write 0", + * 0xffffffff means "don't touch". + * + * In platform data: 0 means "don't touch", + * 0x8000 means "write 0". + * + * Note: WM8903_GPIO_CONFIG_ZERO == 0x8000. + * + * Convert from DT to pdata representation here, + * so no other code needs to change. + */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if (pdata->gpio_cfg[i] == 0) { + pdata->gpio_cfg[i] = WM8903_GPIO_CONFIG_ZERO; + } else if (pdata->gpio_cfg[i] == 0xffffffff) { + pdata->gpio_cfg[i] = 0; + } else if (pdata->gpio_cfg[i] > 0x7fff) { + dev_err(&i2c->dev, "Invalid gpio-cfg[%d] %x\n", + i, pdata->gpio_cfg[i]); + return -EINVAL; + } + } + } + + return 0; +} + static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2111,6 +2154,12 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, if (ret != 0) return ret; } + + if (i2c->dev.of_node) { + ret = wm8903_set_pdata_from_of(i2c, wm8903->pdata); + if (ret != 0) + return ret; + } } ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); -- cgit v1.2.1 From 6664ee115bb45d912d64d1c6b26bd3b96ef7df09 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Dec 2011 10:30:24 +0000 Subject: ASoC: Don't fail if we can't read the IRQ type in WM8903 If we fail to read the IRQ type from the interrupt controller don't fail, just assume a value and solider on - we may fail later when we try to request the IRQ but it's possible we'll succeed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index adfbefaaab21..21b9fdc18319 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2049,6 +2049,7 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, switch (irqd_get_trigger_type(irq_data)) { case IRQ_TYPE_NONE: + default: /* * We assume the controller imposes no restrictions, * so we are able to select active-high @@ -2060,11 +2061,6 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, case IRQ_TYPE_LEVEL_LOW: pdata->irq_active_low = true; break; - default: - dev_err(&i2c->dev, - "Unsupported IRQ_TYPE %x\n", - irqd_get_trigger_type(irq_data)); - return -EINVAL; } return 0; -- cgit v1.2.1 From f18b4e2ee9649c4aa50cc279826d3890f468a80e Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 6 Dec 2011 14:15:41 -0700 Subject: ASoC: WM8903: Add of_match_table This allows the device to be matched against the device tree using the compatible flag directly, as is standard, rather than falling back to matching .id_table against the non-vendor portion of the first compatible property value. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 21b9fdc18319..d88b727d7f99 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2201,6 +2201,12 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id wm8903_of_match[] = { + { .compatible = "wlf,wm8903", }, + {}, +}; +MODULE_DEVICE_TABLE(of, wm8903_of_match); + static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, { } @@ -2211,6 +2217,7 @@ static struct i2c_driver wm8903_i2c_driver = { .driver = { .name = "wm8903", .owner = THIS_MODULE, + .of_match_table = wm8903_of_match, }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), -- cgit v1.2.1 From b960ce74a70477d7d7d3c08669a8f0f52017b4fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:30:37 +0000 Subject: ASoC: Convert Samsung I2S driver to devm_kzalloc() Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/i2s.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 5de500ce5dd4..ff5d9194d11f 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -945,7 +945,7 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; - i2s = kzalloc(sizeof(struct i2s_dai), GFP_KERNEL); + i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) return NULL; @@ -972,10 +972,8 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->pdev = platform_device_register_resndata(NULL, pdev->name, pdev->id + SAMSUNG_I2S_SECOFF, NULL, 0, NULL, 0); - if (IS_ERR(i2s->pdev)) { - kfree(i2s); + if (IS_ERR(i2s->pdev)) return NULL; - } } /* Pre-assign snd_soc_dai_set_drvdata */ @@ -1048,7 +1046,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (!pri_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); ret = -ENOMEM; - goto err1; + goto err; } pri_dai->dma_playback.dma_addr = regs_base + I2STXD; @@ -1073,7 +1071,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); ret = -ENOMEM; - goto err2; + goto err; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = @@ -1092,17 +1090,13 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); ret = -EINVAL; - goto err3; + goto err; } snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); return 0; -err3: - kfree(sec_dai); -err2: - kfree(pri_dai); -err1: +err: release_mem_region(regs_base, resource_size(res)); return ret; @@ -1128,8 +1122,6 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - kfree(i2s); - snd_soc_unregister_dai(&pdev->dev); return 0; -- cgit v1.2.1 From c1496b4ac3c6a1664592351b3530489cd8eff959 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 7 Dec 2011 18:04:10 +0800 Subject: ASoC: Fix a typo in s3c24xx_simtec_tlv320aic23 driver Fix a typo introduced by commit e00c3f55 "ASoC: Convert Samsung directory to module_platform_driver". This fixes the build error: CC sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_init': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: (Each undeclared identifier is reported only once sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: for each function it appears in.) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_exit': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) make[3]: *** [sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I think we had better naming it with *driver, thus I change it to simtec_audio_tlv320aic23_driver. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 7324609833d8..89b57b5c3e17 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -92,7 +92,7 @@ static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic23); } -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { +static struct platform_driver simtec_audio_tlv320aic23_driver = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-simtec-tlv320aic23", @@ -102,7 +102,7 @@ static struct platform_driver simtec_audio_tlv320aic23_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -module_platform_driver(simtec_audio_tlv320aic32_driver); +module_platform_driver(simtec_audio_tlv320aic23_driver); MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); MODULE_AUTHOR("Ben Dooks "); -- cgit v1.2.1 From 5ff7ada748fe2f74f525893577c4418bfdaf6d4f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 7 Dec 2011 10:04:07 +0800 Subject: ASoC: Convert e750_wm9705 to use gpio_request_one() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e750_wm9705.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index ce5f056009a7..55c53d13bea6 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -129,22 +129,16 @@ static int __init e750_init(void) if (!machine_is_e750()) return -ENODEV; - ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp"); + ret = gpio_request_one(GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, + "Headphone amp"); if (ret) return ret; - ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp"); + ret = gpio_request_one(GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, + "Speaker amp"); if (ret) goto free_hp_amp_gpio; - ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1); - if (ret) - goto free_spk_amp_gpio; - - ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1); - if (ret) - goto free_spk_amp_gpio; - e750_snd_device = platform_device_alloc("soc-audio", -1); if (!e750_snd_device) { ret = -ENOMEM; -- cgit v1.2.1 From 8faab941bec7af1c1865db316ac2f37c78071271 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 7 Dec 2011 10:01:30 +0800 Subject: ASoC: Fix error handling in e800_init to free gpios Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e800_wm9712.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 6a8f38b6c379..26e02322f10b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -136,8 +136,10 @@ static int __init e800_init(void) goto free_spk_amp_gpio; e800_snd_device = platform_device_alloc("soc-audio", -1); - if (!e800_snd_device) - return -ENOMEM; + if (!e800_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } platform_set_drvdata(e800_snd_device, &e800); ret = platform_device_add(e800_snd_device); -- cgit v1.2.1 From 209e8cf668d3f421eb6d86eb62451396fb0a737d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 7 Dec 2011 10:03:12 +0800 Subject: ASoC: Convert e800_wm9712 to use gpio_request_one() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e800_wm9712.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 26e02322f10b..478ff191ffb4 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -119,22 +119,16 @@ static int __init e800_init(void) if (!machine_is_e800()) return -ENODEV; - ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp"); + ret = gpio_request_one(GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, + "Headphone amp"); if (ret) return ret; - ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp"); + ret = gpio_request_one(GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, + "Speaker amp"); if (ret) goto free_hp_amp_gpio; - ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1); - if (ret) - goto free_spk_amp_gpio; - - ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1); - if (ret) - goto free_spk_amp_gpio; - e800_snd_device = platform_device_alloc("soc-audio", -1); if (!e800_snd_device) { ret = -ENOMEM; -- cgit v1.2.1 From d6652ef8229e9953543f41d8e081c23e653f0044 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:14:31 +0000 Subject: ASoC: Hold runtime PM references to components of active DAIs Every device that implements runtime power management for DAIs is doing it in pretty much the same way: in the startup callback they take a runtime PM reference and then in the shutdown callback they release that reference, keeping the device active while the DAI is active. Given the frequency with which this is done and the obviousness of the need to keep the device active in this period factor the code out into the core, taking references on the device for each CPU DAI, CODEC DAI and DMA device in the core. As runtime PM is reference counted this shouldn't interfere with any other reference holding by the drivers, and since (in common with the existing implementations) we don't check for errors on enabling it shouldn't matter if the device actually has runtime PM enabled or not. Signed-off-by: Mark Brown Tested-by: Peter Ujfalusi --- sound/soc/soc-pcm.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 49aa71e0d7e6..8aa7cec6eab2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -77,6 +78,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); + pm_runtime_get_sync(codec_dai->dev); + pm_runtime_get_sync(platform->dev); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ @@ -233,6 +238,11 @@ platform_err: cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&rtd->pcm_mutex); + + pm_runtime_put(platform->dev); + pm_runtime_put(codec_dai->dev); + pm_runtime_put(cpu_dai->dev); + return ret; } @@ -339,6 +349,11 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } mutex_unlock(&rtd->pcm_mutex); + + pm_runtime_put(platform->dev); + pm_runtime_put(codec_dai->dev); + pm_runtime_put(cpu_dai->dev); + return 0; } -- cgit v1.2.1 From 06d07b6b1cf46ad1bfd15ab1ba84c0d7ee6dab31 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:20:02 +0000 Subject: ASoC: Use core pm_runtime callbacks for omap-dmic Now that the core holds a pm_runtime reference to the device while the link is active there is no need for the driver to do so. Signed-off-by: Mark Brown Acked-by: Peter Ujfalusi --- sound/soc/omap/omap-dmic.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 9c73c0c70d39..0855c1cfa7fd 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -114,7 +114,6 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); if (!dai->active) { - pm_runtime_get_sync(dmic->dev); snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); dmic->active = 1; } else { @@ -133,10 +132,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); - if (!dai->active) { - pm_runtime_put_sync(dmic->dev); + if (!dai->active) dmic->active = 0; - } mutex_unlock(&dmic->mutex); } -- cgit v1.2.1 From beaff340e04fc3a752aa2cca70195dd506deccef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:20:47 +0000 Subject: ASoC: Use core pm_runtime callbacks for omap-mcpdm Now that the core holds a pm_runtime reference to the device while the link is active there is no need for the driver to do so. Signed-off-by: Mark Brown Acked-by: Peter Ujfalusi --- sound/soc/omap/omap-mcpdm.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index b50ac60be7db..0e25df4fa9e5 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -266,8 +266,6 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&mcpdm->mutex); if (!dai->active) { - pm_runtime_get_sync(mcpdm->dev); - /* Enable watch dog for ES above ES 1.0 to avoid saturation */ if (omap_rev() != OMAP4430_REV_ES1_0) { u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); @@ -295,9 +293,6 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, omap_mcpdm_stop(mcpdm); omap_mcpdm_close_streams(mcpdm); } - - if (!omap_mcpdm_active(mcpdm)) - pm_runtime_put_sync(mcpdm->dev); } mutex_unlock(&mcpdm->mutex); -- cgit v1.2.1 From f1aac484f705007caf0d7c256a1a29506600cae3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Dec 2011 15:17:06 +0000 Subject: ASoC: Take a pm_runtime reference on DAPM devices that are enabled As for PCMs take a runtime power management reference to devices that are in a non-off bias, avoiding the need to do this in individual drivers. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bb327e431a5..e174d0811dae 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include @@ -1206,6 +1207,9 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) /* If we're off and we're not supposed to be go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { + if (d->dev) + pm_runtime_get_sync(d->dev); + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1245,6 +1249,9 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); + + if (d->dev) + pm_runtime_put_sync(d->dev); } /* If we just powered up then move to active bias */ -- cgit v1.2.1 From 4105ab846ca795f03e63fb7bfacafc4217f48ca8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Dec 2011 15:17:36 +0000 Subject: ASoC: Rely on core enabling the wm8994 with runtime PM No need to do this in the driver now. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 02ca2573214d..3eaf56a33964 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2188,8 +2188,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - pm_runtime_get_sync(codec->dev); - switch (control->type) { case WM8994: if (wm8994->revision < 4) { @@ -2256,11 +2254,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) wm8994->cur_fw = NULL; - - pm_runtime_put(codec->dev); - } break; } codec->dapm.bias_level = level; -- cgit v1.2.1 From 16b24881a031a653cd76b83bfd96ef2d30b2491b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 8 Dec 2011 11:09:15 +0800 Subject: ASoC: wm8960: Use snd_soc_update_bits for read-modify-write Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 67 +++++++++++++++-------------------------------- 1 file changed, 21 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6a9c41d351de..2315b866d002 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -543,30 +543,24 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0x8); else - snd_soc_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0); return 0; } static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_STANDBY: @@ -579,23 +573,19 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Enable & ramp VMID at 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x80, 0x80); msleep(100); /* Enable VREF */ - snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF, + WM8960_VREF); /* Disable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x100; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x100); break; case SND_SOC_BIAS_OFF: @@ -787,10 +777,8 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) & ~1); - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) & ~1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0); if (!freq_in || !freq_out) return 0; @@ -809,11 +797,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) | 1); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0x1); msleep(250); - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) | 1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0x1); return 0; } @@ -913,7 +899,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = dev_get_platdata(codec->dev); int ret; - u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; @@ -944,26 +929,16 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8960_LINVOL); - snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RINVOL); - snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LADC); - snd_soc_write(codec, WM8960_LADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RADC); - snd_soc_write(codec, WM8960_RADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LDAC); - snd_soc_write(codec, WM8960_LDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RDAC); - snd_soc_write(codec, WM8960_RDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT1); - snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT1); - snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT2); - snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT2); - snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); + snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); -- cgit v1.2.1 From 7b9b5e11704afb8f05aa6490c3b4bb2cc328647c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 7 Dec 2011 13:58:29 -0700 Subject: ASoC: Tegra: Move DAS configuration into DAS driver Move DAS routing setup into the DAS driver itself. This removes the need to duplicate this in each machine driver, of which we'll soon have three. An added advantage is that the machine drivers no longer call the Tegra20- specific DAS functions by name, so the machine driver no longer needs to be split up into Tegra20 and Tegra30 versions. If individual machine drivers need a different routing setup to this default, they can still call the DAS functions to set that up. Long-term, DAS will be a codec driver, and user-space will be able to control its routing, possibly within constraints that the machine driver sets up. Configuring the DAS routing from the DAS driver is a very slight move in that direction. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.c | 13 +++++++++++++ sound/soc/tegra/tegra_wm8903.c | 13 ------------- sound/soc/tegra/trimslice.c | 23 ----------------------- 3 files changed, 13 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 5b82b4e79231..3b3c1ba4d235 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -202,6 +202,19 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) goto err; } + ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, + TEGRA_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); + goto err; + } + ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, + TEGRA_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); + goto err; + } + tegra_das_debug_add(das); platform_set_drvdata(pdev, das); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2f5b1074a8d9..ba2d23ea6424 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -249,19 +249,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct tegra_wm8903_platform_data *pdata = machine->pdata; int ret; - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(card->dev, "Can't set up DAS DAP connection\n"); - return ret; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(card->dev, "Can't set up DAS DAC connection\n"); - return ret; - } - if (gpio_is_valid(pdata->gpio_spkr_en)) { ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); if (ret) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 043eb7c7eb73..7d95b7697a73 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -115,28 +115,6 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = { {"RLINEIN", NULL, "Line In"}, }; -static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_card *card = codec->card; - int ret; - - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(card->dev, "Can't set up DAS DAP connection\n"); - return ret; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(card->dev, "Can't set up DAS DAC connection\n"); - return ret; - } - - return 0; -} - static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", @@ -144,7 +122,6 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "tlv320aic23-hifi", - .init = trimslice_asoc_init, .ops = &trimslice_asoc_ops, }; -- cgit v1.2.1 From 2610ab7767bba916f65094d71cfed3b8281cba08 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 7 Dec 2011 13:58:27 -0700 Subject: ASoC: Refactor some conditions and loop in soc_bind_dai_link() Transform some loops from: for_each(x) { if (f(x)) { work_on(x); } } to new structure: for_each(x) { if (!f(x)) continue; work_on(x); } This will allow future modification of f(x) with less impact to the code. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 52 ++++++++++++++++++++++++++++++---------------------- 1 file changed, 30 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5195f0653b35..ebb104878c48 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -763,10 +763,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { - rtd->cpu_dai = cpu_dai; - goto find_codec; - } + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + + rtd->cpu_dai = cpu_dai; + goto find_codec; } dev_dbg(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -779,22 +780,28 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (!strcmp(codec->name, dai_link->codec_name)) { - rtd->codec = codec; - - /* CODEC found, so find CODEC DAI from registered DAIs from this CODEC*/ - list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, dai_link->codec_dai_name)) { - rtd->codec_dai = codec_dai; - goto find_platform; - } - } - dev_dbg(card->dev, "CODEC DAI %s not registered\n", - dai_link->codec_dai_name); + if (strcmp(codec->name, dai_link->codec_name)) + continue; + + rtd->codec = codec; - goto find_platform; + /* + * CODEC found, so find CODEC DAI from registered DAIs from + * this CODEC + */ + list_for_each_entry(codec_dai, &dai_list, list) { + if (codec->dev == codec_dai->dev && + !strcmp(codec_dai->name, + dai_link->codec_dai_name)) { + + rtd->codec_dai = codec_dai; + goto find_platform; + } } + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + + goto find_platform; } dev_dbg(card->dev, "CODEC %s not registered\n", dai_link->codec_name); @@ -811,10 +818,11 @@ find_platform: /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (!strcmp(platform->name, platform_name)) { - rtd->platform = platform; - goto out; - } + if (strcmp(platform->name, platform_name)) + continue; + + rtd->platform = platform; + goto out; } dev_dbg(card->dev, "platform %s not registered\n", -- cgit v1.2.1 From bf97ca9a0dabd6110a6aa7b4d1b20274973810af Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:45:22 +0800 Subject: ASoC: Convert WM8776 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index fbf80c5220d0..38b455662195 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -452,7 +452,8 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&spi->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -461,15 +462,13 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static int __devexit wm8776_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -491,7 +490,8 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&i2c->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -500,15 +500,13 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static __devexit int wm8776_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From bc9c040d363f3be17a59024191e9400e5b6205ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:46:59 +0800 Subject: ASoC: Make WM8770 SPI usage unconditional The device only supports SPI. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8770.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 8976eb5796d3..ea6f007a8114 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -690,7 +690,6 @@ static const struct of_device_id wm8770_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8770_of_match); -#if defined(CONFIG_SPI_MASTER) static int __devinit wm8770_spi_probe(struct spi_device *spi) { struct wm8770_priv *wm8770; @@ -726,28 +725,23 @@ static struct spi_driver wm8770_spi_driver = { .probe = wm8770_spi_probe, .remove = __devexit_p(wm8770_spi_remove) }; -#endif static int __init wm8770_modinit(void) { int ret = 0; -#if defined(CONFIG_SPI_MASTER) ret = spi_register_driver(&wm8770_spi_driver); if (ret) { printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n", ret); } -#endif return ret; } module_init(wm8770_modinit); static void __exit wm8770_exit(void) { -#if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8770_spi_driver); -#endif } module_exit(wm8770_exit); -- cgit v1.2.1 From 5a374524216a244d30c42545ab49f743a43b05c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:52:19 +0800 Subject: ASoC: Convert WM8804 to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index ae4b8fb3c3e5..d54a3ca5e19e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -659,8 +659,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8804_snd_controls, - ARRAY_SIZE(wm8804_snd_controls)); return 0; err_reg_enable: @@ -715,7 +713,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .reg_cache_size = ARRAY_SIZE(wm8804_reg_defs), .reg_word_size = sizeof(u8), .reg_cache_default = wm8804_reg_defs, - .volatile_register = wm8804_volatile + .volatile_register = wm8804_volatile, + + .controls = wm8804_snd_controls, + .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; static const struct of_device_id wm8804_of_match[] = { -- cgit v1.2.1 From 46ce904f7d4788ebc2ca7894fb56b9aa5b84af2d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:53:47 +0800 Subject: ASoC: Convert WM8900 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e427a38032cc..f18c554efc98 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -542,7 +542,7 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8900_REG_POWER3, 2, 0, }; /* Target, Path, Source */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8900_dapm_routes[] = { /* Inputs */ {"Left Input PGA", "LINPUT1 Switch", "LINPUT1"}, {"Left Input PGA", "LINPUT2 Switch", "LINPUT2"}, @@ -606,17 +606,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"HP_R", NULL, "Headphone Amplifier"}, }; -static int wm8900_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8900_dapm_widgets, - ARRAY_SIZE(wm8900_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1203,10 +1192,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) /* Set the DAC and mixer output bias */ snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81); - snd_soc_add_controls(codec, wm8900_snd_controls, - ARRAY_SIZE(wm8900_snd_controls)); - wm8900_add_widgets(codec); - return 0; } @@ -1227,6 +1212,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .reg_cache_size = ARRAY_SIZE(wm8900_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8900_reg_defaults, + + .controls = wm8900_snd_controls, + .num_controls = ARRAY_SIZE(wm8900_snd_controls), + .dapm_widgets = wm8900_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8900_dapm_widgets), + .dapm_routes = wm8900_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8900_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.1 From 7fcadfd17699b6b7973ce4f99eae47a11b4c44a7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 9 Dec 2011 18:43:20 +0800 Subject: ASoC: Fix comments for disabling amplifier and PGA Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index dc087c155975..58fbf0a87b6a 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -422,11 +422,11 @@ static int wm8961_spk_event(struct snd_soc_dapm_widget *w, } if (event & SND_SOC_DAPM_PRE_PMD) { - /* Enable the amplifier */ + /* Disable the amplifier */ spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA); snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); - /* Enable the PGA */ + /* Disable the PGA */ pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA); snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); } -- cgit v1.2.1 From 3025ae45d6d905c8e973bba59d6f9a1be0da734d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:24:16 +0800 Subject: ASoC: Convert wm8770 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8770.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index ea6f007a8114..19374a9e5ba6 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -695,7 +695,8 @@ static int __devinit wm8770_spi_probe(struct spi_device *spi) struct wm8770_priv *wm8770; int ret; - wm8770 = kzalloc(sizeof(struct wm8770_priv), GFP_KERNEL); + wm8770 = devm_kzalloc(&spi->dev, sizeof(struct wm8770_priv), + GFP_KERNEL); if (!wm8770) return -ENOMEM; @@ -704,15 +705,13 @@ static int __devinit wm8770_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8770, &wm8770_dai, 1); - if (ret < 0) - kfree(wm8770); + return ret; } static int __devexit wm8770_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- cgit v1.2.1 From 7c08be84f83b23762fb7571ac9a4aea3c34d1a66 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Lothar=20Wa=C3=9Fmann?= Date: Fri, 9 Dec 2011 14:16:29 +0100 Subject: ASoC: Fix an obvious copy paste error in an error message MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The message was obviously copied from soc_init_codec_debugfs() Signed-off-by: Lothar Waßmann Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ebb104878c48..1252ab1ebf69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -412,7 +412,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card) snd_soc_debugfs_root); if (!card->debugfs_card_root) { dev_warn(card->dev, - "ASoC: Failed to create codec debugfs directory\n"); + "ASoC: Failed to create card debugfs directory\n"); return; } -- cgit v1.2.1 From 3628137646e2ee25c9e46ba9d2c20b313e4a1a25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Dec 2011 11:31:22 +0800 Subject: ASoC: Raise Speyside audio system clock rate to 512fs To support advanced system functionality for additional components; the actively used clocks will remain the same for current components. Also factor the rate out to a single #define while we're at it. Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 18e6356e86db..0222d8636323 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -19,6 +19,7 @@ #include "../codecs/wm9081.h" #define WM8996_HPSEL_GPIO 214 +#define MCLK_AUDIO_RATE (512 * 48000) static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, @@ -67,7 +68,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, 0, WM8996_FLL_MCLK2, - 32768, 48000 * 256); + 32768, MCLK_AUDIO_RATE); if (ret < 0) { pr_err("Failed to start FLL\n"); return ret; @@ -75,7 +76,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, ret = snd_soc_dai_set_sysclk(codec_dai, WM8996_SYSCLK_FLL, - 48000 * 256, + MCLK_AUDIO_RATE, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -224,7 +225,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) { /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - 48000 * 256, 0); + MCLK_AUDIO_RATE, 0); } static struct snd_soc_aux_dev speyside_aux_dev[] = { -- cgit v1.2.1 From 0604ca48f1689ad06144b81f5c08f297b6edd831 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 12 Dec 2011 16:01:15 +0800 Subject: ASoC: Add missed MODULE_LICENSE("GPL") for imx-pcm-fiq This driver can be built as module and the file header indicates that the driver is published under the GPL. Thus add MODULE_LICENSE("GPL") for it. Signed-off-by: Axel Lin Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d7ea0b354124..456b7d723d66 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -332,3 +332,5 @@ static struct platform_driver imx_pcm_driver = { }; module_platform_driver(imx_pcm_driver); + +MODULE_LICENSE("GPL"); -- cgit v1.2.1 From 4de45284d3927b5068de6ed972b11627a3428427 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Oct 2011 15:44:12 +0200 Subject: mfd: Define some additional wm8994 registers Add a bunch of definitions for wm8994 registers that are not currently used by software. Signed-off-by: Mark Brown Acked-by: Samuel Ortiz --- sound/soc/codecs/wm8994-tables.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index df5a8b9a250f..6ed19d9e7454 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -78,7 +78,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R74 */ { 0x0000, 0x0000 }, /* R75 */ { 0x8000, 0x8000 }, /* R76 - Charge Pump (1) */ - { 0x0000, 0x0000 }, /* R77 */ + { 0x8000, 0x8000 }, /* R77 - Charge Pump (2) */ { 0x0000, 0x0000 }, /* R78 */ { 0x0000, 0x0000 }, /* R79 */ { 0x0000, 0x0000 }, /* R80 */ @@ -1651,7 +1651,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R74 */ 0x0000, /* R75 */ 0x1F25, /* R76 - Charge Pump (1) */ - 0x0000, /* R77 */ + 0xAB19, /* R77 - Charge Pump (2) */ 0x0000, /* R78 */ 0x0000, /* R79 */ 0x0000, /* R80 */ @@ -2124,8 +2124,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R547 - FLL1 Control (4) */ 0x0C80, /* R548 - FLL1 Control (5) */ 0x0000, /* R549 */ - 0x0000, /* R550 */ - 0x0000, /* R551 */ + 0x0000, /* R550 - FLL1 EFS 1 */ + 0x0006, /* R551 - FLL1 EFS 2 */ 0x0000, /* R552 */ 0x0000, /* R553 */ 0x0000, /* R554 */ @@ -2156,8 +2156,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R579 - FLL2 Control (4) */ 0x0C80, /* R580 - FLL2 Control (5) */ 0x0000, /* R581 */ - 0x0000, /* R582 */ - 0x0000, /* R583 */ + 0x0000, /* R582 - FLL2 EFS 1 */ + 0x0006, /* R583 - FLL2 EFS 2 */ 0x0000, /* R584 */ 0x0000, /* R585 */ 0x0000, /* R586 */ -- cgit v1.2.1 From d9a7666ff3a9e109844bf5aca5f50e3743f65840 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 Jul 2011 12:49:52 +0100 Subject: ASoC: Remove ASoC-specific WM8994 I/O code Just go directly to the regmap API, saving code and making integration that bit more direct. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 63 +++++++++-------------------------------------- 1 file changed, 12 insertions(+), 51 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3eaf56a33964..285890802d62 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -184,44 +184,6 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) } } -static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - BUG_ON(reg > WM8994_MAX_REGISTER); - - if (!wm8994_volatile(codec, reg)) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret != 0) - dev_err(codec->dev, "Cache write to %x failed: %d\n", - reg, ret); - } - - return wm8994_reg_write(codec->control_data, reg, value); -} - -static unsigned int wm8994_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned int val; - int ret; - - BUG_ON(reg > WM8994_MAX_REGISTER); - - if (!wm8994_volatile(codec, reg) && wm8994_readable(codec, reg) && - reg < codec->driver->reg_cache_size) { - ret = snd_soc_cache_read(codec, reg, &val); - if (ret >= 0) - return val; - else - dev_err(codec->dev, "Cache read from %x failed: %d\n", - reg, ret); - } - - return wm8994_reg_read(codec->control_data, reg); -} - static int configure_aif_clock(struct snd_soc_codec *codec, int aif) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -2839,8 +2801,8 @@ static int wm8994_resume(struct snd_soc_codec *codec) if (wm8994->revision < 4) { /* force a HW read */ - val = wm8994_reg_read(codec->control_data, - WM8994_POWER_MANAGEMENT_5); + ret = regmap_read(control->regmap, + WM8994_POWER_MANAGEMENT_5, &val); /* modify the cache only */ codec->cache_only = 1; @@ -3455,13 +3417,13 @@ static irqreturn_t wm8994_temp_shut(int irq, void *data) static int wm8994_codec_probe(struct snd_soc_codec *codec) { - struct wm8994 *control; + struct wm8994 *control = dev_get_drvdata(codec->dev->parent); struct wm8994_priv *wm8994; struct snd_soc_dapm_context *dapm = &codec->dapm; + unsigned int reg; int ret, i; - codec->control_data = dev_get_drvdata(codec->dev->parent); - control = codec->control_data; + codec->control_data = control->regmap; wm8994 = devm_kzalloc(codec->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -3469,6 +3431,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm8994); + snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); wm8994->wm8994 = dev_get_drvdata(codec->dev->parent); wm8994->pdata = dev_get_platdata(codec->dev->parent); @@ -3494,11 +3457,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i)) continue; - ret = wm8994_reg_read(codec->control_data, i); + ret = regmap_read(control->regmap, i, ®); if (ret <= 0) continue; - ret = snd_soc_cache_write(codec, i, ret); + ret = snd_soc_cache_write(codec, i, reg); if (ret != 0) { dev_err(codec->dev, "Failed to initialise cache for 0x%x: %d\n", @@ -3653,24 +3616,24 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. */ - ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_1); + ret = regmap_read(control->regmap, WM8994_GPIO_1, ®); if (ret < 0) { dev_err(codec->dev, "Failed to read GPIO1 state: %d\n", ret); goto err_irq; } - if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + if ((reg & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { wm8994->lrclk_shared[0] = 1; wm8994_dai[0].symmetric_rates = 1; } else { wm8994->lrclk_shared[0] = 0; } - ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_6); + ret = regmap_read(control->regmap, WM8994_GPIO_6, ®); if (ret < 0) { dev_err(codec->dev, "Failed to read GPIO6 state: %d\n", ret); goto err_irq; } - if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + if ((reg & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { wm8994->lrclk_shared[1] = 1; wm8994_dai[1].symmetric_rates = 1; } else { @@ -3937,8 +3900,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .remove = wm8994_codec_remove, .suspend = wm8994_suspend, .resume = wm8994_resume, - .read = wm8994_read, - .write = wm8994_write, .readable_register = wm8994_readable, .volatile_register = wm8994_volatile, .set_bias_level = wm8994_set_bias_level, -- cgit v1.2.1 From cae59c7b2185856522822e40260174c088ca5b11 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Oct 2011 16:13:12 +0200 Subject: ASoC: Remove WM8994 register cache Now that the mfd is using the register map cache there's no need for the CODEC driver to do any register cache management or any funny dances to interact with the other drivers using the device so just remove the cache initialisation and volatility information. Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wm8994-tables.c | 3147 -------------------------------------- sound/soc/codecs/wm8994.c | 87 -- sound/soc/codecs/wm8994.h | 10 - 4 files changed, 1 insertion(+), 3245 deletions(-) delete mode 100644 sound/soc/codecs/wm8994-tables.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9aa6e669e6ef..de8078178f86 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -87,7 +87,7 @@ snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o -snd-soc-wm8994-objs := wm8994.o wm8994-tables.o wm8958-dsp2.o +snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c deleted file mode 100644 index 6ed19d9e7454..000000000000 --- a/sound/soc/codecs/wm8994-tables.c +++ /dev/null @@ -1,3147 +0,0 @@ -#include "wm8994.h" - -const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { - { 0xFFFF, 0xFFFF }, /* R0 - Software Reset */ - { 0x3B37, 0x3B37 }, /* R1 - Power Management (1) */ - { 0x6BF0, 0x6BF0 }, /* R2 - Power Management (2) */ - { 0x3FF0, 0x3FF0 }, /* R3 - Power Management (3) */ - { 0x3F3F, 0x3F3F }, /* R4 - Power Management (4) */ - { 0x3F0F, 0x3F0F }, /* R5 - Power Management (5) */ - { 0x003F, 0x003F }, /* R6 - Power Management (6) */ - { 0x0000, 0x0000 }, /* R7 */ - { 0x0000, 0x0000 }, /* R8 */ - { 0x0000, 0x0000 }, /* R9 */ - { 0x0000, 0x0000 }, /* R10 */ - { 0x0000, 0x0000 }, /* R11 */ - { 0x0000, 0x0000 }, /* R12 */ - { 0x0000, 0x0000 }, /* R13 */ - { 0x0000, 0x0000 }, /* R14 */ - { 0x0000, 0x0000 }, /* R15 */ - { 0x0000, 0x0000 }, /* R16 */ - { 0x0000, 0x0000 }, /* R17 */ - { 0x0000, 0x0000 }, /* R18 */ - { 0x0000, 0x0000 }, /* R19 */ - { 0x0000, 0x0000 }, /* R20 */ - { 0x01C0, 0x01C0 }, /* R21 - Input Mixer (1) */ - { 0x0000, 0x0000 }, /* R22 */ - { 0x0000, 0x0000 }, /* R23 */ - { 0x00DF, 0x01DF }, /* R24 - Left Line Input 1&2 Volume */ - { 0x00DF, 0x01DF }, /* R25 - Left Line Input 3&4 Volume */ - { 0x00DF, 0x01DF }, /* R26 - Right Line Input 1&2 Volume */ - { 0x00DF, 0x01DF }, /* R27 - Right Line Input 3&4 Volume */ - { 0x00FF, 0x01FF }, /* R28 - Left Output Volume */ - { 0x00FF, 0x01FF }, /* R29 - Right Output Volume */ - { 0x0077, 0x0077 }, /* R30 - Line Outputs Volume */ - { 0x0030, 0x0030 }, /* R31 - HPOUT2 Volume */ - { 0x00FF, 0x01FF }, /* R32 - Left OPGA Volume */ - { 0x00FF, 0x01FF }, /* R33 - Right OPGA Volume */ - { 0x007F, 0x007F }, /* R34 - SPKMIXL Attenuation */ - { 0x017F, 0x017F }, /* R35 - SPKMIXR Attenuation */ - { 0x003F, 0x003F }, /* R36 - SPKOUT Mixers */ - { 0x003F, 0x003F }, /* R37 - ClassD */ - { 0x00FF, 0x01FF }, /* R38 - Speaker Volume Left */ - { 0x00FF, 0x01FF }, /* R39 - Speaker Volume Right */ - { 0x00FF, 0x00FF }, /* R40 - Input Mixer (2) */ - { 0x01B7, 0x01B7 }, /* R41 - Input Mixer (3) */ - { 0x01B7, 0x01B7 }, /* R42 - Input Mixer (4) */ - { 0x01C7, 0x01C7 }, /* R43 - Input Mixer (5) */ - { 0x01C7, 0x01C7 }, /* R44 - Input Mixer (6) */ - { 0x01FF, 0x01FF }, /* R45 - Output Mixer (1) */ - { 0x01FF, 0x01FF }, /* R46 - Output Mixer (2) */ - { 0x0FFF, 0x0FFF }, /* R47 - Output Mixer (3) */ - { 0x0FFF, 0x0FFF }, /* R48 - Output Mixer (4) */ - { 0x0FFF, 0x0FFF }, /* R49 - Output Mixer (5) */ - { 0x0FFF, 0x0FFF }, /* R50 - Output Mixer (6) */ - { 0x0038, 0x0038 }, /* R51 - HPOUT2 Mixer */ - { 0x0077, 0x0077 }, /* R52 - Line Mixer (1) */ - { 0x0077, 0x0077 }, /* R53 - Line Mixer (2) */ - { 0x03FF, 0x03FF }, /* R54 - Speaker Mixer */ - { 0x00C1, 0x00C1 }, /* R55 - Additional Control */ - { 0x00F0, 0x00F0 }, /* R56 - AntiPOP (1) */ - { 0x01EF, 0x01EF }, /* R57 - AntiPOP (2) */ - { 0x00FF, 0x00FF }, /* R58 - MICBIAS */ - { 0x000F, 0x000F }, /* R59 - LDO 1 */ - { 0x0007, 0x0007 }, /* R60 - LDO 2 */ - { 0xFFFF, 0xFFFF }, /* R61 */ - { 0xFFFF, 0xFFFF }, /* R62 */ - { 0x0000, 0x0000 }, /* R63 */ - { 0x0000, 0x0000 }, /* R64 */ - { 0x0000, 0x0000 }, /* R65 */ - { 0x0000, 0x0000 }, /* R66 */ - { 0x0000, 0x0000 }, /* R67 */ - { 0x0000, 0x0000 }, /* R68 */ - { 0x0000, 0x0000 }, /* R69 */ - { 0x0000, 0x0000 }, /* R70 */ - { 0x0000, 0x0000 }, /* R71 */ - { 0x0000, 0x0000 }, /* R72 */ - { 0x0000, 0x0000 }, /* R73 */ - { 0x0000, 0x0000 }, /* R74 */ - { 0x0000, 0x0000 }, /* R75 */ - { 0x8000, 0x8000 }, /* R76 - Charge Pump (1) */ - { 0x8000, 0x8000 }, /* R77 - Charge Pump (2) */ - { 0x0000, 0x0000 }, /* R78 */ - { 0x0000, 0x0000 }, /* R79 */ - { 0x0000, 0x0000 }, /* R80 */ - { 0x0301, 0x0301 }, /* R81 - Class W (1) */ - { 0x0000, 0x0000 }, /* R82 */ - { 0x0000, 0x0000 }, /* R83 */ - { 0x333F, 0x333F }, /* R84 - DC Servo (1) */ - { 0x0FEF, 0x0FEF }, /* R85 - DC Servo (2) */ - { 0x0000, 0x0000 }, /* R86 */ - { 0xFFFF, 0xFFFF }, /* R87 - DC Servo (4) */ - { 0x0333, 0x0000 }, /* R88 - DC Servo Readback */ - { 0x0000, 0x0000 }, /* R89 */ - { 0x0000, 0x0000 }, /* R90 */ - { 0x0000, 0x0000 }, /* R91 */ - { 0x0000, 0x0000 }, /* R92 */ - { 0x0000, 0x0000 }, /* R93 */ - { 0x0000, 0x0000 }, /* R94 */ - { 0x0000, 0x0000 }, /* R95 */ - { 0x00EE, 0x00EE }, /* R96 - Analogue HP (1) */ - { 0x0000, 0x0000 }, /* R97 */ - { 0x0000, 0x0000 }, /* R98 */ - { 0x0000, 0x0000 }, /* R99 */ - { 0x0000, 0x0000 }, /* R100 */ - { 0x0000, 0x0000 }, /* R101 */ - { 0x0000, 0x0000 }, /* R102 */ - { 0x0000, 0x0000 }, /* R103 */ - { 0x0000, 0x0000 }, /* R104 */ - { 0x0000, 0x0000 }, /* R105 */ - { 0x0000, 0x0000 }, /* R106 */ - { 0x0000, 0x0000 }, /* R107 */ - { 0x0000, 0x0000 }, /* R108 */ - { 0x0000, 0x0000 }, /* R109 */ - { 0x0000, 0x0000 }, /* R110 */ - { 0x0000, 0x0000 }, /* R111 */ - { 0x0000, 0x0000 }, /* R112 */ - { 0x0000, 0x0000 }, /* R113 */ - { 0x0000, 0x0000 }, /* R114 */ - { 0x0000, 0x0000 }, /* R115 */ - { 0x0000, 0x0000 }, /* R116 */ - { 0x0000, 0x0000 }, /* R117 */ - { 0x0000, 0x0000 }, /* R118 */ - { 0x0000, 0x0000 }, /* R119 */ - { 0x0000, 0x0000 }, /* R120 */ - { 0x0000, 0x0000 }, /* R121 */ - { 0x0000, 0x0000 }, /* R122 */ - { 0x0000, 0x0000 }, /* R123 */ - { 0x0000, 0x0000 }, /* R124 */ - { 0x0000, 0x0000 }, /* R125 */ - { 0x0000, 0x0000 }, /* R126 */ - { 0x0000, 0x0000 }, /* R127 */ - { 0x0000, 0x0000 }, /* R128 */ - { 0x0000, 0x0000 }, /* R129 */ - { 0x0000, 0x0000 }, /* R130 */ - { 0x0000, 0x0000 }, /* R131 */ - { 0x0000, 0x0000 }, /* R132 */ - { 0x0000, 0x0000 }, /* R133 */ - { 0x0000, 0x0000 }, /* R134 */ - { 0x0000, 0x0000 }, /* R135 */ - { 0x0000, 0x0000 }, /* R136 */ - { 0x0000, 0x0000 }, /* R137 */ - { 0x0000, 0x0000 }, /* R138 */ - { 0x0000, 0x0000 }, /* R139 */ - { 0x0000, 0x0000 }, /* R140 */ - { 0x0000, 0x0000 }, /* R141 */ - { 0x0000, 0x0000 }, /* R142 */ - { 0x0000, 0x0000 }, /* R143 */ - { 0x0000, 0x0000 }, /* R144 */ - { 0x0000, 0x0000 }, /* R145 */ - { 0x0000, 0x0000 }, /* R146 */ - { 0x0000, 0x0000 }, /* R147 */ - { 0x0000, 0x0000 }, /* R148 */ - { 0x0000, 0x0000 }, /* R149 */ - { 0x0000, 0x0000 }, /* R150 */ - { 0x0000, 0x0000 }, /* R151 */ - { 0x0000, 0x0000 }, /* R152 */ - { 0x0000, 0x0000 }, /* R153 */ - { 0x0000, 0x0000 }, /* R154 */ - { 0x0000, 0x0000 }, /* R155 */ - { 0x0000, 0x0000 }, /* R156 */ - { 0x0000, 0x0000 }, /* R157 */ - { 0x0000, 0x0000 }, /* R158 */ - { 0x0000, 0x0000 }, /* R159 */ - { 0x0000, 0x0000 }, /* R160 */ - { 0x0000, 0x0000 }, /* R161 */ - { 0x0000, 0x0000 }, /* R162 */ - { 0x0000, 0x0000 }, /* R163 */ - { 0x0000, 0x0000 }, /* R164 */ - { 0x0000, 0x0000 }, /* R165 */ - { 0x0000, 0x0000 }, /* R166 */ - { 0x0000, 0x0000 }, /* R167 */ - { 0x0000, 0x0000 }, /* R168 */ - { 0x0000, 0x0000 }, /* R169 */ - { 0x0000, 0x0000 }, /* R170 */ - { 0x0000, 0x0000 }, /* R171 */ - { 0x0000, 0x0000 }, /* R172 */ - { 0x0000, 0x0000 }, /* R173 */ - { 0x0000, 0x0000 }, /* R174 */ - { 0x0000, 0x0000 }, /* R175 */ - { 0x0000, 0x0000 }, /* R176 */ - { 0x0000, 0x0000 }, /* R177 */ - { 0x0000, 0x0000 }, /* R178 */ - { 0x0000, 0x0000 }, /* R179 */ - { 0x0000, 0x0000 }, /* R180 */ - { 0x0000, 0x0000 }, /* R181 */ - { 0x0000, 0x0000 }, /* R182 */ - { 0x0000, 0x0000 }, /* R183 */ - { 0x0000, 0x0000 }, /* R184 */ - { 0x0000, 0x0000 }, /* R185 */ - { 0x0000, 0x0000 }, /* R186 */ - { 0x0000, 0x0000 }, /* R187 */ - { 0x0000, 0x0000 }, /* R188 */ - { 0x0000, 0x0000 }, /* R189 */ - { 0x0000, 0x0000 }, /* R190 */ - { 0x0000, 0x0000 }, /* R191 */ - { 0x0000, 0x0000 }, /* R192 */ - { 0x0000, 0x0000 }, /* R193 */ - { 0x0000, 0x0000 }, /* R194 */ - { 0x0000, 0x0000 }, /* R195 */ - { 0x0000, 0x0000 }, /* R196 */ - { 0x0000, 0x0000 }, /* R197 */ - { 0x0000, 0x0000 }, /* R198 */ - { 0x0000, 0x0000 }, /* R199 */ - { 0x0000, 0x0000 }, /* R200 */ - { 0x0000, 0x0000 }, /* R201 */ - { 0x0000, 0x0000 }, /* R202 */ - { 0x0000, 0x0000 }, /* R203 */ - { 0x0000, 0x0000 }, /* R204 */ - { 0x0000, 0x0000 }, /* R205 */ - { 0x0000, 0x0000 }, /* R206 */ - { 0x0000, 0x0000 }, /* R207 */ - { 0xFFFF, 0xFFFF }, /* R208 */ - { 0xFFFF, 0xFFFF }, /* R209 */ - { 0xFFFF, 0xFFFF }, /* R210 */ - { 0x0000, 0x0000 }, /* R211 */ - { 0x0000, 0x0000 }, /* R212 */ - { 0x0000, 0x0000 }, /* R213 */ - { 0x0000, 0x0000 }, /* R214 */ - { 0x0000, 0x0000 }, /* R215 */ - { 0x0000, 0x0000 }, /* R216 */ - { 0x0000, 0x0000 }, /* R217 */ - { 0x0000, 0x0000 }, /* R218 */ - { 0x0000, 0x0000 }, /* R219 */ - { 0x0000, 0x0000 }, /* R220 */ - { 0x0000, 0x0000 }, /* R221 */ - { 0x0000, 0x0000 }, /* R222 */ - { 0x0000, 0x0000 }, /* R223 */ - { 0x0000, 0x0000 }, /* R224 */ - { 0x0000, 0x0000 }, /* R225 */ - { 0x0000, 0x0000 }, /* R226 */ - { 0x0000, 0x0000 }, /* R227 */ - { 0x0000, 0x0000 }, /* R228 */ - { 0x0000, 0x0000 }, /* R229 */ - { 0x0000, 0x0000 }, /* R230 */ - { 0x0000, 0x0000 }, /* R231 */ - { 0x0000, 0x0000 }, /* R232 */ - { 0x0000, 0x0000 }, /* R233 */ - { 0x0000, 0x0000 }, /* R234 */ - { 0x0000, 0x0000 }, /* R235 */ - { 0x0000, 0x0000 }, /* R236 */ - { 0x0000, 0x0000 }, /* R237 */ - { 0x0000, 0x0000 }, /* R238 */ - { 0x0000, 0x0000 }, /* R239 */ - { 0x0000, 0x0000 }, /* R240 */ - { 0x0000, 0x0000 }, /* R241 */ - { 0x0000, 0x0000 }, /* R242 */ - { 0x0000, 0x0000 }, /* R243 */ - { 0x0000, 0x0000 }, /* R244 */ - { 0x0000, 0x0000 }, /* R245 */ - { 0x0000, 0x0000 }, /* R246 */ - { 0x0000, 0x0000 }, /* R247 */ - { 0x0000, 0x0000 }, /* R248 */ - { 0x0000, 0x0000 }, /* R249 */ - { 0x0000, 0x0000 }, /* R250 */ - { 0x0000, 0x0000 }, /* R251 */ - { 0x0000, 0x0000 }, /* R252 */ - { 0x0000, 0x0000 }, /* R253 */ - { 0x0000, 0x0000 }, /* R254 */ - { 0x0000, 0x0000 }, /* R255 */ - { 0x000F, 0x0000 }, /* R256 - Chip Revision */ - { 0x0074, 0x0074 }, /* R257 - Control Interface */ - { 0x0000, 0x0000 }, /* R258 */ - { 0x0000, 0x0000 }, /* R259 */ - { 0x0000, 0x0000 }, /* R260 */ - { 0x0000, 0x0000 }, /* R261 */ - { 0x0000, 0x0000 }, /* R262 */ - { 0x0000, 0x0000 }, /* R263 */ - { 0x0000, 0x0000 }, /* R264 */ - { 0x0000, 0x0000 }, /* R265 */ - { 0x0000, 0x0000 }, /* R266 */ - { 0x0000, 0x0000 }, /* R267 */ - { 0x0000, 0x0000 }, /* R268 */ - { 0x0000, 0x0000 }, /* R269 */ - { 0x0000, 0x0000 }, /* R270 */ - { 0x0000, 0x0000 }, /* R271 */ - { 0x807F, 0x837F }, /* R272 - Write Sequencer Ctrl (1) */ - { 0x017F, 0x0000 }, /* R273 - Write Sequencer Ctrl (2) */ - { 0x0000, 0x0000 }, /* R274 */ - { 0x0000, 0x0000 }, /* R275 */ - { 0x0000, 0x0000 }, /* R276 */ - { 0x0000, 0x0000 }, /* R277 */ - { 0x0000, 0x0000 }, /* R278 */ - { 0x0000, 0x0000 }, /* R279 */ - { 0x0000, 0x0000 }, /* R280 */ - { 0x0000, 0x0000 }, /* R281 */ - { 0x0000, 0x0000 }, /* R282 */ - { 0x0000, 0x0000 }, /* R283 */ - { 0x0000, 0x0000 }, /* R284 */ - { 0x0000, 0x0000 }, /* R285 */ - { 0x0000, 0x0000 }, /* R286 */ - { 0x0000, 0x0000 }, /* R287 */ - { 0x0000, 0x0000 }, /* R288 */ - { 0x0000, 0x0000 }, /* R289 */ - { 0x0000, 0x0000 }, /* R290 */ - { 0x0000, 0x0000 }, /* R291 */ - { 0x0000, 0x0000 }, /* R292 */ - { 0x0000, 0x0000 }, /* R293 */ - { 0x0000, 0x0000 }, /* R294 */ - { 0x0000, 0x0000 }, /* R295 */ - { 0x0000, 0x0000 }, /* R296 */ - { 0x0000, 0x0000 }, /* R297 */ - { 0x0000, 0x0000 }, /* R298 */ - { 0x0000, 0x0000 }, /* R299 */ - { 0x0000, 0x0000 }, /* R300 */ - { 0x0000, 0x0000 }, /* R301 */ - { 0x0000, 0x0000 }, /* R302 */ - { 0x0000, 0x0000 }, /* R303 */ - { 0x0000, 0x0000 }, /* R304 */ - { 0x0000, 0x0000 }, /* R305 */ - { 0x0000, 0x0000 }, /* R306 */ - { 0x0000, 0x0000 }, /* R307 */ - { 0x0000, 0x0000 }, /* R308 */ - { 0x0000, 0x0000 }, /* R309 */ - { 0x0000, 0x0000 }, /* R310 */ - { 0x0000, 0x0000 }, /* R311 */ - { 0x0000, 0x0000 }, /* R312 */ - { 0x0000, 0x0000 }, /* R313 */ - { 0x0000, 0x0000 }, /* R314 */ - { 0x0000, 0x0000 }, /* R315 */ - { 0x0000, 0x0000 }, /* R316 */ - { 0x0000, 0x0000 }, /* R317 */ - { 0x0000, 0x0000 }, /* R318 */ - { 0x0000, 0x0000 }, /* R319 */ - { 0x0000, 0x0000 }, /* R320 */ - { 0x0000, 0x0000 }, /* R321 */ - { 0x0000, 0x0000 }, /* R322 */ - { 0x0000, 0x0000 }, /* R323 */ - { 0x0000, 0x0000 }, /* R324 */ - { 0x0000, 0x0000 }, /* R325 */ - { 0x0000, 0x0000 }, /* R326 */ - { 0x0000, 0x0000 }, /* R327 */ - { 0x0000, 0x0000 }, /* R328 */ - { 0x0000, 0x0000 }, /* R329 */ - { 0x0000, 0x0000 }, /* R330 */ - { 0x0000, 0x0000 }, /* R331 */ - { 0x0000, 0x0000 }, /* R332 */ - { 0x0000, 0x0000 }, /* R333 */ - { 0x0000, 0x0000 }, /* R334 */ - { 0x0000, 0x0000 }, /* R335 */ - { 0x0000, 0x0000 }, /* R336 */ - { 0x0000, 0x0000 }, /* R337 */ - { 0x0000, 0x0000 }, /* R338 */ - { 0x0000, 0x0000 }, /* R339 */ - { 0x0000, 0x0000 }, /* R340 */ - { 0x0000, 0x0000 }, /* R341 */ - { 0x0000, 0x0000 }, /* R342 */ - { 0x0000, 0x0000 }, /* R343 */ - { 0x0000, 0x0000 }, /* R344 */ - { 0x0000, 0x0000 }, /* R345 */ - { 0x0000, 0x0000 }, /* R346 */ - { 0x0000, 0x0000 }, /* R347 */ - { 0x0000, 0x0000 }, /* R348 */ - { 0x0000, 0x0000 }, /* R349 */ - { 0x0000, 0x0000 }, /* R350 */ - { 0x0000, 0x0000 }, /* R351 */ - { 0x0000, 0x0000 }, /* R352 */ - { 0x0000, 0x0000 }, /* R353 */ - { 0x0000, 0x0000 }, /* R354 */ - { 0x0000, 0x0000 }, /* R355 */ - { 0x0000, 0x0000 }, /* R356 */ - { 0x0000, 0x0000 }, /* R357 */ - { 0x0000, 0x0000 }, /* R358 */ - { 0x0000, 0x0000 }, /* R359 */ - { 0x0000, 0x0000 }, /* R360 */ - { 0x0000, 0x0000 }, /* R361 */ - { 0x0000, 0x0000 }, /* R362 */ - { 0x0000, 0x0000 }, /* R363 */ - { 0x0000, 0x0000 }, /* R364 */ - { 0x0000, 0x0000 }, /* R365 */ - { 0x0000, 0x0000 }, /* R366 */ - { 0x0000, 0x0000 }, /* R367 */ - { 0x0000, 0x0000 }, /* R368 */ - { 0x0000, 0x0000 }, /* R369 */ - { 0x0000, 0x0000 }, /* R370 */ - { 0x0000, 0x0000 }, /* R371 */ - { 0x0000, 0x0000 }, /* R372 */ - { 0x0000, 0x0000 }, /* R373 */ - { 0x0000, 0x0000 }, /* R374 */ - { 0x0000, 0x0000 }, /* R375 */ - { 0x0000, 0x0000 }, /* R376 */ - { 0x0000, 0x0000 }, /* R377 */ - { 0x0000, 0x0000 }, /* R378 */ - { 0x0000, 0x0000 }, /* R379 */ - { 0x0000, 0x0000 }, /* R380 */ - { 0x0000, 0x0000 }, /* R381 */ - { 0x0000, 0x0000 }, /* R382 */ - { 0x0000, 0x0000 }, /* R383 */ - { 0x0000, 0x0000 }, /* R384 */ - { 0x0000, 0x0000 }, /* R385 */ - { 0x0000, 0x0000 }, /* R386 */ - { 0x0000, 0x0000 }, /* R387 */ - { 0x0000, 0x0000 }, /* R388 */ - { 0x0000, 0x0000 }, /* R389 */ - { 0x0000, 0x0000 }, /* R390 */ - { 0x0000, 0x0000 }, /* R391 */ - { 0x0000, 0x0000 }, /* R392 */ - { 0x0000, 0x0000 }, /* R393 */ - { 0x0000, 0x0000 }, /* R394 */ - { 0x0000, 0x0000 }, /* R395 */ - { 0x0000, 0x0000 }, /* R396 */ - { 0x0000, 0x0000 }, /* R397 */ - { 0x0000, 0x0000 }, /* R398 */ - { 0x0000, 0x0000 }, /* R399 */ - { 0x0000, 0x0000 }, /* R400 */ - { 0x0000, 0x0000 }, /* R401 */ - { 0x0000, 0x0000 }, /* R402 */ - { 0x0000, 0x0000 }, /* R403 */ - { 0x0000, 0x0000 }, /* R404 */ - { 0x0000, 0x0000 }, /* R405 */ - { 0x0000, 0x0000 }, /* R406 */ - { 0x0000, 0x0000 }, /* R407 */ - { 0x0000, 0x0000 }, /* R408 */ - { 0x0000, 0x0000 }, /* R409 */ - { 0x0000, 0x0000 }, /* R410 */ - { 0x0000, 0x0000 }, /* R411 */ - { 0x0000, 0x0000 }, /* R412 */ - { 0x0000, 0x0000 }, /* R413 */ - { 0x0000, 0x0000 }, /* R414 */ - { 0x0000, 0x0000 }, /* R415 */ - { 0x0000, 0x0000 }, /* R416 */ - { 0x0000, 0x0000 }, /* R417 */ - { 0x0000, 0x0000 }, /* R418 */ - { 0x0000, 0x0000 }, /* R419 */ - { 0x0000, 0x0000 }, /* R420 */ - { 0x0000, 0x0000 }, /* R421 */ - { 0x0000, 0x0000 }, /* R422 */ - { 0x0000, 0x0000 }, /* R423 */ - { 0x0000, 0x0000 }, /* R424 */ - { 0x0000, 0x0000 }, /* R425 */ - { 0x0000, 0x0000 }, /* R426 */ - { 0x0000, 0x0000 }, /* R427 */ - { 0x0000, 0x0000 }, /* R428 */ - { 0x0000, 0x0000 }, /* R429 */ - { 0x0000, 0x0000 }, /* R430 */ - { 0x0000, 0x0000 }, /* R431 */ - { 0x0000, 0x0000 }, /* R432 */ - { 0x0000, 0x0000 }, /* R433 */ - { 0x0000, 0x0000 }, /* R434 */ - { 0x0000, 0x0000 }, /* R435 */ - { 0x0000, 0x0000 }, /* R436 */ - { 0x0000, 0x0000 }, /* R437 */ - { 0x0000, 0x0000 }, /* R438 */ - { 0x0000, 0x0000 }, /* R439 */ - { 0x0000, 0x0000 }, /* R440 */ - { 0x0000, 0x0000 }, /* R441 */ - { 0x0000, 0x0000 }, /* R442 */ - { 0x0000, 0x0000 }, /* R443 */ - { 0x0000, 0x0000 }, /* R444 */ - { 0x0000, 0x0000 }, /* R445 */ - { 0x0000, 0x0000 }, /* R446 */ - { 0x0000, 0x0000 }, /* R447 */ - { 0x0000, 0x0000 }, /* R448 */ - { 0x0000, 0x0000 }, /* R449 */ - { 0x0000, 0x0000 }, /* R450 */ - { 0x0000, 0x0000 }, /* R451 */ - { 0x0000, 0x0000 }, /* R452 */ - { 0x0000, 0x0000 }, /* R453 */ - { 0x0000, 0x0000 }, /* R454 */ - { 0x0000, 0x0000 }, /* R455 */ - { 0x0000, 0x0000 }, /* R456 */ - { 0x0000, 0x0000 }, /* R457 */ - { 0x0000, 0x0000 }, /* R458 */ - { 0x0000, 0x0000 }, /* R459 */ - { 0x0000, 0x0000 }, /* R460 */ - { 0x0000, 0x0000 }, /* R461 */ - { 0x0000, 0x0000 }, /* R462 */ - { 0x0000, 0x0000 }, /* R463 */ - { 0x0000, 0x0000 }, /* R464 */ - { 0x0000, 0x0000 }, /* R465 */ - { 0x0000, 0x0000 }, /* R466 */ - { 0x0000, 0x0000 }, /* R467 */ - { 0x0000, 0x0000 }, /* R468 */ - { 0x0000, 0x0000 }, /* R469 */ - { 0x0000, 0x0000 }, /* R470 */ - { 0x0000, 0x0000 }, /* R471 */ - { 0x0000, 0x0000 }, /* R472 */ - { 0x0000, 0x0000 }, /* R473 */ - { 0x0000, 0x0000 }, /* R474 */ - { 0x0000, 0x0000 }, /* R475 */ - { 0x0000, 0x0000 }, /* R476 */ - { 0x0000, 0x0000 }, /* R477 */ - { 0x0000, 0x0000 }, /* R478 */ - { 0x0000, 0x0000 }, /* R479 */ - { 0x0000, 0x0000 }, /* R480 */ - { 0x0000, 0x0000 }, /* R481 */ - { 0x0000, 0x0000 }, /* R482 */ - { 0x0000, 0x0000 }, /* R483 */ - { 0x0000, 0x0000 }, /* R484 */ - { 0x0000, 0x0000 }, /* R485 */ - { 0x0000, 0x0000 }, /* R486 */ - { 0x0000, 0x0000 }, /* R487 */ - { 0x0000, 0x0000 }, /* R488 */ - { 0x0000, 0x0000 }, /* R489 */ - { 0x0000, 0x0000 }, /* R490 */ - { 0x0000, 0x0000 }, /* R491 */ - { 0x0000, 0x0000 }, /* R492 */ - { 0x0000, 0x0000 }, /* R493 */ - { 0x0000, 0x0000 }, /* R494 */ - { 0x0000, 0x0000 }, /* R495 */ - { 0x0000, 0x0000 }, /* R496 */ - { 0x0000, 0x0000 }, /* R497 */ - { 0x0000, 0x0000 }, /* R498 */ - { 0x0000, 0x0000 }, /* R499 */ - { 0x0000, 0x0000 }, /* R500 */ - { 0x0000, 0x0000 }, /* R501 */ - { 0x0000, 0x0000 }, /* R502 */ - { 0x0000, 0x0000 }, /* R503 */ - { 0x0000, 0x0000 }, /* R504 */ - { 0x0000, 0x0000 }, /* R505 */ - { 0x0000, 0x0000 }, /* R506 */ - { 0x0000, 0x0000 }, /* R507 */ - { 0x0000, 0x0000 }, /* R508 */ - { 0x0000, 0x0000 }, /* R509 */ - { 0x0000, 0x0000 }, /* R510 */ - { 0x0000, 0x0000 }, /* R511 */ - { 0x001F, 0x001F }, /* R512 - AIF1 Clocking (1) */ - { 0x003F, 0x003F }, /* R513 - AIF1 Clocking (2) */ - { 0x0000, 0x0000 }, /* R514 */ - { 0x0000, 0x0000 }, /* R515 */ - { 0x001F, 0x001F }, /* R516 - AIF2 Clocking (1) */ - { 0x003F, 0x003F }, /* R517 - AIF2 Clocking (2) */ - { 0x0000, 0x0000 }, /* R518 */ - { 0x0000, 0x0000 }, /* R519 */ - { 0x001F, 0x001F }, /* R520 - Clocking (1) */ - { 0x0777, 0x0777 }, /* R521 - Clocking (2) */ - { 0x0000, 0x0000 }, /* R522 */ - { 0x0000, 0x0000 }, /* R523 */ - { 0x0000, 0x0000 }, /* R524 */ - { 0x0000, 0x0000 }, /* R525 */ - { 0x0000, 0x0000 }, /* R526 */ - { 0x0000, 0x0000 }, /* R527 */ - { 0x00FF, 0x00FF }, /* R528 - AIF1 Rate */ - { 0x00FF, 0x00FF }, /* R529 - AIF2 Rate */ - { 0x000F, 0x0000 }, /* R530 - Rate Status */ - { 0x0000, 0x0000 }, /* R531 */ - { 0x0000, 0x0000 }, /* R532 */ - { 0x0000, 0x0000 }, /* R533 */ - { 0x0000, 0x0000 }, /* R534 */ - { 0x0000, 0x0000 }, /* R535 */ - { 0x0000, 0x0000 }, /* R536 */ - { 0x0000, 0x0000 }, /* R537 */ - { 0x0000, 0x0000 }, /* R538 */ - { 0x0000, 0x0000 }, /* R539 */ - { 0x0000, 0x0000 }, /* R540 */ - { 0x0000, 0x0000 }, /* R541 */ - { 0x0000, 0x0000 }, /* R542 */ - { 0x0000, 0x0000 }, /* R543 */ - { 0x0007, 0x0007 }, /* R544 - FLL1 Control (1) */ - { 0x3F77, 0x3F77 }, /* R545 - FLL1 Control (2) */ - { 0xFFFF, 0xFFFF }, /* R546 - FLL1 Control (3) */ - { 0x7FEF, 0x7FEF }, /* R547 - FLL1 Control (4) */ - { 0x1FDB, 0x1FDB }, /* R548 - FLL1 Control (5) */ - { 0x0000, 0x0000 }, /* R549 */ - { 0x0000, 0x0000 }, /* R550 */ - { 0x0000, 0x0000 }, /* R551 */ - { 0x0000, 0x0000 }, /* R552 */ - { 0x0000, 0x0000 }, /* R553 */ - { 0x0000, 0x0000 }, /* R554 */ - { 0x0000, 0x0000 }, /* R555 */ - { 0x0000, 0x0000 }, /* R556 */ - { 0x0000, 0x0000 }, /* R557 */ - { 0x0000, 0x0000 }, /* R558 */ - { 0x0000, 0x0000 }, /* R559 */ - { 0x0000, 0x0000 }, /* R560 */ - { 0x0000, 0x0000 }, /* R561 */ - { 0x0000, 0x0000 }, /* R562 */ - { 0x0000, 0x0000 }, /* R563 */ - { 0x0000, 0x0000 }, /* R564 */ - { 0x0000, 0x0000 }, /* R565 */ - { 0x0000, 0x0000 }, /* R566 */ - { 0x0000, 0x0000 }, /* R567 */ - { 0x0000, 0x0000 }, /* R568 */ - { 0x0000, 0x0000 }, /* R569 */ - { 0x0000, 0x0000 }, /* R570 */ - { 0x0000, 0x0000 }, /* R571 */ - { 0x0000, 0x0000 }, /* R572 */ - { 0x0000, 0x0000 }, /* R573 */ - { 0x0000, 0x0000 }, /* R574 */ - { 0x0000, 0x0000 }, /* R575 */ - { 0x0007, 0x0007 }, /* R576 - FLL2 Control (1) */ - { 0x3F77, 0x3F77 }, /* R577 - FLL2 Control (2) */ - { 0xFFFF, 0xFFFF }, /* R578 - FLL2 Control (3) */ - { 0x7FEF, 0x7FEF }, /* R579 - FLL2 Control (4) */ - { 0x1FDB, 0x1FDB }, /* R580 - FLL2 Control (5) */ - { 0x0000, 0x0000 }, /* R581 */ - { 0x0000, 0x0000 }, /* R582 */ - { 0x0000, 0x0000 }, /* R583 */ - { 0x0000, 0x0000 }, /* R584 */ - { 0x0000, 0x0000 }, /* R585 */ - { 0x0000, 0x0000 }, /* R586 */ - { 0x0000, 0x0000 }, /* R587 */ - { 0x0000, 0x0000 }, /* R588 */ - { 0x0000, 0x0000 }, /* R589 */ - { 0x0000, 0x0000 }, /* R590 */ - { 0x0000, 0x0000 }, /* R591 */ - { 0x0000, 0x0000 }, /* R592 */ - { 0x0000, 0x0000 }, /* R593 */ - { 0x0000, 0x0000 }, /* R594 */ - { 0x0000, 0x0000 }, /* R595 */ - { 0x0000, 0x0000 }, /* R596 */ - { 0x0000, 0x0000 }, /* R597 */ - { 0x0000, 0x0000 }, /* R598 */ - { 0x0000, 0x0000 }, /* R599 */ - { 0x0000, 0x0000 }, /* R600 */ - { 0x0000, 0x0000 }, /* R601 */ - { 0x0000, 0x0000 }, /* R602 */ - { 0x0000, 0x0000 }, /* R603 */ - { 0x0000, 0x0000 }, /* R604 */ - { 0x0000, 0x0000 }, /* R605 */ - { 0x0000, 0x0000 }, /* R606 */ - { 0x0000, 0x0000 }, /* R607 */ - { 0x0000, 0x0000 }, /* R608 */ - { 0x0000, 0x0000 }, /* R609 */ - { 0x0000, 0x0000 }, /* R610 */ - { 0x0000, 0x0000 }, /* R611 */ - { 0x0000, 0x0000 }, /* R612 */ - { 0x0000, 0x0000 }, /* R613 */ - { 0x0000, 0x0000 }, /* R614 */ - { 0x0000, 0x0000 }, /* R615 */ - { 0x0000, 0x0000 }, /* R616 */ - { 0x0000, 0x0000 }, /* R617 */ - { 0x0000, 0x0000 }, /* R618 */ - { 0x0000, 0x0000 }, /* R619 */ - { 0x0000, 0x0000 }, /* R620 */ - { 0x0000, 0x0000 }, /* R621 */ - { 0x0000, 0x0000 }, /* R622 */ - { 0x0000, 0x0000 }, /* R623 */ - { 0x0000, 0x0000 }, /* R624 */ - { 0x0000, 0x0000 }, /* R625 */ - { 0x0000, 0x0000 }, /* R626 */ - { 0x0000, 0x0000 }, /* R627 */ - { 0x0000, 0x0000 }, /* R628 */ - { 0x0000, 0x0000 }, /* R629 */ - { 0x0000, 0x0000 }, /* R630 */ - { 0x0000, 0x0000 }, /* R631 */ - { 0x0000, 0x0000 }, /* R632 */ - { 0x0000, 0x0000 }, /* R633 */ - { 0x0000, 0x0000 }, /* R634 */ - { 0x0000, 0x0000 }, /* R635 */ - { 0x0000, 0x0000 }, /* R636 */ - { 0x0000, 0x0000 }, /* R637 */ - { 0x0000, 0x0000 }, /* R638 */ - { 0x0000, 0x0000 }, /* R639 */ - { 0x0000, 0x0000 }, /* R640 */ - { 0x0000, 0x0000 }, /* R641 */ - { 0x0000, 0x0000 }, /* R642 */ - { 0x0000, 0x0000 }, /* R643 */ - { 0x0000, 0x0000 }, /* R644 */ - { 0x0000, 0x0000 }, /* R645 */ - { 0x0000, 0x0000 }, /* R646 */ - { 0x0000, 0x0000 }, /* R647 */ - { 0x0000, 0x0000 }, /* R648 */ - { 0x0000, 0x0000 }, /* R649 */ - { 0x0000, 0x0000 }, /* R650 */ - { 0x0000, 0x0000 }, /* R651 */ - { 0x0000, 0x0000 }, /* R652 */ - { 0x0000, 0x0000 }, /* R653 */ - { 0x0000, 0x0000 }, /* R654 */ - { 0x0000, 0x0000 }, /* R655 */ - { 0x0000, 0x0000 }, /* R656 */ - { 0x0000, 0x0000 }, /* R657 */ - { 0x0000, 0x0000 }, /* R658 */ - { 0x0000, 0x0000 }, /* R659 */ - { 0x0000, 0x0000 }, /* R660 */ - { 0x0000, 0x0000 }, /* R661 */ - { 0x0000, 0x0000 }, /* R662 */ - { 0x0000, 0x0000 }, /* R663 */ - { 0x0000, 0x0000 }, /* R664 */ - { 0x0000, 0x0000 }, /* R665 */ - { 0x0000, 0x0000 }, /* R666 */ - { 0x0000, 0x0000 }, /* R667 */ - { 0x0000, 0x0000 }, /* R668 */ - { 0x0000, 0x0000 }, /* R669 */ - { 0x0000, 0x0000 }, /* R670 */ - { 0x0000, 0x0000 }, /* R671 */ - { 0x0000, 0x0000 }, /* R672 */ - { 0x0000, 0x0000 }, /* R673 */ - { 0x0000, 0x0000 }, /* R674 */ - { 0x0000, 0x0000 }, /* R675 */ - { 0x0000, 0x0000 }, /* R676 */ - { 0x0000, 0x0000 }, /* R677 */ - { 0x0000, 0x0000 }, /* R678 */ - { 0x0000, 0x0000 }, /* R679 */ - { 0x0000, 0x0000 }, /* R680 */ - { 0x0000, 0x0000 }, /* R681 */ - { 0x0000, 0x0000 }, /* R682 */ - { 0x0000, 0x0000 }, /* R683 */ - { 0x0000, 0x0000 }, /* R684 */ - { 0x0000, 0x0000 }, /* R685 */ - { 0x0000, 0x0000 }, /* R686 */ - { 0x0000, 0x0000 }, /* R687 */ - { 0x0000, 0x0000 }, /* R688 */ - { 0x0000, 0x0000 }, /* R689 */ - { 0x0000, 0x0000 }, /* R690 */ - { 0x0000, 0x0000 }, /* R691 */ - { 0x0000, 0x0000 }, /* R692 */ - { 0x0000, 0x0000 }, /* R693 */ - { 0x0000, 0x0000 }, /* R694 */ - { 0x0000, 0x0000 }, /* R695 */ - { 0x0000, 0x0000 }, /* R696 */ - { 0x0000, 0x0000 }, /* R697 */ - { 0x0000, 0x0000 }, /* R698 */ - { 0x0000, 0x0000 }, /* R699 */ - { 0x0000, 0x0000 }, /* R700 */ - { 0x0000, 0x0000 }, /* R701 */ - { 0x0000, 0x0000 }, /* R702 */ - { 0x0000, 0x0000 }, /* R703 */ - { 0x0000, 0x0000 }, /* R704 */ - { 0x0000, 0x0000 }, /* R705 */ - { 0x0000, 0x0000 }, /* R706 */ - { 0x0000, 0x0000 }, /* R707 */ - { 0x0000, 0x0000 }, /* R708 */ - { 0x0000, 0x0000 }, /* R709 */ - { 0x0000, 0x0000 }, /* R710 */ - { 0x0000, 0x0000 }, /* R711 */ - { 0x0000, 0x0000 }, /* R712 */ - { 0x0000, 0x0000 }, /* R713 */ - { 0x0000, 0x0000 }, /* R714 */ - { 0x0000, 0x0000 }, /* R715 */ - { 0x0000, 0x0000 }, /* R716 */ - { 0x0000, 0x0000 }, /* R717 */ - { 0x0000, 0x0000 }, /* R718 */ - { 0x0000, 0x0000 }, /* R719 */ - { 0x0000, 0x0000 }, /* R720 */ - { 0x0000, 0x0000 }, /* R721 */ - { 0x0000, 0x0000 }, /* R722 */ - { 0x0000, 0x0000 }, /* R723 */ - { 0x0000, 0x0000 }, /* R724 */ - { 0x0000, 0x0000 }, /* R725 */ - { 0x0000, 0x0000 }, /* R726 */ - { 0x0000, 0x0000 }, /* R727 */ - { 0x0000, 0x0000 }, /* R728 */ - { 0x0000, 0x0000 }, /* R729 */ - { 0x0000, 0x0000 }, /* R730 */ - { 0x0000, 0x0000 }, /* R731 */ - { 0x0000, 0x0000 }, /* R732 */ - { 0x0000, 0x0000 }, /* R733 */ - { 0x0000, 0x0000 }, /* R734 */ - { 0x0000, 0x0000 }, /* R735 */ - { 0x0000, 0x0000 }, /* R736 */ - { 0x0000, 0x0000 }, /* R737 */ - { 0x0000, 0x0000 }, /* R738 */ - { 0x0000, 0x0000 }, /* R739 */ - { 0x0000, 0x0000 }, /* R740 */ - { 0x0000, 0x0000 }, /* R741 */ - { 0x0000, 0x0000 }, /* R742 */ - { 0x0000, 0x0000 }, /* R743 */ - { 0x0000, 0x0000 }, /* R744 */ - { 0x0000, 0x0000 }, /* R745 */ - { 0x0000, 0x0000 }, /* R746 */ - { 0x0000, 0x0000 }, /* R747 */ - { 0x0000, 0x0000 }, /* R748 */ - { 0x0000, 0x0000 }, /* R749 */ - { 0x0000, 0x0000 }, /* R750 */ - { 0x0000, 0x0000 }, /* R751 */ - { 0x0000, 0x0000 }, /* R752 */ - { 0x0000, 0x0000 }, /* R753 */ - { 0x0000, 0x0000 }, /* R754 */ - { 0x0000, 0x0000 }, /* R755 */ - { 0x0000, 0x0000 }, /* R756 */ - { 0x0000, 0x0000 }, /* R757 */ - { 0x0000, 0x0000 }, /* R758 */ - { 0x0000, 0x0000 }, /* R759 */ - { 0x0000, 0x0000 }, /* R760 */ - { 0x0000, 0x0000 }, /* R761 */ - { 0x0000, 0x0000 }, /* R762 */ - { 0x0000, 0x0000 }, /* R763 */ - { 0x0000, 0x0000 }, /* R764 */ - { 0x0000, 0x0000 }, /* R765 */ - { 0x0000, 0x0000 }, /* R766 */ - { 0x0000, 0x0000 }, /* R767 */ - { 0xE1F8, 0xE1F8 }, /* R768 - AIF1 Control (1) */ - { 0xCD1F, 0xCD1F }, /* R769 - AIF1 Control (2) */ - { 0xF000, 0xF000 }, /* R770 - AIF1 Master/Slave */ - { 0x01F0, 0x01F0 }, /* R771 - AIF1 BCLK */ - { 0x0FFF, 0x0FFF }, /* R772 - AIF1ADC LRCLK */ - { 0x0FFF, 0x0FFF }, /* R773 - AIF1DAC LRCLK */ - { 0x0003, 0x0003 }, /* R774 - AIF1DAC Data */ - { 0x0003, 0x0003 }, /* R775 - AIF1ADC Data */ - { 0x0000, 0x0000 }, /* R776 */ - { 0x0000, 0x0000 }, /* R777 */ - { 0x0000, 0x0000 }, /* R778 */ - { 0x0000, 0x0000 }, /* R779 */ - { 0x0000, 0x0000 }, /* R780 */ - { 0x0000, 0x0000 }, /* R781 */ - { 0x0000, 0x0000 }, /* R782 */ - { 0x0000, 0x0000 }, /* R783 */ - { 0xF1F8, 0xF1F8 }, /* R784 - AIF2 Control (1) */ - { 0xFD1F, 0xFD1F }, /* R785 - AIF2 Control (2) */ - { 0xF000, 0xF000 }, /* R786 - AIF2 Master/Slave */ - { 0x01F0, 0x01F0 }, /* R787 - AIF2 BCLK */ - { 0x0FFF, 0x0FFF }, /* R788 - AIF2ADC LRCLK */ - { 0x0FFF, 0x0FFF }, /* R789 - AIF2DAC LRCLK */ - { 0x0003, 0x0003 }, /* R790 - AIF2DAC Data */ - { 0x0003, 0x0003 }, /* R791 - AIF2ADC Data */ - { 0x0000, 0x0000 }, /* R792 */ - { 0x0000, 0x0000 }, /* R793 */ - { 0x0000, 0x0000 }, /* R794 */ - { 0x0000, 0x0000 }, /* R795 */ - { 0x0000, 0x0000 }, /* R796 */ - { 0x0000, 0x0000 }, /* R797 */ - { 0x0000, 0x0000 }, /* R798 */ - { 0x0000, 0x0000 }, /* R799 */ - { 0x0000, 0x0000 }, /* R800 */ - { 0x0000, 0x0000 }, /* R801 */ - { 0x0000, 0x0000 }, /* R802 */ - { 0x0000, 0x0000 }, /* R803 */ - { 0x0000, 0x0000 }, /* R804 */ - { 0x0000, 0x0000 }, /* R805 */ - { 0x0000, 0x0000 }, /* R806 */ - { 0x0000, 0x0000 }, /* R807 */ - { 0x0000, 0x0000 }, /* R808 */ - { 0x0000, 0x0000 }, /* R809 */ - { 0x0000, 0x0000 }, /* R810 */ - { 0x0000, 0x0000 }, /* R811 */ - { 0x0000, 0x0000 }, /* R812 */ - { 0x0000, 0x0000 }, /* R813 */ - { 0x0000, 0x0000 }, /* R814 */ - { 0x0000, 0x0000 }, /* R815 */ - { 0x0000, 0x0000 }, /* R816 */ - { 0x0000, 0x0000 }, /* R817 */ - { 0x0000, 0x0000 }, /* R818 */ - { 0x0000, 0x0000 }, /* R819 */ - { 0x0000, 0x0000 }, /* R820 */ - { 0x0000, 0x0000 }, /* R821 */ - { 0x0000, 0x0000 }, /* R822 */ - { 0x0000, 0x0000 }, /* R823 */ - { 0x0000, 0x0000 }, /* R824 */ - { 0x0000, 0x0000 }, /* R825 */ - { 0x0000, 0x0000 }, /* R826 */ - { 0x0000, 0x0000 }, /* R827 */ - { 0x0000, 0x0000 }, /* R828 */ - { 0x0000, 0x0000 }, /* R829 */ - { 0x0000, 0x0000 }, /* R830 */ - { 0x0000, 0x0000 }, /* R831 */ - { 0x0000, 0x0000 }, /* R832 */ - { 0x0000, 0x0000 }, /* R833 */ - { 0x0000, 0x0000 }, /* R834 */ - { 0x0000, 0x0000 }, /* R835 */ - { 0x0000, 0x0000 }, /* R836 */ - { 0x0000, 0x0000 }, /* R837 */ - { 0x0000, 0x0000 }, /* R838 */ - { 0x0000, 0x0000 }, /* R839 */ - { 0x0000, 0x0000 }, /* R840 */ - { 0x0000, 0x0000 }, /* R841 */ - { 0x0000, 0x0000 }, /* R842 */ - { 0x0000, 0x0000 }, /* R843 */ - { 0x0000, 0x0000 }, /* R844 */ - { 0x0000, 0x0000 }, /* R845 */ - { 0x0000, 0x0000 }, /* R846 */ - { 0x0000, 0x0000 }, /* R847 */ - { 0x0000, 0x0000 }, /* R848 */ - { 0x0000, 0x0000 }, /* R849 */ - { 0x0000, 0x0000 }, /* R850 */ - { 0x0000, 0x0000 }, /* R851 */ - { 0x0000, 0x0000 }, /* R852 */ - { 0x0000, 0x0000 }, /* R853 */ - { 0x0000, 0x0000 }, /* R854 */ - { 0x0000, 0x0000 }, /* R855 */ - { 0x0000, 0x0000 }, /* R856 */ - { 0x0000, 0x0000 }, /* R857 */ - { 0x0000, 0x0000 }, /* R858 */ - { 0x0000, 0x0000 }, /* R859 */ - { 0x0000, 0x0000 }, /* R860 */ - { 0x0000, 0x0000 }, /* R861 */ - { 0x0000, 0x0000 }, /* R862 */ - { 0x0000, 0x0000 }, /* R863 */ - { 0x0000, 0x0000 }, /* R864 */ - { 0x0000, 0x0000 }, /* R865 */ - { 0x0000, 0x0000 }, /* R866 */ - { 0x0000, 0x0000 }, /* R867 */ - { 0x0000, 0x0000 }, /* R868 */ - { 0x0000, 0x0000 }, /* R869 */ - { 0x0000, 0x0000 }, /* R870 */ - { 0x0000, 0x0000 }, /* R871 */ - { 0x0000, 0x0000 }, /* R872 */ - { 0x0000, 0x0000 }, /* R873 */ - { 0x0000, 0x0000 }, /* R874 */ - { 0x0000, 0x0000 }, /* R875 */ - { 0x0000, 0x0000 }, /* R876 */ - { 0x0000, 0x0000 }, /* R877 */ - { 0x0000, 0x0000 }, /* R878 */ - { 0x0000, 0x0000 }, /* R879 */ - { 0x0000, 0x0000 }, /* R880 */ - { 0x0000, 0x0000 }, /* R881 */ - { 0x0000, 0x0000 }, /* R882 */ - { 0x0000, 0x0000 }, /* R883 */ - { 0x0000, 0x0000 }, /* R884 */ - { 0x0000, 0x0000 }, /* R885 */ - { 0x0000, 0x0000 }, /* R886 */ - { 0x0000, 0x0000 }, /* R887 */ - { 0x0000, 0x0000 }, /* R888 */ - { 0x0000, 0x0000 }, /* R889 */ - { 0x0000, 0x0000 }, /* R890 */ - { 0x0000, 0x0000 }, /* R891 */ - { 0x0000, 0x0000 }, /* R892 */ - { 0x0000, 0x0000 }, /* R893 */ - { 0x0000, 0x0000 }, /* R894 */ - { 0x0000, 0x0000 }, /* R895 */ - { 0x0000, 0x0000 }, /* R896 */ - { 0x0000, 0x0000 }, /* R897 */ - { 0x0000, 0x0000 }, /* R898 */ - { 0x0000, 0x0000 }, /* R899 */ - { 0x0000, 0x0000 }, /* R900 */ - { 0x0000, 0x0000 }, /* R901 */ - { 0x0000, 0x0000 }, /* R902 */ - { 0x0000, 0x0000 }, /* R903 */ - { 0x0000, 0x0000 }, /* R904 */ - { 0x0000, 0x0000 }, /* R905 */ - { 0x0000, 0x0000 }, /* R906 */ - { 0x0000, 0x0000 }, /* R907 */ - { 0x0000, 0x0000 }, /* R908 */ - { 0x0000, 0x0000 }, /* R909 */ - { 0x0000, 0x0000 }, /* R910 */ - { 0x0000, 0x0000 }, /* R911 */ - { 0x0000, 0x0000 }, /* R912 */ - { 0x0000, 0x0000 }, /* R913 */ - { 0x0000, 0x0000 }, /* R914 */ - { 0x0000, 0x0000 }, /* R915 */ - { 0x0000, 0x0000 }, /* R916 */ - { 0x0000, 0x0000 }, /* R917 */ - { 0x0000, 0x0000 }, /* R918 */ - { 0x0000, 0x0000 }, /* R919 */ - { 0x0000, 0x0000 }, /* R920 */ - { 0x0000, 0x0000 }, /* R921 */ - { 0x0000, 0x0000 }, /* R922 */ - { 0x0000, 0x0000 }, /* R923 */ - { 0x0000, 0x0000 }, /* R924 */ - { 0x0000, 0x0000 }, /* R925 */ - { 0x0000, 0x0000 }, /* R926 */ - { 0x0000, 0x0000 }, /* R927 */ - { 0x0000, 0x0000 }, /* R928 */ - { 0x0000, 0x0000 }, /* R929 */ - { 0x0000, 0x0000 }, /* R930 */ - { 0x0000, 0x0000 }, /* R931 */ - { 0x0000, 0x0000 }, /* R932 */ - { 0x0000, 0x0000 }, /* R933 */ - { 0x0000, 0x0000 }, /* R934 */ - { 0x0000, 0x0000 }, /* R935 */ - { 0x0000, 0x0000 }, /* R936 */ - { 0x0000, 0x0000 }, /* R937 */ - { 0x0000, 0x0000 }, /* R938 */ - { 0x0000, 0x0000 }, /* R939 */ - { 0x0000, 0x0000 }, /* R940 */ - { 0x0000, 0x0000 }, /* R941 */ - { 0x0000, 0x0000 }, /* R942 */ - { 0x0000, 0x0000 }, /* R943 */ - { 0x0000, 0x0000 }, /* R944 */ - { 0x0000, 0x0000 }, /* R945 */ - { 0x0000, 0x0000 }, /* R946 */ - { 0x0000, 0x0000 }, /* R947 */ - { 0x0000, 0x0000 }, /* R948 */ - { 0x0000, 0x0000 }, /* R949 */ - { 0x0000, 0x0000 }, /* R950 */ - { 0x0000, 0x0000 }, /* R951 */ - { 0x0000, 0x0000 }, /* R952 */ - { 0x0000, 0x0000 }, /* R953 */ - { 0x0000, 0x0000 }, /* R954 */ - { 0x0000, 0x0000 }, /* R955 */ - { 0x0000, 0x0000 }, /* R956 */ - { 0x0000, 0x0000 }, /* R957 */ - { 0x0000, 0x0000 }, /* R958 */ - { 0x0000, 0x0000 }, /* R959 */ - { 0x0000, 0x0000 }, /* R960 */ - { 0x0000, 0x0000 }, /* R961 */ - { 0x0000, 0x0000 }, /* R962 */ - { 0x0000, 0x0000 }, /* R963 */ - { 0x0000, 0x0000 }, /* R964 */ - { 0x0000, 0x0000 }, /* R965 */ - { 0x0000, 0x0000 }, /* R966 */ - { 0x0000, 0x0000 }, /* R967 */ - { 0x0000, 0x0000 }, /* R968 */ - { 0x0000, 0x0000 }, /* R969 */ - { 0x0000, 0x0000 }, /* R970 */ - { 0x0000, 0x0000 }, /* R971 */ - { 0x0000, 0x0000 }, /* R972 */ - { 0x0000, 0x0000 }, /* R973 */ - { 0x0000, 0x0000 }, /* R974 */ - { 0x0000, 0x0000 }, /* R975 */ - { 0x0000, 0x0000 }, /* R976 */ - { 0x0000, 0x0000 }, /* R977 */ - { 0x0000, 0x0000 }, /* R978 */ - { 0x0000, 0x0000 }, /* R979 */ - { 0x0000, 0x0000 }, /* R980 */ - { 0x0000, 0x0000 }, /* R981 */ - { 0x0000, 0x0000 }, /* R982 */ - { 0x0000, 0x0000 }, /* R983 */ - { 0x0000, 0x0000 }, /* R984 */ - { 0x0000, 0x0000 }, /* R985 */ - { 0x0000, 0x0000 }, /* R986 */ - { 0x0000, 0x0000 }, /* R987 */ - { 0x0000, 0x0000 }, /* R988 */ - { 0x0000, 0x0000 }, /* R989 */ - { 0x0000, 0x0000 }, /* R990 */ - { 0x0000, 0x0000 }, /* R991 */ - { 0x0000, 0x0000 }, /* R992 */ - { 0x0000, 0x0000 }, /* R993 */ - { 0x0000, 0x0000 }, /* R994 */ - { 0x0000, 0x0000 }, /* R995 */ - { 0x0000, 0x0000 }, /* R996 */ - { 0x0000, 0x0000 }, /* R997 */ - { 0x0000, 0x0000 }, /* R998 */ - { 0x0000, 0x0000 }, /* R999 */ - { 0x0000, 0x0000 }, /* R1000 */ - { 0x0000, 0x0000 }, /* R1001 */ - { 0x0000, 0x0000 }, /* R1002 */ - { 0x0000, 0x0000 }, /* R1003 */ - { 0x0000, 0x0000 }, /* R1004 */ - { 0x0000, 0x0000 }, /* R1005 */ - { 0x0000, 0x0000 }, /* R1006 */ - { 0x0000, 0x0000 }, /* R1007 */ - { 0x0000, 0x0000 }, /* R1008 */ - { 0x0000, 0x0000 }, /* R1009 */ - { 0x0000, 0x0000 }, /* R1010 */ - { 0x0000, 0x0000 }, /* R1011 */ - { 0x0000, 0x0000 }, /* R1012 */ - { 0x0000, 0x0000 }, /* R1013 */ - { 0x0000, 0x0000 }, /* R1014 */ - { 0x0000, 0x0000 }, /* R1015 */ - { 0x0000, 0x0000 }, /* R1016 */ - { 0x0000, 0x0000 }, /* R1017 */ - { 0x0000, 0x0000 }, /* R1018 */ - { 0x0000, 0x0000 }, /* R1019 */ - { 0x0000, 0x0000 }, /* R1020 */ - { 0x0000, 0x0000 }, /* R1021 */ - { 0x0000, 0x0000 }, /* R1022 */ - { 0x0000, 0x0000 }, /* R1023 */ - { 0x00FF, 0x01FF }, /* R1024 - AIF1 ADC1 Left Volume */ - { 0x00FF, 0x01FF }, /* R1025 - AIF1 ADC1 Right Volume */ - { 0x00FF, 0x01FF }, /* R1026 - AIF1 DAC1 Left Volume */ - { 0x00FF, 0x01FF }, /* R1027 - AIF1 DAC1 Right Volume */ - { 0x00FF, 0x01FF }, /* R1028 - AIF1 ADC2 Left Volume */ - { 0x00FF, 0x01FF }, /* R1029 - AIF1 ADC2 Right Volume */ - { 0x00FF, 0x01FF }, /* R1030 - AIF1 DAC2 Left Volume */ - { 0x00FF, 0x01FF }, /* R1031 - AIF1 DAC2 Right Volume */ - { 0x0000, 0x0000 }, /* R1032 */ - { 0x0000, 0x0000 }, /* R1033 */ - { 0x0000, 0x0000 }, /* R1034 */ - { 0x0000, 0x0000 }, /* R1035 */ - { 0x0000, 0x0000 }, /* R1036 */ - { 0x0000, 0x0000 }, /* R1037 */ - { 0x0000, 0x0000 }, /* R1038 */ - { 0x0000, 0x0000 }, /* R1039 */ - { 0xF800, 0xF800 }, /* R1040 - AIF1 ADC1 Filters */ - { 0x7800, 0x7800 }, /* R1041 - AIF1 ADC2 Filters */ - { 0x0000, 0x0000 }, /* R1042 */ - { 0x0000, 0x0000 }, /* R1043 */ - { 0x0000, 0x0000 }, /* R1044 */ - { 0x0000, 0x0000 }, /* R1045 */ - { 0x0000, 0x0000 }, /* R1046 */ - { 0x0000, 0x0000 }, /* R1047 */ - { 0x0000, 0x0000 }, /* R1048 */ - { 0x0000, 0x0000 }, /* R1049 */ - { 0x0000, 0x0000 }, /* R1050 */ - { 0x0000, 0x0000 }, /* R1051 */ - { 0x0000, 0x0000 }, /* R1052 */ - { 0x0000, 0x0000 }, /* R1053 */ - { 0x0000, 0x0000 }, /* R1054 */ - { 0x0000, 0x0000 }, /* R1055 */ - { 0x02B6, 0x02B6 }, /* R1056 - AIF1 DAC1 Filters (1) */ - { 0x3F00, 0x3F00 }, /* R1057 - AIF1 DAC1 Filters (2) */ - { 0x02B6, 0x02B6 }, /* R1058 - AIF1 DAC2 Filters (1) */ - { 0x3F00, 0x3F00 }, /* R1059 - AIF1 DAC2 Filters (2) */ - { 0x0000, 0x0000 }, /* R1060 */ - { 0x0000, 0x0000 }, /* R1061 */ - { 0x0000, 0x0000 }, /* R1062 */ - { 0x0000, 0x0000 }, /* R1063 */ - { 0x0000, 0x0000 }, /* R1064 */ - { 0x0000, 0x0000 }, /* R1065 */ - { 0x0000, 0x0000 }, /* R1066 */ - { 0x0000, 0x0000 }, /* R1067 */ - { 0x0000, 0x0000 }, /* R1068 */ - { 0x0000, 0x0000 }, /* R1069 */ - { 0x0000, 0x0000 }, /* R1070 */ - { 0x0000, 0x0000 }, /* R1071 */ - { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ - { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ - { 0x0000, 0x0000 }, /* R1074 */ - { 0x0000, 0x0000 }, /* R1075 */ - { 0x0000, 0x0000 }, /* R1076 */ - { 0x0000, 0x0000 }, /* R1077 */ - { 0x0000, 0x0000 }, /* R1078 */ - { 0x0000, 0x0000 }, /* R1079 */ - { 0x0000, 0x0000 }, /* R1080 */ - { 0x0000, 0x0000 }, /* R1081 */ - { 0x0000, 0x0000 }, /* R1082 */ - { 0x0000, 0x0000 }, /* R1083 */ - { 0x0000, 0x0000 }, /* R1084 */ - { 0x0000, 0x0000 }, /* R1085 */ - { 0x0000, 0x0000 }, /* R1086 */ - { 0x0000, 0x0000 }, /* R1087 */ - { 0xFFFF, 0xFFFF }, /* R1088 - AIF1 DRC1 (1) */ - { 0x1FFF, 0x1FFF }, /* R1089 - AIF1 DRC1 (2) */ - { 0xFFFF, 0xFFFF }, /* R1090 - AIF1 DRC1 (3) */ - { 0x07FF, 0x07FF }, /* R1091 - AIF1 DRC1 (4) */ - { 0x03FF, 0x03FF }, /* R1092 - AIF1 DRC1 (5) */ - { 0x0000, 0x0000 }, /* R1093 */ - { 0x0000, 0x0000 }, /* R1094 */ - { 0x0000, 0x0000 }, /* R1095 */ - { 0x0000, 0x0000 }, /* R1096 */ - { 0x0000, 0x0000 }, /* R1097 */ - { 0x0000, 0x0000 }, /* R1098 */ - { 0x0000, 0x0000 }, /* R1099 */ - { 0x0000, 0x0000 }, /* R1100 */ - { 0x0000, 0x0000 }, /* R1101 */ - { 0x0000, 0x0000 }, /* R1102 */ - { 0x0000, 0x0000 }, /* R1103 */ - { 0xFFFF, 0xFFFF }, /* R1104 - AIF1 DRC2 (1) */ - { 0x1FFF, 0x1FFF }, /* R1105 - AIF1 DRC2 (2) */ - { 0xFFFF, 0xFFFF }, /* R1106 - AIF1 DRC2 (3) */ - { 0x07FF, 0x07FF }, /* R1107 - AIF1 DRC2 (4) */ - { 0x03FF, 0x03FF }, /* R1108 - AIF1 DRC2 (5) */ - { 0x0000, 0x0000 }, /* R1109 */ - { 0x0000, 0x0000 }, /* R1110 */ - { 0x0000, 0x0000 }, /* R1111 */ - { 0x0000, 0x0000 }, /* R1112 */ - { 0x0000, 0x0000 }, /* R1113 */ - { 0x0000, 0x0000 }, /* R1114 */ - { 0x0000, 0x0000 }, /* R1115 */ - { 0x0000, 0x0000 }, /* R1116 */ - { 0x0000, 0x0000 }, /* R1117 */ - { 0x0000, 0x0000 }, /* R1118 */ - { 0x0000, 0x0000 }, /* R1119 */ - { 0x0000, 0x0000 }, /* R1120 */ - { 0x0000, 0x0000 }, /* R1121 */ - { 0x0000, 0x0000 }, /* R1122 */ - { 0x0000, 0x0000 }, /* R1123 */ - { 0x0000, 0x0000 }, /* R1124 */ - { 0x0000, 0x0000 }, /* R1125 */ - { 0x0000, 0x0000 }, /* R1126 */ - { 0x0000, 0x0000 }, /* R1127 */ - { 0x0000, 0x0000 }, /* R1128 */ - { 0x0000, 0x0000 }, /* R1129 */ - { 0x0000, 0x0000 }, /* R1130 */ - { 0x0000, 0x0000 }, /* R1131 */ - { 0x0000, 0x0000 }, /* R1132 */ - { 0x0000, 0x0000 }, /* R1133 */ - { 0x0000, 0x0000 }, /* R1134 */ - { 0x0000, 0x0000 }, /* R1135 */ - { 0x0000, 0x0000 }, /* R1136 */ - { 0x0000, 0x0000 }, /* R1137 */ - { 0x0000, 0x0000 }, /* R1138 */ - { 0x0000, 0x0000 }, /* R1139 */ - { 0x0000, 0x0000 }, /* R1140 */ - { 0x0000, 0x0000 }, /* R1141 */ - { 0x0000, 0x0000 }, /* R1142 */ - { 0x0000, 0x0000 }, /* R1143 */ - { 0x0000, 0x0000 }, /* R1144 */ - { 0x0000, 0x0000 }, /* R1145 */ - { 0x0000, 0x0000 }, /* R1146 */ - { 0x0000, 0x0000 }, /* R1147 */ - { 0x0000, 0x0000 }, /* R1148 */ - { 0x0000, 0x0000 }, /* R1149 */ - { 0x0000, 0x0000 }, /* R1150 */ - { 0x0000, 0x0000 }, /* R1151 */ - { 0xFFFF, 0xFFFF }, /* R1152 - AIF1 DAC1 EQ Gains (1) */ - { 0xFFC0, 0xFFC0 }, /* R1153 - AIF1 DAC1 EQ Gains (2) */ - { 0xFFFF, 0xFFFF }, /* R1154 - AIF1 DAC1 EQ Band 1 A */ - { 0xFFFF, 0xFFFF }, /* R1155 - AIF1 DAC1 EQ Band 1 B */ - { 0xFFFF, 0xFFFF }, /* R1156 - AIF1 DAC1 EQ Band 1 PG */ - { 0xFFFF, 0xFFFF }, /* R1157 - AIF1 DAC1 EQ Band 2 A */ - { 0xFFFF, 0xFFFF }, /* R1158 - AIF1 DAC1 EQ Band 2 B */ - { 0xFFFF, 0xFFFF }, /* R1159 - AIF1 DAC1 EQ Band 2 C */ - { 0xFFFF, 0xFFFF }, /* R1160 - AIF1 DAC1 EQ Band 2 PG */ - { 0xFFFF, 0xFFFF }, /* R1161 - AIF1 DAC1 EQ Band 3 A */ - { 0xFFFF, 0xFFFF }, /* R1162 - AIF1 DAC1 EQ Band 3 B */ - { 0xFFFF, 0xFFFF }, /* R1163 - AIF1 DAC1 EQ Band 3 C */ - { 0xFFFF, 0xFFFF }, /* R1164 - AIF1 DAC1 EQ Band 3 PG */ - { 0xFFFF, 0xFFFF }, /* R1165 - AIF1 DAC1 EQ Band 4 A */ - { 0xFFFF, 0xFFFF }, /* R1166 - AIF1 DAC1 EQ Band 4 B */ - { 0xFFFF, 0xFFFF }, /* R1167 - AIF1 DAC1 EQ Band 4 C */ - { 0xFFFF, 0xFFFF }, /* R1168 - AIF1 DAC1 EQ Band 4 PG */ - { 0xFFFF, 0xFFFF }, /* R1169 - AIF1 DAC1 EQ Band 5 A */ - { 0xFFFF, 0xFFFF }, /* R1170 - AIF1 DAC1 EQ Band 5 B */ - { 0xFFFF, 0xFFFF }, /* R1171 - AIF1 DAC1 EQ Band 5 PG */ - { 0x0000, 0x0000 }, /* R1172 */ - { 0x0000, 0x0000 }, /* R1173 */ - { 0x0000, 0x0000 }, /* R1174 */ - { 0x0000, 0x0000 }, /* R1175 */ - { 0x0000, 0x0000 }, /* R1176 */ - { 0x0000, 0x0000 }, /* R1177 */ - { 0x0000, 0x0000 }, /* R1178 */ - { 0x0000, 0x0000 }, /* R1179 */ - { 0x0000, 0x0000 }, /* R1180 */ - { 0x0000, 0x0000 }, /* R1181 */ - { 0x0000, 0x0000 }, /* R1182 */ - { 0x0000, 0x0000 }, /* R1183 */ - { 0xFFFF, 0xFFFF }, /* R1184 - AIF1 DAC2 EQ Gains (1) */ - { 0xFFC0, 0xFFC0 }, /* R1185 - AIF1 DAC2 EQ Gains (2) */ - { 0xFFFF, 0xFFFF }, /* R1186 - AIF1 DAC2 EQ Band 1 A */ - { 0xFFFF, 0xFFFF }, /* R1187 - AIF1 DAC2 EQ Band 1 B */ - { 0xFFFF, 0xFFFF }, /* R1188 - AIF1 DAC2 EQ Band 1 PG */ - { 0xFFFF, 0xFFFF }, /* R1189 - AIF1 DAC2 EQ Band 2 A */ - { 0xFFFF, 0xFFFF }, /* R1190 - AIF1 DAC2 EQ Band 2 B */ - { 0xFFFF, 0xFFFF }, /* R1191 - AIF1 DAC2 EQ Band 2 C */ - { 0xFFFF, 0xFFFF }, /* R1192 - AIF1 DAC2 EQ Band 2 PG */ - { 0xFFFF, 0xFFFF }, /* R1193 - AIF1 DAC2 EQ Band 3 A */ - { 0xFFFF, 0xFFFF }, /* R1194 - AIF1 DAC2 EQ Band 3 B */ - { 0xFFFF, 0xFFFF }, /* R1195 - AIF1 DAC2 EQ Band 3 C */ - { 0xFFFF, 0xFFFF }, /* R1196 - AIF1 DAC2 EQ Band 3 PG */ - { 0xFFFF, 0xFFFF }, /* R1197 - AIF1 DAC2 EQ Band 4 A */ - { 0xFFFF, 0xFFFF }, /* R1198 - AIF1 DAC2 EQ Band 4 B */ - { 0xFFFF, 0xFFFF }, /* R1199 - AIF1 DAC2 EQ Band 4 C */ - { 0xFFFF, 0xFFFF }, /* R1200 - AIF1 DAC2 EQ Band 4 PG */ - { 0xFFFF, 0xFFFF }, /* R1201 - AIF1 DAC2 EQ Band 5 A */ - { 0xFFFF, 0xFFFF }, /* R1202 - AIF1 DAC2 EQ Band 5 B */ - { 0xFFFF, 0xFFFF }, /* R1203 - AIF1 DAC2 EQ Band 5 PG */ - { 0x0000, 0x0000 }, /* R1204 */ - { 0x0000, 0x0000 }, /* R1205 */ - { 0x0000, 0x0000 }, /* R1206 */ - { 0x0000, 0x0000 }, /* R1207 */ - { 0x0000, 0x0000 }, /* R1208 */ - { 0x0000, 0x0000 }, /* R1209 */ - { 0x0000, 0x0000 }, /* R1210 */ - { 0x0000, 0x0000 }, /* R1211 */ - { 0x0000, 0x0000 }, /* R1212 */ - { 0x0000, 0x0000 }, /* R1213 */ - { 0x0000, 0x0000 }, /* R1214 */ - { 0x0000, 0x0000 }, /* R1215 */ - { 0x0000, 0x0000 }, /* R1216 */ - { 0x0000, 0x0000 }, /* R1217 */ - { 0x0000, 0x0000 }, /* R1218 */ - { 0x0000, 0x0000 }, /* R1219 */ - { 0x0000, 0x0000 }, /* R1220 */ - { 0x0000, 0x0000 }, /* R1221 */ - { 0x0000, 0x0000 }, /* R1222 */ - { 0x0000, 0x0000 }, /* R1223 */ - { 0x0000, 0x0000 }, /* R1224 */ - { 0x0000, 0x0000 }, /* R1225 */ - { 0x0000, 0x0000 }, /* R1226 */ - { 0x0000, 0x0000 }, /* R1227 */ - { 0x0000, 0x0000 }, /* R1228 */ - { 0x0000, 0x0000 }, /* R1229 */ - { 0x0000, 0x0000 }, /* R1230 */ - { 0x0000, 0x0000 }, /* R1231 */ - { 0x0000, 0x0000 }, /* R1232 */ - { 0x0000, 0x0000 }, /* R1233 */ - { 0x0000, 0x0000 }, /* R1234 */ - { 0x0000, 0x0000 }, /* R1235 */ - { 0x0000, 0x0000 }, /* R1236 */ - { 0x0000, 0x0000 }, /* R1237 */ - { 0x0000, 0x0000 }, /* R1238 */ - { 0x0000, 0x0000 }, /* R1239 */ - { 0x0000, 0x0000 }, /* R1240 */ - { 0x0000, 0x0000 }, /* R1241 */ - { 0x0000, 0x0000 }, /* R1242 */ - { 0x0000, 0x0000 }, /* R1243 */ - { 0x0000, 0x0000 }, /* R1244 */ - { 0x0000, 0x0000 }, /* R1245 */ - { 0x0000, 0x0000 }, /* R1246 */ - { 0x0000, 0x0000 }, /* R1247 */ - { 0x0000, 0x0000 }, /* R1248 */ - { 0x0000, 0x0000 }, /* R1249 */ - { 0x0000, 0x0000 }, /* R1250 */ - { 0x0000, 0x0000 }, /* R1251 */ - { 0x0000, 0x0000 }, /* R1252 */ - { 0x0000, 0x0000 }, /* R1253 */ - { 0x0000, 0x0000 }, /* R1254 */ - { 0x0000, 0x0000 }, /* R1255 */ - { 0x0000, 0x0000 }, /* R1256 */ - { 0x0000, 0x0000 }, /* R1257 */ - { 0x0000, 0x0000 }, /* R1258 */ - { 0x0000, 0x0000 }, /* R1259 */ - { 0x0000, 0x0000 }, /* R1260 */ - { 0x0000, 0x0000 }, /* R1261 */ - { 0x0000, 0x0000 }, /* R1262 */ - { 0x0000, 0x0000 }, /* R1263 */ - { 0x0000, 0x0000 }, /* R1264 */ - { 0x0000, 0x0000 }, /* R1265 */ - { 0x0000, 0x0000 }, /* R1266 */ - { 0x0000, 0x0000 }, /* R1267 */ - { 0x0000, 0x0000 }, /* R1268 */ - { 0x0000, 0x0000 }, /* R1269 */ - { 0x0000, 0x0000 }, /* R1270 */ - { 0x0000, 0x0000 }, /* R1271 */ - { 0x0000, 0x0000 }, /* R1272 */ - { 0x0000, 0x0000 }, /* R1273 */ - { 0x0000, 0x0000 }, /* R1274 */ - { 0x0000, 0x0000 }, /* R1275 */ - { 0x0000, 0x0000 }, /* R1276 */ - { 0x0000, 0x0000 }, /* R1277 */ - { 0x0000, 0x0000 }, /* R1278 */ - { 0x0000, 0x0000 }, /* R1279 */ - { 0x00FF, 0x01FF }, /* R1280 - AIF2 ADC Left Volume */ - { 0x00FF, 0x01FF }, /* R1281 - AIF2 ADC Right Volume */ - { 0x00FF, 0x01FF }, /* R1282 - AIF2 DAC Left Volume */ - { 0x00FF, 0x01FF }, /* R1283 - AIF2 DAC Right Volume */ - { 0x0000, 0x0000 }, /* R1284 */ - { 0x0000, 0x0000 }, /* R1285 */ - { 0x0000, 0x0000 }, /* R1286 */ - { 0x0000, 0x0000 }, /* R1287 */ - { 0x0000, 0x0000 }, /* R1288 */ - { 0x0000, 0x0000 }, /* R1289 */ - { 0x0000, 0x0000 }, /* R1290 */ - { 0x0000, 0x0000 }, /* R1291 */ - { 0x0000, 0x0000 }, /* R1292 */ - { 0x0000, 0x0000 }, /* R1293 */ - { 0x0000, 0x0000 }, /* R1294 */ - { 0x0000, 0x0000 }, /* R1295 */ - { 0xF800, 0xF800 }, /* R1296 - AIF2 ADC Filters */ - { 0x0000, 0x0000 }, /* R1297 */ - { 0x0000, 0x0000 }, /* R1298 */ - { 0x0000, 0x0000 }, /* R1299 */ - { 0x0000, 0x0000 }, /* R1300 */ - { 0x0000, 0x0000 }, /* R1301 */ - { 0x0000, 0x0000 }, /* R1302 */ - { 0x0000, 0x0000 }, /* R1303 */ - { 0x0000, 0x0000 }, /* R1304 */ - { 0x0000, 0x0000 }, /* R1305 */ - { 0x0000, 0x0000 }, /* R1306 */ - { 0x0000, 0x0000 }, /* R1307 */ - { 0x0000, 0x0000 }, /* R1308 */ - { 0x0000, 0x0000 }, /* R1309 */ - { 0x0000, 0x0000 }, /* R1310 */ - { 0x0000, 0x0000 }, /* R1311 */ - { 0x02B6, 0x02B6 }, /* R1312 - AIF2 DAC Filters (1) */ - { 0x3F00, 0x3F00 }, /* R1313 - AIF2 DAC Filters (2) */ - { 0x0000, 0x0000 }, /* R1314 */ - { 0x0000, 0x0000 }, /* R1315 */ - { 0x0000, 0x0000 }, /* R1316 */ - { 0x0000, 0x0000 }, /* R1317 */ - { 0x0000, 0x0000 }, /* R1318 */ - { 0x0000, 0x0000 }, /* R1319 */ - { 0x0000, 0x0000 }, /* R1320 */ - { 0x0000, 0x0000 }, /* R1321 */ - { 0x0000, 0x0000 }, /* R1322 */ - { 0x0000, 0x0000 }, /* R1323 */ - { 0x0000, 0x0000 }, /* R1324 */ - { 0x0000, 0x0000 }, /* R1325 */ - { 0x0000, 0x0000 }, /* R1326 */ - { 0x0000, 0x0000 }, /* R1327 */ - { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ - { 0x0000, 0x0000 }, /* R1329 */ - { 0x0000, 0x0000 }, /* R1330 */ - { 0x0000, 0x0000 }, /* R1331 */ - { 0x0000, 0x0000 }, /* R1332 */ - { 0x0000, 0x0000 }, /* R1333 */ - { 0x0000, 0x0000 }, /* R1334 */ - { 0x0000, 0x0000 }, /* R1335 */ - { 0x0000, 0x0000 }, /* R1336 */ - { 0x0000, 0x0000 }, /* R1337 */ - { 0x0000, 0x0000 }, /* R1338 */ - { 0x0000, 0x0000 }, /* R1339 */ - { 0x0000, 0x0000 }, /* R1340 */ - { 0x0000, 0x0000 }, /* R1341 */ - { 0x0000, 0x0000 }, /* R1342 */ - { 0x0000, 0x0000 }, /* R1343 */ - { 0xFFFF, 0xFFFF }, /* R1344 - AIF2 DRC (1) */ - { 0x1FFF, 0x1FFF }, /* R1345 - AIF2 DRC (2) */ - { 0xFFFF, 0xFFFF }, /* R1346 - AIF2 DRC (3) */ - { 0x07FF, 0x07FF }, /* R1347 - AIF2 DRC (4) */ - { 0x03FF, 0x03FF }, /* R1348 - AIF2 DRC (5) */ - { 0x0000, 0x0000 }, /* R1349 */ - { 0x0000, 0x0000 }, /* R1350 */ - { 0x0000, 0x0000 }, /* R1351 */ - { 0x0000, 0x0000 }, /* R1352 */ - { 0x0000, 0x0000 }, /* R1353 */ - { 0x0000, 0x0000 }, /* R1354 */ - { 0x0000, 0x0000 }, /* R1355 */ - { 0x0000, 0x0000 }, /* R1356 */ - { 0x0000, 0x0000 }, /* R1357 */ - { 0x0000, 0x0000 }, /* R1358 */ - { 0x0000, 0x0000 }, /* R1359 */ - { 0x0000, 0x0000 }, /* R1360 */ - { 0x0000, 0x0000 }, /* R1361 */ - { 0x0000, 0x0000 }, /* R1362 */ - { 0x0000, 0x0000 }, /* R1363 */ - { 0x0000, 0x0000 }, /* R1364 */ - { 0x0000, 0x0000 }, /* R1365 */ - { 0x0000, 0x0000 }, /* R1366 */ - { 0x0000, 0x0000 }, /* R1367 */ - { 0x0000, 0x0000 }, /* R1368 */ - { 0x0000, 0x0000 }, /* R1369 */ - { 0x0000, 0x0000 }, /* R1370 */ - { 0x0000, 0x0000 }, /* R1371 */ - { 0x0000, 0x0000 }, /* R1372 */ - { 0x0000, 0x0000 }, /* R1373 */ - { 0x0000, 0x0000 }, /* R1374 */ - { 0x0000, 0x0000 }, /* R1375 */ - { 0x0000, 0x0000 }, /* R1376 */ - { 0x0000, 0x0000 }, /* R1377 */ - { 0x0000, 0x0000 }, /* R1378 */ - { 0x0000, 0x0000 }, /* R1379 */ - { 0x0000, 0x0000 }, /* R1380 */ - { 0x0000, 0x0000 }, /* R1381 */ - { 0x0000, 0x0000 }, /* R1382 */ - { 0x0000, 0x0000 }, /* R1383 */ - { 0x0000, 0x0000 }, /* R1384 */ - { 0x0000, 0x0000 }, /* R1385 */ - { 0x0000, 0x0000 }, /* R1386 */ - { 0x0000, 0x0000 }, /* R1387 */ - { 0x0000, 0x0000 }, /* R1388 */ - { 0x0000, 0x0000 }, /* R1389 */ - { 0x0000, 0x0000 }, /* R1390 */ - { 0x0000, 0x0000 }, /* R1391 */ - { 0x0000, 0x0000 }, /* R1392 */ - { 0x0000, 0x0000 }, /* R1393 */ - { 0x0000, 0x0000 }, /* R1394 */ - { 0x0000, 0x0000 }, /* R1395 */ - { 0x0000, 0x0000 }, /* R1396 */ - { 0x0000, 0x0000 }, /* R1397 */ - { 0x0000, 0x0000 }, /* R1398 */ - { 0x0000, 0x0000 }, /* R1399 */ - { 0x0000, 0x0000 }, /* R1400 */ - { 0x0000, 0x0000 }, /* R1401 */ - { 0x0000, 0x0000 }, /* R1402 */ - { 0x0000, 0x0000 }, /* R1403 */ - { 0x0000, 0x0000 }, /* R1404 */ - { 0x0000, 0x0000 }, /* R1405 */ - { 0x0000, 0x0000 }, /* R1406 */ - { 0x0000, 0x0000 }, /* R1407 */ - { 0xFFFF, 0xFFFF }, /* R1408 - AIF2 EQ Gains (1) */ - { 0xFFC0, 0xFFC0 }, /* R1409 - AIF2 EQ Gains (2) */ - { 0xFFFF, 0xFFFF }, /* R1410 - AIF2 EQ Band 1 A */ - { 0xFFFF, 0xFFFF }, /* R1411 - AIF2 EQ Band 1 B */ - { 0xFFFF, 0xFFFF }, /* R1412 - AIF2 EQ Band 1 PG */ - { 0xFFFF, 0xFFFF }, /* R1413 - AIF2 EQ Band 2 A */ - { 0xFFFF, 0xFFFF }, /* R1414 - AIF2 EQ Band 2 B */ - { 0xFFFF, 0xFFFF }, /* R1415 - AIF2 EQ Band 2 C */ - { 0xFFFF, 0xFFFF }, /* R1416 - AIF2 EQ Band 2 PG */ - { 0xFFFF, 0xFFFF }, /* R1417 - AIF2 EQ Band 3 A */ - { 0xFFFF, 0xFFFF }, /* R1418 - AIF2 EQ Band 3 B */ - { 0xFFFF, 0xFFFF }, /* R1419 - AIF2 EQ Band 3 C */ - { 0xFFFF, 0xFFFF }, /* R1420 - AIF2 EQ Band 3 PG */ - { 0xFFFF, 0xFFFF }, /* R1421 - AIF2 EQ Band 4 A */ - { 0xFFFF, 0xFFFF }, /* R1422 - AIF2 EQ Band 4 B */ - { 0xFFFF, 0xFFFF }, /* R1423 - AIF2 EQ Band 4 C */ - { 0xFFFF, 0xFFFF }, /* R1424 - AIF2 EQ Band 4 PG */ - { 0xFFFF, 0xFFFF }, /* R1425 - AIF2 EQ Band 5 A */ - { 0xFFFF, 0xFFFF }, /* R1426 - AIF2 EQ Band 5 B */ - { 0xFFFF, 0xFFFF }, /* R1427 - AIF2 EQ Band 5 PG */ - { 0x0000, 0x0000 }, /* R1428 */ - { 0x0000, 0x0000 }, /* R1429 */ - { 0x0000, 0x0000 }, /* R1430 */ - { 0x0000, 0x0000 }, /* R1431 */ - { 0x0000, 0x0000 }, /* R1432 */ - { 0x0000, 0x0000 }, /* R1433 */ - { 0x0000, 0x0000 }, /* R1434 */ - { 0x0000, 0x0000 }, /* R1435 */ - { 0x0000, 0x0000 }, /* R1436 */ - { 0x0000, 0x0000 }, /* R1437 */ - { 0x0000, 0x0000 }, /* R1438 */ - { 0x0000, 0x0000 }, /* R1439 */ - { 0x0000, 0x0000 }, /* R1440 */ - { 0x0000, 0x0000 }, /* R1441 */ - { 0x0000, 0x0000 }, /* R1442 */ - { 0x0000, 0x0000 }, /* R1443 */ - { 0x0000, 0x0000 }, /* R1444 */ - { 0x0000, 0x0000 }, /* R1445 */ - { 0x0000, 0x0000 }, /* R1446 */ - { 0x0000, 0x0000 }, /* R1447 */ - { 0x0000, 0x0000 }, /* R1448 */ - { 0x0000, 0x0000 }, /* R1449 */ - { 0x0000, 0x0000 }, /* R1450 */ - { 0x0000, 0x0000 }, /* R1451 */ - { 0x0000, 0x0000 }, /* R1452 */ - { 0x0000, 0x0000 }, /* R1453 */ - { 0x0000, 0x0000 }, /* R1454 */ - { 0x0000, 0x0000 }, /* R1455 */ - { 0x0000, 0x0000 }, /* R1456 */ - { 0x0000, 0x0000 }, /* R1457 */ - { 0x0000, 0x0000 }, /* R1458 */ - { 0x0000, 0x0000 }, /* R1459 */ - { 0x0000, 0x0000 }, /* R1460 */ - { 0x0000, 0x0000 }, /* R1461 */ - { 0x0000, 0x0000 }, /* R1462 */ - { 0x0000, 0x0000 }, /* R1463 */ - { 0x0000, 0x0000 }, /* R1464 */ - { 0x0000, 0x0000 }, /* R1465 */ - { 0x0000, 0x0000 }, /* R1466 */ - { 0x0000, 0x0000 }, /* R1467 */ - { 0x0000, 0x0000 }, /* R1468 */ - { 0x0000, 0x0000 }, /* R1469 */ - { 0x0000, 0x0000 }, /* R1470 */ - { 0x0000, 0x0000 }, /* R1471 */ - { 0x0000, 0x0000 }, /* R1472 */ - { 0x0000, 0x0000 }, /* R1473 */ - { 0x0000, 0x0000 }, /* R1474 */ - { 0x0000, 0x0000 }, /* R1475 */ - { 0x0000, 0x0000 }, /* R1476 */ - { 0x0000, 0x0000 }, /* R1477 */ - { 0x0000, 0x0000 }, /* R1478 */ - { 0x0000, 0x0000 }, /* R1479 */ - { 0x0000, 0x0000 }, /* R1480 */ - { 0x0000, 0x0000 }, /* R1481 */ - { 0x0000, 0x0000 }, /* R1482 */ - { 0x0000, 0x0000 }, /* R1483 */ - { 0x0000, 0x0000 }, /* R1484 */ - { 0x0000, 0x0000 }, /* R1485 */ - { 0x0000, 0x0000 }, /* R1486 */ - { 0x0000, 0x0000 }, /* R1487 */ - { 0x0000, 0x0000 }, /* R1488 */ - { 0x0000, 0x0000 }, /* R1489 */ - { 0x0000, 0x0000 }, /* R1490 */ - { 0x0000, 0x0000 }, /* R1491 */ - { 0x0000, 0x0000 }, /* R1492 */ - { 0x0000, 0x0000 }, /* R1493 */ - { 0x0000, 0x0000 }, /* R1494 */ - { 0x0000, 0x0000 }, /* R1495 */ - { 0x0000, 0x0000 }, /* R1496 */ - { 0x0000, 0x0000 }, /* R1497 */ - { 0x0000, 0x0000 }, /* R1498 */ - { 0x0000, 0x0000 }, /* R1499 */ - { 0x0000, 0x0000 }, /* R1500 */ - { 0x0000, 0x0000 }, /* R1501 */ - { 0x0000, 0x0000 }, /* R1502 */ - { 0x0000, 0x0000 }, /* R1503 */ - { 0x0000, 0x0000 }, /* R1504 */ - { 0x0000, 0x0000 }, /* R1505 */ - { 0x0000, 0x0000 }, /* R1506 */ - { 0x0000, 0x0000 }, /* R1507 */ - { 0x0000, 0x0000 }, /* R1508 */ - { 0x0000, 0x0000 }, /* R1509 */ - { 0x0000, 0x0000 }, /* R1510 */ - { 0x0000, 0x0000 }, /* R1511 */ - { 0x0000, 0x0000 }, /* R1512 */ - { 0x0000, 0x0000 }, /* R1513 */ - { 0x0000, 0x0000 }, /* R1514 */ - { 0x0000, 0x0000 }, /* R1515 */ - { 0x0000, 0x0000 }, /* R1516 */ - { 0x0000, 0x0000 }, /* R1517 */ - { 0x0000, 0x0000 }, /* R1518 */ - { 0x0000, 0x0000 }, /* R1519 */ - { 0x0000, 0x0000 }, /* R1520 */ - { 0x0000, 0x0000 }, /* R1521 */ - { 0x0000, 0x0000 }, /* R1522 */ - { 0x0000, 0x0000 }, /* R1523 */ - { 0x0000, 0x0000 }, /* R1524 */ - { 0x0000, 0x0000 }, /* R1525 */ - { 0x0000, 0x0000 }, /* R1526 */ - { 0x0000, 0x0000 }, /* R1527 */ - { 0x0000, 0x0000 }, /* R1528 */ - { 0x0000, 0x0000 }, /* R1529 */ - { 0x0000, 0x0000 }, /* R1530 */ - { 0x0000, 0x0000 }, /* R1531 */ - { 0x0000, 0x0000 }, /* R1532 */ - { 0x0000, 0x0000 }, /* R1533 */ - { 0x0000, 0x0000 }, /* R1534 */ - { 0x0000, 0x0000 }, /* R1535 */ - { 0x01EF, 0x01EF }, /* R1536 - DAC1 Mixer Volumes */ - { 0x0037, 0x0037 }, /* R1537 - DAC1 Left Mixer Routing */ - { 0x0037, 0x0037 }, /* R1538 - DAC1 Right Mixer Routing */ - { 0x01EF, 0x01EF }, /* R1539 - DAC2 Mixer Volumes */ - { 0x0037, 0x0037 }, /* R1540 - DAC2 Left Mixer Routing */ - { 0x0037, 0x0037 }, /* R1541 - DAC2 Right Mixer Routing */ - { 0x0003, 0x0003 }, /* R1542 - AIF1 ADC1 Left Mixer Routing */ - { 0x0003, 0x0003 }, /* R1543 - AIF1 ADC1 Right Mixer Routing */ - { 0x0003, 0x0003 }, /* R1544 - AIF1 ADC2 Left Mixer Routing */ - { 0x0003, 0x0003 }, /* R1545 - AIF1 ADC2 Right mixer Routing */ - { 0x0000, 0x0000 }, /* R1546 */ - { 0x0000, 0x0000 }, /* R1547 */ - { 0x0000, 0x0000 }, /* R1548 */ - { 0x0000, 0x0000 }, /* R1549 */ - { 0x0000, 0x0000 }, /* R1550 */ - { 0x0000, 0x0000 }, /* R1551 */ - { 0x02FF, 0x03FF }, /* R1552 - DAC1 Left Volume */ - { 0x02FF, 0x03FF }, /* R1553 - DAC1 Right Volume */ - { 0x02FF, 0x03FF }, /* R1554 - DAC2 Left Volume */ - { 0x02FF, 0x03FF }, /* R1555 - DAC2 Right Volume */ - { 0x0003, 0x0003 }, /* R1556 - DAC Softmute */ - { 0x0000, 0x0000 }, /* R1557 */ - { 0x0000, 0x0000 }, /* R1558 */ - { 0x0000, 0x0000 }, /* R1559 */ - { 0x0000, 0x0000 }, /* R1560 */ - { 0x0000, 0x0000 }, /* R1561 */ - { 0x0000, 0x0000 }, /* R1562 */ - { 0x0000, 0x0000 }, /* R1563 */ - { 0x0000, 0x0000 }, /* R1564 */ - { 0x0000, 0x0000 }, /* R1565 */ - { 0x0000, 0x0000 }, /* R1566 */ - { 0x0000, 0x0000 }, /* R1567 */ - { 0x0003, 0x0003 }, /* R1568 - Oversampling */ - { 0x03C3, 0x03C3 }, /* R1569 - Sidetone */ -}; - -const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { - 0x8994, /* R0 - Software Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x0000, /* R4 - Power Management (4) */ - 0x0000, /* R5 - Power Management (5) */ - 0x0000, /* R6 - Power Management (6) */ - 0x0000, /* R7 */ - 0x0000, /* R8 */ - 0x0000, /* R9 */ - 0x0000, /* R10 */ - 0x0000, /* R11 */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0000, /* R14 */ - 0x0000, /* R15 */ - 0x0000, /* R16 */ - 0x0000, /* R17 */ - 0x0000, /* R18 */ - 0x0000, /* R19 */ - 0x0000, /* R20 */ - 0x0000, /* R21 - Input Mixer (1) */ - 0x0000, /* R22 */ - 0x0000, /* R23 */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x006D, /* R28 - Left Output Volume */ - 0x006D, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0020, /* R31 - HPOUT2 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - SPKMIXL Attenuation */ - 0x0003, /* R35 - SPKMIXR Attenuation */ - 0x0011, /* R36 - SPKOUT Mixers */ - 0x0140, /* R37 - ClassD */ - 0x0079, /* R38 - Speaker Volume Left */ - 0x0079, /* R39 - Speaker Volume Right */ - 0x0000, /* R40 - Input Mixer (2) */ - 0x0000, /* R41 - Input Mixer (3) */ - 0x0000, /* R42 - Input Mixer (4) */ - 0x0000, /* R43 - Input Mixer (5) */ - 0x0000, /* R44 - Input Mixer (6) */ - 0x0000, /* R45 - Output Mixer (1) */ - 0x0000, /* R46 - Output Mixer (2) */ - 0x0000, /* R47 - Output Mixer (3) */ - 0x0000, /* R48 - Output Mixer (4) */ - 0x0000, /* R49 - Output Mixer (5) */ - 0x0000, /* R50 - Output Mixer (6) */ - 0x0000, /* R51 - HPOUT2 Mixer */ - 0x0000, /* R52 - Line Mixer (1) */ - 0x0000, /* R53 - Line Mixer (2) */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP (1) */ - 0x0000, /* R57 - AntiPOP (2) */ - 0x0000, /* R58 - MICBIAS */ - 0x000D, /* R59 - LDO 1 */ - 0x0003, /* R60 - LDO 2 */ - 0x0039, /* R61 - MICBIAS1 */ - 0x0039, /* R62 - MICBIAS2 */ - 0x0000, /* R63 */ - 0x0000, /* R64 */ - 0x0000, /* R65 */ - 0x0000, /* R66 */ - 0x0000, /* R67 */ - 0x0000, /* R68 */ - 0x0000, /* R69 */ - 0x0000, /* R70 */ - 0x0000, /* R71 */ - 0x0000, /* R72 */ - 0x0000, /* R73 */ - 0x0000, /* R74 */ - 0x0000, /* R75 */ - 0x1F25, /* R76 - Charge Pump (1) */ - 0xAB19, /* R77 - Charge Pump (2) */ - 0x0000, /* R78 */ - 0x0000, /* R79 */ - 0x0000, /* R80 */ - 0x0004, /* R81 - Class W (1) */ - 0x0000, /* R82 */ - 0x0000, /* R83 */ - 0x0000, /* R84 - DC Servo (1) */ - 0x054A, /* R85 - DC Servo (2) */ - 0x0000, /* R86 */ - 0x0000, /* R87 - DC Servo (4) */ - 0x0000, /* R88 - DC Servo Readback */ - 0x0000, /* R89 */ - 0x0000, /* R90 */ - 0x0000, /* R91 */ - 0x0000, /* R92 */ - 0x0000, /* R93 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0000, /* R96 - Analogue HP (1) */ - 0x0000, /* R97 */ - 0x0000, /* R98 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0000, /* R104 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 */ - 0x0000, /* R109 */ - 0x0000, /* R110 */ - 0x0000, /* R111 */ - 0x0000, /* R112 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 */ - 0x0000, /* R117 */ - 0x0000, /* R118 */ - 0x0000, /* R119 */ - 0x0000, /* R120 */ - 0x0000, /* R121 */ - 0x0000, /* R122 */ - 0x0000, /* R123 */ - 0x0000, /* R124 */ - 0x0000, /* R125 */ - 0x0000, /* R126 */ - 0x0000, /* R127 */ - 0x0000, /* R128 */ - 0x0000, /* R129 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x0000, /* R135 */ - 0x0000, /* R136 */ - 0x0000, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x0000, /* R148 */ - 0x0000, /* R149 */ - 0x0000, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0000, /* R196 */ - 0x0000, /* R197 */ - 0x0000, /* R198 */ - 0x0000, /* R199 */ - 0x0000, /* R200 */ - 0x0000, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 */ - 0x0000, /* R205 */ - 0x0000, /* R206 */ - 0x0000, /* R207 */ - 0x0000, /* R208 */ - 0x0000, /* R209 */ - 0x0000, /* R210 */ - 0x0000, /* R211 */ - 0x0000, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0000, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0000, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0000, /* R226 */ - 0x0000, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0000, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0000, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0000, /* R245 */ - 0x0000, /* R246 */ - 0x0000, /* R247 */ - 0x0000, /* R248 */ - 0x0000, /* R249 */ - 0x0000, /* R250 */ - 0x0000, /* R251 */ - 0x0000, /* R252 */ - 0x0000, /* R253 */ - 0x0000, /* R254 */ - 0x0000, /* R255 */ - 0x0003, /* R256 - Chip Revision */ - 0x8004, /* R257 - Control Interface */ - 0x0000, /* R258 */ - 0x0000, /* R259 */ - 0x0000, /* R260 */ - 0x0000, /* R261 */ - 0x0000, /* R262 */ - 0x0000, /* R263 */ - 0x0000, /* R264 */ - 0x0000, /* R265 */ - 0x0000, /* R266 */ - 0x0000, /* R267 */ - 0x0000, /* R268 */ - 0x0000, /* R269 */ - 0x0000, /* R270 */ - 0x0000, /* R271 */ - 0x0000, /* R272 - Write Sequencer Ctrl (1) */ - 0x0000, /* R273 - Write Sequencer Ctrl (2) */ - 0x0000, /* R274 */ - 0x0000, /* R275 */ - 0x0000, /* R276 */ - 0x0000, /* R277 */ - 0x0000, /* R278 */ - 0x0000, /* R279 */ - 0x0000, /* R280 */ - 0x0000, /* R281 */ - 0x0000, /* R282 */ - 0x0000, /* R283 */ - 0x0000, /* R284 */ - 0x0000, /* R285 */ - 0x0000, /* R286 */ - 0x0000, /* R287 */ - 0x0000, /* R288 */ - 0x0000, /* R289 */ - 0x0000, /* R290 */ - 0x0000, /* R291 */ - 0x0000, /* R292 */ - 0x0000, /* R293 */ - 0x0000, /* R294 */ - 0x0000, /* R295 */ - 0x0000, /* R296 */ - 0x0000, /* R297 */ - 0x0000, /* R298 */ - 0x0000, /* R299 */ - 0x0000, /* R300 */ - 0x0000, /* R301 */ - 0x0000, /* R302 */ - 0x0000, /* R303 */ - 0x0000, /* R304 */ - 0x0000, /* R305 */ - 0x0000, /* R306 */ - 0x0000, /* R307 */ - 0x0000, /* R308 */ - 0x0000, /* R309 */ - 0x0000, /* R310 */ - 0x0000, /* R311 */ - 0x0000, /* R312 */ - 0x0000, /* R313 */ - 0x0000, /* R314 */ - 0x0000, /* R315 */ - 0x0000, /* R316 */ - 0x0000, /* R317 */ - 0x0000, /* R318 */ - 0x0000, /* R319 */ - 0x0000, /* R320 */ - 0x0000, /* R321 */ - 0x0000, /* R322 */ - 0x0000, /* R323 */ - 0x0000, /* R324 */ - 0x0000, /* R325 */ - 0x0000, /* R326 */ - 0x0000, /* R327 */ - 0x0000, /* R328 */ - 0x0000, /* R329 */ - 0x0000, /* R330 */ - 0x0000, /* R331 */ - 0x0000, /* R332 */ - 0x0000, /* R333 */ - 0x0000, /* R334 */ - 0x0000, /* R335 */ - 0x0000, /* R336 */ - 0x0000, /* R337 */ - 0x0000, /* R338 */ - 0x0000, /* R339 */ - 0x0000, /* R340 */ - 0x0000, /* R341 */ - 0x0000, /* R342 */ - 0x0000, /* R343 */ - 0x0000, /* R344 */ - 0x0000, /* R345 */ - 0x0000, /* R346 */ - 0x0000, /* R347 */ - 0x0000, /* R348 */ - 0x0000, /* R349 */ - 0x0000, /* R350 */ - 0x0000, /* R351 */ - 0x0000, /* R352 */ - 0x0000, /* R353 */ - 0x0000, /* R354 */ - 0x0000, /* R355 */ - 0x0000, /* R356 */ - 0x0000, /* R357 */ - 0x0000, /* R358 */ - 0x0000, /* R359 */ - 0x0000, /* R360 */ - 0x0000, /* R361 */ - 0x0000, /* R362 */ - 0x0000, /* R363 */ - 0x0000, /* R364 */ - 0x0000, /* R365 */ - 0x0000, /* R366 */ - 0x0000, /* R367 */ - 0x0000, /* R368 */ - 0x0000, /* R369 */ - 0x0000, /* R370 */ - 0x0000, /* R371 */ - 0x0000, /* R372 */ - 0x0000, /* R373 */ - 0x0000, /* R374 */ - 0x0000, /* R375 */ - 0x0000, /* R376 */ - 0x0000, /* R377 */ - 0x0000, /* R378 */ - 0x0000, /* R379 */ - 0x0000, /* R380 */ - 0x0000, /* R381 */ - 0x0000, /* R382 */ - 0x0000, /* R383 */ - 0x0000, /* R384 */ - 0x0000, /* R385 */ - 0x0000, /* R386 */ - 0x0000, /* R387 */ - 0x0000, /* R388 */ - 0x0000, /* R389 */ - 0x0000, /* R390 */ - 0x0000, /* R391 */ - 0x0000, /* R392 */ - 0x0000, /* R393 */ - 0x0000, /* R394 */ - 0x0000, /* R395 */ - 0x0000, /* R396 */ - 0x0000, /* R397 */ - 0x0000, /* R398 */ - 0x0000, /* R399 */ - 0x0000, /* R400 */ - 0x0000, /* R401 */ - 0x0000, /* R402 */ - 0x0000, /* R403 */ - 0x0000, /* R404 */ - 0x0000, /* R405 */ - 0x0000, /* R406 */ - 0x0000, /* R407 */ - 0x0000, /* R408 */ - 0x0000, /* R409 */ - 0x0000, /* R410 */ - 0x0000, /* R411 */ - 0x0000, /* R412 */ - 0x0000, /* R413 */ - 0x0000, /* R414 */ - 0x0000, /* R415 */ - 0x0000, /* R416 */ - 0x0000, /* R417 */ - 0x0000, /* R418 */ - 0x0000, /* R419 */ - 0x0000, /* R420 */ - 0x0000, /* R421 */ - 0x0000, /* R422 */ - 0x0000, /* R423 */ - 0x0000, /* R424 */ - 0x0000, /* R425 */ - 0x0000, /* R426 */ - 0x0000, /* R427 */ - 0x0000, /* R428 */ - 0x0000, /* R429 */ - 0x0000, /* R430 */ - 0x0000, /* R431 */ - 0x0000, /* R432 */ - 0x0000, /* R433 */ - 0x0000, /* R434 */ - 0x0000, /* R435 */ - 0x0000, /* R436 */ - 0x0000, /* R437 */ - 0x0000, /* R438 */ - 0x0000, /* R439 */ - 0x0000, /* R440 */ - 0x0000, /* R441 */ - 0x0000, /* R442 */ - 0x0000, /* R443 */ - 0x0000, /* R444 */ - 0x0000, /* R445 */ - 0x0000, /* R446 */ - 0x0000, /* R447 */ - 0x0000, /* R448 */ - 0x0000, /* R449 */ - 0x0000, /* R450 */ - 0x0000, /* R451 */ - 0x0000, /* R452 */ - 0x0000, /* R453 */ - 0x0000, /* R454 */ - 0x0000, /* R455 */ - 0x0000, /* R456 */ - 0x0000, /* R457 */ - 0x0000, /* R458 */ - 0x0000, /* R459 */ - 0x0000, /* R460 */ - 0x0000, /* R461 */ - 0x0000, /* R462 */ - 0x0000, /* R463 */ - 0x0000, /* R464 */ - 0x0000, /* R465 */ - 0x0000, /* R466 */ - 0x0000, /* R467 */ - 0x0000, /* R468 */ - 0x0000, /* R469 */ - 0x0000, /* R470 */ - 0x0000, /* R471 */ - 0x0000, /* R472 */ - 0x0000, /* R473 */ - 0x0000, /* R474 */ - 0x0000, /* R475 */ - 0x0000, /* R476 */ - 0x0000, /* R477 */ - 0x0000, /* R478 */ - 0x0000, /* R479 */ - 0x0000, /* R480 */ - 0x0000, /* R481 */ - 0x0000, /* R482 */ - 0x0000, /* R483 */ - 0x0000, /* R484 */ - 0x0000, /* R485 */ - 0x0000, /* R486 */ - 0x0000, /* R487 */ - 0x0000, /* R488 */ - 0x0000, /* R489 */ - 0x0000, /* R490 */ - 0x0000, /* R491 */ - 0x0000, /* R492 */ - 0x0000, /* R493 */ - 0x0000, /* R494 */ - 0x0000, /* R495 */ - 0x0000, /* R496 */ - 0x0000, /* R497 */ - 0x0000, /* R498 */ - 0x0000, /* R499 */ - 0x0000, /* R500 */ - 0x0000, /* R501 */ - 0x0000, /* R502 */ - 0x0000, /* R503 */ - 0x0000, /* R504 */ - 0x0000, /* R505 */ - 0x0000, /* R506 */ - 0x0000, /* R507 */ - 0x0000, /* R508 */ - 0x0000, /* R509 */ - 0x0000, /* R510 */ - 0x0000, /* R511 */ - 0x0000, /* R512 - AIF1 Clocking (1) */ - 0x0000, /* R513 - AIF1 Clocking (2) */ - 0x0000, /* R514 */ - 0x0000, /* R515 */ - 0x0000, /* R516 - AIF2 Clocking (1) */ - 0x0000, /* R517 - AIF2 Clocking (2) */ - 0x0000, /* R518 */ - 0x0000, /* R519 */ - 0x0000, /* R520 - Clocking (1) */ - 0x0000, /* R521 - Clocking (2) */ - 0x0000, /* R522 */ - 0x0000, /* R523 */ - 0x0000, /* R524 */ - 0x0000, /* R525 */ - 0x0000, /* R526 */ - 0x0000, /* R527 */ - 0x0083, /* R528 - AIF1 Rate */ - 0x0083, /* R529 - AIF2 Rate */ - 0x0000, /* R530 - Rate Status */ - 0x0000, /* R531 */ - 0x0000, /* R532 */ - 0x0000, /* R533 */ - 0x0000, /* R534 */ - 0x0000, /* R535 */ - 0x0000, /* R536 */ - 0x0000, /* R537 */ - 0x0000, /* R538 */ - 0x0000, /* R539 */ - 0x0000, /* R540 */ - 0x0000, /* R541 */ - 0x0000, /* R542 */ - 0x0000, /* R543 */ - 0x0000, /* R544 - FLL1 Control (1) */ - 0x0000, /* R545 - FLL1 Control (2) */ - 0x0000, /* R546 - FLL1 Control (3) */ - 0x0000, /* R547 - FLL1 Control (4) */ - 0x0C80, /* R548 - FLL1 Control (5) */ - 0x0000, /* R549 */ - 0x0000, /* R550 - FLL1 EFS 1 */ - 0x0006, /* R551 - FLL1 EFS 2 */ - 0x0000, /* R552 */ - 0x0000, /* R553 */ - 0x0000, /* R554 */ - 0x0000, /* R555 */ - 0x0000, /* R556 */ - 0x0000, /* R557 */ - 0x0000, /* R558 */ - 0x0000, /* R559 */ - 0x0000, /* R560 */ - 0x0000, /* R561 */ - 0x0000, /* R562 */ - 0x0000, /* R563 */ - 0x0000, /* R564 */ - 0x0000, /* R565 */ - 0x0000, /* R566 */ - 0x0000, /* R567 */ - 0x0000, /* R568 */ - 0x0000, /* R569 */ - 0x0000, /* R570 */ - 0x0000, /* R571 */ - 0x0000, /* R572 */ - 0x0000, /* R573 */ - 0x0000, /* R574 */ - 0x0000, /* R575 */ - 0x0000, /* R576 - FLL2 Control (1) */ - 0x0000, /* R577 - FLL2 Control (2) */ - 0x0000, /* R578 - FLL2 Control (3) */ - 0x0000, /* R579 - FLL2 Control (4) */ - 0x0C80, /* R580 - FLL2 Control (5) */ - 0x0000, /* R581 */ - 0x0000, /* R582 - FLL2 EFS 1 */ - 0x0006, /* R583 - FLL2 EFS 2 */ - 0x0000, /* R584 */ - 0x0000, /* R585 */ - 0x0000, /* R586 */ - 0x0000, /* R587 */ - 0x0000, /* R588 */ - 0x0000, /* R589 */ - 0x0000, /* R590 */ - 0x0000, /* R591 */ - 0x0000, /* R592 */ - 0x0000, /* R593 */ - 0x0000, /* R594 */ - 0x0000, /* R595 */ - 0x0000, /* R596 */ - 0x0000, /* R597 */ - 0x0000, /* R598 */ - 0x0000, /* R599 */ - 0x0000, /* R600 */ - 0x0000, /* R601 */ - 0x0000, /* R602 */ - 0x0000, /* R603 */ - 0x0000, /* R604 */ - 0x0000, /* R605 */ - 0x0000, /* R606 */ - 0x0000, /* R607 */ - 0x0000, /* R608 */ - 0x0000, /* R609 */ - 0x0000, /* R610 */ - 0x0000, /* R611 */ - 0x0000, /* R612 */ - 0x0000, /* R613 */ - 0x0000, /* R614 */ - 0x0000, /* R615 */ - 0x0000, /* R616 */ - 0x0000, /* R617 */ - 0x0000, /* R618 */ - 0x0000, /* R619 */ - 0x0000, /* R620 */ - 0x0000, /* R621 */ - 0x0000, /* R622 */ - 0x0000, /* R623 */ - 0x0000, /* R624 */ - 0x0000, /* R625 */ - 0x0000, /* R626 */ - 0x0000, /* R627 */ - 0x0000, /* R628 */ - 0x0000, /* R629 */ - 0x0000, /* R630 */ - 0x0000, /* R631 */ - 0x0000, /* R632 */ - 0x0000, /* R633 */ - 0x0000, /* R634 */ - 0x0000, /* R635 */ - 0x0000, /* R636 */ - 0x0000, /* R637 */ - 0x0000, /* R638 */ - 0x0000, /* R639 */ - 0x0000, /* R640 */ - 0x0000, /* R641 */ - 0x0000, /* R642 */ - 0x0000, /* R643 */ - 0x0000, /* R644 */ - 0x0000, /* R645 */ - 0x0000, /* R646 */ - 0x0000, /* R647 */ - 0x0000, /* R648 */ - 0x0000, /* R649 */ - 0x0000, /* R650 */ - 0x0000, /* R651 */ - 0x0000, /* R652 */ - 0x0000, /* R653 */ - 0x0000, /* R654 */ - 0x0000, /* R655 */ - 0x0000, /* R656 */ - 0x0000, /* R657 */ - 0x0000, /* R658 */ - 0x0000, /* R659 */ - 0x0000, /* R660 */ - 0x0000, /* R661 */ - 0x0000, /* R662 */ - 0x0000, /* R663 */ - 0x0000, /* R664 */ - 0x0000, /* R665 */ - 0x0000, /* R666 */ - 0x0000, /* R667 */ - 0x0000, /* R668 */ - 0x0000, /* R669 */ - 0x0000, /* R670 */ - 0x0000, /* R671 */ - 0x0000, /* R672 */ - 0x0000, /* R673 */ - 0x0000, /* R674 */ - 0x0000, /* R675 */ - 0x0000, /* R676 */ - 0x0000, /* R677 */ - 0x0000, /* R678 */ - 0x0000, /* R679 */ - 0x0000, /* R680 */ - 0x0000, /* R681 */ - 0x0000, /* R682 */ - 0x0000, /* R683 */ - 0x0000, /* R684 */ - 0x0000, /* R685 */ - 0x0000, /* R686 */ - 0x0000, /* R687 */ - 0x0000, /* R688 */ - 0x0000, /* R689 */ - 0x0000, /* R690 */ - 0x0000, /* R691 */ - 0x0000, /* R692 */ - 0x0000, /* R693 */ - 0x0000, /* R694 */ - 0x0000, /* R695 */ - 0x0000, /* R696 */ - 0x0000, /* R697 */ - 0x0000, /* R698 */ - 0x0000, /* R699 */ - 0x0000, /* R700 */ - 0x0000, /* R701 */ - 0x0000, /* R702 */ - 0x0000, /* R703 */ - 0x0000, /* R704 */ - 0x0000, /* R705 */ - 0x0000, /* R706 */ - 0x0000, /* R707 */ - 0x0000, /* R708 */ - 0x0000, /* R709 */ - 0x0000, /* R710 */ - 0x0000, /* R711 */ - 0x0000, /* R712 */ - 0x0000, /* R713 */ - 0x0000, /* R714 */ - 0x0000, /* R715 */ - 0x0000, /* R716 */ - 0x0000, /* R717 */ - 0x0000, /* R718 */ - 0x0000, /* R719 */ - 0x0000, /* R720 */ - 0x0000, /* R721 */ - 0x0000, /* R722 */ - 0x0000, /* R723 */ - 0x0000, /* R724 */ - 0x0000, /* R725 */ - 0x0000, /* R726 */ - 0x0000, /* R727 */ - 0x0000, /* R728 */ - 0x0000, /* R729 */ - 0x0000, /* R730 */ - 0x0000, /* R731 */ - 0x0000, /* R732 */ - 0x0000, /* R733 */ - 0x0000, /* R734 */ - 0x0000, /* R735 */ - 0x0000, /* R736 */ - 0x0000, /* R737 */ - 0x0000, /* R738 */ - 0x0000, /* R739 */ - 0x0000, /* R740 */ - 0x0000, /* R741 */ - 0x0000, /* R742 */ - 0x0000, /* R743 */ - 0x0000, /* R744 */ - 0x0000, /* R745 */ - 0x0000, /* R746 */ - 0x0000, /* R747 */ - 0x0000, /* R748 */ - 0x0000, /* R749 */ - 0x0000, /* R750 */ - 0x0000, /* R751 */ - 0x0000, /* R752 */ - 0x0000, /* R753 */ - 0x0000, /* R754 */ - 0x0000, /* R755 */ - 0x0000, /* R756 */ - 0x0000, /* R757 */ - 0x0000, /* R758 */ - 0x0000, /* R759 */ - 0x0000, /* R760 */ - 0x0000, /* R761 */ - 0x0000, /* R762 */ - 0x0000, /* R763 */ - 0x0000, /* R764 */ - 0x0000, /* R765 */ - 0x0000, /* R766 */ - 0x0000, /* R767 */ - 0x4050, /* R768 - AIF1 Control (1) */ - 0x4000, /* R769 - AIF1 Control (2) */ - 0x0000, /* R770 - AIF1 Master/Slave */ - 0x0040, /* R771 - AIF1 BCLK */ - 0x0040, /* R772 - AIF1ADC LRCLK */ - 0x0040, /* R773 - AIF1DAC LRCLK */ - 0x0004, /* R774 - AIF1DAC Data */ - 0x0100, /* R775 - AIF1ADC Data */ - 0x0000, /* R776 */ - 0x0000, /* R777 */ - 0x0000, /* R778 */ - 0x0000, /* R779 */ - 0x0000, /* R780 */ - 0x0000, /* R781 */ - 0x0000, /* R782 */ - 0x0000, /* R783 */ - 0x4050, /* R784 - AIF2 Control (1) */ - 0x4000, /* R785 - AIF2 Control (2) */ - 0x0000, /* R786 - AIF2 Master/Slave */ - 0x0040, /* R787 - AIF2 BCLK */ - 0x0040, /* R788 - AIF2ADC LRCLK */ - 0x0040, /* R789 - AIF2DAC LRCLK */ - 0x0000, /* R790 - AIF2DAC Data */ - 0x0000, /* R791 - AIF2ADC Data */ - 0x0000, /* R792 */ - 0x0000, /* R793 */ - 0x0000, /* R794 */ - 0x0000, /* R795 */ - 0x0000, /* R796 */ - 0x0000, /* R797 */ - 0x0000, /* R798 */ - 0x0000, /* R799 */ - 0x0000, /* R800 */ - 0x0000, /* R801 */ - 0x0000, /* R802 */ - 0x0000, /* R803 */ - 0x0000, /* R804 */ - 0x0000, /* R805 */ - 0x0000, /* R806 */ - 0x0000, /* R807 */ - 0x0000, /* R808 */ - 0x0000, /* R809 */ - 0x0000, /* R810 */ - 0x0000, /* R811 */ - 0x0000, /* R812 */ - 0x0000, /* R813 */ - 0x0000, /* R814 */ - 0x0000, /* R815 */ - 0x0000, /* R816 */ - 0x0000, /* R817 */ - 0x0000, /* R818 */ - 0x0000, /* R819 */ - 0x0000, /* R820 */ - 0x0000, /* R821 */ - 0x0000, /* R822 */ - 0x0000, /* R823 */ - 0x0000, /* R824 */ - 0x0000, /* R825 */ - 0x0000, /* R826 */ - 0x0000, /* R827 */ - 0x0000, /* R828 */ - 0x0000, /* R829 */ - 0x0000, /* R830 */ - 0x0000, /* R831 */ - 0x0000, /* R832 */ - 0x0000, /* R833 */ - 0x0000, /* R834 */ - 0x0000, /* R835 */ - 0x0000, /* R836 */ - 0x0000, /* R837 */ - 0x0000, /* R838 */ - 0x0000, /* R839 */ - 0x0000, /* R840 */ - 0x0000, /* R841 */ - 0x0000, /* R842 */ - 0x0000, /* R843 */ - 0x0000, /* R844 */ - 0x0000, /* R845 */ - 0x0000, /* R846 */ - 0x0000, /* R847 */ - 0x0000, /* R848 */ - 0x0000, /* R849 */ - 0x0000, /* R850 */ - 0x0000, /* R851 */ - 0x0000, /* R852 */ - 0x0000, /* R853 */ - 0x0000, /* R854 */ - 0x0000, /* R855 */ - 0x0000, /* R856 */ - 0x0000, /* R857 */ - 0x0000, /* R858 */ - 0x0000, /* R859 */ - 0x0000, /* R860 */ - 0x0000, /* R861 */ - 0x0000, /* R862 */ - 0x0000, /* R863 */ - 0x0000, /* R864 */ - 0x0000, /* R865 */ - 0x0000, /* R866 */ - 0x0000, /* R867 */ - 0x0000, /* R868 */ - 0x0000, /* R869 */ - 0x0000, /* R870 */ - 0x0000, /* R871 */ - 0x0000, /* R872 */ - 0x0000, /* R873 */ - 0x0000, /* R874 */ - 0x0000, /* R875 */ - 0x0000, /* R876 */ - 0x0000, /* R877 */ - 0x0000, /* R878 */ - 0x0000, /* R879 */ - 0x0000, /* R880 */ - 0x0000, /* R881 */ - 0x0000, /* R882 */ - 0x0000, /* R883 */ - 0x0000, /* R884 */ - 0x0000, /* R885 */ - 0x0000, /* R886 */ - 0x0000, /* R887 */ - 0x0000, /* R888 */ - 0x0000, /* R889 */ - 0x0000, /* R890 */ - 0x0000, /* R891 */ - 0x0000, /* R892 */ - 0x0000, /* R893 */ - 0x0000, /* R894 */ - 0x0000, /* R895 */ - 0x0000, /* R896 */ - 0x0000, /* R897 */ - 0x0000, /* R898 */ - 0x0000, /* R899 */ - 0x0000, /* R900 */ - 0x0000, /* R901 */ - 0x0000, /* R902 */ - 0x0000, /* R903 */ - 0x0000, /* R904 */ - 0x0000, /* R905 */ - 0x0000, /* R906 */ - 0x0000, /* R907 */ - 0x0000, /* R908 */ - 0x0000, /* R909 */ - 0x0000, /* R910 */ - 0x0000, /* R911 */ - 0x0000, /* R912 */ - 0x0000, /* R913 */ - 0x0000, /* R914 */ - 0x0000, /* R915 */ - 0x0000, /* R916 */ - 0x0000, /* R917 */ - 0x0000, /* R918 */ - 0x0000, /* R919 */ - 0x0000, /* R920 */ - 0x0000, /* R921 */ - 0x0000, /* R922 */ - 0x0000, /* R923 */ - 0x0000, /* R924 */ - 0x0000, /* R925 */ - 0x0000, /* R926 */ - 0x0000, /* R927 */ - 0x0000, /* R928 */ - 0x0000, /* R929 */ - 0x0000, /* R930 */ - 0x0000, /* R931 */ - 0x0000, /* R932 */ - 0x0000, /* R933 */ - 0x0000, /* R934 */ - 0x0000, /* R935 */ - 0x0000, /* R936 */ - 0x0000, /* R937 */ - 0x0000, /* R938 */ - 0x0000, /* R939 */ - 0x0000, /* R940 */ - 0x0000, /* R941 */ - 0x0000, /* R942 */ - 0x0000, /* R943 */ - 0x0000, /* R944 */ - 0x0000, /* R945 */ - 0x0000, /* R946 */ - 0x0000, /* R947 */ - 0x0000, /* R948 */ - 0x0000, /* R949 */ - 0x0000, /* R950 */ - 0x0000, /* R951 */ - 0x0000, /* R952 */ - 0x0000, /* R953 */ - 0x0000, /* R954 */ - 0x0000, /* R955 */ - 0x0000, /* R956 */ - 0x0000, /* R957 */ - 0x0000, /* R958 */ - 0x0000, /* R959 */ - 0x0000, /* R960 */ - 0x0000, /* R961 */ - 0x0000, /* R962 */ - 0x0000, /* R963 */ - 0x0000, /* R964 */ - 0x0000, /* R965 */ - 0x0000, /* R966 */ - 0x0000, /* R967 */ - 0x0000, /* R968 */ - 0x0000, /* R969 */ - 0x0000, /* R970 */ - 0x0000, /* R971 */ - 0x0000, /* R972 */ - 0x0000, /* R973 */ - 0x0000, /* R974 */ - 0x0000, /* R975 */ - 0x0000, /* R976 */ - 0x0000, /* R977 */ - 0x0000, /* R978 */ - 0x0000, /* R979 */ - 0x0000, /* R980 */ - 0x0000, /* R981 */ - 0x0000, /* R982 */ - 0x0000, /* R983 */ - 0x0000, /* R984 */ - 0x0000, /* R985 */ - 0x0000, /* R986 */ - 0x0000, /* R987 */ - 0x0000, /* R988 */ - 0x0000, /* R989 */ - 0x0000, /* R990 */ - 0x0000, /* R991 */ - 0x0000, /* R992 */ - 0x0000, /* R993 */ - 0x0000, /* R994 */ - 0x0000, /* R995 */ - 0x0000, /* R996 */ - 0x0000, /* R997 */ - 0x0000, /* R998 */ - 0x0000, /* R999 */ - 0x0000, /* R1000 */ - 0x0000, /* R1001 */ - 0x0000, /* R1002 */ - 0x0000, /* R1003 */ - 0x0000, /* R1004 */ - 0x0000, /* R1005 */ - 0x0000, /* R1006 */ - 0x0000, /* R1007 */ - 0x0000, /* R1008 */ - 0x0000, /* R1009 */ - 0x0000, /* R1010 */ - 0x0000, /* R1011 */ - 0x0000, /* R1012 */ - 0x0000, /* R1013 */ - 0x0000, /* R1014 */ - 0x0000, /* R1015 */ - 0x0000, /* R1016 */ - 0x0000, /* R1017 */ - 0x0000, /* R1018 */ - 0x0000, /* R1019 */ - 0x0000, /* R1020 */ - 0x0000, /* R1021 */ - 0x0000, /* R1022 */ - 0x0000, /* R1023 */ - 0x00C0, /* R1024 - AIF1 ADC1 Left Volume */ - 0x00C0, /* R1025 - AIF1 ADC1 Right Volume */ - 0x00C0, /* R1026 - AIF1 DAC1 Left Volume */ - 0x00C0, /* R1027 - AIF1 DAC1 Right Volume */ - 0x00C0, /* R1028 - AIF1 ADC2 Left Volume */ - 0x00C0, /* R1029 - AIF1 ADC2 Right Volume */ - 0x00C0, /* R1030 - AIF1 DAC2 Left Volume */ - 0x00C0, /* R1031 - AIF1 DAC2 Right Volume */ - 0x0000, /* R1032 */ - 0x0000, /* R1033 */ - 0x0000, /* R1034 */ - 0x0000, /* R1035 */ - 0x0000, /* R1036 */ - 0x0000, /* R1037 */ - 0x0000, /* R1038 */ - 0x0000, /* R1039 */ - 0x0000, /* R1040 - AIF1 ADC1 Filters */ - 0x0000, /* R1041 - AIF1 ADC2 Filters */ - 0x0000, /* R1042 */ - 0x0000, /* R1043 */ - 0x0000, /* R1044 */ - 0x0000, /* R1045 */ - 0x0000, /* R1046 */ - 0x0000, /* R1047 */ - 0x0000, /* R1048 */ - 0x0000, /* R1049 */ - 0x0000, /* R1050 */ - 0x0000, /* R1051 */ - 0x0000, /* R1052 */ - 0x0000, /* R1053 */ - 0x0000, /* R1054 */ - 0x0000, /* R1055 */ - 0x0200, /* R1056 - AIF1 DAC1 Filters (1) */ - 0x0010, /* R1057 - AIF1 DAC1 Filters (2) */ - 0x0200, /* R1058 - AIF1 DAC2 Filters (1) */ - 0x0010, /* R1059 - AIF1 DAC2 Filters (2) */ - 0x0000, /* R1060 */ - 0x0000, /* R1061 */ - 0x0000, /* R1062 */ - 0x0000, /* R1063 */ - 0x0000, /* R1064 */ - 0x0000, /* R1065 */ - 0x0000, /* R1066 */ - 0x0000, /* R1067 */ - 0x0000, /* R1068 */ - 0x0000, /* R1069 */ - 0x0000, /* R1070 */ - 0x0000, /* R1071 */ - 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ - 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ - 0x0000, /* R1074 */ - 0x0000, /* R1075 */ - 0x0000, /* R1076 */ - 0x0000, /* R1077 */ - 0x0000, /* R1078 */ - 0x0000, /* R1079 */ - 0x0000, /* R1080 */ - 0x0000, /* R1081 */ - 0x0000, /* R1082 */ - 0x0000, /* R1083 */ - 0x0000, /* R1084 */ - 0x0000, /* R1085 */ - 0x0000, /* R1086 */ - 0x0000, /* R1087 */ - 0x0098, /* R1088 - AIF1 DRC1 (1) */ - 0x0845, /* R1089 - AIF1 DRC1 (2) */ - 0x0000, /* R1090 - AIF1 DRC1 (3) */ - 0x0000, /* R1091 - AIF1 DRC1 (4) */ - 0x0000, /* R1092 - AIF1 DRC1 (5) */ - 0x0000, /* R1093 */ - 0x0000, /* R1094 */ - 0x0000, /* R1095 */ - 0x0000, /* R1096 */ - 0x0000, /* R1097 */ - 0x0000, /* R1098 */ - 0x0000, /* R1099 */ - 0x0000, /* R1100 */ - 0x0000, /* R1101 */ - 0x0000, /* R1102 */ - 0x0000, /* R1103 */ - 0x0098, /* R1104 - AIF1 DRC2 (1) */ - 0x0845, /* R1105 - AIF1 DRC2 (2) */ - 0x0000, /* R1106 - AIF1 DRC2 (3) */ - 0x0000, /* R1107 - AIF1 DRC2 (4) */ - 0x0000, /* R1108 - AIF1 DRC2 (5) */ - 0x0000, /* R1109 */ - 0x0000, /* R1110 */ - 0x0000, /* R1111 */ - 0x0000, /* R1112 */ - 0x0000, /* R1113 */ - 0x0000, /* R1114 */ - 0x0000, /* R1115 */ - 0x0000, /* R1116 */ - 0x0000, /* R1117 */ - 0x0000, /* R1118 */ - 0x0000, /* R1119 */ - 0x0000, /* R1120 */ - 0x0000, /* R1121 */ - 0x0000, /* R1122 */ - 0x0000, /* R1123 */ - 0x0000, /* R1124 */ - 0x0000, /* R1125 */ - 0x0000, /* R1126 */ - 0x0000, /* R1127 */ - 0x0000, /* R1128 */ - 0x0000, /* R1129 */ - 0x0000, /* R1130 */ - 0x0000, /* R1131 */ - 0x0000, /* R1132 */ - 0x0000, /* R1133 */ - 0x0000, /* R1134 */ - 0x0000, /* R1135 */ - 0x0000, /* R1136 */ - 0x0000, /* R1137 */ - 0x0000, /* R1138 */ - 0x0000, /* R1139 */ - 0x0000, /* R1140 */ - 0x0000, /* R1141 */ - 0x0000, /* R1142 */ - 0x0000, /* R1143 */ - 0x0000, /* R1144 */ - 0x0000, /* R1145 */ - 0x0000, /* R1146 */ - 0x0000, /* R1147 */ - 0x0000, /* R1148 */ - 0x0000, /* R1149 */ - 0x0000, /* R1150 */ - 0x0000, /* R1151 */ - 0x6318, /* R1152 - AIF1 DAC1 EQ Gains (1) */ - 0x6300, /* R1153 - AIF1 DAC1 EQ Gains (2) */ - 0x0FCA, /* R1154 - AIF1 DAC1 EQ Band 1 A */ - 0x0400, /* R1155 - AIF1 DAC1 EQ Band 1 B */ - 0x00D8, /* R1156 - AIF1 DAC1 EQ Band 1 PG */ - 0x1EB5, /* R1157 - AIF1 DAC1 EQ Band 2 A */ - 0xF145, /* R1158 - AIF1 DAC1 EQ Band 2 B */ - 0x0B75, /* R1159 - AIF1 DAC1 EQ Band 2 C */ - 0x01C5, /* R1160 - AIF1 DAC1 EQ Band 2 PG */ - 0x1C58, /* R1161 - AIF1 DAC1 EQ Band 3 A */ - 0xF373, /* R1162 - AIF1 DAC1 EQ Band 3 B */ - 0x0A54, /* R1163 - AIF1 DAC1 EQ Band 3 C */ - 0x0558, /* R1164 - AIF1 DAC1 EQ Band 3 PG */ - 0x168E, /* R1165 - AIF1 DAC1 EQ Band 4 A */ - 0xF829, /* R1166 - AIF1 DAC1 EQ Band 4 B */ - 0x07AD, /* R1167 - AIF1 DAC1 EQ Band 4 C */ - 0x1103, /* R1168 - AIF1 DAC1 EQ Band 4 PG */ - 0x0564, /* R1169 - AIF1 DAC1 EQ Band 5 A */ - 0x0559, /* R1170 - AIF1 DAC1 EQ Band 5 B */ - 0x4000, /* R1171 - AIF1 DAC1 EQ Band 5 PG */ - 0x0000, /* R1172 */ - 0x0000, /* R1173 */ - 0x0000, /* R1174 */ - 0x0000, /* R1175 */ - 0x0000, /* R1176 */ - 0x0000, /* R1177 */ - 0x0000, /* R1178 */ - 0x0000, /* R1179 */ - 0x0000, /* R1180 */ - 0x0000, /* R1181 */ - 0x0000, /* R1182 */ - 0x0000, /* R1183 */ - 0x6318, /* R1184 - AIF1 DAC2 EQ Gains (1) */ - 0x6300, /* R1185 - AIF1 DAC2 EQ Gains (2) */ - 0x0FCA, /* R1186 - AIF1 DAC2 EQ Band 1 A */ - 0x0400, /* R1187 - AIF1 DAC2 EQ Band 1 B */ - 0x00D8, /* R1188 - AIF1 DAC2 EQ Band 1 PG */ - 0x1EB5, /* R1189 - AIF1 DAC2 EQ Band 2 A */ - 0xF145, /* R1190 - AIF1 DAC2 EQ Band 2 B */ - 0x0B75, /* R1191 - AIF1 DAC2 EQ Band 2 C */ - 0x01C5, /* R1192 - AIF1 DAC2 EQ Band 2 PG */ - 0x1C58, /* R1193 - AIF1 DAC2 EQ Band 3 A */ - 0xF373, /* R1194 - AIF1 DAC2 EQ Band 3 B */ - 0x0A54, /* R1195 - AIF1 DAC2 EQ Band 3 C */ - 0x0558, /* R1196 - AIF1 DAC2 EQ Band 3 PG */ - 0x168E, /* R1197 - AIF1 DAC2 EQ Band 4 A */ - 0xF829, /* R1198 - AIF1 DAC2 EQ Band 4 B */ - 0x07AD, /* R1199 - AIF1 DAC2 EQ Band 4 C */ - 0x1103, /* R1200 - AIF1 DAC2 EQ Band 4 PG */ - 0x0564, /* R1201 - AIF1 DAC2 EQ Band 5 A */ - 0x0559, /* R1202 - AIF1 DAC2 EQ Band 5 B */ - 0x4000, /* R1203 - AIF1 DAC2 EQ Band 5 PG */ - 0x0000, /* R1204 */ - 0x0000, /* R1205 */ - 0x0000, /* R1206 */ - 0x0000, /* R1207 */ - 0x0000, /* R1208 */ - 0x0000, /* R1209 */ - 0x0000, /* R1210 */ - 0x0000, /* R1211 */ - 0x0000, /* R1212 */ - 0x0000, /* R1213 */ - 0x0000, /* R1214 */ - 0x0000, /* R1215 */ - 0x0000, /* R1216 */ - 0x0000, /* R1217 */ - 0x0000, /* R1218 */ - 0x0000, /* R1219 */ - 0x0000, /* R1220 */ - 0x0000, /* R1221 */ - 0x0000, /* R1222 */ - 0x0000, /* R1223 */ - 0x0000, /* R1224 */ - 0x0000, /* R1225 */ - 0x0000, /* R1226 */ - 0x0000, /* R1227 */ - 0x0000, /* R1228 */ - 0x0000, /* R1229 */ - 0x0000, /* R1230 */ - 0x0000, /* R1231 */ - 0x0000, /* R1232 */ - 0x0000, /* R1233 */ - 0x0000, /* R1234 */ - 0x0000, /* R1235 */ - 0x0000, /* R1236 */ - 0x0000, /* R1237 */ - 0x0000, /* R1238 */ - 0x0000, /* R1239 */ - 0x0000, /* R1240 */ - 0x0000, /* R1241 */ - 0x0000, /* R1242 */ - 0x0000, /* R1243 */ - 0x0000, /* R1244 */ - 0x0000, /* R1245 */ - 0x0000, /* R1246 */ - 0x0000, /* R1247 */ - 0x0000, /* R1248 */ - 0x0000, /* R1249 */ - 0x0000, /* R1250 */ - 0x0000, /* R1251 */ - 0x0000, /* R1252 */ - 0x0000, /* R1253 */ - 0x0000, /* R1254 */ - 0x0000, /* R1255 */ - 0x0000, /* R1256 */ - 0x0000, /* R1257 */ - 0x0000, /* R1258 */ - 0x0000, /* R1259 */ - 0x0000, /* R1260 */ - 0x0000, /* R1261 */ - 0x0000, /* R1262 */ - 0x0000, /* R1263 */ - 0x0000, /* R1264 */ - 0x0000, /* R1265 */ - 0x0000, /* R1266 */ - 0x0000, /* R1267 */ - 0x0000, /* R1268 */ - 0x0000, /* R1269 */ - 0x0000, /* R1270 */ - 0x0000, /* R1271 */ - 0x0000, /* R1272 */ - 0x0000, /* R1273 */ - 0x0000, /* R1274 */ - 0x0000, /* R1275 */ - 0x0000, /* R1276 */ - 0x0000, /* R1277 */ - 0x0000, /* R1278 */ - 0x0000, /* R1279 */ - 0x00C0, /* R1280 - AIF2 ADC Left Volume */ - 0x00C0, /* R1281 - AIF2 ADC Right Volume */ - 0x00C0, /* R1282 - AIF2 DAC Left Volume */ - 0x00C0, /* R1283 - AIF2 DAC Right Volume */ - 0x0000, /* R1284 */ - 0x0000, /* R1285 */ - 0x0000, /* R1286 */ - 0x0000, /* R1287 */ - 0x0000, /* R1288 */ - 0x0000, /* R1289 */ - 0x0000, /* R1290 */ - 0x0000, /* R1291 */ - 0x0000, /* R1292 */ - 0x0000, /* R1293 */ - 0x0000, /* R1294 */ - 0x0000, /* R1295 */ - 0x0000, /* R1296 - AIF2 ADC Filters */ - 0x0000, /* R1297 */ - 0x0000, /* R1298 */ - 0x0000, /* R1299 */ - 0x0000, /* R1300 */ - 0x0000, /* R1301 */ - 0x0000, /* R1302 */ - 0x0000, /* R1303 */ - 0x0000, /* R1304 */ - 0x0000, /* R1305 */ - 0x0000, /* R1306 */ - 0x0000, /* R1307 */ - 0x0000, /* R1308 */ - 0x0000, /* R1309 */ - 0x0000, /* R1310 */ - 0x0000, /* R1311 */ - 0x0200, /* R1312 - AIF2 DAC Filters (1) */ - 0x0010, /* R1313 - AIF2 DAC Filters (2) */ - 0x0000, /* R1314 */ - 0x0000, /* R1315 */ - 0x0000, /* R1316 */ - 0x0000, /* R1317 */ - 0x0000, /* R1318 */ - 0x0000, /* R1319 */ - 0x0000, /* R1320 */ - 0x0000, /* R1321 */ - 0x0000, /* R1322 */ - 0x0000, /* R1323 */ - 0x0000, /* R1324 */ - 0x0000, /* R1325 */ - 0x0000, /* R1326 */ - 0x0000, /* R1327 */ - 0x0068, /* R1328 - AIF2 DAC Noise Gate */ - 0x0000, /* R1329 */ - 0x0000, /* R1330 */ - 0x0000, /* R1331 */ - 0x0000, /* R1332 */ - 0x0000, /* R1333 */ - 0x0000, /* R1334 */ - 0x0000, /* R1335 */ - 0x0000, /* R1336 */ - 0x0000, /* R1337 */ - 0x0000, /* R1338 */ - 0x0000, /* R1339 */ - 0x0000, /* R1340 */ - 0x0000, /* R1341 */ - 0x0000, /* R1342 */ - 0x0000, /* R1343 */ - 0x0098, /* R1344 - AIF2 DRC (1) */ - 0x0845, /* R1345 - AIF2 DRC (2) */ - 0x0000, /* R1346 - AIF2 DRC (3) */ - 0x0000, /* R1347 - AIF2 DRC (4) */ - 0x0000, /* R1348 - AIF2 DRC (5) */ - 0x0000, /* R1349 */ - 0x0000, /* R1350 */ - 0x0000, /* R1351 */ - 0x0000, /* R1352 */ - 0x0000, /* R1353 */ - 0x0000, /* R1354 */ - 0x0000, /* R1355 */ - 0x0000, /* R1356 */ - 0x0000, /* R1357 */ - 0x0000, /* R1358 */ - 0x0000, /* R1359 */ - 0x0000, /* R1360 */ - 0x0000, /* R1361 */ - 0x0000, /* R1362 */ - 0x0000, /* R1363 */ - 0x0000, /* R1364 */ - 0x0000, /* R1365 */ - 0x0000, /* R1366 */ - 0x0000, /* R1367 */ - 0x0000, /* R1368 */ - 0x0000, /* R1369 */ - 0x0000, /* R1370 */ - 0x0000, /* R1371 */ - 0x0000, /* R1372 */ - 0x0000, /* R1373 */ - 0x0000, /* R1374 */ - 0x0000, /* R1375 */ - 0x0000, /* R1376 */ - 0x0000, /* R1377 */ - 0x0000, /* R1378 */ - 0x0000, /* R1379 */ - 0x0000, /* R1380 */ - 0x0000, /* R1381 */ - 0x0000, /* R1382 */ - 0x0000, /* R1383 */ - 0x0000, /* R1384 */ - 0x0000, /* R1385 */ - 0x0000, /* R1386 */ - 0x0000, /* R1387 */ - 0x0000, /* R1388 */ - 0x0000, /* R1389 */ - 0x0000, /* R1390 */ - 0x0000, /* R1391 */ - 0x0000, /* R1392 */ - 0x0000, /* R1393 */ - 0x0000, /* R1394 */ - 0x0000, /* R1395 */ - 0x0000, /* R1396 */ - 0x0000, /* R1397 */ - 0x0000, /* R1398 */ - 0x0000, /* R1399 */ - 0x0000, /* R1400 */ - 0x0000, /* R1401 */ - 0x0000, /* R1402 */ - 0x0000, /* R1403 */ - 0x0000, /* R1404 */ - 0x0000, /* R1405 */ - 0x0000, /* R1406 */ - 0x0000, /* R1407 */ - 0x6318, /* R1408 - AIF2 EQ Gains (1) */ - 0x6300, /* R1409 - AIF2 EQ Gains (2) */ - 0x0FCA, /* R1410 - AIF2 EQ Band 1 A */ - 0x0400, /* R1411 - AIF2 EQ Band 1 B */ - 0x00D8, /* R1412 - AIF2 EQ Band 1 PG */ - 0x1EB5, /* R1413 - AIF2 EQ Band 2 A */ - 0xF145, /* R1414 - AIF2 EQ Band 2 B */ - 0x0B75, /* R1415 - AIF2 EQ Band 2 C */ - 0x01C5, /* R1416 - AIF2 EQ Band 2 PG */ - 0x1C58, /* R1417 - AIF2 EQ Band 3 A */ - 0xF373, /* R1418 - AIF2 EQ Band 3 B */ - 0x0A54, /* R1419 - AIF2 EQ Band 3 C */ - 0x0558, /* R1420 - AIF2 EQ Band 3 PG */ - 0x168E, /* R1421 - AIF2 EQ Band 4 A */ - 0xF829, /* R1422 - AIF2 EQ Band 4 B */ - 0x07AD, /* R1423 - AIF2 EQ Band 4 C */ - 0x1103, /* R1424 - AIF2 EQ Band 4 PG */ - 0x0564, /* R1425 - AIF2 EQ Band 5 A */ - 0x0559, /* R1426 - AIF2 EQ Band 5 B */ - 0x4000, /* R1427 - AIF2 EQ Band 5 PG */ - 0x0000, /* R1428 */ - 0x0000, /* R1429 */ - 0x0000, /* R1430 */ - 0x0000, /* R1431 */ - 0x0000, /* R1432 */ - 0x0000, /* R1433 */ - 0x0000, /* R1434 */ - 0x0000, /* R1435 */ - 0x0000, /* R1436 */ - 0x0000, /* R1437 */ - 0x0000, /* R1438 */ - 0x0000, /* R1439 */ - 0x0000, /* R1440 */ - 0x0000, /* R1441 */ - 0x0000, /* R1442 */ - 0x0000, /* R1443 */ - 0x0000, /* R1444 */ - 0x0000, /* R1445 */ - 0x0000, /* R1446 */ - 0x0000, /* R1447 */ - 0x0000, /* R1448 */ - 0x0000, /* R1449 */ - 0x0000, /* R1450 */ - 0x0000, /* R1451 */ - 0x0000, /* R1452 */ - 0x0000, /* R1453 */ - 0x0000, /* R1454 */ - 0x0000, /* R1455 */ - 0x0000, /* R1456 */ - 0x0000, /* R1457 */ - 0x0000, /* R1458 */ - 0x0000, /* R1459 */ - 0x0000, /* R1460 */ - 0x0000, /* R1461 */ - 0x0000, /* R1462 */ - 0x0000, /* R1463 */ - 0x0000, /* R1464 */ - 0x0000, /* R1465 */ - 0x0000, /* R1466 */ - 0x0000, /* R1467 */ - 0x0000, /* R1468 */ - 0x0000, /* R1469 */ - 0x0000, /* R1470 */ - 0x0000, /* R1471 */ - 0x0000, /* R1472 */ - 0x0000, /* R1473 */ - 0x0000, /* R1474 */ - 0x0000, /* R1475 */ - 0x0000, /* R1476 */ - 0x0000, /* R1477 */ - 0x0000, /* R1478 */ - 0x0000, /* R1479 */ - 0x0000, /* R1480 */ - 0x0000, /* R1481 */ - 0x0000, /* R1482 */ - 0x0000, /* R1483 */ - 0x0000, /* R1484 */ - 0x0000, /* R1485 */ - 0x0000, /* R1486 */ - 0x0000, /* R1487 */ - 0x0000, /* R1488 */ - 0x0000, /* R1489 */ - 0x0000, /* R1490 */ - 0x0000, /* R1491 */ - 0x0000, /* R1492 */ - 0x0000, /* R1493 */ - 0x0000, /* R1494 */ - 0x0000, /* R1495 */ - 0x0000, /* R1496 */ - 0x0000, /* R1497 */ - 0x0000, /* R1498 */ - 0x0000, /* R1499 */ - 0x0000, /* R1500 */ - 0x0000, /* R1501 */ - 0x0000, /* R1502 */ - 0x0000, /* R1503 */ - 0x0000, /* R1504 */ - 0x0000, /* R1505 */ - 0x0000, /* R1506 */ - 0x0000, /* R1507 */ - 0x0000, /* R1508 */ - 0x0000, /* R1509 */ - 0x0000, /* R1510 */ - 0x0000, /* R1511 */ - 0x0000, /* R1512 */ - 0x0000, /* R1513 */ - 0x0000, /* R1514 */ - 0x0000, /* R1515 */ - 0x0000, /* R1516 */ - 0x0000, /* R1517 */ - 0x0000, /* R1518 */ - 0x0000, /* R1519 */ - 0x0000, /* R1520 */ - 0x0000, /* R1521 */ - 0x0000, /* R1522 */ - 0x0000, /* R1523 */ - 0x0000, /* R1524 */ - 0x0000, /* R1525 */ - 0x0000, /* R1526 */ - 0x0000, /* R1527 */ - 0x0000, /* R1528 */ - 0x0000, /* R1529 */ - 0x0000, /* R1530 */ - 0x0000, /* R1531 */ - 0x0000, /* R1532 */ - 0x0000, /* R1533 */ - 0x0000, /* R1534 */ - 0x0000, /* R1535 */ - 0x0000, /* R1536 - DAC1 Mixer Volumes */ - 0x0000, /* R1537 - DAC1 Left Mixer Routing */ - 0x0000, /* R1538 - DAC1 Right Mixer Routing */ - 0x0000, /* R1539 - DAC2 Mixer Volumes */ - 0x0000, /* R1540 - DAC2 Left Mixer Routing */ - 0x0000, /* R1541 - DAC2 Right Mixer Routing */ - 0x0000, /* R1542 - AIF1 ADC1 Left Mixer Routing */ - 0x0000, /* R1543 - AIF1 ADC1 Right Mixer Routing */ - 0x0000, /* R1544 - AIF1 ADC2 Left Mixer Routing */ - 0x0000, /* R1545 - AIF1 ADC2 Right mixer Routing */ - 0x0000, /* R1546 */ - 0x0000, /* R1547 */ - 0x0000, /* R1548 */ - 0x0000, /* R1549 */ - 0x0000, /* R1550 */ - 0x0000, /* R1551 */ - 0x02C0, /* R1552 - DAC1 Left Volume */ - 0x02C0, /* R1553 - DAC1 Right Volume */ - 0x02C0, /* R1554 - DAC2 Left Volume */ - 0x02C0, /* R1555 - DAC2 Right Volume */ - 0x0000, /* R1556 - DAC Softmute */ - 0x0000, /* R1557 */ - 0x0000, /* R1558 */ - 0x0000, /* R1559 */ - 0x0000, /* R1560 */ - 0x0000, /* R1561 */ - 0x0000, /* R1562 */ - 0x0000, /* R1563 */ - 0x0000, /* R1564 */ - 0x0000, /* R1565 */ - 0x0000, /* R1566 */ - 0x0000, /* R1567 */ - 0x0002, /* R1568 - Oversampling */ - 0x0000, /* R1569 - Sidetone */ -}; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 285890802d62..a9936904d1a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -123,67 +123,6 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) WM8958_MICD_RATE_MASK, val); } -static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) -{ - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; - - switch (reg) { - case WM8994_GPIO_1: - case WM8994_GPIO_2: - case WM8994_GPIO_3: - case WM8994_GPIO_4: - case WM8994_GPIO_5: - case WM8994_GPIO_6: - case WM8994_GPIO_7: - case WM8994_GPIO_8: - case WM8994_GPIO_9: - case WM8994_GPIO_10: - case WM8994_GPIO_11: - case WM8994_INTERRUPT_STATUS_1: - case WM8994_INTERRUPT_STATUS_2: - case WM8994_INTERRUPT_RAW_STATUS_2: - return 1; - - case WM8958_DSP2_PROGRAM: - case WM8958_DSP2_CONFIG: - case WM8958_DSP2_EXECCONTROL: - if (control->type == WM8958) - return 1; - else - return 0; - - default: - break; - } - - if (reg >= WM8994_CACHE_SIZE) - return 0; - return wm8994_access_masks[reg].readable != 0; -} - -static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) -{ - if (reg >= WM8994_CACHE_SIZE) - return 1; - - switch (reg) { - case WM8994_SOFTWARE_RESET: - case WM8994_CHIP_REVISION: - case WM8994_DC_SERVO_1: - case WM8994_DC_SERVO_READBACK: - case WM8994_RATE_STATUS: - case WM8994_LDO_1: - case WM8994_LDO_2: - case WM8958_DSP2_EXECCONTROL: - case WM8958_MIC_DETECT_3: - case WM8994_DC_SERVO_4E: - return 1; - default: - return 0; - } -} - static int configure_aif_clock(struct snd_soc_codec *codec, int aif) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3451,25 +3390,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); - /* Read our current status back from the chip - we don't want to - * reset as this may interfere with the GPIO or LDO operation. */ - for (i = 0; i < WM8994_CACHE_SIZE; i++) { - if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i)) - continue; - - ret = regmap_read(control->regmap, i, ®); - if (ret <= 0) - continue; - - ret = snd_soc_cache_write(codec, i, reg); - if (ret != 0) { - dev_err(codec->dev, - "Failed to initialise cache for 0x%x: %d\n", - i, ret); - goto err; - } - } - /* Set revision-specific configuration */ wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); switch (control->type) { @@ -3900,14 +3820,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .remove = wm8994_codec_remove, .suspend = wm8994_suspend, .resume = wm8994_resume, - .readable_register = wm8994_readable, - .volatile_register = wm8994_volatile, .set_bias_level = wm8994_set_bias_level, - - .reg_cache_size = WM8994_CACHE_SIZE, - .reg_cache_default = wm8994_reg_defaults, - .reg_word_size = 2, - .compress_type = SND_SOC_RBTREE_COMPRESSION, }; static int __devinit wm8994_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 6ef3f11878c6..c3a42474ab19 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -39,16 +39,6 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micdet_cb cb, void *cb_data); -#define WM8994_CACHE_SIZE 1570 - -struct wm8994_access_mask { - unsigned short readable; /* Mask of readable bits */ - unsigned short writable; /* Mask of writable bits */ -}; - -extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE]; -extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; - int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.1 From 1dfb6efd87d63d2efef6e985770d5dd642f83146 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Nov 2011 17:39:40 +0000 Subject: ASoC: Remove rbtree register cache All users now use regmap directly so delete the ASoC version of the code. Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 381 -------------------------------------------------- 1 file changed, 381 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 18bb6b3335e0..9d56f0218f41 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -66,378 +66,6 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, return -1; } -struct snd_soc_rbtree_node { - struct rb_node node; /* the actual rbtree node holding this block */ - unsigned int base_reg; /* base register handled by this block */ - unsigned int word_size; /* number of bytes needed to represent the register index */ - void *block; /* block of adjacent registers */ - unsigned int blklen; /* number of registers available in the block */ -} __attribute__ ((packed)); - -struct snd_soc_rbtree_ctx { - struct rb_root root; - struct snd_soc_rbtree_node *cached_rbnode; -}; - -static inline void snd_soc_rbtree_get_base_top_reg( - struct snd_soc_rbtree_node *rbnode, - unsigned int *base, unsigned int *top) -{ - *base = rbnode->base_reg; - *top = rbnode->base_reg + rbnode->blklen - 1; -} - -static unsigned int snd_soc_rbtree_get_register( - struct snd_soc_rbtree_node *rbnode, unsigned int idx) -{ - unsigned int val; - - switch (rbnode->word_size) { - case 1: { - u8 *p = rbnode->block; - val = p[idx]; - return val; - } - case 2: { - u16 *p = rbnode->block; - val = p[idx]; - return val; - } - default: - BUG(); - break; - } - return -1; -} - -static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode, - unsigned int idx, unsigned int val) -{ - switch (rbnode->word_size) { - case 1: { - u8 *p = rbnode->block; - p[idx] = val; - break; - } - case 2: { - u16 *p = rbnode->block; - p[idx] = val; - break; - } - default: - BUG(); - break; - } -} - -static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup( - struct rb_root *root, unsigned int reg) -{ - struct rb_node *node; - struct snd_soc_rbtree_node *rbnode; - unsigned int base_reg, top_reg; - - node = root->rb_node; - while (node) { - rbnode = container_of(node, struct snd_soc_rbtree_node, node); - snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); - if (reg >= base_reg && reg <= top_reg) - return rbnode; - else if (reg > top_reg) - node = node->rb_right; - else if (reg < base_reg) - node = node->rb_left; - } - - return NULL; -} - -static int snd_soc_rbtree_insert(struct rb_root *root, - struct snd_soc_rbtree_node *rbnode) -{ - struct rb_node **new, *parent; - struct snd_soc_rbtree_node *rbnode_tmp; - unsigned int base_reg_tmp, top_reg_tmp; - unsigned int base_reg; - - parent = NULL; - new = &root->rb_node; - while (*new) { - rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node, - node); - /* base and top registers of the current rbnode */ - snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp, - &top_reg_tmp); - /* base register of the rbnode to be added */ - base_reg = rbnode->base_reg; - parent = *new; - /* if this register has already been inserted, just return */ - if (base_reg >= base_reg_tmp && - base_reg <= top_reg_tmp) - return 0; - else if (base_reg > top_reg_tmp) - new = &((*new)->rb_right); - else if (base_reg < base_reg_tmp) - new = &((*new)->rb_left); - } - - /* insert the node into the rbtree */ - rb_link_node(&rbnode->node, parent, new); - rb_insert_color(&rbnode->node, root); - - return 1; -} - -static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) -{ - struct snd_soc_rbtree_ctx *rbtree_ctx; - struct rb_node *node; - struct snd_soc_rbtree_node *rbnode; - unsigned int regtmp; - unsigned int val, def; - int ret; - int i; - - rbtree_ctx = codec->reg_cache; - for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) { - rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); - for (i = 0; i < rbnode->blklen; ++i) { - regtmp = rbnode->base_reg + i; - val = snd_soc_rbtree_get_register(rbnode, i); - def = snd_soc_get_cache_val(codec->reg_def_copy, i, - rbnode->word_size); - if (val == def) - continue; - - WARN_ON(!snd_soc_codec_writable_register(codec, regtmp)); - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, regtmp, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - regtmp, val); - } - } - - return 0; -} - -static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode, - unsigned int pos, unsigned int reg, - unsigned int value) -{ - u8 *blk; - - blk = krealloc(rbnode->block, - (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL); - if (!blk) - return -ENOMEM; - - /* insert the register value in the correct place in the rbnode block */ - memmove(blk + (pos + 1) * rbnode->word_size, - blk + pos * rbnode->word_size, - (rbnode->blklen - pos) * rbnode->word_size); - - /* update the rbnode block, its size and the base register */ - rbnode->block = blk; - rbnode->blklen++; - if (!pos) - rbnode->base_reg = reg; - - snd_soc_rbtree_set_register(rbnode, pos, value); - return 0; -} - -static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct snd_soc_rbtree_ctx *rbtree_ctx; - struct snd_soc_rbtree_node *rbnode, *rbnode_tmp; - struct rb_node *node; - unsigned int val; - unsigned int reg_tmp; - unsigned int base_reg, top_reg; - unsigned int pos; - int i; - int ret; - - rbtree_ctx = codec->reg_cache; - /* look up the required register in the cached rbnode */ - rbnode = rbtree_ctx->cached_rbnode; - if (rbnode) { - snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); - if (reg >= base_reg && reg <= top_reg) { - reg_tmp = reg - base_reg; - val = snd_soc_rbtree_get_register(rbnode, reg_tmp); - if (val == value) - return 0; - snd_soc_rbtree_set_register(rbnode, reg_tmp, value); - return 0; - } - } - /* if we can't locate it in the cached rbnode we'll have - * to traverse the rbtree looking for it. - */ - rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); - if (rbnode) { - reg_tmp = reg - rbnode->base_reg; - val = snd_soc_rbtree_get_register(rbnode, reg_tmp); - if (val == value) - return 0; - snd_soc_rbtree_set_register(rbnode, reg_tmp, value); - rbtree_ctx->cached_rbnode = rbnode; - } else { - /* bail out early, no need to create the rbnode yet */ - if (!value) - return 0; - /* look for an adjacent register to the one we are about to add */ - for (node = rb_first(&rbtree_ctx->root); node; - node = rb_next(node)) { - rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node); - for (i = 0; i < rbnode_tmp->blklen; ++i) { - reg_tmp = rbnode_tmp->base_reg + i; - if (abs(reg_tmp - reg) != 1) - continue; - /* decide where in the block to place our register */ - if (reg_tmp + 1 == reg) - pos = i + 1; - else - pos = i; - ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos, - reg, value); - if (ret) - return ret; - rbtree_ctx->cached_rbnode = rbnode_tmp; - return 0; - } - } - /* we did not manage to find a place to insert it in an existing - * block so create a new rbnode with a single register in its block. - * This block will get populated further if any other adjacent - * registers get modified in the future. - */ - rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL); - if (!rbnode) - return -ENOMEM; - rbnode->blklen = 1; - rbnode->base_reg = reg; - rbnode->word_size = codec->driver->reg_word_size; - rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size, - GFP_KERNEL); - if (!rbnode->block) { - kfree(rbnode); - return -ENOMEM; - } - snd_soc_rbtree_set_register(rbnode, 0, value); - snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode); - rbtree_ctx->cached_rbnode = rbnode; - } - - return 0; -} - -static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - struct snd_soc_rbtree_ctx *rbtree_ctx; - struct snd_soc_rbtree_node *rbnode; - unsigned int base_reg, top_reg; - unsigned int reg_tmp; - - rbtree_ctx = codec->reg_cache; - /* look up the required register in the cached rbnode */ - rbnode = rbtree_ctx->cached_rbnode; - if (rbnode) { - snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); - if (reg >= base_reg && reg <= top_reg) { - reg_tmp = reg - base_reg; - *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); - return 0; - } - } - /* if we can't locate it in the cached rbnode we'll have - * to traverse the rbtree looking for it. - */ - rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); - if (rbnode) { - reg_tmp = reg - rbnode->base_reg; - *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); - rbtree_ctx->cached_rbnode = rbnode; - } else { - /* uninitialized registers default to 0 */ - *value = 0; - } - - return 0; -} - -static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) -{ - struct rb_node *next; - struct snd_soc_rbtree_ctx *rbtree_ctx; - struct snd_soc_rbtree_node *rbtree_node; - - /* if we've already been called then just return */ - rbtree_ctx = codec->reg_cache; - if (!rbtree_ctx) - return 0; - - /* free up the rbtree */ - next = rb_first(&rbtree_ctx->root); - while (next) { - rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node); - next = rb_next(&rbtree_node->node); - rb_erase(&rbtree_node->node, &rbtree_ctx->root); - kfree(rbtree_node->block); - kfree(rbtree_node); - } - - /* release the resources */ - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - return 0; -} - -static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) -{ - struct snd_soc_rbtree_ctx *rbtree_ctx; - unsigned int word_size; - unsigned int val; - int i; - int ret; - - codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL); - if (!codec->reg_cache) - return -ENOMEM; - - rbtree_ctx = codec->reg_cache; - rbtree_ctx->root = RB_ROOT; - rbtree_ctx->cached_rbnode = NULL; - - if (!codec->reg_def_copy) - return 0; - - word_size = codec->driver->reg_word_size; - for (i = 0; i < codec->driver->reg_cache_size; ++i) { - val = snd_soc_get_cache_val(codec->reg_def_copy, i, - word_size); - if (!val) - continue; - ret = snd_soc_rbtree_cache_write(codec, i, val); - if (ret) - goto err; - } - - return 0; - -err: - snd_soc_cache_exit(codec); - return ret; -} - static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; @@ -516,15 +144,6 @@ static const struct snd_soc_cache_ops cache_types[] = { .write = snd_soc_flat_cache_write, .sync = snd_soc_flat_cache_sync }, - { - .id = SND_SOC_RBTREE_COMPRESSION, - .name = "rbtree", - .init = snd_soc_rbtree_cache_init, - .exit = snd_soc_rbtree_cache_exit, - .read = snd_soc_rbtree_cache_read, - .write = snd_soc_rbtree_cache_write, - .sync = snd_soc_rbtree_cache_sync - } }; int snd_soc_cache_init(struct snd_soc_codec *codec) -- cgit v1.2.1 From a0e17b4e3e09e49d218958fdce09da407573a574 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 13 Dec 2011 10:13:13 +0800 Subject: ASoC: Staticise rx51_aux_dev Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 4cabb74d97e9..ad16db536320 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -365,7 +365,7 @@ static struct snd_soc_dai_link rx51_dai[] = { }, }; -struct snd_soc_aux_dev rx51_aux_dev[] = { +static struct snd_soc_aux_dev rx51_aux_dev[] = { { .name = "TLV320AIC34b", .codec_name = "tlv320aic3x-codec.2-0019", -- cgit v1.2.1 From bba59f332687884e98c920e6c27278824d194c24 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 13 Dec 2011 10:14:30 +0800 Subject: ASoC: Staticise au1xpsc_soc_platform Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 09699de9b337..92bc1b0346fa 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -341,7 +341,7 @@ static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) } /* au1xpsc audio platform */ -struct snd_soc_platform_driver au1xpsc_soc_platform = { +static struct snd_soc_platform_driver au1xpsc_soc_platform = { .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, -- cgit v1.2.1 From 796ba9001dfd5cdf19926f374015e514ea2eaa51 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 13 Dec 2011 10:15:40 +0800 Subject: ASoC: Staticise alchemy_pcm_soc_platform Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/au1x/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index dc4dae48aed9..c4017bd56ab8 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -316,7 +316,7 @@ static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -struct snd_soc_platform_driver alchemy_pcm_soc_platform = { +static struct snd_soc_platform_driver alchemy_pcm_soc_platform = { .ops = &alchemy_pcm_ops, .pcm_new = alchemy_pcm_new, .pcm_free = alchemy_pcm_free_dma_buffers, -- cgit v1.2.1 From 9215aa4d96add60e95adccbcb11b1dc16a8c3422 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Mon, 12 Dec 2011 21:43:45 +0200 Subject: ASoC: Rename ALC5632 MICBIAS to common name convention. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 08613c7e1091..390e437d7c5e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -379,8 +379,8 @@ SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -- cgit v1.2.1 From 63a9332b232bdab0df6ef18a9f39e8d58a82bda4 Mon Sep 17 00:00:00 2001 From: Andrew Lunn Date: Wed, 7 Dec 2011 21:48:07 +0100 Subject: ARM: Orion: Get address map from plat-orion instead of via platform_data Use an getter function in plat-orion/addr-map.c to get the address map structure, rather than pass it to drivers in the platform_data structures. When the drivers are built for none orion platforms, a dummy function is provided instead which returns NULL. Signed-off-by: Andrew Lunn Tested-by: Michael Walle Acked-by: Nicolas Pitre Signed-off-by: Nicolas Pitre --- sound/soc/kirkwood/kirkwood-dma.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cd33de1c5b7a..df12e0993f5a 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -94,9 +94,10 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) return IRQ_HANDLED; } -static void kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, - unsigned long dma, - struct mbus_dram_target_info *dram) +static void +kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, + unsigned long dma, + const struct mbus_dram_target_info *dram) { int i; @@ -106,7 +107,7 @@ static void kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = dram->cs + i; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); @@ -127,6 +128,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; struct kirkwood_dma_data *priv; struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + const struct mbus_dram_target_info *dram; unsigned long addr; priv = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -175,15 +177,16 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); } + dram = mv_mbus_dram_info(); addr = virt_to_phys(substream->dma_buffer.area); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { prdata->play_stream = substream; kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_PLAYBACK_WIN, addr, priv->dram); + KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { prdata->rec_stream = substream; kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_RECORD_WIN, addr, priv->dram); + KIRKWOOD_RECORD_WIN, addr, dram); } return 0; -- cgit v1.2.1 From 6f8d272a440ca7c61f5727300cbae642759d6765 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 11:20:42 +0800 Subject: ASoC: Fix wm8995 regmap usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 5863406b459d..c8aada597d70 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2051,6 +2051,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; + codec->control_data = wm8995->regmap; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); -- cgit v1.2.1 From 8858d21891ad6aecced34c31ae961584ad418522 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Dec 2011 22:47:25 +0800 Subject: ASoC: Staticise asoc_idma_platform Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/idma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index baf97ebadd48..2bcf75815624 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -410,7 +410,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.lp_tx_addr = addr; } -struct snd_soc_platform_driver asoc_idma_platform = { +static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, .pcm_new = idma_new, .pcm_free = idma_free, -- cgit v1.2.1 From 0c9f110574bdde21ac62b948272a90f6e72b94d8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Dec 2011 19:05:58 +0800 Subject: ASoC: Complete initialisation before registering Samsung PCM DAI Otherwise there's a race where the DAI might get used without everything having been set up. Signed-off-by: Mark Brown Acked-by: Sangbeom Kim --- sound/soc/samsung/pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index beef63fca052..3a29c268ea5d 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -570,12 +570,6 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) } clk_enable(pcm->pclk); - ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + S3C_PCM_RXFIFO; s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start @@ -587,6 +581,12 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); + goto err5; + } + return 0; err5: -- cgit v1.2.1 From 92f6d63bf1366a9dae18965cdb4a950b5ece3a64 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 13 Dec 2011 20:23:50 +0800 Subject: ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform Both sh4_hac_dai and sh7760_soc_platform are changed to static by multi-component patch and they are not used in sh7760-ac97.c now. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/sh/sh7760-ac97.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index c62ae689c4a1..df651e8e38de 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -16,10 +16,6 @@ #define IPSEL 0xFE400034 -/* platform specific structs can be declared here */ -extern struct snd_soc_dai_driver sh4_hac_dai[2]; -extern struct snd_soc_platform_driver sh7760_soc_platform; - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", -- cgit v1.2.1 From 8faa8c1a5520c1e21372f2016355f4b5c2349bb2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 13 Dec 2011 17:13:45 +0800 Subject: ASoC: Staticise sst_pcm_new and sst_soc_platform_drv Signed-off-by: Axel Lin Acked-by Vinod Koul Acked-by: Lu Guanqun Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f947146947..5b936d5029fe 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -443,7 +443,7 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; @@ -463,7 +463,7 @@ int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) } return retval; } -struct snd_soc_platform_driver sst_soc_platform_drv = { +static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, -- cgit v1.2.1 From 42f3b0109ea61aee0541a02f1802fd7939b9853a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Dec 2011 18:30:03 +0800 Subject: ASoC: Remove cache default for volatile wm9081 reset register Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 1f2672b1e03e..a6bab392700e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -31,7 +31,6 @@ #include "wm9081.h" static struct reg_default wm9081_reg[] = { - { 0, 0x9081 }, /* R0 - Software Reset */ { 2, 0x00B9 }, /* R2 - Analogue Lineout */ { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ { 4, 0x0001 }, /* R4 - VMID Control */ -- cgit v1.2.1 From ffbf2a363e1867ba5f5869236dda944ec12fe99b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Dec 2011 21:04:26 +0800 Subject: ASoC: Use standard snd_soc_cache_sync() for WM9090 Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index d1d2c703eab2..41ebe0dce772 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -513,18 +513,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Restore the register cache */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (reg_cache[i] == wm9090_reg_defaults[i]) - continue; - if (wm9090_volatile(codec, i)) - continue; - - ret = snd_soc_write(codec, i, reg_cache[i]); - if (ret != 0) - dev_warn(codec->dev, - "Failed to restore register %d: %d\n", - i, ret); - } + snd_soc_cache_sync(codec); } /* We keep VMID off during standby since the combination of -- cgit v1.2.1 From f6a9336879caeed63e77bc10097966fa3a6ba20c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 11:11:52 +0800 Subject: ASoC: Convert wm8993 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f472ea6ecf6b..b966f6979ade 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1613,7 +1613,8 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, struct wm8993_priv *wm8993; int ret; - wm8993 = kzalloc(sizeof(struct wm8993_priv), GFP_KERNEL); + wm8993 = devm_kzalloc(&i2c-dev, sizeof(struct wm8993_priv), + GFP_KERNEL); if (wm8993 == NULL) return -ENOMEM; @@ -1621,8 +1622,6 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8993, &wm8993_dai, 1); - if (ret < 0) - kfree(wm8993); return ret; } -- cgit v1.2.1 From d0616bbed18884cb2475ca0abb5a596105444b96 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 11:40:59 +0800 Subject: ASoC: Use standard register cache sync in wm8993 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index b966f6979ade..2835e7d5d60b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -934,28 +934,6 @@ static const struct snd_soc_dapm_route routes[] = { { "Right Headphone Mux", "DAC", "DACR" }, }; -static void wm8993_cache_restore(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - /* Reenable hardware writes */ - codec->cache_only = 0; - - /* Restore the register settings */ - for (i = 1; i < WM8993_MAX_REGISTER; i++) { - if (cache[i] == wm8993_reg_defaults[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } - - /* We're in sync again */ - codec->cache_sync = 0; -} - static int wm8993_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -979,7 +957,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, if (ret != 0) return ret; - wm8993_cache_restore(codec); + snd_soc_cache_sync(codec); /* Tune DC servo configuration */ snd_soc_write(codec, 0x44, 3); -- cgit v1.2.1 From 45ba82d81741398ec4f097fedf2c204704d53b6b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 19:23:37 +0800 Subject: ASoC: Tune the accessory detection rates for WM8996 Use longer intervals when the microphone is not inserted to increase robustness against leisurely insertion. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8f88f5a9c985..da7acaefa9d9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2572,8 +2572,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) SND_JACK_BTN_0); snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - WM8996_MICD_RATE_MASK); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + WM8996_MICD_RATE_MASK | + 9 << WM8996_MICD_BIAS_STARTTIME_SHIFT); return; } @@ -2590,8 +2592,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 5 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } else { dev_dbg(codec->dev, "Mic button up\n"); snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); @@ -2639,8 +2643,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) * responsiveness. */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 7 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } } } -- cgit v1.2.1 From 20e757f79b9956a91f4ba0f33cc3f34efe6eb188 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:21:32 +0000 Subject: ASoC: Use core pm_runtime callbacks for siu_dai Now that the core holds a pm_runtime reference to the device while the link is active there is no need for the driver to do so. Signed-off-by: Mark Brown --- sound/soc/sh/siu_dai.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 11c608570820..52d4c17b1232 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -112,9 +112,6 @@ static void siu_dai_start(struct siu_port *port_info) dev_dbg(port_info->pcm->card->dev, "%s\n", __func__); - /* Turn on SIU clock */ - pm_runtime_get_sync(info->dev); - /* Issue software reset to siu */ siu_write32(base + SIU_SRCTL, 0); @@ -158,9 +155,6 @@ static void siu_dai_stop(struct siu_port *port_info) /* SIU software reset */ siu_write32(base + SIU_SRCTL, 0); - - /* Turn off SIU clock */ - pm_runtime_put_sync(info->dev); } static void siu_dai_spbAselect(struct siu_port *port_info) -- cgit v1.2.1 From 27f478a65ff7b67b843250f0a2d1e8b306bf57b6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 20:22:18 +0000 Subject: ASoC: Use core pm_runtime callbacks for fsi Now that the core holds a pm_runtime reference to the device while the link is active there is no need for the driver to do so. Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a27c30636b82..db6c89a28bda 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -893,8 +893,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, u32 flags = fsi_get_info_flags(fsi); u32 data = 0; - pm_runtime_get_sync(dev); - /* clock setting */ if (fsi_is_clk_master(fsi)) data = DIMD | DOMD; @@ -951,8 +949,6 @@ static void fsi_hw_shutdown(struct fsi_priv *fsi, { if (fsi_is_clk_master(fsi)) fsi_set_master_clk(dev, fsi, fsi->rate, 0); - - pm_runtime_put_sync(dev); } static int fsi_dai_startup(struct snd_pcm_substream *substream, -- cgit v1.2.1 From ec641c459048f30e378e1386ba9eff1f7ada522f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Dec 2011 11:54:00 +0800 Subject: ASoC: Fix partial cherry pick in wm8993 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 2835e7d5d60b..2b40c93601ed 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1591,7 +1591,7 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, struct wm8993_priv *wm8993; int ret; - wm8993 = devm_kzalloc(&i2c-dev, sizeof(struct wm8993_priv), + wm8993 = devm_kzalloc(&i2c->dev, sizeof(struct wm8993_priv), GFP_KERNEL); if (wm8993 == NULL) return -ENOMEM; -- cgit v1.2.1 From 00c651612ef2b4b6b953059ace6b50afcb8d88c4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 14 Dec 2011 19:23:01 +0800 Subject: ASoC: Staticise mfld_msic_dailink Signed-off-by: Axel Lin Acked by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index e53f8e473a78..8ae057433968 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -281,7 +281,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } -struct snd_soc_dai_link mfld_msic_dailink[] = { +static struct snd_soc_dai_link mfld_msic_dailink[] = { { .name = "Medfield Headset", .stream_name = "Headset", -- cgit v1.2.1 From c45471eac2bdc271df40963ac8448d76ac434872 Mon Sep 17 00:00:00 2001 From: Joerg Roedel Date: Thu, 15 Dec 2011 18:24:54 +0100 Subject: ASoC: Fix compile error in sound/soc/mid-x86/sst_platform.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The variable ret_val is used but not declared. This causes the following compile error: sound/soc/mid-x86/sst_platform.c: In function ‘sst_platform_open’: sound/soc/mid-x86/sst_platform.c:274:2: error: ‘ret_val’ undeclared (first use in this function) sound/soc/mid-x86/sst_platform.c:274:2: note: each undeclared identifier is reported only once for each function it appears in make[1]: *** [sound/soc/mid-x86/sst_platform.o] Error 1 Fix this. Signed-off-by: Joerg Roedel Signed-off-by: Takashi Iwai --- sound/soc/mid-x86/sst_platform.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f947146947..11c39c548a06 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -267,6 +267,7 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; + int ret_val; pr_debug("sst_platform_open called\n"); -- cgit v1.2.1 From 7243a4b1eb377b3c5376cbf0ebdf87972b969127 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Dec 2011 11:32:27 +0200 Subject: ASoC: omap-mcbsp: Enable FIFO usage on OMAP4 Allow McBSP FIFO configuration from ASoC dai driver on OMAP4 platform. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index bd11d2568584..017371913ec3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap34xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == -- cgit v1.2.1 From 62133829fa12a55902ac400b74e424c1ecd161b3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 10:52:42 +0800 Subject: ASoC: pxa: Convert e740_wm9705 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e740_wm9705.c | 75 +++++++++++++++++++-------------------------- 1 file changed, 32 insertions(+), 43 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 818dc57b0b2f..203ab78a4d68 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -137,66 +137,55 @@ static struct snd_soc_card e740 = { .num_links = ARRAY_SIZE(e740_dai), }; -static struct platform_device *e740_snd_device; +static struct gpio e740_audio_gpios[] = { + { GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp" }, + { GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, "Output amp" }, + { GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, "Audio power" }, +}; -static int __init e740_init(void) +static int __devinit e740_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &e740; int ret; - if (!machine_is_e740()) - return -ENODEV; - - /* Disable audio */ - ret = gpio_request_one(GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp"); + ret = gpio_request_array(e740_audio_gpios, + ARRAY_SIZE(e740_audio_gpios)); if (ret) return ret; - ret = gpio_request_one(GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, - "Output amp"); - if (ret) - goto free_mic_amp_gpio; + card->dev = &pdev->dev; - ret = gpio_request_one(GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, - "Audio power"); - if (ret) - goto free_op_amp_gpio; - - e740_snd_device = platform_device_alloc("soc-audio", -1); - if (!e740_snd_device) { - ret = -ENOMEM; - goto free_apwr_gpio; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); } - - platform_set_drvdata(e740_snd_device, &e740); - ret = platform_device_add(e740_snd_device); - - if (!ret) - return 0; - -/* Fail gracefully */ - platform_device_put(e740_snd_device); -free_apwr_gpio: - gpio_free(GPIO_E740_WM9705_nAVDD2); -free_op_amp_gpio: - gpio_free(GPIO_E740_AMP_ON); -free_mic_amp_gpio: - gpio_free(GPIO_E740_MIC_ON); - return ret; } -static void __exit e740_exit(void) +static int __devexit e740_remove(struct platform_device *pdev) { - platform_device_unregister(e740_snd_device); - gpio_free(GPIO_E740_WM9705_nAVDD2); - gpio_free(GPIO_E740_AMP_ON); - gpio_free(GPIO_E740_MIC_ON); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); + snd_soc_unregister_card(card); + return 0; } -module_init(e740_init); -module_exit(e740_exit); +static struct platform_driver e740_driver = { + .driver = { + .name = "e740-audio", + .owner = THIS_MODULE, + }, + .probe = e740_probe, + .remove = __devexit_p(e740_remove), +}; + +module_platform_driver(e740_driver); /* Module information */ MODULE_AUTHOR("Ian Molton "); MODULE_DESCRIPTION("ALSA SoC driver for e740"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:e740-audio"); -- cgit v1.2.1 From 5eb2c3d9273ae63d6b347cde38fe15bda8be1361 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 10:53:29 +0800 Subject: ASoC: pxa: Convert e750_wm9705 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e750_wm9705.c | 66 +++++++++++++++++++++------------------------ 1 file changed, 31 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 55c53d13bea6..27f90cc44234 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -120,58 +120,54 @@ static struct snd_soc_card e750 = { .num_links = ARRAY_SIZE(e750_dai), }; -static struct platform_device *e750_snd_device; +static struct gpio e750_audio_gpios[] = { + { GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Headphone amp" }, + { GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" }, +}; -static int __init e750_init(void) +static int __devinit e750_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &e750; int ret; - if (!machine_is_e750()) - return -ENODEV; - - ret = gpio_request_one(GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, - "Headphone amp"); + ret = gpio_request_array(e750_audio_gpios, + ARRAY_SIZE(e750_audio_gpios)); if (ret) return ret; - ret = gpio_request_one(GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, - "Speaker amp"); - if (ret) - goto free_hp_amp_gpio; + card->dev = &pdev->dev; - e750_snd_device = platform_device_alloc("soc-audio", -1); - if (!e750_snd_device) { - ret = -ENOMEM; - goto free_spk_amp_gpio; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); } - - platform_set_drvdata(e750_snd_device, &e750); - ret = platform_device_add(e750_snd_device); - - if (!ret) - return 0; - -/* Fail gracefully */ - platform_device_put(e750_snd_device); -free_spk_amp_gpio: - gpio_free(GPIO_E750_SPK_AMP_OFF); -free_hp_amp_gpio: - gpio_free(GPIO_E750_HP_AMP_OFF); - return ret; } -static void __exit e750_exit(void) +static int __devexit e750_remove(struct platform_device *pdev) { - platform_device_unregister(e750_snd_device); - gpio_free(GPIO_E750_SPK_AMP_OFF); - gpio_free(GPIO_E750_HP_AMP_OFF); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); + snd_soc_unregister_card(card); + return 0; } -module_init(e750_init); -module_exit(e750_exit); +static struct platform_driver e750_driver = { + .driver = { + .name = "e750-audio", + .owner = THIS_MODULE, + }, + .probe = e750_probe, + .remove = __devexit_p(e750_remove), +}; + +module_platform_driver(e750_driver); /* Module information */ MODULE_AUTHOR("Ian Molton "); MODULE_DESCRIPTION("ALSA SoC driver for e750"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:e750-audio"); -- cgit v1.2.1 From ac1e89860a89c9d91174bf5439689bba2e4f83bb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 10:55:24 +0800 Subject: ASoC: pxa: Convert imote2 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/imote2.c | 41 ++++++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 154fc6f23438..97d3aecfc203 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -70,39 +70,46 @@ static struct snd_soc_dai_link imote2_dai = { .ops = &imote2_asoc_ops, }; -static struct snd_soc_card snd_soc_imote2 = { +static struct snd_soc_card imote2 = { .name = "Imote2", .dai_link = &imote2_dai, .num_links = 1, }; -static struct platform_device *imote2_snd_device; - -static int __init imote2_asoc_init(void) +static int __devinit imote2_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &imote2; int ret; - if (!machine_is_intelmote2()) - return -ENODEV; - imote2_snd_device = platform_device_alloc("soc-audio", -1); - if (!imote2_snd_device) - return -ENOMEM; + card->dev = &pdev->dev; - platform_set_drvdata(imote2_snd_device, &snd_soc_imote2); - ret = platform_device_add(imote2_snd_device); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(imote2_snd_device); - + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -module_init(imote2_asoc_init); -static void __exit imote2_asoc_exit(void) +static int __devexit imote2_remove(struct platform_device *pdev) { - platform_device_unregister(imote2_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_exit(imote2_asoc_exit); + +static struct platform_driver imote2_driver = { + .driver = { + .name = "imote2-audio", + .owner = THIS_MODULE, + }, + .probe = imote2_probe, + .remove = __devexit_p(imote2_remove), +}; + +module_platform_driver(imote2_driver); MODULE_AUTHOR("Jonathan Cameron"); MODULE_DESCRIPTION("ALSA SoC Imote 2"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imote2-audio"); -- cgit v1.2.1 From f285b8c83a8dccc70f168bb1eb6f04c8e36450a6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 10:57:22 +0800 Subject: ASoC: pxa: Convert tosa to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 77 ++++++++++++++++++++-------------------------------- 1 file changed, 30 insertions(+), 47 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 620fc69ae632..3f394de297a2 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -34,8 +34,6 @@ #include "../codecs/wm9712.h" #include "pxa2xx-ac97.h" -static struct snd_soc_card tosa; - #define TOSA_HP 0 #define TOSA_MIC_INT 1 #define TOSA_HEADSET 2 @@ -236,70 +234,55 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; -static int tosa_probe(struct snd_soc_card *card) -{ - int ret; - - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - if (ret) - gpio_free(TOSA_GPIO_L_MUTE); - - return ret; -} - -static int tosa_remove(struct snd_soc_card *card) -{ - gpio_free(TOSA_GPIO_L_MUTE); - return 0; -} - static struct snd_soc_card tosa = { .name = "Tosa", .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), - .probe = tosa_probe, - .remove = tosa_remove, }; -static struct platform_device *tosa_snd_device; - -static int __init tosa_init(void) +static int __devinit tosa_probe(struct platform_device *pdev) { + struct snd_soc_card *card = ⤩ int ret; - if (!machine_is_tosa()) - return -ENODEV; - - tosa_snd_device = platform_device_alloc("soc-audio", -1); - if (!tosa_snd_device) { - ret = -ENOMEM; - goto err_alloc; - } - - platform_set_drvdata(tosa_snd_device, &tosa); - ret = platform_device_add(tosa_snd_device); - - if (!ret) - return 0; + ret = gpio_request_one(TOSA_GPIO_L_MUTE, GPIOF_OUT_INIT_LOW, + "Headphone Jack"); + if (ret) + return ret; - platform_device_put(tosa_snd_device); + card->dev = &pdev->dev; -err_alloc: + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free(TOSA_GPIO_L_MUTE); + } return ret; } -static void __exit tosa_exit(void) +static int __devexit tosa_remove(struct platform_device *pdev) { - platform_device_unregister(tosa_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + gpio_free(TOSA_GPIO_L_MUTE); + snd_soc_unregister_card(card); + return 0; } -module_init(tosa_init); -module_exit(tosa_exit); +static struct platform_driver tosa_driver = { + .driver = { + .name = "tosa-audio", + .owner = THIS_MODULE, + }, + .probe = tosa_probe, + .remove = __devexit_p(tosa_remove), +}; + +module_platform_driver(tosa_driver); /* Module information */ MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Tosa"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:tosa-audio"); -- cgit v1.2.1 From 1eb0202dc7e45be5996416bc41489ae5a75485e5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 10:54:25 +0800 Subject: ASoC: pxa: Convert e800_wm9712 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/e800_wm9712.c | 66 +++++++++++++++++++++------------------------ 1 file changed, 31 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 478ff191ffb4..858bf94160c5 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -110,58 +110,54 @@ static struct snd_soc_card e800 = { .num_links = ARRAY_SIZE(e800_dai), }; -static struct platform_device *e800_snd_device; +static struct gpio e800_audio_gpios[] = { + { GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, "Headphone amp" }, + { GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" }, +}; -static int __init e800_init(void) +static int __devinit e800_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &e800; int ret; - if (!machine_is_e800()) - return -ENODEV; - - ret = gpio_request_one(GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, - "Headphone amp"); + ret = gpio_request_array(e800_audio_gpios, + ARRAY_SIZE(e800_audio_gpios)); if (ret) return ret; - ret = gpio_request_one(GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, - "Speaker amp"); - if (ret) - goto free_hp_amp_gpio; + card->dev = &pdev->dev; - e800_snd_device = platform_device_alloc("soc-audio", -1); - if (!e800_snd_device) { - ret = -ENOMEM; - goto free_spk_amp_gpio; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); } - - platform_set_drvdata(e800_snd_device, &e800); - ret = platform_device_add(e800_snd_device); - - if (!ret) - return 0; - -/* Fail gracefully */ - platform_device_put(e800_snd_device); -free_spk_amp_gpio: - gpio_free(GPIO_E800_SPK_AMP_ON); -free_hp_amp_gpio: - gpio_free(GPIO_E800_HP_AMP_OFF); - return ret; } -static void __exit e800_exit(void) +static int __devexit e800_remove(struct platform_device *pdev) { - platform_device_unregister(e800_snd_device); - gpio_free(GPIO_E800_SPK_AMP_ON); - gpio_free(GPIO_E800_HP_AMP_OFF); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); + snd_soc_unregister_card(card); + return 0; } -module_init(e800_init); -module_exit(e800_exit); +static struct platform_driver e800_driver = { + .driver = { + .name = "e800-audio", + .owner = THIS_MODULE, + }, + .probe = e800_probe, + .remove = __devexit_p(e800_remove), +}; + +module_platform_driver(e800_driver); /* Module information */ MODULE_AUTHOR("Ian Molton "); MODULE_DESCRIPTION("ALSA SoC driver for e800"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:e800-audio"); -- cgit v1.2.1 From b9791c010966207a9f111e9339d6087a1a7269ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Dec 2011 07:42:58 +0100 Subject: ASoC: Convert WM8960 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 2315b866d002..e5caae32e541 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -973,7 +973,8 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, struct wm8960_priv *wm8960; int ret; - wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL); + wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv), + GFP_KERNEL); if (wm8960 == NULL) return -ENOMEM; @@ -982,15 +983,13 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); - if (ret < 0) - kfree(wm8960); + return ret; } static __devexit int wm8960_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 69be6660f30b79410111f4e7c55307d775cfb274 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Dec 2011 07:55:02 +0100 Subject: ASoC: Remove I2C ifdefs from wm8961 driver The driver only supports I2C so no need to conditionalise its use. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 58fbf0a87b6a..13a085040900 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1081,7 +1081,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .volatile_register = wm8961_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1123,27 +1122,22 @@ static struct i2c_driver wm8961_i2c_driver = { .remove = __devexit_p(wm8961_i2c_remove), .id_table = wm8961_i2c_id, }; -#endif static int __init wm8961_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8961_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8961 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8961_modinit); static void __exit wm8961_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8961_i2c_driver); -#endif } module_exit(wm8961_exit); -- cgit v1.2.1 From 2ec2a9061dac94ca4c5af13566fe107d84c30d4e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Dec 2011 07:56:02 +0100 Subject: ASoC: Convert wm8961 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 13a085040900..8bcc17a61329 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1087,7 +1087,8 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, struct wm8961_priv *wm8961; int ret; - wm8961 = kzalloc(sizeof(struct wm8961_priv), GFP_KERNEL); + wm8961 = devm_kzalloc(&i2c->dev, sizeof(struct wm8961_priv), + GFP_KERNEL); if (wm8961 == NULL) return -ENOMEM; @@ -1095,15 +1096,14 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8961, &wm8961_dai, 1); - if (ret < 0) - kfree(wm8961); + return ret; } static __devexit int wm8961_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.1 From 202a51a8d9c1fddea9eca5953e6e7d7d504a4343 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Dec 2011 07:57:11 +0100 Subject: ASoC: Use standard cache sync code in wm8961 We write the reset register with the default value so it should not be mistakenly written. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 8bcc17a61329..4f20c72a0f1d 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1047,18 +1047,7 @@ static int wm8961_suspend(struct snd_soc_codec *codec) static int wm8961_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (reg_cache[i] == wm8961_reg_defaults[i]) - continue; - - if (i == WM8961_SOFTWARE_RESET) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } + snd_soc_cache_sync(codec); wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.1 From 5ee65ec628090a3dbfbd900e4174f56e92e70945 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 19 Dec 2011 13:13:31 +0800 Subject: ASoC: Convert max9850 to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 7dfd6e84796d..47060d2afe90 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -86,7 +86,7 @@ SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route max9850_dapm_routes[] = { /* output mixer */ {"Output Mixer", NULL, "DAC"}, {"Output Mixer", "Line In Switch", "Line Input"}, @@ -293,7 +293,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -309,13 +308,6 @@ static int max9850_probe(struct snd_soc_codec *codec) /* set slew-rate 125ms */ snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); - snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, - ARRAY_SIZE(max9850_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - snd_soc_add_controls(codec, max9850_controls, - ARRAY_SIZE(max9850_controls)); - return 0; } @@ -328,6 +320,13 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .reg_word_size = sizeof(u8), .reg_cache_default = max9850_reg, .volatile_register = max9850_volatile_register, + + .controls = max9850_controls, + .num_controls = ARRAY_SIZE(max9850_controls), + .dapm_widgets = max9850_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9850_dapm_widgets), + .dapm_routes = max9850_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9850_dapm_routes), }; static int __devinit max9850_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.1 From 58fa8e456c8e8329fe829994202b43286eb0de3f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 19 Dec 2011 13:54:38 +0800 Subject: ASoC: Convert uda1380 to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 83e45d2b3e84..8f734d69f651 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -373,7 +373,7 @@ static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route uda1380_dapm_routes[] = { /* output mux */ {"HeadPhone Driver", NULL, "Output Mux"}, @@ -410,17 +410,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right PGA", NULL, "VINR"}, }; -static int uda1380_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -764,10 +753,6 @@ static int uda1380_probe(struct snd_soc_codec *codec) break; } - snd_soc_add_controls(codec, uda1380_snd_controls, - ARRAY_SIZE(uda1380_snd_controls)); - uda1380_add_widgets(codec); - return 0; err_free_gpio: @@ -802,6 +787,13 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .reg_word_size = sizeof(u16), .reg_cache_default = uda1380_reg, .reg_cache_step = 1, + + .controls = uda1380_snd_controls, + .num_controls = ARRAY_SIZE(uda1380_snd_controls), + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = uda1380_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.1 From 58783faf281559379871d85faf2ef53e97d075e0 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Mon, 19 Dec 2011 21:51:52 +0200 Subject: ASoC: Tegra machine ASoC driver for boards using ALC5332 codec At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky Signed-off-by: Andrey Danin Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 213 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 224 insertions(+) create mode 100644 sound/soc/tegra/tegra_alc5632.c (limited to 'sound/soc') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd707f5..ce1b773c351f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3fe150..8e584b8fcfba 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 000000000000..9287eb8028fd --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,213 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team +* +* Authors: Leon Romanovsky +* Andrey Danin +* Marc Dietrich +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +*/ + +#include + +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 512 * srate; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, + .controls = tegra_alc5632_controls, + .num_controls = ARRAY_SIZE(tegra_alc5632_controls), + .dapm_widgets = tegra_alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), + .dapm_routes = tegra_alc5632_audio_map, + .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), + .fully_routed = true, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + alc5632 = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + tegra_asoc_utils_fini(&alc5632->util_data); + return ret; + } + + return 0; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&alc5632->util_data); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; +module_platform_driver(tegra_alc5632_driver); + +MODULE_AUTHOR("Leon Romanovsky "); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.1 From 5ab2ab6a432e93f083db3904f9bcbaa97d4b1d35 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 14 Dec 2011 19:13:26 +0800 Subject: ASoC: Remove export of s3c_pcm_dai We don't need to export s3c_pcm_dai after multi-component patch. Thus remove export of s3c_pcm_dai and make it static. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 3a29c268ea5d..5776addd1f94 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -478,7 +478,7 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .formats = SNDRV_PCM_FMTBIT_S16_LE, \ } -struct snd_soc_dai_driver s3c_pcm_dai[] = { +static struct snd_soc_dai_driver s3c_pcm_dai[] = { [0] = { .name = "samsung-pcm.0", S3C_PCM_DAI_DECLARE, @@ -488,7 +488,6 @@ struct snd_soc_dai_driver s3c_pcm_dai[] = { S3C_PCM_DAI_DECLARE, }, }; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) { -- cgit v1.2.1 From dbec3b30a601791717bc5bb827e210c3b5d6e067 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 14 Dec 2011 15:47:46 +0800 Subject: ASoC: mxs: correct 'direction' of device_prep_dma_cyclic The commit 49920bc (dmaengine: add new enum dma_transfer_direction) changes the type of parameter 'direction' of device_prep_dma_cyclic from dma_data_direction to dma_transfer_direction. Signed-off-by: Shawn Guo Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index f39d7dd9fbcb..5dfd3250ddf1 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_TO_DEVICE : DMA_FROM_DEVICE); + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); if (!iprtd->desc) { dev_err(&chan->dev->device, "cannot prepare slave dma\n"); return -EINVAL; -- cgit v1.2.1 From f521812b03eb906ece7433ffd681fd5a35c0aced Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 17 Dec 2011 15:36:52 +0800 Subject: ASoC: Use dai_fmt in edb93xx machine driver Signed-off-by: Axel Lin Acked-by: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 6b90c757cf4c..9f6fecdf49e7 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -48,18 +48,6 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, else mclk_rate = rate * 64 * 2; - err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - - err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, SND_SOC_CLOCK_IN); if (err) @@ -80,6 +68,8 @@ static struct snd_soc_dai_link edb93xx_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &edb93xx_ops, }; -- cgit v1.2.1 From f49f85108b2b3b1aaa7632418411c401fbb6741c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 17 Dec 2011 15:41:11 +0800 Subject: ASoC: Use dai_fmt in snappercl15 machine driver Signed-off-by: Axel Lin Reviewed-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/snappercl15.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index 33901d647b72..e97cd5701f51 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -33,16 +33,6 @@ static int snappercl15_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - - err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (err) @@ -96,6 +86,8 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-pcm-audio", .init = snappercl15_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, }; -- cgit v1.2.1 From 6048ef768e7bec7e1e17f48fe8d5360021928b4a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Dec 2011 11:57:57 +0800 Subject: ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls The module desciption says this is ASoC alc5621/2/3 driver. Make the naming consistent with the reset of the code. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index da97f024ec74..6a9b621ef32d 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -99,7 +99,7 @@ static const unsigned int boost_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); -static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -110,7 +110,7 @@ static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; -static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -925,12 +925,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, rt5621_vol_snd_controls, - ARRAY_SIZE(rt5621_vol_snd_controls)); + snd_soc_add_controls(codec, alc5621_vol_snd_controls, + ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, rt5622_vol_snd_controls, - ARRAY_SIZE(rt5622_vol_snd_controls)); + snd_soc_add_controls(codec, alc5622_vol_snd_controls, + ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: snd_soc_add_controls(codec, alc5623_vol_snd_controls, -- cgit v1.2.1 From bec4fa05e25f7e78ec67df389539acc6bb352a2a Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 12 Dec 2011 15:55:34 -0700 Subject: ASoC: Add utility to set a card's name from device tree Implement snd_soc_of_parse_card_name(), a utility function that sets a card's name from device tree. The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1252ab1ebf69..51eef9b7b53f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include @@ -3317,6 +3318,30 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); +/* Retrieve a card's name from device tree */ +int snd_soc_of_parse_card_name(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + int ret; + + ret = of_property_read_string_index(np, propname, 0, &card->name); + /* + * EINVAL means the property does not exist. This is fine providing + * card->name was previously set, which is checked later in + * snd_soc_register_card. + */ + if (ret < 0 && ret != -EINVAL) { + dev_err(card->dev, + "Property '%s' could not be read: %d\n", + propname, ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit v1.2.1 From a4a54dd5bb1bb01010f46147d6d8b452255957bf Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 12 Dec 2011 15:55:35 -0700 Subject: ASoC: Add utility to parse DAPM routes from device tree Implement snd_soc_of_parse_audio_routing(), a utility function that can parses a simple DAPM route table from device tree.The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 57 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 51eef9b7b53f..42ad2db8f082 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3342,6 +3342,63 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); +int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + int num_routes; + struct snd_soc_dapm_route *routes; + int i, ret; + + num_routes = of_property_count_strings(np, propname); + if (num_routes & 1) { + dev_err(card->dev, + "Property '%s's length is not even\n", + propname); + return -EINVAL; + } + num_routes /= 2; + if (!num_routes) { + dev_err(card->dev, + "Property '%s's length is zero\n", + propname); + return -EINVAL; + } + + routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + GFP_KERNEL); + if (!routes) { + dev_err(card->dev, + "Could not allocate DAPM route table\n"); + return -EINVAL; + } + + for (i = 0; i < num_routes; i++) { + ret = of_property_read_string_index(np, propname, + 2 * i, &routes[i].sink); + if (ret) { + dev_err(card->dev, + "Property '%s' index %d could not be read: %d\n", + propname, 2 * i, ret); + return -EINVAL; + } + ret = of_property_read_string_index(np, propname, + (2 * i) + 1, &routes[i].source); + if (ret) { + dev_err(card->dev, + "Property '%s' index %d could not be read: %d\n", + propname, (2 * i) + 1, ret); + return -EINVAL; + } + } + + card->num_dapm_routes = num_routes; + card->dapm_routes = routes; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit v1.2.1 From 07cdf36d8c4ba4ad0db13228eb25bcd3d5138b29 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 12 Dec 2011 15:55:36 -0700 Subject: ASoC: Tegra+WM8903 machine: Add device tree binding This driver is parameterized in two ways: a) Platform data, which supplies the set of GPIOs used by the driver. These GPIOs can now be parsed out of device tree. b) Machine-specific DAPM route arrays embedded into the ASoC machine driver itself. Historically, the driver picks the appropriate array to use using machine_is_*(). The driver now requires this array to be parsed from device tree when instantiated through device tree, using the core ASoC support for this parsing. Based on work by John Bonesio, but significantly reworked since then. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 128 +++++++++++++++++++++++++++++++++-------- 1 file changed, 103 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index ba2d23ea6424..4677f2666300 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,6 +34,7 @@ #include #include #include +#include #include @@ -59,8 +60,9 @@ #define GPIO_HP_DET BIT(4) struct tegra_wm8903 { + struct tegra_wm8903_platform_data pdata; + struct platform_device *pcm_dev; struct tegra_asoc_utils_data util_data; - struct tegra_wm8903_platform_data *pdata; int gpio_requested; }; @@ -160,7 +162,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = machine->pdata; + struct tegra_wm8903_platform_data *pdata = &machine->pdata; if (!(machine->gpio_requested & GPIO_SPKR_EN)) return 0; @@ -177,7 +179,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = machine->pdata; + struct tegra_wm8903_platform_data *pdata = &machine->pdata; if (!(machine->gpio_requested & GPIO_HP_MUTE)) return 0; @@ -246,9 +248,36 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = machine->pdata; + struct tegra_wm8903_platform_data *pdata = &machine->pdata; + struct device_node *np = card->dev->of_node; int ret; + if (card->dev->platform_data) { + memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); + } else if (np) { + /* + * This part must be in init() rather than probe() in order to + * guarantee that the WM8903 has been probed, and hence its + * GPIO controller registered, which is a pre-condition for + * of_get_named_gpio() to be able to map the phandles in the + * properties to the controller node. Given this, all + * pdata handling is in init() for consistency. + */ + pdata->gpio_spkr_en = of_get_named_gpio(np, + "nvidia,spkr-en-gpios", 0); + pdata->gpio_hp_mute = of_get_named_gpio(np, + "nvidia,hp-mute-gpios", 0); + pdata->gpio_hp_det = of_get_named_gpio(np, + "nvidia,hp-det-gpios", 0); + pdata->gpio_int_mic_en = of_get_named_gpio(np, + "nvidia,int-mic-en-gpios", 0); + pdata->gpio_ext_mic_en = of_get_named_gpio(np, + "nvidia,ext-mic-en-gpios", 0); + } else { + dev_err(card->dev, "No platform data supplied\n"); + return -EINVAL; + } + if (gpio_is_valid(pdata->gpio_spkr_en)) { ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); if (ret) { @@ -348,11 +377,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_tegra_wm8903; struct tegra_wm8903 *machine; - struct tegra_wm8903_platform_data *pdata; int ret; - pdata = pdev->dev.platform_data; - if (!pdata) { + if (!pdev->dev.platform_data && !pdev->dev.of_node) { dev_err(&pdev->dev, "No platform data supplied\n"); return -EINVAL; } @@ -364,31 +391,70 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } - - machine->pdata = pdata; - - ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); - if (ret) - goto err; + machine->pcm_dev = ERR_PTR(-EINVAL); card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony()) { - card->dapm_routes = harmony_audio_map; - card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); - } else if (machine_is_seaboard()) { - card->dapm_routes = seaboard_audio_map; - card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); - } else if (machine_is_kaen()) { - card->dapm_routes = kaen_audio_map; - card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); + if (pdev->dev.of_node) { + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_wm8903_dai.codec_name = NULL; + tegra_wm8903_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!tegra_wm8903_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8903_dai.cpu_dai_name = NULL; + tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_wm8903_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + machine->pcm_dev = platform_device_register_simple( + "tegra-pcm-audio", -1, NULL, 0); + if (IS_ERR(machine->pcm_dev)) { + dev_err(&pdev->dev, + "Can't instantiate tegra-pcm-audio\n"); + ret = PTR_ERR(machine->pcm_dev); + goto err; + } } else { - card->dapm_routes = aebl_audio_map; - card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); + if (machine_is_harmony()) { + card->dapm_routes = harmony_audio_map; + card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); + } else if (machine_is_seaboard()) { + card->dapm_routes = seaboard_audio_map; + card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); + } else if (machine_is_kaen()) { + card->dapm_routes = kaen_audio_map; + card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); + } else { + card->dapm_routes = aebl_audio_map; + card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); + } } + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err_unregister; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", @@ -400,6 +466,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_unregister: + if (!IS_ERR(machine->pcm_dev)) + platform_device_unregister(machine->pcm_dev); err: return ret; } @@ -408,7 +477,7 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = machine->pdata; + struct tegra_wm8903_platform_data *pdata = &machine->pdata; if (machine->gpio_requested & GPIO_HP_DET) snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, @@ -427,15 +496,23 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); + if (!IS_ERR(machine->pcm_dev)) + platform_device_unregister(machine->pcm_dev); return 0; } +static const struct of_device_id tegra_wm8903_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-wm8903", }, + {}, +}; + static struct platform_driver tegra_wm8903_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = tegra_wm8903_of_match, }, .probe = tegra_wm8903_driver_probe, .remove = __devexit_p(tegra_wm8903_driver_remove), @@ -446,3 +523,4 @@ MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra+WM8903 machine ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_wm8903_of_match); -- cgit v1.2.1 From 3922d5180ffa605b08c50c13a7c9db68ab6bbc19 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 14:37:12 +0800 Subject: ASoC: Convert ak4104 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 152420ca78b8..d27b5e4cce99 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -261,7 +261,8 @@ static int ak4104_spi_probe(struct spi_device *spi) if (ret < 0) return ret; - ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + ak4104 = devm_kzalloc(&spi->dev, sizeof(struct ak4104_private), + GFP_KERNEL); if (ak4104 == NULL) return -ENOMEM; @@ -271,15 +272,12 @@ static int ak4104_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_device_ak4104, &ak4104_dai, 1); - if (ret < 0) - kfree(ak4104); return ret; } static int __devexit ak4104_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- cgit v1.2.1 From 7246492dcf0e56d23f71194f8cd8722a681dfa47 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 14:38:09 +0800 Subject: ASoC: Convert ak4535 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 96296fd172f9..9e809e05d066 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -416,7 +416,8 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, struct ak4535_priv *ak4535; int ret; - ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + ak4535 = devm_kzalloc(&i2c->dev, sizeof(struct ak4535_priv), + GFP_KERNEL); if (ak4535 == NULL) return -ENOMEM; @@ -425,15 +426,12 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4535, &ak4535_dai, 1); - if (ret < 0) - kfree(ak4535); return ret; } static __devexit int ak4535_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 4273fcfd71285b4ab6a5d3ce3943e30c2975b797 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 14:39:20 +0800 Subject: ASoC: Convert ak4641 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 90184701480d..266ebea2b65a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -602,7 +602,8 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, struct ak4641_priv *ak4641; int ret; - ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL); + ak4641 = devm_kzalloc(&i2c->dev, sizeof(struct ak4641_priv), + GFP_KERNEL); if (!ak4641) return -ENOMEM; @@ -610,16 +611,12 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, ak4641_dai, ARRAY_SIZE(ak4641_dai)); - if (ret < 0) - kfree(ak4641); - return ret; } static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); - kfree(i2c_get_clientdata(i2c)); return 0; } -- cgit v1.2.1 From 2ff49eea9b8a1d92c2ab09d803dfdc06f4f8e74b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 14:40:12 +0800 Subject: ASoC: Convert ak4642 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 9b4ee6c63d28..5ef70b5d27e4 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -525,7 +525,8 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, struct ak4642_priv *ak4642; int ret; - ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + ak4642 = devm_kzalloc(&i2c->dev, sizeof(struct ak4642_priv), + GFP_KERNEL); if (!ak4642) return -ENOMEM; @@ -535,15 +536,12 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, (struct snd_soc_codec_driver *)id->driver_data, &ak4642_dai, 1); - if (ret < 0) - kfree(ak4642); return ret; } static __devexit int ak4642_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 5b48a5a6dfd44ac80775d94e4ec573f4edda9144 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 14:41:19 +0800 Subject: ASoC: Convert ak4671 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 4f5c69f735a9..a53b152e6a07 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -661,7 +661,8 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client, struct ak4671_priv *ak4671; int ret; - ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + ak4671 = devm_kzalloc(&client->dev, sizeof(struct ak4671_priv), + GFP_KERNEL); if (ak4671 == NULL) return -ENOMEM; @@ -670,15 +671,12 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ak4671, &ak4671_dai, 1); - if (ret < 0) - kfree(ak4671); return ret; } static __devexit int ak4671_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 673847cfb0b07ba42d23f32d42e59eeda81c3b2f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 16:13:26 +0800 Subject: ASoC: Use dai_fmt in hx4700 machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/hx4700.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 03ef9f393434..8260207818a5 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -65,20 +65,6 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the I2S system clock as output */ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_OUT); @@ -175,6 +161,8 @@ static struct snd_soc_dai_link hx4700_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, + .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &hx4700_ops, }; -- cgit v1.2.1 From 52ec35f64ecdd7ef759ce594061be780ca4b324b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 16:27:28 +0800 Subject: ASoC: Use dai_fmt in imote2 machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/imote2.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 97d3aecfc203..dc905aec6294 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -30,20 +30,6 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* CPU should be clock master */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -67,6 +53,8 @@ static struct snd_soc_dai_link imote2_dai = { .codec_dai_name = "wm8940-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &imote2_asoc_ops, }; -- cgit v1.2.1 From f4f8e4c32c5064b292303b270999a87fe11f4ba4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 10:14:25 +0800 Subject: ASoC: Convert 88pm860x-codec to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 2d39123dd21a..99ca53c01676 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -861,7 +861,7 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route pm860x_dapm_routes[] = { /* supply */ {"Left DAC", NULL, "VCODEC"}, {"Right DAC", NULL, "VCODEC"}, @@ -1361,7 +1361,6 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect); static int pm860x_probe(struct snd_soc_codec *codec) { struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, ret; pm860x->codec = codec; @@ -1388,11 +1387,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) goto out; } - snd_soc_add_controls(codec, pm860x_snd_controls, - ARRAY_SIZE(pm860x_snd_controls)); - snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets, - ARRAY_SIZE(pm860x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; out: @@ -1420,6 +1414,13 @@ static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .reg_cache_size = REG_CACHE_SIZE, .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, + + .controls = pm860x_snd_controls, + .num_controls = ARRAY_SIZE(pm860x_snd_controls), + .dapm_widgets = pm860x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pm860x_dapm_widgets), + .dapm_routes = pm860x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pm860x_dapm_routes), }; static int __devinit pm860x_codec_probe(struct platform_device *pdev) -- cgit v1.2.1 From 3f7cec0493eec1d0139a20716b1ce34815a446c3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Dec 2011 10:19:54 +0800 Subject: ASoC: Convert cs42l51 to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 528510b8e5de..ffce9f2a6643 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -511,7 +511,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; ret = cs42l51_fill_cache(codec); @@ -539,20 +538,20 @@ static int cs42l51_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - snd_soc_add_controls(codec, cs42l51_snd_controls, - ARRAY_SIZE(cs42l51_snd_controls)); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets, - ARRAY_SIZE(cs42l51_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, cs42l51_routes, - ARRAY_SIZE(cs42l51_routes)); - return 0; } static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { - .probe = cs42l51_probe, + .probe = cs42l51_probe, .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), + + .controls = cs42l51_snd_controls, + .num_controls = ARRAY_SIZE(cs42l51_snd_controls), + .dapm_widgets = cs42l51_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), + .dapm_routes = cs42l51_routes, + .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; static int cs42l51_i2c_probe(struct i2c_client *i2c_client, -- cgit v1.2.1 From 82150101df27c0f3d315b597081b9fa0e23cd002 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 20 Dec 2011 23:59:41 +0000 Subject: ASoC: Remove ifdefs for GPIO_SYSFS It is part of the GPIO API so should be stubbed appropriately. Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 6c5ebd38c1b0..ee4353f843ea 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -341,10 +341,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpios[i].gpio, ret); } -#ifdef CONFIG_GPIO_SYSFS /* Expose GPIO value over sysfs for diagnostic purposes */ gpio_export(gpios[i].gpio, false); -#endif /* Update initial jack status */ snd_soc_jack_gpio_detect(&gpios[i]); @@ -376,9 +374,7 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, int i; for (i = 0; i < count; i++) { -#ifdef CONFIG_GPIO_SYSFS gpio_unexport(gpios[i].gpio); -#endif free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); gpio_free(gpios[i].gpio); -- cgit v1.2.1 From 5a5049637cf08c4c17805be679c19544bb27fb92 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 21 Dec 2011 10:40:59 -0700 Subject: ASoC: Allow DAI links to be specified using device tree nodes DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 57 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42ad2db8f082..a4592cbee49b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -764,8 +764,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; + if (dai_link->cpu_dai_of_node) { + if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) + continue; + } else { + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + } rtd->cpu_dai = cpu_dai; goto find_codec; @@ -781,8 +786,13 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } rtd->codec = codec; @@ -814,13 +824,19 @@ find_platform: /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; - if (!platform_name) + if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (strcmp(platform->name, platform_name)) - continue; + if (dai_link->platform_of_node) { + if (platform->dev->of_node != + dai_link->platform_of_node) + continue; + } else { + if (strcmp(platform->name, platform_name)) + continue; + } rtd->platform = platform; goto out; @@ -2831,6 +2847,40 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai_link *link = &card->dai_link[i]; + + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codec_name == !!link->codec_of_node) { + dev_err(card->dev, + "Neither/both codec name/of_node are set\n"); + return -EINVAL; + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "Both platform name/of_node are set\n"); + return -EINVAL; + } + + /* + * CPU DAI must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + dev_err(card->dev, + "Neither/both cpu_dai name/of_node are set\n"); + return -EINVAL; + } + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- cgit v1.2.1 From 561c6a172f065fa918d0ff3cecdca1b22dca893f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 09:44:43 +0800 Subject: ASoC: pxa: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/em-x270.c | 1 + sound/soc/pxa/hx4700.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/magician.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/raumfeld.c | 2 ++ sound/soc/pxa/saarb.c | 1 + sound/soc/pxa/spitz.c | 1 + sound/soc/pxa/tavorevb3.c | 1 + sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/z2.c | 1 + sound/soc/pxa/zylonite.c | 1 + 17 files changed, 18 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b0e2fb720910..5e5004a84073 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -317,6 +317,7 @@ static struct snd_soc_dai_link corgi_dai = { /* corgi audio machine driver */ static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", + .owner = THIS_MODULE, .dai_link = &corgi_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 203ab78a4d68..7b1bc2390039 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -133,6 +133,7 @@ static struct snd_soc_dai_link e740_dai[] = { static struct snd_soc_card e740 = { .name = "Toshiba e740", + .owner = THIS_MODULE, .dai_link = e740_dai, .num_links = ARRAY_SIZE(e740_dai), }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 27f90cc44234..47b89d71e287 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -116,6 +116,7 @@ static struct snd_soc_dai_link e750_dai[] = { static struct snd_soc_card e750 = { .name = "Toshiba e750", + .owner = THIS_MODULE, .dai_link = e750_dai, .num_links = ARRAY_SIZE(e750_dai), }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 858bf94160c5..ea9707ec6f28 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -106,6 +106,7 @@ static struct snd_soc_dai_link e800_dai[] = { static struct snd_soc_card e800 = { .name = "Toshiba e800", + .owner = THIS_MODULE, .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index b13a4252812d..64743a05aeae 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -54,6 +54,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { static struct snd_soc_card em_x270 = { .name = "EM-X270", + .owner = THIS_MODULE, .dai_link = em_x270_dai, .num_links = ARRAY_SIZE(em_x270_dai), }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 8260207818a5..2a342c92d829 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -169,6 +169,7 @@ static struct snd_soc_dai_link hx4700_dai = { /* hx4700 audio machine driver */ static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", + .owner = THIS_MODULE, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index dc905aec6294..b93dafd32b80 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -60,6 +60,7 @@ static struct snd_soc_dai_link imote2_dai = { static struct snd_soc_card imote2 = { .name = "Imote2", + .owner = THIS_MODULE, .dai_link = &imote2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index e79f516c400e..3f7a8ecb9720 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -452,6 +452,7 @@ static struct snd_soc_dai_link magician_dai[] = { /* magician audio machine driver */ static struct snd_soc_card snd_soc_card_magician = { .name = "Magician", + .owner = THIS_MODULE, .dai_link = magician_dai, .num_links = ARRAY_SIZE(magician_dai), diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0e73a7f718e4..9c585af59b5f 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,6 +181,7 @@ static struct snd_soc_dai_link mioa701_dai[] = { static struct snd_soc_card mioa701 = { .name = "MioA701", + .owner = THIS_MODULE, .dai_link = mioa701_dai, .num_links = ARRAY_SIZE(mioa701_dai), }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index f313eca40fdc..db24bc685bd3 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { static struct snd_soc_card palm27x_asoc = { .name = "Palm/PXA27x", + .owner = THIS_MODULE, .dai_link = palm27x_dai, .num_links = ARRAY_SIZE(palm27x_dai), }; diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index b899a3bc8f42..ba1545188ec6 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -260,6 +260,7 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_connector_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), .suspend_post = raumfeld_analog_suspend, @@ -268,6 +269,7 @@ static struct snd_soc_card snd_soc_raumfeld_connector = { static struct snd_soc_card snd_soc_raumfeld_speaker = { .name = "Raumfeld Speaker", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_speaker_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), .suspend_post = raumfeld_analog_suspend, diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index d9467a2c6de0..2e21712cec67 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link saarb_dai[] = { static struct snd_soc_card snd_soc_card_saarb = { .name = "Saarb", + .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2d6ff9b1588..bb060482c00d 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -319,6 +319,7 @@ static struct snd_soc_dai_link spitz_dai = { /* spitz audio machine driver */ static struct snd_soc_card snd_soc_spitz = { .name = "Spitz", + .owner = THIS_MODULE, .dai_link = &spitz_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index eeec892e0e04..4bef12cd424d 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link evb3_dai[] = { static struct snd_soc_card snd_soc_card_evb3 = { .name = "Tavor EVB3", + .owner = THIS_MODULE, .dai_link = evb3_dai, .num_links = ARRAY_SIZE(evb3_dai), }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 3f394de297a2..564ef08a89f2 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -236,6 +236,7 @@ static struct snd_soc_dai_link tosa_dai[] = { static struct snd_soc_card tosa = { .name = "Tosa", + .owner = THIS_MODULE, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index b311ffe04b71..d6807e0372bd 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -202,6 +202,7 @@ static struct snd_soc_dai_link z2_dai = { /* z2 audio machine driver */ static struct snd_soc_card snd_soc_z2 = { .name = "Z2", + .owner = THIS_MODULE, .dai_link = &z2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 580aae38e502..ceb656695b0f 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -249,6 +249,7 @@ static int zylonite_resume_pre(struct snd_soc_card *card) static struct snd_soc_card zylonite = { .name = "Zylonite", + .owner = THIS_MODULE, .probe = &zylonite_probe, .remove = &zylonite_remove, .suspend_post = &zylonite_suspend_post, -- cgit v1.2.1 From 095d79dc491dab1311978e0efb252bc23da88b32 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 10:53:15 +0800 Subject: ASoC: samsung: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/samsung/goni_wm8994.c | 1 + sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/jive_wm8750.c | 1 + sound/soc/samsung/littlemill.c | 1 + sound/soc/samsung/ln2440sbc_alc650.c | 1 + sound/soc/samsung/lowland.c | 1 + sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 + sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 + sound/soc/samsung/s3c24xx_uda134x.c | 1 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk2443_wm9710.c | 1 + sound/soc/samsung/smdk_spdif.c | 1 + sound/soc/samsung/smdk_wm8580.c | 1 + sound/soc/samsung/smdk_wm8580pcm.c | 1 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/smdk_wm8994pcm.c | 1 + sound/soc/samsung/smdk_wm9713.c | 1 + sound/soc/samsung/speyside.c | 1 + sound/soc/samsung/tobermory.c | 1 + 21 files changed, 21 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 84f9c3cf7f3e..c23c2ae91f58 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -244,6 +244,7 @@ static struct snd_soc_dai_link goni_dai[] = { static struct snd_soc_card goni = { .name = "goni", + .owner = THIS_MODULE, .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 03cfa5fcdcca..6e3257717c54 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -215,6 +215,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", + .owner = THIS_MODULE, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 8e523fd9189e..1578663a1faa 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -127,6 +127,7 @@ static struct snd_soc_dai_link jive_dai = { /* jive audio machine driver */ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .owner = THIS_MODULE, .dai_link = &jive_dai, .num_links = 1, diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5cea59beec9f..9dd818bde06f 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -194,6 +194,7 @@ static int littlemill_late_probe(struct snd_soc_card *card) static struct snd_soc_card littlemill = { .name = "Littlemill", + .owner = THIS_MODULE, .dai_link = littlemill_dai, .num_links = ARRAY_SIZE(littlemill_dai), diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c index cde38b8e9dc2..69c4a5934a4d 100644 --- a/sound/soc/samsung/ln2440sbc_alc650.c +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -34,6 +34,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { static struct snd_soc_card ln2440sbc = { .name = "LN2440SBC", + .owner = THIS_MODULE, .dai_link = ln2440sbc_dai, .num_links = ARRAY_SIZE(ln2440sbc_dai), }; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4216a06b45f5..4adff934f771 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -177,6 +177,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card lowland = { .name = "Lowland", + .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), .aux_dev = lowland_aux_dev, diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7207189cd211..7ac0ba2025c3 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -465,6 +465,7 @@ static const struct gpio neo1973_gta02_gpios[] = {}; static struct snd_soc_card neo1973 = { .name = "neo1973", + .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), .aux_dev = neo1973_aux_devs, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 71b4c029fc35..21e12361a9cd 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -114,6 +114,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", + .owner = THIS_MODULE, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 502798100f21..7ace6a87f41b 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -89,6 +89,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic33, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 89b57b5c3e17..c42d5f00b0e1 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -78,6 +78,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic23, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 62b69fb6a085..d731042e51b0 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -229,6 +229,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { static struct snd_soc_card snd_soc_s3c24xx_uda134x = { .name = "S3C24XX_UDA134X", + .owner = THIS_MODULE, .dai_link = &s3c24xx_uda134x_dai_link, .num_links = 1, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index a22fc4402802..f2dcb424ea25 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -198,6 +198,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", + .owner = THIS_MODULE, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 8bd1dc5706bf..720ba29bb7e4 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -30,6 +30,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { static struct snd_soc_card smdk2443 = { .name = "SMDK2443", + .owner = THIS_MODULE, .dai_link = smdk2443_dai, .num_links = ARRAY_SIZE(smdk2443_dai), }; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index e0fd8ad23552..beaa9c15d697 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -160,6 +160,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK-S/PDIF", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 81b447823992..bff8758e7f20 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -203,6 +203,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 49dfafbf3df6..fab5322e9f05 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -143,6 +143,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ad9ac42522e2..8eb309f23d18 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -144,6 +144,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = ARRAY_SIZE(smdk_dai), }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 23c7fb71ddfa..77ecba935119 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -124,6 +124,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 31c6daf6d4d0..8e26a730fcdc 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -50,6 +50,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK WM9713", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 0222d8636323..f9ab7707a3e4 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -291,6 +291,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card speyside = { .name = "Speyside", + .owner = THIS_MODULE, .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), .aux_dev = speyside_aux_dev, diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 6f91c65c5a0e..9199649bf786 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -196,6 +196,7 @@ static int tobermory_late_probe(struct snd_soc_card *card) static struct snd_soc_card tobermory = { .name = "Tobermory", + .owner = THIS_MODULE, .dai_link = tobermory_dai, .num_links = ARRAY_SIZE(tobermory_dai), -- cgit v1.2.1 From da403f87a2599db34f611f198d744d54ed964e26 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 22 Dec 2011 11:37:03 +0000 Subject: Revert "ASoC: mxs: correct 'direction' of device_prep_dma_cyclic" This reverts commit dbec3b30a601791717bc5bb827e210c3b5d6e067 as it should never have been applied to the ASoC tree at all, let alone 3.2. --- sound/soc/mxs/mxs-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 5dfd3250ddf1..f39d7dd9fbcb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); + DMA_TO_DEVICE : DMA_FROM_DEVICE); if (!iprtd->desc) { dev_err(&chan->dev->device, "cannot prepare slave dma\n"); return -EINVAL; -- cgit v1.2.1 From 354a21423d09c2a6afe0fcea9dbbda9cdada6e45 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 22 Dec 2011 12:16:39 +0000 Subject: ASoC: Declare soc_new_pcm() properly Ensure that everything is seeing the same declaration by moving it to a header file rather than putting the declaration in soc-core.c Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4592cbee49b..acbb96005a69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -59,8 +59,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); - /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. -- cgit v1.2.1 From 4c3c5df05e02bfa774517cebc5b91b07c2a365dc Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:04:54 +0800 Subject: ASoC: fsl: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/fsl/efika-audio-fabric.c | 14 +++++++------- sound/soc/fsl/pcm030-audio-fabric.c | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 108b5d8bd0e9..b2acd3293ea8 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -31,8 +31,6 @@ #define DRV_NAME "efika-audio-fabric" -static struct snd_soc_card card; - static struct snd_soc_dai_link efika_fabric_dai[] = { { .name = "AC97", @@ -52,6 +50,13 @@ static struct snd_soc_dai_link efika_fabric_dai[] = { }, }; +static struct snd_soc_card card = { + .name = "Efika", + .owner = THIS_MODULE, + .dai_link = efika_fabric_dai, + .num_links = ARRAY_SIZE(efika_fabric_dai), +}; + static __init int efika_fabric_init(void) { struct platform_device *pdev; @@ -60,11 +65,6 @@ static __init int efika_fabric_init(void) if (!of_machine_is_compatible("bplan,efika")) return -ENODEV; - card.name = "Efika"; - card.dai_link = efika_fabric_dai; - card.num_links = ARRAY_SIZE(efika_fabric_dai); - - pdev = platform_device_alloc("soc-audio", 1); if (!pdev) { pr_err("efika_fabric_init: platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ba4d85e317ed..b3af55dcde9d 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -31,8 +31,6 @@ #define DRV_NAME "pcm030-audio-fabric" -static struct snd_soc_card card; - static struct snd_soc_dai_link pcm030_fabric_dai[] = { { .name = "AC97", @@ -52,6 +50,13 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { }, }; +static struct snd_soc_card card = { + .name = "pcm030", + .owner = THIS_MODULE, + .dai_link = pcm030_fabric_dai, + .num_links = ARRAY_SIZE(pcm030_fabric_dai), +}; + static __init int pcm030_fabric_init(void) { struct platform_device *pdev; @@ -60,11 +65,6 @@ static __init int pcm030_fabric_init(void) if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; - - card.name = "pcm030"; - card.dai_link = pcm030_fabric_dai; - card.num_links = ARRAY_SIZE(pcm030_fabric_dai); - pdev = platform_device_alloc("soc-audio", 1); if (!pdev) { pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); -- cgit v1.2.1 From 338d68db77aaf90eea9e06517272b7d7a83d95a4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:14:58 +0800 Subject: ASoC: atmel: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 0377c5451aed..c88351488f45 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -189,6 +189,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { static struct snd_soc_card snd_soc_at91sam9g20ek = { .name = "AT91SAMG20-EK", + .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, .set_bias_level = at91sam9g20ek_set_bias_level, diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index d427e9217ce4..4ca667d477f9 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -135,6 +135,7 @@ static struct snd_soc_dai_link afeb9260_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_machine_afeb9260 = { .name = "AFEB9260", + .owner = THIS_MODULE, .dai_link = &afeb9260_dai, .num_links = 1, }; -- cgit v1.2.1 From 30e4953011fd7a22044a62b9cf77252493b1bd17 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:17:22 +0800 Subject: ASoC: blackfin: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1836.c | 1 + sound/soc/blackfin/bf5xx-ad193x.c | 1 + sound/soc/blackfin/bf5xx-ad1980.c | 1 + sound/soc/blackfin/bf5xx-ad73311.c | 1 + sound/soc/blackfin/bf5xx-ssm2602.c | 1 + sound/soc/blackfin/bfin-eval-adau1373.c | 1 + sound/soc/blackfin/bfin-eval-adau1701.c | 1 + sound/soc/blackfin/bfin-eval-adav80x.c | 1 + 8 files changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index f79d1655e035..60962ce6cd4d 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -91,6 +91,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", + .owner = THIS_MODULE, .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 5956584ea3a4..2d8d82dbc159 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -119,6 +119,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { static struct snd_soc_card bf5xx_ad193x = { .name = "bfin-ad193x", + .owner = THIS_MODULE, .dai_link = &bf5xx_ad193x_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index 06a84b211b52..b30f88bbd703 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link bf5xx_board_dai[] = { static struct snd_soc_card bf5xx_board = { .name = "bfin-ad1980", + .owner = THIS_MODULE, .dai_link = &bf5xx_board_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index b94eb7ef7d16..8e49508596da 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -192,6 +192,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { static struct snd_soc_card bf5xx_ad73311 = { .name = "bfin-ad73311", + .owner = THIS_MODULE, .probe = bf5xx_probe, .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 767e772a815d..030303238042 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { static struct snd_soc_card bf5xx_ssm2602 = { .name = "bfin-ssm2602", + .owner = THIS_MODULE, .dai_link = &bf5xx_ssm2602_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 85ed39abe10e..26b271c62efa 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -147,6 +147,7 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = { static struct snd_soc_card bfin_eval_adau1373 = { .name = "bfin-eval-adau1373", + .owner = THIS_MODULE, .dai_link = &bfin_eval_adau1373_dai, .num_links = 1, diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index 1a88fe9ce34c..c0064fa1dca6 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -84,6 +84,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { static struct snd_soc_card bfin_eval_adau1701 = { .name = "bfin-eval-adau1701", + .owner = THIS_MODULE, .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 0bc995fb6283..4ef079f95e2e 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -93,6 +93,7 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { static struct snd_soc_card bfin_eval_adav80x = { .name = "bfin-eval-adav80x", + .owner = THIS_MODULE, .dai_link = bfin_eval_adav80x_dais, .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais), -- cgit v1.2.1 From 36a16d1ae0735bd95ab86fdf5983ddbaa20c648d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:19:42 +0800 Subject: ASoC: davinci: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 6 ++++++ sound/soc/davinci/davinci-sffsdr.c | 1 + 2 files changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f78c3f0f280c..10a2d8c788b7 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -242,6 +242,7 @@ static struct snd_soc_dai_link da850_evm_dai = { /* davinci dm6446 evm audio machine driver */ static struct snd_soc_card dm6446_snd_soc_card_evm = { .name = "DaVinci DM6446 EVM", + .owner = THIS_MODULE, .dai_link = &dm6446_evm_dai, .num_links = 1, }; @@ -249,6 +250,7 @@ static struct snd_soc_card dm6446_snd_soc_card_evm = { /* davinci dm355 evm audio machine driver */ static struct snd_soc_card dm355_snd_soc_card_evm = { .name = "DaVinci DM355 EVM", + .owner = THIS_MODULE, .dai_link = &dm355_evm_dai, .num_links = 1, }; @@ -256,6 +258,7 @@ static struct snd_soc_card dm355_snd_soc_card_evm = { /* davinci dm365 evm audio machine driver */ static struct snd_soc_card dm365_snd_soc_card_evm = { .name = "DaVinci DM365 EVM", + .owner = THIS_MODULE, .dai_link = &dm365_evm_dai, .num_links = 1, }; @@ -263,18 +266,21 @@ static struct snd_soc_card dm365_snd_soc_card_evm = { /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", + .owner = THIS_MODULE, .dai_link = dm6467_evm_dai, .num_links = ARRAY_SIZE(dm6467_evm_dai), }; static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", + .owner = THIS_MODULE, .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", + .owner = THIS_MODULE, .dai_link = &da850_evm_dai, .num_links = 1, }; diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 0fe558c65145..f71175b29e38 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -93,6 +93,7 @@ static struct snd_soc_dai_link sffsdr_dai = { /* davinci-sffsdr audio machine driver */ static struct snd_soc_card snd_soc_sffsdr = { .name = "DaVinci SFFSDR", + .owner = THIS_MODULE, .dai_link = &sffsdr_dai, .num_links = 1, }; -- cgit v1.2.1 From a76a70232914902e47d289b6d3853ac850097573 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:21:37 +0800 Subject: ASoC: ep93xx: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 1 + sound/soc/ep93xx/simone.c | 1 + sound/soc/ep93xx/snappercl15.c | 1 + 3 files changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 9f6fecdf49e7..bae5cbbbd2b2 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -75,6 +75,7 @@ static struct snd_soc_dai_link edb93xx_dai = { static struct snd_soc_card snd_soc_edb93xx = { .name = "EDB93XX", + .owner = THIS_MODULE, .dai_link = &edb93xx_dai, .num_links = 1, }; diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 1e00b33cc508..dd997094eb30 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -34,6 +34,7 @@ static struct snd_soc_dai_link simone_dai = { static struct snd_soc_card snd_soc_simone = { .name = "Sim.One", + .owner = THIS_MODULE, .dai_link = &simone_dai, .num_links = 1, }; diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index e97cd5701f51..ccae34a3f280 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -93,6 +93,7 @@ static struct snd_soc_dai_link snappercl15_dai = { static struct snd_soc_card snd_soc_snappercl15 = { .name = "Snapper CL15", + .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, }; -- cgit v1.2.1 From b16eaf9fd324a70ecca48faa7ef3f349baf7f0cd Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 21:23:01 +0800 Subject: ASoC: tegra: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 1 + sound/soc/tegra/tegra_wm8903.c | 1 + sound/soc/tegra/trimslice.c | 1 + 3 files changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 9287eb8028fd..4a0e805c4edd 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -141,6 +141,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = { static struct snd_soc_card snd_soc_tegra_alc5632 = { .name = "tegra-alc5632", + .owner = THIS_MODULE, .dai_link = &tegra_alc5632_dai, .num_links = 1, .controls = tegra_alc5632_controls, diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4677f2666300..566655e23b7d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -363,6 +363,7 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { static struct snd_soc_card snd_soc_tegra_wm8903 = { .name = "tegra-wm8903", + .owner = THIS_MODULE, .dai_link = &tegra_wm8903_dai, .num_links = 1, diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 7d95b7697a73..2bdfc550cff8 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -127,6 +127,7 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { static struct snd_soc_card snd_soc_trimslice = { .name = "tegra-trimslice", + .owner = THIS_MODULE, .dai_link = &trimslice_tlv320aic23_dai, .num_links = 1, -- cgit v1.2.1 From 662d4e5c247e9feb2814d9eafc160f42d9035978 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 09:53:55 +0800 Subject: ASoC: au1x: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/au1x/db1000.c | 1 + sound/soc/au1x/db1200.c | 2 ++ 2 files changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 094a20723bc6..511d83c11a9a 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -29,6 +29,7 @@ static struct snd_soc_dai_link db1000_ac97_dai = { static struct snd_soc_card db1000_ac97 = { .name = "DB1000_AC97", + .owner = THIS_MODULE, .dai_link = &db1000_ac97_dai, .num_links = 1, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 80733331733f..1c629393df78 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -45,6 +45,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { static struct snd_soc_card db1200_ac97_machine = { .name = "DB1200_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -94,6 +95,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { static struct snd_soc_card db1200_i2s_machine = { .name = "DB1200_I2S", + .owner = THIS_MODULE, .dai_link = &db1200_i2s_dai, .num_links = 1, }; -- cgit v1.2.1 From 6aff8ccb0cdd14be2acaa7cf397842005752b0f3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:47:08 +0800 Subject: ASoC: imx: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Acked-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/imx/eukrea-tlv320.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 1 + sound/soc/imx/phycore-ac97.c | 1 + sound/soc/imx/wm1133-ev1.c | 1 + 4 files changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 75fb4b83548b..1c1fdd10f73f 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -87,6 +87,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { static struct snd_soc_card eukrea_tlv320 = { .name = "cpuimx-audio", + .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, }; diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b91d42..3c2eed9094d5 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -86,6 +86,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { static struct snd_soc_card mx27vis_aic32x4 = { .name = "visstrim_m10-audio", + .owner = THIS_MODULE, .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, }; diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index a7deb5cb2433..6ac12111de6a 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -38,6 +38,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { static struct snd_soc_card imx_phycore = { .name = "PhyCORE-ac97-audio", + .owner = THIS_MODULE, .dai_link = imx_phycore_dai_ac97, .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), }; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 490a1260c228..37480c90e997 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -255,6 +255,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { static struct snd_soc_card wm1133_ev1 = { .name = "WM1133-EV1", + .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, }; -- cgit v1.2.1 From 1d9d25b35261af7892df7d339b6c34ed648ccd57 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:48:19 +0800 Subject: ASoC: jz4740: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index c5fc339f68f1..0097c3b13a1a 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -81,6 +81,7 @@ static struct snd_soc_dai_link qi_lb60_dai = { static struct snd_soc_card qi_lb60 = { .name = "QI LB60", + .owner = THIS_MODULE, .dai_link = &qi_lb60_dai, .num_links = 1, -- cgit v1.2.1 From b5a67048d012cc69618140ce31316eedc9c57e8c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:51:17 +0800 Subject: ASoC: nuc900: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-audio.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c index 38a2d0d883b5..2f6e6fd6e05c 100644 --- a/sound/soc/nuc900/nuc900-audio.c +++ b/sound/soc/nuc900/nuc900-audio.c @@ -32,6 +32,7 @@ static struct snd_soc_dai_link nuc900evb_ac97_dai = { static struct snd_soc_card nuc900evb_audio_machine = { .name = "NUC900EVB_AC97", + .owner = THIS_MODULE, .dai_link = &nuc900evb_ac97_dai, .num_links = 1, }; -- cgit v1.2.1 From 23bd1ce48f0b2721f0f37087d8acd9fe57f895d7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:52:22 +0800 Subject: ASoC: s6000: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Acked-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6105-ipcam.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 5890e431852f..58cfb1eb7dd3 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -187,6 +187,7 @@ static struct snd_soc_dai_link s6105_dai = { /* s6105 audio machine driver */ static struct snd_soc_card snd_soc_card_s6105 = { .name = "Stretch IP Camera", + .owner = THIS_MODULE, .dai_link = &s6105_dai, .num_links = 1, }; -- cgit v1.2.1 From 4a7042e59908231fc046aac88bd98454d886052d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:53:32 +0800 Subject: ASoC: sh: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 1 + sound/soc/sh/fsi-da7210.c | 1 + sound/soc/sh/fsi-hdmi.c | 1 + sound/soc/sh/migor.c | 1 + sound/soc/sh/sh7760-ac97.c | 1 + 5 files changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index eb52778d0f90..97f540aabbdd 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -49,6 +49,7 @@ static struct snd_soc_dai_link fsi_dai_link = { }; static struct snd_soc_card fsi_soc_card = { + .owner = THIS_MODULE, .dai_link = &fsi_dai_link, .num_links = 1, }; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index f5586b5b0c3b..1dd3354c7411 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -44,6 +44,7 @@ static struct snd_soc_dai_link fsi_da7210_dai = { static struct snd_soc_card fsi_soc_card = { .name = "FSI-DA7210", + .owner = THIS_MODULE, .dai_link = &fsi_da7210_dai, .num_links = 1, }; diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index 621aea155ac1..6e41908323e8 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -39,6 +39,7 @@ static struct snd_soc_dai_link fsi_dai_link = { }; static struct snd_soc_card fsi_soc_card = { + .owner = THIS_MODULE, .dai_link = &fsi_dai_link, .num_links = 1, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 6088a6a3238a..9d9ad8d61c0a 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -164,6 +164,7 @@ static struct snd_soc_dai_link migor_dai = { /* migor audio machine driver */ static struct snd_soc_card snd_soc_migor = { .name = "Migo-R", + .owner = THIS_MODULE, .dai_link = &migor_dai, .num_links = 1, }; diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index df651e8e38de..4a3568a9bf59 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -28,6 +28,7 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { static struct snd_soc_card sh7760_ac97_soc_machine = { .name = "SH7760 AC97", + .owner = THIS_MODULE, .dai_link = &sh7760_ac97_dai, .num_links = 1, }; -- cgit v1.2.1 From e181d14ac3f304086e41f8e6601f0bd3c75570b2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:54:30 +0800 Subject: ASoC: txx9: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-generic.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 9b5e283af51c..b056a1431ed4 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -32,6 +32,7 @@ static struct snd_soc_dai_link txx9aclc_generic_dai = { static struct snd_soc_card txx9aclc_generic_card = { .name = "Generic TXx9 ACLC Audio", + .owner = THIS_MODULE, .dai_link = &txx9aclc_generic_dai, .num_links = 1, }; -- cgit v1.2.1 From 9b344ce80f8bdf5887d9b5f6d9566d336a8c6ab9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:49:28 +0800 Subject: ASoC: kirkwood: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-openrd.c | 1 + sound/soc/kirkwood/kirkwood-t5325.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index d863afb3ee52..8a5a3ddaa5e2 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -76,6 +76,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { static struct snd_soc_card openrd_client = { .name = "OpenRD Client", + .owner = THIS_MODULE, .dai_link = openrd_client_dai, .num_links = ARRAY_SIZE(openrd_client_dai), }; diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index c772b3cf4039..a8930c7c9d1e 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -98,6 +98,7 @@ static struct snd_soc_dai_link t5325_dai[] = { static struct snd_soc_card t5325 = { .name = "t5325", + .owner = THIS_MODULE, .dai_link = t5325_dai, .num_links = ARRAY_SIZE(t5325_dai), }; -- cgit v1.2.1 From 8270ba0c96702864cb0451079384e5060537d345 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 23 Dec 2011 14:50:17 +0800 Subject: ASoC: mid-x86: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 8ae057433968..6f77eef0f131 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -323,6 +323,7 @@ static struct snd_soc_dai_link mfld_msic_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_mfld = { .name = "medfield_audio", + .owner = THIS_MODULE, .dai_link = mfld_msic_dailink, .num_links = ARRAY_SIZE(mfld_msic_dailink), }; -- cgit v1.2.1 From c5cf4dbc7f804bb4ff02a065b927bd8688204253 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Dec 2011 16:45:03 +0800 Subject: ASoC: Add trivial pm_runtime usage to Samsung DAI drivers Currently this won't actually do anything but using this will help the core SoC code track when the system is idle. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 6 +++++- sound/soc/samsung/pcm.c | 5 +++++ 2 files changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ff5d9194d11f..87a874dc7a35 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include @@ -1095,6 +1096,8 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); + pm_runtime_enable(&pdev->dev); + return 0; err: release_mem_region(regs_base, resource_size(res)); @@ -1105,6 +1108,7 @@ err: static __devexit int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; + struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1113,7 +1117,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) other->pri_dai = NULL; other->sec_dai = NULL; } else { - struct resource *res; + pm_runtime_disable(&pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res) release_mem_region(res->start, resource_size(res)); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 5776addd1f94..56780206c000 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include @@ -580,6 +581,8 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + pm_runtime_enable(&pdev->dev); + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); if (ret != 0) { dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); @@ -609,6 +612,8 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); + pm_runtime_disable(&pdev->dev); + iounmap(pcm->regs); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); -- cgit v1.2.1 From b425b88418e302caf27e9cf44aa987b83c04cb2d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 22 Dec 2011 11:08:59 +0800 Subject: ASoC: omap: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann Signed-off-by: Axel Lin Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/am3517evm.c | 1 + sound/soc/omap/ams-delta.c | 1 + sound/soc/omap/igep0020.c | 1 + sound/soc/omap/n810.c | 1 + sound/soc/omap/omap3evm.c | 1 + sound/soc/omap/omap3pandora.c | 1 + sound/soc/omap/omap4-hdmi-card.c | 1 + sound/soc/omap/osk5912.c | 1 + sound/soc/omap/overo.c | 1 + sound/soc/omap/rx51.c | 1 + sound/soc/omap/sdp3430.c | 1 + sound/soc/omap/sdp4430.c | 1 + sound/soc/omap/zoom2.c | 1 + 13 files changed, 13 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1cd4a0cbe9e..add4866d7e67 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,6 +107,7 @@ static struct snd_soc_dai_link am3517evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", + .owner = THIS_MODULE, .dai_link = &am3517evm_dai, .num_links = 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a4338fdac..3e523a7a9efb 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -597,6 +597,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { /* Audio card driver */ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", + .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 591fbf8f7cd9..ccae58a1339c 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link igep2_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_igep2 = { .name = "igep2", + .owner = THIS_MODULE, .dai_link = &igep2_dai, .num_links = 1, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b3f20c..597be412f1e4 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -289,6 +289,7 @@ static struct snd_soc_dai_link n810_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .owner = THIS_MODULE, .dai_link = &n810_dai, .num_links = 1, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 68578959e4aa..071fcb09b8b2 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -70,6 +70,7 @@ static struct snd_soc_dai_link omap3evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3evm = { .name = "omap3evm", + .owner = THIS_MODULE, .dai_link = &omap3evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 7605c37c91e7..07794bd10952 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -233,6 +233,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_omap3pandora = { .name = "omap3pandora", + .owner = THIS_MODULE, .dai_link = omap3pandora_dai, .num_links = ARRAY_SIZE(omap3pandora_dai), }; diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 52d471c1eeed..28d689b2714d 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link omap4_hdmi_dai = { static struct snd_soc_card snd_soc_omap4_hdmi = { .name = "OMAP4HDMI", + .owner = THIS_MODULE, .dai_link = &omap4_hdmi_dai, .num_links = 1, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 351ec9db384d..d859b597e7ec 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link osk_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index c3550aeee533..2ee889c50256 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link overo_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_overo = { .name = "overo", + .owner = THIS_MODULE, .dai_link = &overo_dai, .num_links = 1, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db536320..fada6ef43eea 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", + .owner = THIS_MODULE, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e8fbf8efdbb8..2c850662ea7e 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -213,6 +213,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", + .owner = THIS_MODULE, .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 2735fa03b74b..175ba9a04edf 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", + .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 7641a7fa8f97..981616d61f67 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -157,6 +157,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", + .owner = THIS_MODULE, .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), -- cgit v1.2.1 From 306bf6b19ee3da824fbdbdb2dc4e5d62a8983a2c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:54:04 +0800 Subject: ASoC: Convert da7210 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e4ca61c18605..62e6a9cc82bd 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -944,7 +944,8 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, struct da7210_priv *da7210; int ret; - da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + da7210 = devm_kzalloc(&i2c->dev, sizeof(struct da7210_priv), + GFP_KERNEL); if (!da7210) return -ENOMEM; @@ -953,16 +954,12 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); - if (ret < 0) - kfree(da7210); - return ret; } static int __devexit da7210_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From eb3bb97ce73ac666d9c3d16fc250fa0b78e3b8f2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:56:25 +0800 Subject: ASoC: Convert lm4857 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index c387dafc6ab6..319039240e0f 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -215,7 +215,7 @@ static int __devinit lm4857_i2c_probe(struct i2c_client *i2c, struct lm4857 *lm4857; int ret; - lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL); + lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); if (!lm4857) return -ENOMEM; @@ -225,21 +225,12 @@ static int __devinit lm4857_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); - if (ret) { - kfree(lm4857); - return ret; - } - - return 0; + return ret; } static int __devexit lm4857_i2c_remove(struct i2c_client *i2c) { - struct lm4857 *lm4857 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); - kfree(lm4857); - return 0; } -- cgit v1.2.1 From 6ce91ad4d8d7370be4f9ca3d7ded866cb1e2430d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:58:14 +0800 Subject: ASoC: Convert uda1380 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 8f734d69f651..4f1b23d7e404 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -803,7 +803,8 @@ static __devinit int uda1380_i2c_probe(struct i2c_client *i2c, struct uda1380_priv *uda1380; int ret; - uda1380 = kzalloc(sizeof(struct uda1380_priv), GFP_KERNEL); + uda1380 = devm_kzalloc(&i2c->dev, sizeof(struct uda1380_priv), + GFP_KERNEL); if (uda1380 == NULL) return -ENOMEM; @@ -812,15 +813,12 @@ static __devinit int uda1380_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_uda1380, uda1380_dai, ARRAY_SIZE(uda1380_dai)); - if (ret < 0) - kfree(uda1380); return ret; } static int __devexit uda1380_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); - kfree(i2c_get_clientdata(i2c)); return 0; } -- cgit v1.2.1 From 6ab7e71a9cbcd31f5ee09da384bcfcf0fa11b8c9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:48:48 +0800 Subject: ASoC: Convert 88pm860x-codec to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 99ca53c01676..9fd3b6827bba 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1430,7 +1430,8 @@ static int __devinit pm860x_codec_probe(struct platform_device *pdev) struct resource *res; int i, ret; - pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL); + pm860x = devm_kzalloc(&pdev->dev, sizeof(struct pm860x_priv), + GFP_KERNEL); if (pm860x == NULL) return -ENOMEM; @@ -1459,17 +1460,13 @@ static int __devinit pm860x_codec_probe(struct platform_device *pdev) out: platform_set_drvdata(pdev, NULL); - kfree(pm860x); return -EINVAL; } static int __devexit pm860x_codec_remove(struct platform_device *pdev) { - struct pm860x_priv *pm860x = platform_get_drvdata(pdev); - snd_soc_unregister_codec(&pdev->dev); platform_set_drvdata(pdev, NULL); - kfree(pm860x); return 0; } -- cgit v1.2.1 From d4d9820b4ad6d6227cad090a9f695eea37814215 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 16:05:22 +0800 Subject: ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c Since commit db33f4de "ARM: Orion: Remove address map info from all platform data structures", the dram is removed from struct kirkwood_asoc_platform_data. This patch fixes below build error: CC sound/soc/kirkwood/kirkwood-i2s.o sound/soc/kirkwood/kirkwood-i2s.c: In function 'kirkwood_i2s_dev_probe': sound/soc/kirkwood/kirkwood-i2s.c:444: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' sound/soc/kirkwood/kirkwood-i2s.c:450: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' make[3]: *** [sound/soc/kirkwood/kirkwood-i2s.o] Error 1 make[2]: *** [sound/soc/kirkwood] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Cc: Andrew Lunn Cc: Nicolas Pitre Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 3 +-- sound/soc/kirkwood/kirkwood.h | 1 - 2 files changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index f6bb21156876..3cb9aa4299d3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -441,13 +441,12 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) goto err_ioremap; } - if (!data || !data->dram) { + if (!data) { dev_err(&pdev->dev, "no platform data ?!\n"); err = -EINVAL; goto err_ioremap; } - priv->dram = data->dram; priv->burst = data->burst; return snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bb6e6a5648c9..9047436b3937 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,7 +123,6 @@ struct kirkwood_dma_data { void __iomem *io; int irq; int burst; - struct mbus_dram_target_info *dram; }; #endif -- cgit v1.2.1 From 077a2ba4c8e40e6256948c3b4cc60608a284f555 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 18:58:17 +0800 Subject: ASoC: Use dai_fmt in kirkwood-openrd machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-openrd.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 8a5a3ddaa5e2..55d2ed3df30d 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -26,18 +26,7 @@ static int openrd_client_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - unsigned int freq, fmt; - - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) - return ret; + unsigned int freq; switch (params_rate(params)) { default: @@ -69,6 +58,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .platform_name = "kirkwood-pcm-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, .ops = &openrd_client_ops, }, }; -- cgit v1.2.1 From 7e0d6ac0d894ebae2ff85ea6108dd065a274dfb9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 18:59:30 +0800 Subject: ASoC: Use dai_fmt in kirkwood-t5325 machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-t5325.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index a8930c7c9d1e..6e992307c252 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -25,18 +25,7 @@ static int t5325_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - unsigned int freq, fmt; - - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) - return ret; + unsigned int freq; freq = params_rate(params) * 256; @@ -90,6 +79,7 @@ static struct snd_soc_dai_link t5325_dai[] = { .platform_name = "kirkwood-pcm-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, .ops = &t5325_ops, .init = t5325_dai_init, }, -- cgit v1.2.1 From 5f5de18a7f81382fead581efc489dfcfa34601af Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 19:03:23 +0800 Subject: ASoC: Convert kirkwood-t5325 to table based DAPM init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-t5325.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 6e992307c252..b47cc4e9b746 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -59,11 +59,6 @@ static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets, - ARRAY_SIZE(t5325_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route)); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); @@ -91,6 +86,11 @@ static struct snd_soc_card t5325 = { .owner = THIS_MODULE, .dai_link = t5325_dai, .num_links = ARRAY_SIZE(t5325_dai), + + .dapm_widgets = t5325_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), + .dapm_routes = t5325_route, + .num_dapm_routes = ARRAY_SIZE(t5325_route), }; static struct platform_device *t5325_snd_device; -- cgit v1.2.1 From 67939b22dadd6c5cbf7a4abe9eaec3a8415569a7 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Tue, 20 Dec 2011 14:15:44 +0800 Subject: ASoC: mxs-saif: convert to clk_prepare/clk_unprepare The patch converts mxs-saif driver to clk_prepare/clk_unprepare by using helper functions clk_prepare_enable/clk_disable_unprepare. Signed-off-by: Shawn Guo Cc: Dong Aisheng Cc: Liam Girdwood Acked-by: Mark Brown Acked-by: Marek Vasut --- sound/soc/mxs/mxs-saif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d24fc2..ef1abb53af95 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -210,7 +210,7 @@ int mxs_saif_put_mclk(unsigned int saif_id) return -EBUSY; } - clk_disable(saif->clk); + clk_disable_unprepare(saif->clk); /* disable MCLK output */ __raw_writel(BM_SAIF_CTRL_CLKGATE, @@ -264,7 +264,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, if (ret) return ret; - ret = clk_enable(saif->clk); + ret = clk_prepare_enable(saif->clk); if (ret) return ret; -- cgit v1.2.1 From cf1ee98d800459e6f055742f84355b1aa9e937ae Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 28 Dec 2011 09:55:15 -0200 Subject: ASoC: sgtl5000: Fix voltage units in dev_err message vdda, vddio and vddd are voltages expressed in milivolts (mV), so use the proper annotation. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 250175755eb2..827a43bec531 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1076,7 +1076,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* according to datasheet, maximum voltage of supplies */ if (vdda > 3600 || vddio > 3600 || vddd > 1980) { dev_err(codec->dev, - "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + "exceed max voltage vdda %dmV vddio %dmV vddd %dmV\n", vdda, vddio, vddd); return -EINVAL; -- cgit v1.2.1 From 512fa7c40b9e808000eac31458668369e131a243 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 28 Dec 2011 11:30:11 -0200 Subject: ASoC: Convert sgtl5000 to use devm_kzalloc() Convert sgtl5000 codec driver to use devm_kzalloc(). Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 827a43bec531..fc9b127206e2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1401,7 +1401,8 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, struct sgtl5000_priv *sgtl5000; int ret; - sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); if (!sgtl5000) return -ENOMEM; @@ -1409,22 +1410,13 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); - if (ret) { - dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(sgtl5000); - return ret; - } - - return 0; + return ret; } static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) { - struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(sgtl5000); return 0; } -- cgit v1.2.1 From a5b683489fd3c83f3951eccdc6aee14f50474dda Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Lothar=20Wa=C3=9Fmann?= Date: Wed, 28 Dec 2011 20:06:21 +0800 Subject: ASoC: mxs: Add .owner to struct snd_soc_card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Lothar Waßmann Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 259278f95cdf..60f052b7cf22 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -105,6 +105,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { static struct snd_soc_card mxs_sgtl5000 = { .name = "mxs_sgtl5000", + .owner = THIS_MODULE, .dai_link = mxs_sgtl5000_dai, .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -- cgit v1.2.1 From 30c88f2ca89d6c0706ab585beca3730c9d7524de Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:51:16 +0800 Subject: ASoC: Convert ad193x to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c1b7d928c347..a4a6bef2c0bb 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -385,14 +385,15 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) struct ad193x_priv *ad193x; int ret; - ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); + ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), + GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); if (IS_ERR(ad193x->regmap)) { ret = PTR_ERR(ad193x->regmap); - goto err_free; + goto err_out; } spi_set_drvdata(spi, ad193x); @@ -406,9 +407,7 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) err_regmap_exit: regmap_exit(ad193x->regmap); -err_free: - kfree(ad193x); - +err_out: return ret; } @@ -418,7 +417,6 @@ static int __devexit ad193x_spi_remove(struct spi_device *spi) snd_soc_unregister_codec(&spi->dev); regmap_exit(ad193x->regmap); - kfree(ad193x); return 0; } @@ -455,14 +453,15 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, struct ad193x_priv *ad193x; int ret; - ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); + ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), + GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); if (IS_ERR(ad193x->regmap)) { ret = PTR_ERR(ad193x->regmap); - goto err_free; + goto err_out; } i2c_set_clientdata(client, ad193x); @@ -476,8 +475,7 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, err_regmap_exit: regmap_exit(ad193x->regmap); -err_free: - kfree(ad193x); +err_out: return ret; } @@ -487,7 +485,6 @@ static int __devexit ad193x_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(ad193x->regmap); - kfree(ad193x); return 0; } -- cgit v1.2.1 From 80c2f9da4ecba2ba2ab65ddc058190b1be28d9e5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:52:13 +0800 Subject: ASoC: Convert adau1373 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 637b114bea7f..971ba4529171 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1360,7 +1360,7 @@ static int __devinit adau1373_i2c_probe(struct i2c_client *client, struct adau1373 *adau1373; int ret; - adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL); + adau1373 = devm_kzalloc(&client->dev, sizeof(*adau1373), GFP_KERNEL); if (!adau1373) return -ENOMEM; @@ -1368,16 +1368,12 @@ static int __devinit adau1373_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver)); - if (ret < 0) - kfree(adau1373); - return ret; } static int __devexit adau1373_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(dev_get_drvdata(&client->dev)); return 0; } -- cgit v1.2.1 From 6e4f17cb2b7e8a5327ccc5a6a32442acd408c190 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:50:02 +0800 Subject: ASoC: Convert ad1836 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 919322daf6dd..982d201c2e86 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -341,7 +341,8 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) struct ad1836_priv *ad1836; int ret; - ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); + ad1836 = devm_kzalloc(&spi->dev, sizeof(struct ad1836_priv), + GFP_KERNEL); if (ad1836 == NULL) return -ENOMEM; @@ -351,17 +352,15 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1); - if (ret < 0) - kfree(ad1836); return ret; } static int __devexit ad1836_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } + static const struct spi_device_id ad1836_ids[] = { { "ad1835", AD1835 }, { "ad1836", AD1836 }, -- cgit v1.2.1 From 38b81c1d2517d7c4a6685d49136474bcd0105ab9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:53:09 +0800 Subject: ASoC: Convert adau1701 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6a6af567f02a..6b325ea03869 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -496,23 +496,19 @@ static __devinit int adau1701_i2c_probe(struct i2c_client *client, struct adau1701 *adau1701; int ret; - adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL); + adau1701 = devm_kzalloc(&client->dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); - if (ret < 0) - kfree(adau1701); - return ret; } static __devexit int adau1701_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 558460c65af17f70a0d7adece0f73b8ab968a0f8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:55:01 +0800 Subject: ASoC: Convert jz4740 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/jz4740.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index d73d28317c00..4624e752a188 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -353,7 +353,8 @@ static int __devinit jz4740_codec_probe(struct platform_device *pdev) struct jz4740_codec *jz4740_codec; struct resource *mem; - jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL); + jz4740_codec = devm_kzalloc(&pdev->dev, sizeof(*jz4740_codec), + GFP_KERNEL); if (!jz4740_codec) return -ENOMEM; @@ -361,14 +362,14 @@ static int __devinit jz4740_codec_probe(struct platform_device *pdev) if (!mem) { dev_err(&pdev->dev, "Failed to get mmio memory resource\n"); ret = -ENOENT; - goto err_free_codec; + goto err_out; } mem = request_mem_region(mem->start, resource_size(mem), pdev->name); if (!mem) { dev_err(&pdev->dev, "Failed to request mmio memory region\n"); ret = -EBUSY; - goto err_free_codec; + goto err_out; } jz4740_codec->base = ioremap(mem->start, resource_size(mem)); @@ -394,9 +395,7 @@ err_iounmap: iounmap(jz4740_codec->base); err_release_mem_region: release_mem_region(mem->start, resource_size(mem)); -err_free_codec: - kfree(jz4740_codec); - +err_out: return ret; } @@ -411,7 +410,6 @@ static int __devexit jz4740_codec_remove(struct platform_device *pdev) release_mem_region(mem->start, resource_size(mem)); platform_set_drvdata(pdev, NULL); - kfree(jz4740_codec); return 0; } -- cgit v1.2.1 From 658ecf7784e9bd081ed49a17274ff36bc15ff4d3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 26 Dec 2011 20:57:24 +0800 Subject: ASoC: Convert tlv320aic32x4 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 81a26e1090b3..eb401ef021fb 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -709,7 +709,8 @@ static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, struct aic32x4_priv *aic32x4; int ret; - aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), + GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; @@ -728,15 +729,12 @@ static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); - if (ret < 0) - kfree(aic32x4); return ret; } static __devexit int aic32x4_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From a421a0e41c28ec4bfc719194e95065ec1cb4aee4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:08:34 +0000 Subject: ASoC: Remove unused label from wm8994 probe() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a9936904d1a0..71472b626750 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3759,7 +3759,7 @@ err_irq: wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); -err: + return ret; } -- cgit v1.2.1 From 1b39bf3468e03016ffdcadef3dac1fd75d2db6fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 12:18:53 +0000 Subject: ASoC: Enable ASoC register map dump for some regmap CODECs It's still useful to be able to poke around in the register map at runtime. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 9 +++++++++ sound/soc/codecs/wm8962.c | 9 +++++++++ sound/soc/codecs/wm8994.c | 8 ++++++++ sound/soc/codecs/wm8996.c | 8 ++++++++ 4 files changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index a234b70377fc..8b24323d6b2c 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2568,6 +2568,13 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } +static int wm5100_soc_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + return true; +} + + static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2576,6 +2583,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_pll = wm5100_set_fll, .set_bias_level = wm5100_set_bias_level, .idle_bias_off = 1, + .reg_cache_size = WM5100_MAX_REGISTER, + .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index be35b6468cb1..1be4eb364128 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4123,11 +4123,20 @@ static int wm8962_remove(struct snd_soc_codec *codec) return 0; } +static int wm8962_soc_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + return true; +} + + static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, .set_bias_level = wm8962_set_bias_level, .set_pll = wm8962_set_fll, + .reg_cache_size = WM8962_MAX_REGISTER, + .volatile_register = wm8962_soc_volatile, }; static const struct regmap_config wm8962_regmap = { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 71472b626750..93d27b660257 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3815,12 +3815,20 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) return 0; } +static int wm8994_soc_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + return true; +} + static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, .suspend = wm8994_suspend, .resume = wm8994_resume, .set_bias_level = wm8994_set_bias_level, + .reg_cache_size = WM8994_MAX_REGISTER, + .volatile_register = wm8994_soc_volatile, }; static int __devinit wm8994_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index da7acaefa9d9..d8da10fe5b52 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3039,6 +3039,12 @@ static int wm8996_remove(struct snd_soc_codec *codec) return 0; } +static int wm8996_soc_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + return true; +} + static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, @@ -3051,6 +3057,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .dapm_routes = wm8996_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8996_dapm_routes), .set_pll = wm8996_set_fll, + .reg_cache_size = WM8996_MAX_REGISTER, + .volatile_register = wm8996_soc_volatile_register, }; #define WM8996_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ -- cgit v1.2.1 From 2445ecc3c036ae5f1cc0c3dfed4731d9519a3811 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 09:16:11 +0800 Subject: ASoC: pxa: Convert poodle to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/poodle.c | 42 +++++++++++++++++++++++------------------- 1 file changed, 23 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4c29bc1f9cfe..c9e24bf51763 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -281,22 +281,18 @@ static struct snd_soc_dai_link poodle_dai = { }; /* poodle audio machine driver */ -static struct snd_soc_card snd_soc_poodle = { +static struct snd_soc_card poodle = { .name = "Poodle", .dai_link = &poodle_dai, .num_links = 1, .owner = THIS_MODULE, }; -static struct platform_device *poodle_snd_device; - -static int __init poodle_init(void) +static int __devinit poodle_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &poodle; int ret; - if (!machine_is_poodle()) - return -ENODEV; - locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_AMP_ON, 0); /* should we mute HP at startup - burning power ?*/ @@ -305,28 +301,36 @@ static int __init poodle_init(void) locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - poodle_snd_device = platform_device_alloc("soc-audio", -1); - if (!poodle_snd_device) - return -ENOMEM; - - platform_set_drvdata(poodle_snd_device, &snd_soc_poodle); - ret = platform_device_add(poodle_snd_device); + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); if (ret) - platform_device_put(poodle_snd_device); - + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -static void __exit poodle_exit(void) +static int __devexit poodle_remove(struct platform_device *pdev) { - platform_device_unregister(poodle_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(poodle_init); -module_exit(poodle_exit); +static struct platform_driver poodle_driver = { + .driver = { + .name = "poodle-audio", + .owner = THIS_MODULE, + }, + .probe = poodle_probe, + .remove = __devexit_p(poodle_remove), +}; + +module_platform_driver(poodle_driver); /* Module information */ MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Poodle"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:poodle-audio"); -- cgit v1.2.1 From fe366d067409d7633ca1186b533289828e5f4161 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 09:19:32 +0800 Subject: ASoC: Convert poodle to table based DAPM and control init Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/poodle.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index c9e24bf51763..a321b4df534c 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -214,7 +214,7 @@ SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), }; /* Corgi machine connections to the codec pins */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route poodle_audio_map[] = { /* headphone connected to LHPOUT1, RHPOUT1 */ {"Headphone Jack", NULL, "LHPOUT"}, @@ -246,25 +246,11 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_nc_pin(dapm, "LLINEIN"); snd_soc_dapm_nc_pin(dapm, "RLINEIN"); snd_soc_dapm_enable_pin(dapm, "MICIN"); - /* Add poodle specific controls */ - err = snd_soc_add_controls(codec, wm8731_poodle_controls, - ARRAY_SIZE(wm8731_poodle_controls)); - if (err < 0) - return err; - - /* Add poodle specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, - ARRAY_SIZE(wm8731_dapm_widgets)); - - /* Set up poodle specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -286,6 +272,13 @@ static struct snd_soc_card poodle = { .dai_link = &poodle_dai, .num_links = 1, .owner = THIS_MODULE, + + .controls = wm8731_poodle_controls, + .num_controls = ARRAY_SIZE(wm8731_poodle_controls), + .dapm_widgets = wm8731_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), + .dapm_routes = poodle_audio_map, + .num_dapm_routes = ARRAY_SIZE(poodle_audio_map), }; static int __devinit poodle_probe(struct platform_device *pdev) -- cgit v1.2.1 From 87063dfcea80316ff4647b9906d146b5c7969766 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 09:20:40 +0800 Subject: ASoC: Use dai_fmt in poodle machine driver Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/poodle.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index a321b4df534c..fd0ed10c6fe7 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -121,18 +121,6 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); @@ -263,6 +251,8 @@ static struct snd_soc_dai_link poodle_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .init = poodle_wm8731_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &poodle_ops, }; -- cgit v1.2.1 From 74f4dd56ffab34f280b83d18a5343ac96a7b91ad Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 11:13:24 +0800 Subject: ASoC: Use dai_fmt in corgi machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5e5004a84073..5ff6dac4794d 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -142,18 +142,6 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); @@ -311,6 +299,8 @@ static struct snd_soc_dai_link corgi_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .init = corgi_wm8731_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, }; -- cgit v1.2.1 From 32696af13724aaf7651d1cf95bc1a7a8af97a5c8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 11:16:32 +0800 Subject: ASoC: Convert corgi to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5ff6dac4794d..30ebce26dd9d 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -227,7 +227,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL), }; /* Corgi machine audio map (connections to the codec pins) */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route corgi_audio_map[] = { /* headset Jack - in = micin, out = LHPOUT*/ {"Headset Jack", NULL, "LHPOUT"}, @@ -269,24 +269,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_nc_pin(dapm, "LLINEIN"); snd_soc_dapm_nc_pin(dapm, "RLINEIN"); - /* Add corgi specific controls */ - err = snd_soc_add_controls(codec, wm8731_corgi_controls, - ARRAY_SIZE(wm8731_corgi_controls)); - if (err < 0) - return err; - - /* Add corgi specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, - ARRAY_SIZE(wm8731_dapm_widgets)); - - /* Set up corgi specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -310,6 +296,13 @@ static struct snd_soc_card snd_soc_corgi = { .owner = THIS_MODULE, .dai_link = &corgi_dai, .num_links = 1, + + .controls = wm8731_corgi_controls, + .num_controls = ARRAY_SIZE(wm8731_corgi_controls), + .dapm_widgets = wm8731_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), + .dapm_routes = corgi_audio_map, + .num_dapm_routes = ARRAY_SIZE(corgi_audio_map), }; static struct platform_device *corgi_snd_device; -- cgit v1.2.1 From 24b6f263d97cd2f1f2d579021af97fcd1d632a98 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Mon, 2 Jan 2012 17:35:52 +0530 Subject: ASoC: da7210: Add support for line input and mic DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 62e6a9cc82bd..ab38e93c3543 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char * const da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- cgit v1.2.1 From 021b918efb204b1deda7cfc7edef2972d98ffc46 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:00:13 +0800 Subject: ASoC: Convert cs42l51 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index ffce9f2a6643..a8bf588e8740 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -577,7 +577,8 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", ret & 7); - cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL); + cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), + GFP_KERNEL); if (!cs42l51) { dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; @@ -588,18 +589,13 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); - if (ret < 0) - kfree(cs42l51); error: return ret; } static int cs42l51_i2c_remove(struct i2c_client *client) { - struct cs42l51_private *cs42l51 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(cs42l51); return 0; } -- cgit v1.2.1 From 49ba7673243013103bde4706c506bda2c631a39b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:01:07 +0800 Subject: ASoC: Convert max98088 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ba4f6f167a13..006efcfe6dda 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2069,7 +2069,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c, struct max98088_priv *max98088; int ret; - max98088 = kzalloc(sizeof(struct max98088_priv), GFP_KERNEL); + max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), + GFP_KERNEL); if (max98088 == NULL) return -ENOMEM; @@ -2080,15 +2081,12 @@ static int max98088_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98088, &max98088_dai[0], 2); - if (ret < 0) - kfree(max98088); return ret; } static int __devexit max98088_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From b1b548824ba7e182424da6e4655c1904be7dc6fa Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:02:21 +0800 Subject: ASoC: Convert max98095 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index c69dd022bea8..fcfa7497d7b7 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2340,7 +2340,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c, struct max98095_priv *max98095; int ret; - max98095 = kzalloc(sizeof(struct max98095_priv), GFP_KERNEL); + max98095 = devm_kzalloc(&i2c->dev, sizeof(struct max98095_priv), + GFP_KERNEL); if (max98095 == NULL) return -ENOMEM; @@ -2350,16 +2351,12 @@ static int max98095_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, max98095_dai, ARRAY_SIZE(max98095_dai)); - if (ret < 0) - kfree(max98095); return ret; } static int __devexit max98095_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); - return 0; } -- cgit v1.2.1 From bf791bdb383fdf159c2282d958682b92a2601170 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:03:16 +0800 Subject: ASoC: Convert max9850 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 47060d2afe90..a1913091f56c 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -335,7 +335,8 @@ static int __devinit max9850_i2c_probe(struct i2c_client *i2c, struct max9850_priv *max9850; int ret; - max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL); + max9850 = devm_kzalloc(&i2c->dev, sizeof(struct max9850_priv), + GFP_KERNEL); if (max9850 == NULL) return -ENOMEM; @@ -343,15 +344,12 @@ static int __devinit max9850_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max9850, &max9850_dai, 1); - if (ret < 0) - kfree(max9850); return ret; } static __devexit int max9850_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From a92b0a0803a40f91689fa479b7a169d0467ba33f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:04:15 +0800 Subject: ASoC: Convert rt5631 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index f6e4f5ed9286..20c324c7c349 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1724,7 +1724,8 @@ static int rt5631_i2c_probe(struct i2c_client *i2c, struct rt5631_priv *rt5631; int ret; - rt5631 = kzalloc(sizeof(struct rt5631_priv), GFP_KERNEL); + rt5631 = devm_kzalloc(&i2c->dev, sizeof(struct rt5631_priv), + GFP_KERNEL); if (NULL == rt5631) return -ENOMEM; @@ -1732,16 +1733,12 @@ static int rt5631_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, rt5631_dai, ARRAY_SIZE(rt5631_dai)); - if (ret < 0) - kfree(rt5631); - return ret; } static __devexit int rt5631_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 8eeffe9891dbb74aedcb9a82da4733961d7b432f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:05:20 +0800 Subject: ASoC: Convert ssm2602 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 7dfc7b08114c..333dd98af39c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -652,7 +652,8 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) struct ssm2602_priv *ssm2602; int ret; - ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv), + GFP_KERNEL); if (ssm2602 == NULL) return -ENOMEM; @@ -662,15 +663,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - if (ret < 0) - kfree(ssm2602); return ret; } static int __devexit ssm2602_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -697,7 +695,8 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, struct ssm2602_priv *ssm2602; int ret; - ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv), + GFP_KERNEL); if (ssm2602 == NULL) return -ENOMEM; @@ -707,15 +706,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - if (ret < 0) - kfree(ssm2602); return ret; } static int __devexit ssm2602_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From d999c021b64289b571e5d295deade44e40cbcc4f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:06:39 +0800 Subject: ASoC: Convert sta32x to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 6648af6656c8..fbd145091356 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -968,28 +968,23 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, struct sta32x_priv *sta32x; int ret; - sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + sta32x = devm_kzalloc(&i2c->dev, sizeof(struct sta32x_priv), + GFP_KERNEL); if (!sta32x) return -ENOMEM; i2c_set_clientdata(i2c, sta32x); ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); - if (ret != 0) { + if (ret != 0) dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); - kfree(sta32x); - return ret; - } - return 0; + return ret; } static __devexit int sta32x_i2c_remove(struct i2c_client *client) { - struct sta32x_priv *sta32x = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(sta32x); return 0; } -- cgit v1.2.1 From 099830608a04a7194d00228084bb08130f761084 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:07:30 +0800 Subject: ASoC: Convert tlv320aic23 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 60d08aeac22a..dfa41a96599b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -634,7 +634,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) return -EINVAL; - aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + aic23 = devm_kzalloc(&i2c->dev, sizeof(struct aic23), GFP_KERNEL); if (aic23 == NULL) return -ENOMEM; @@ -643,14 +643,11 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); - if (ret < 0) - kfree(aic23); return ret; } static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); - kfree(i2c_get_clientdata(i2c)); return 0; } -- cgit v1.2.1 From a8163023d29c1439a2447f5203694bef3ed1c61c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:08:36 +0800 Subject: ASoC: Convert tlv320aic26 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 86d1fa38ed2e..a038daec682b 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -416,7 +416,7 @@ static int aic26_spi_probe(struct spi_device *spi) dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n"); /* Allocate driver data */ - aic26 = kzalloc(sizeof *aic26, GFP_KERNEL); + aic26 = devm_kzalloc(&spi->dev, sizeof *aic26, GFP_KERNEL); if (!aic26) return -ENOMEM; @@ -427,18 +427,12 @@ static int aic26_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &aic26_soc_codec_dev, &aic26_dai, 1); - if (ret < 0) - kfree(aic26); return ret; - - dev_dbg(&spi->dev, "SPI device initialized\n"); - return 0; } static int aic26_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- cgit v1.2.1 From e2257db325e8031a149c0f8e3f228d02d08ae578 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:10:04 +0800 Subject: ASoC: Convert tlv320aic3x to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6f963c50e76e..492f22f8a4d7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1504,7 +1504,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_priv *aic3x; int ret; - aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); + aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { dev_err(&i2c->dev, "failed to create private data\n"); return -ENOMEM; @@ -1524,15 +1524,12 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); - if (ret < 0) - kfree(aic3x); return ret; } static int aic3x_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.1 From 360b70ca5e4668c9b9e24d8b200e7069bec83b4e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 11:56:23 +0800 Subject: ASoC: Convert alc5623 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a9b621ef32d..3feee569ceea 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1022,7 +1022,8 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); if (alc5623 == NULL) return -ENOMEM; @@ -1044,7 +1045,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, alc5623_dai.name = "alc5623-hifi"; break; default: - kfree(alc5623); return -EINVAL; } @@ -1053,20 +1053,15 @@ static int alc5623_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); - if (ret != 0) { + if (ret != 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(alc5623); - } return ret; } static int alc5623_i2c_remove(struct i2c_client *client) { - struct alc5623_priv *alc5623 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(alc5623); return 0; } -- cgit v1.2.1 From 7fd8a67446aded9d25e0ae1d94d19105f1620af5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 11:58:22 +0800 Subject: ASoC: Convert cs4270 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index fef0f48330e4..055536645da9 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -671,7 +671,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, i2c_client->addr); dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); - cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); + cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), + GFP_KERNEL); if (!cs4270) { dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; @@ -682,8 +683,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs4270, &cs4270_dai, 1); - if (ret < 0) - kfree(cs4270); return ret; } @@ -696,7 +695,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, static int cs4270_i2c_remove(struct i2c_client *i2c_client) { snd_soc_unregister_codec(&i2c_client->dev); - kfree(i2c_get_clientdata(i2c_client)); return 0; } -- cgit v1.2.1 From a54877d7456ffa88c95d7eb587971792cb1892d6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:11:00 +0800 Subject: ASoC: Convert tlv320dac33 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c7a61fbdae4b..f0aad26cdb31 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1532,7 +1532,8 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, } pdata = client->dev.platform_data; - dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), + GFP_KERNEL); if (dac33 == NULL) return -ENOMEM; @@ -1587,7 +1588,6 @@ err_get: if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); err_gpio: - kfree(dac33); return ret; } @@ -1604,8 +1604,6 @@ static int __devexit dac33_i2c_remove(struct i2c_client *client) regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); snd_soc_unregister_codec(&client->dev); - kfree(dac33); - return 0; } -- cgit v1.2.1 From 6945e9f9dfee897891a8ac620ce1621a2daf7e02 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 29 Dec 2011 12:12:29 +0800 Subject: ASoC: Convert tpa6130a2 to devm_kzalloc() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 7eeca79d7387..363b99dad8e9 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -376,7 +376,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return -ENODEV; } - data = kzalloc(sizeof(*data), GFP_KERNEL); + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); if (data == NULL) { dev_err(dev, "Can not allocate memory\n"); return -ENOMEM; @@ -450,7 +450,6 @@ err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); err_gpio: - kfree(data); tpa6130a2_client = NULL; return ret; @@ -466,8 +465,6 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) gpio_free(data->power_gpio); regulator_put(data->supply); - - kfree(data); tpa6130a2_client = NULL; return 0; -- cgit v1.2.1 From a3bb8f3f818667872728085497b3a3ab3caba371 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:20 +0100 Subject: ASoC: davinci-vcif.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-vcif.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 70ce10c5d998..da030ff883d5 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -210,7 +210,9 @@ static int davinci_vcif_probe(struct platform_device *pdev) struct davinci_vcif_dev *davinci_vcif_dev; int ret; - davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + davinci_vcif_dev = devm_kzalloc(&pdev->dev, + sizeof(struct davinci_vcif_dev), + GFP_KERNEL); if (!davinci_vcif_dev) { dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); @@ -235,23 +237,15 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_vcif_dai); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); - goto fail; + return ret; } return 0; - -fail: - kfree(davinci_vcif_dev); - - return ret; } static int davinci_vcif_remove(struct platform_device *pdev) { - struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - kfree(davinci_vcif_dev); return 0; } -- cgit v1.2.1 From 96d31e2b128e2adc7c4907e259a2d58b2f5edb32 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:21 +0100 Subject: ASoC: davinci-mcasp.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. In this case, the original code did not contain a call to iounmap, nor does one appear anywhere else in the file. I have assumed that it is safe to use devm_ioremap for the allocation in any case. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 40 ++++++++++++--------------------------- 1 file changed, 12 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 2152ff5c04f6..95441bfc8190 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -865,38 +865,35 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res; struct snd_platform_data *pdata; struct davinci_audio_dev *dev; - int ret = 0; + int ret; - dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), + GFP_KERNEL); if (!dev) return -ENOMEM; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "no mem resource?\n"); - ret = -ENODEV; - goto err_release_data; + return -ENODEV; } - ioarea = request_mem_region(mem->start, + ioarea = devm_request_mem_region(&pdev->dev, mem->start, resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "Audio region already claimed\n"); - ret = -EBUSY; - goto err_release_data; + return -EBUSY; } pdata = pdev->dev.platform_data; dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) { - ret = -ENODEV; - goto err_release_region; - } + if (IS_ERR(dev->clk)) + return -ENODEV; clk_enable(dev->clk); dev->clk_active = 1; - dev->base = ioremap(mem->start, resource_size(mem)); + dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!dev->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; @@ -924,7 +921,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_iounmap; + goto err_release_clk; } dma_data->channel = res->start; @@ -940,7 +937,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_iounmap; + goto err_release_clk; } dma_data->channel = res->start; @@ -948,37 +945,24 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) - goto err_iounmap; + goto err_release_clk; return 0; -err_iounmap: - iounmap(dev->base); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); -err_release_region: - release_mem_region(mem->start, resource_size(mem)); -err_release_data: - kfree(dev); - return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { struct davinci_audio_dev *dev = dev_get_drvdata(&pdev->dev); - struct resource *mem; snd_soc_unregister_dai(&pdev->dev); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, resource_size(mem)); - - kfree(dev); - return 0; } -- cgit v1.2.1 From cd0ff7eff08e7daeba278cf58392aac519edff60 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:22 +0100 Subject: ASoC: davinci-i2s.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. In this case, the original code did not contain a call to iounmap, nor does one appear anywhere else in the file. I have assumed that it is safe to use devm_ioremap for the allocation in any case. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 39 +++++++++++++-------------------------- 1 file changed, 13 insertions(+), 26 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ec187100367e..0a74b9587a2c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -661,18 +661,18 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; } - ioarea = request_mem_region(mem->start, resource_size(mem), - pdev->name); + ioarea = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), + pdev->name); if (!ioarea) { dev_err(&pdev->dev, "McBSP region already claimed\n"); return -EBUSY; } - dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); - if (!dev) { - ret = -ENOMEM; - goto err_release_region; - } + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = @@ -691,13 +691,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) { - ret = -ENODEV; - goto err_free_mem; - } + if (IS_ERR(dev->clk)) + return -ENODEV; clk_enable(dev->clk); - dev->base = ioremap(mem->start, resource_size(mem)); + dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!dev->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; @@ -715,7 +713,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_iounmap; + goto err_release_clk; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; @@ -723,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_iounmap; + goto err_release_clk; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; dev->dev = &pdev->dev; @@ -732,35 +730,24 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai); if (ret != 0) - goto err_iounmap; + goto err_release_clk; return 0; -err_iounmap: - iounmap(dev->base); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); -err_free_mem: - kfree(dev); -err_release_region: - release_mem_region(mem->start, resource_size(mem)); - return ret; } static int davinci_i2s_remove(struct platform_device *pdev) { struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev); - struct resource *mem; snd_soc_unregister_dai(&pdev->dev); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, resource_size(mem)); return 0; } -- cgit v1.2.1 From aa4079c110133e5ed86895a07bffb20dd20ed40e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:24 +0100 Subject: ASoC: psc-i2s.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/psc-i2s.c | 42 ++++++++++++++---------------------------- 1 file changed, 14 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 5c1dc8a141ab..0607ba3d9258 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -295,33 +295,34 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) int ret; struct au1xpsc_audio_data *wd; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); if (!wd) return -ENOMEM; iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - goto out0; - } + if (!iores) + return -ENODEV; ret = -EBUSY; - if (!request_mem_region(iores->start, resource_size(iores), - pdev->name)) - goto out0; + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; - wd->mmio = ioremap(iores->start, resource_size(iores)); + wd->mmio = devm_ioremap(&pdev->dev, iores->start, + resource_size(iores)); if (!wd->mmio) - goto out1; + return -EBUSY; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) - goto out2; + return -EBUSY; wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!dmares) - goto out2; + return -EBUSY; wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* preserve PSC clock source set up by platform (dev.platform_data @@ -349,23 +350,12 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); - ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); - if (!ret) - return 0; - -out2: - iounmap(wd->mmio); -out1: - release_mem_region(iores->start, resource_size(iores)); -out0: - kfree(wd); - return ret; + return snd_soc_register_dai(&pdev->dev, &wd->dai_drv); } static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); - struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); snd_soc_unregister_dai(&pdev->dev); @@ -374,10 +364,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(wd->mmio); - release_mem_region(r->start, resource_size(r)); - kfree(wd); - return 0; } -- cgit v1.2.1 From 8d9626d72833bf68791e4cf9ac151c96c44c0f87 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:25 +0100 Subject: ASoC: psc-ac97.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 87daf456b1c9..476b79a1c11a 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -368,35 +368,35 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) unsigned long sel; struct au1xpsc_audio_data *wd; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); if (!wd) return -ENOMEM; mutex_init(&wd->lock); iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - goto out0; - } + if (!iores) + return -ENODEV; - ret = -EBUSY; - if (!request_mem_region(iores->start, resource_size(iores), - pdev->name)) - goto out0; + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; - wd->mmio = ioremap(iores->start, resource_size(iores)); + wd->mmio = devm_ioremap(&pdev->dev, iores->start, + resource_size(iores)); if (!wd->mmio) - goto out1; + return -EBUSY; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) - goto out2; + return -EBUSY; wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!dmares) - goto out2; + return -EBUSY; wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* configuration: max dma trigger threshold, enable ac97 */ @@ -421,24 +421,15 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) - goto out2; + return ret; au1xpsc_ac97_workdata = wd; return 0; - -out2: - iounmap(wd->mmio); -out1: - release_mem_region(iores->start, resource_size(iores)); -out0: - kfree(wd); - return ret; } static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); - struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); snd_soc_unregister_dai(&pdev->dev); @@ -448,10 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(wd->mmio); - release_mem_region(r->start, resource_size(r)); - kfree(wd); - au1xpsc_ac97_workdata = NULL; /* MDEV */ return 0; -- cgit v1.2.1 From 6d8955262ab4cbfcb3ddaca4f978d27d9c088a75 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:26 +0100 Subject: ASoC: i2sc.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/i2sc.c | 45 +++++++++++++-------------------------------- 1 file changed, 13 insertions(+), 32 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index cb53ad87d0a9..d4b9e364a47a 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -227,69 +227,50 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { - int ret; struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - goto out0; - } + if (!iores) + return -ENODEV; - ret = -EBUSY; - if (!request_mem_region(iores->start, resource_size(iores), - pdev->name)) - goto out0; + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; - ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); + ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start, + resource_size(iores)); if (!ctx->mmio) - goto out1; + return -EBUSY; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) - goto out2; + return -EBUSY; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!dmares) - goto out2; + return -EBUSY; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; platform_set_drvdata(pdev, ctx); - ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); - if (ret) - goto out2; - - return 0; - -out2: - iounmap(ctx->mmio); -out1: - release_mem_region(iores->start, resource_size(iores)); -out0: - kfree(ctx); - return ret; + return snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); } static int __devexit au1xi2s_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); - struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); snd_soc_unregister_dai(&pdev->dev); WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ - iounmap(ctx->mmio); - release_mem_region(r->start, resource_size(r)); - kfree(ctx); - return 0; } -- cgit v1.2.1 From 46c3a02cc93083cb946872896428798cfb8609c0 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:27 +0100 Subject: ASoC: dma.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/dma.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index c4017bd56ab8..0a91b186a86f 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -325,27 +325,19 @@ static struct snd_soc_platform_driver alchemy_pcm_soc_platform = { static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev) { struct alchemy_pcm_ctx *ctx; - int ret; - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; platform_set_drvdata(pdev, ctx); - ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); - if (ret) - kfree(ctx); - - return ret; + return snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); } static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev) { - struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev); - snd_soc_unregister_platform(&pdev->dev); - kfree(ctx); return 0; } -- cgit v1.2.1 From be547dd1727fce22ec001006ea4da169df32b6c6 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:28 +0100 Subject: ASoC: dbdma2.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 92bc1b0346fa..8372cd35f0d6 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -350,27 +350,21 @@ static struct snd_soc_platform_driver au1xpsc_soc_platform = { static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct au1xpsc_audio_dmadata *dmadata; - int ret; - dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + dmadata = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct au1xpsc_audio_dmadata), + GFP_KERNEL); if (!dmadata) return -ENOMEM; platform_set_drvdata(pdev, dmadata); - ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); - if (ret) - kfree(dmadata); - - return ret; + return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); } static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { - struct au1xpsc_audio_dmadata *dmadata = platform_get_drvdata(pdev); - snd_soc_unregister_platform(&pdev->dev); - kfree(dmadata); return 0; } -- cgit v1.2.1 From 6065abf5ce8ba0ad945d21255a1d581ca30f2e18 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 29 Dec 2011 17:51:29 +0100 Subject: ASoC: ac97c.c: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses devm_kzalloc, devm_request_mem_region and devm_ioremap for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 40 +++++++++++++--------------------------- 1 file changed, 13 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 7771934b93e2..c5ac2449563a 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -229,35 +229,34 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; mutex_init(&ctx->lock); iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - goto out0; - } + if (!iores) + return -ENODEV; - ret = -EBUSY; - if (!request_mem_region(iores->start, resource_size(iores), - pdev->name)) - goto out0; + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; - ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); + ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start, + resource_size(iores)); if (!ctx->mmio) - goto out1; + return -EBUSY; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) - goto out2; + return -EBUSY; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!dmares) - goto out2; + return -EBUSY; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* switch it on */ @@ -271,33 +270,20 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out2; + return ret; ac97c_workdata = ctx; return 0; - -out2: - iounmap(ctx->mmio); -out1: - release_mem_region(iores->start, resource_size(iores)); -out0: - kfree(ctx); - return ret; } static int __devexit au1xac97c_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); - struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); snd_soc_unregister_dai(&pdev->dev); WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ - iounmap(ctx->mmio); - release_mem_region(r->start, resource_size(r)); - kfree(ctx); - ac97c_workdata = NULL; /* MDEV */ return 0; -- cgit v1.2.1 From 16aff769d73c6b66a79450d7218f31dc46962536 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:34:54 +0800 Subject: ASoC: Fix return value of ak4641_pcm_set_dai_fmt() We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 266ebea2b65a..c4d165a4bddf 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -339,6 +339,7 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 btif; + int ret; /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -358,7 +359,11 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + ret = snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + if (ret < 0) + return ret; + + return 0; } static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, -- cgit v1.2.1 From fe75fe0e041bd5badc6a0be0c3918590198df2a0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:38:03 +0800 Subject: ASoC: Fix return value of wm8962_gpio_direction_out() We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1be4eb364128..296de4e30d26 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3878,13 +3878,17 @@ static int wm8962_gpio_direction_out(struct gpio_chip *chip, { struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); struct snd_soc_codec *codec = wm8962->codec; - int val; + int ret, val; /* Force function 1 (logic output) */ val = (1 << WM8962_GP2_FN_SHIFT) | (value << WM8962_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, - WM8962_GP2_FN_MASK | WM8962_GP2_LVL, val); + ret = snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, + WM8962_GP2_FN_MASK | WM8962_GP2_LVL, val); + if (ret < 0) + return ret; + + return 0; } static struct gpio_chip wm8962_template_chip = { -- cgit v1.2.1 From 2ce7f207c33578ba147c359ccf173b88271d992b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:41:54 +0800 Subject: ASoC: Use dai_fmt in saarb machine driver Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/saarb.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 2e21712cec67..b2be225b1463 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -92,15 +92,6 @@ static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; @@ -119,6 +110,8 @@ static struct snd_soc_dai_link saarb_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = saarb_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &saarb_i2s_ops, }, }; -- cgit v1.2.1 From c0e942310a0a8881ace0a8bf0aa9e7efbb988309 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:42:55 +0800 Subject: ASoC: Use dai_fmt in spitz machine driver Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index bb060482c00d..76288ad905e2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -143,18 +143,6 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); @@ -313,6 +301,8 @@ static struct snd_soc_dai_link spitz_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .init = spitz_wm8750_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &spitz_ops, }; -- cgit v1.2.1 From 36c1b400188266be737392c1ce9b74e3e7136be2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:44:37 +0800 Subject: ASoC: Use dai_fmt in z2 machine driver Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index d6807e0372bd..e8f15ce83749 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -56,18 +56,6 @@ static int z2_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); @@ -196,6 +184,8 @@ static struct snd_soc_dai_link z2_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &z2_ops, }; -- cgit v1.2.1 From 38b437be0b16517c8b6db66d82d63f5c20927116 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 31 Dec 2011 08:40:30 +0800 Subject: ASoC: Convert saarb to table based DAPM init Also remove a unused ret variable to silence the build warning. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/saarb.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index b2be225b1463..c34146b776b4 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { }; /* saarb machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route saarb_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -121,17 +121,17 @@ static struct snd_soc_card snd_soc_card_saarb = { .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), + + .dapm_widgets = saarb_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), + .dapm_routes = saarb_audio_map, + .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), }; static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, - ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- cgit v1.2.1 From 7c27426356c185ac9f8af8c77889b51d1442a2ac Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 31 Dec 2011 08:44:04 +0800 Subject: ASoC: Convert spitz to table based DAPM and control init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 76288ad905e2..90c5245c4742 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -222,7 +222,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { }; /* Spitz machine audio_map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route spitz_audio_map[] = { /* headphone connected to LOUT1, ROUT1 */ {"Headphone Jack", NULL, "LOUT1"}, @@ -265,7 +265,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* NC codec pins */ snd_soc_dapm_nc_pin(dapm, "RINPUT1"); @@ -276,19 +275,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "MONO1"); - /* Add spitz specific controls */ - err = snd_soc_add_controls(codec, wm8750_spitz_controls, - ARRAY_SIZE(wm8750_spitz_controls)); - if (err < 0) - return err; - - /* Add spitz specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - - /* Set up spitz specific audio paths */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -312,6 +298,13 @@ static struct snd_soc_card snd_soc_spitz = { .owner = THIS_MODULE, .dai_link = &spitz_dai, .num_links = 1, + + .controls = wm8750_spitz_controls, + .num_controls = ARRAY_SIZE(wm8750_spitz_controls), + .dapm_widgets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = spitz_audio_map, + .num_dapm_routes = ARRAY_SIZE(spitz_audio_map), }; static struct platform_device *spitz_snd_device; -- cgit v1.2.1 From 1a2dbcbe0491bdfd4fc2484a9853a019f9e21a9c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 31 Dec 2011 08:45:01 +0800 Subject: ASoC: Convert tavorevb3 to table based DAPM init Also remove a unsued ret variable to silence the build warning. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/tavorevb3.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index 4bef12cd424d..56ee82f61189 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { }; /* tavorevb3 machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route evb3_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -128,17 +128,17 @@ static struct snd_soc_card snd_soc_card_evb3 = { .owner = THIS_MODULE, .dai_link = evb3_dai, .num_links = ARRAY_SIZE(evb3_dai), + + .dapm_widgets = evb3_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets), + .dapm_routes = evb3_audio_map, + .num_dapm_routes = ARRAY_SIZE(evb3_audio_map), }; static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, evb3_dapm_widgets, - ARRAY_SIZE(evb3_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- cgit v1.2.1 From 3c3f51f6a37ff9c3f8ffef2ab600d1482a9f30c8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 31 Dec 2011 08:45:49 +0800 Subject: ASoC: Convert z2 to table based DAPM init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index e8f15ce83749..76ccb172d0a7 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -112,7 +112,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { }; /* Z2 machine audio_map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route z2_audio_map[] = { /* headphone connected to LOUT1, ROUT1 */ {"Headphone Jack", NULL, "LOUT1"}, @@ -142,13 +142,6 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "OUT3"); snd_soc_dapm_disable_pin(dapm, "MONO1"); - /* Add z2 specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - - /* Set up z2 specific audio paths */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* Jack detection API stuff */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); @@ -195,6 +188,11 @@ static struct snd_soc_card snd_soc_z2 = { .owner = THIS_MODULE, .dai_link = &z2_dai, .num_links = 1, + + .dapm_widgets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = z2_audio_map, + .num_dapm_routes = ARRAY_SIZE(z2_audio_map), }; static struct platform_device *z2_snd_device; -- cgit v1.2.1 From 385bd9379babaf0982c76e4c073d928e830df6ad Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 31 Dec 2011 11:01:41 +0800 Subject: ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out() We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d88b727d7f99..c91fb2f99c13 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1777,13 +1777,18 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) @@ -1803,13 +1808,18 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) -- cgit v1.2.1 From c49c7f0cf91c8506d0a0ed61227a0da3b243384d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:43:44 +0800 Subject: ASoC: Use dai_fmt in tavorevb3 machine driver Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/tavorevb3.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index 56ee82f61189..8b5ab8f72726 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -92,16 +92,6 @@ static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; } @@ -119,6 +109,8 @@ static struct snd_soc_dai_link evb3_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = evb3_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &evb3_i2s_ops, }, }; -- cgit v1.2.1 From 748b217827974d34a7341142599f0db631a3e45a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 23:36:23 +0800 Subject: ASoC: Fix return value of wm8580_set_sysclk() We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index b1c8d3de08b2..211285164d70 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -670,7 +670,7 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); - int sel, sel_mask, sel_shift; + int ret, sel, sel_mask, sel_shift; switch (dai->driver->id) { case WM8580_DAI_PAIFRX: @@ -711,7 +711,11 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id, /* We really should validate PLL settings but not yet */ wm8580->sysclk[dai->driver->id] = freq; - return snd_soc_update_bits(codec, WM8580_CLKSEL, sel_mask, sel); + ret = snd_soc_update_bits(codec, WM8580_CLKSEL, sel_mask, sel); + if (ret < 0) + return ret; + + return 0; } static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) -- cgit v1.2.1 From 34be9244c7d8107ab9a46af53869f826648fccc8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 30 Dec 2011 11:18:13 +0800 Subject: ASoC: pxa: Convert corgi to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 43 +++++++++++++++++++++++-------------------- 1 file changed, 23 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 30ebce26dd9d..bc21944851c4 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -291,7 +291,7 @@ static struct snd_soc_dai_link corgi_dai = { }; /* corgi audio machine driver */ -static struct snd_soc_card snd_soc_corgi = { +static struct snd_soc_card corgi = { .name = "Corgi", .owner = THIS_MODULE, .dai_link = &corgi_dai, @@ -305,38 +305,41 @@ static struct snd_soc_card snd_soc_corgi = { .num_dapm_routes = ARRAY_SIZE(corgi_audio_map), }; -static struct platform_device *corgi_snd_device; - -static int __init corgi_init(void) +static int __devinit corgi_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &corgi; int ret; - if (!(machine_is_corgi() || machine_is_shepherd() || - machine_is_husky())) - return -ENODEV; - - corgi_snd_device = platform_device_alloc("soc-audio", -1); - if (!corgi_snd_device) - return -ENOMEM; - - platform_set_drvdata(corgi_snd_device, &snd_soc_corgi); - ret = platform_device_add(corgi_snd_device); + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); if (ret) - platform_device_put(corgi_snd_device); - + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -static void __exit corgi_exit(void) +static int __devexit corgi_remove(struct platform_device *pdev) { - platform_device_unregister(corgi_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(corgi_init); -module_exit(corgi_exit); +static struct platform_driver corgi_driver = { + .driver = { + .name = "corgi-audio", + .owner = THIS_MODULE, + }, + .probe = corgi_probe, + .remove = __devexit_p(corgi_remove), +}; + +module_platform_driver(corgi_driver); /* Module information */ MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Corgi"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:corgi-audio"); -- cgit v1.2.1 From a500231da461cfe29541cb4b8422eb9bf59aa6ac Mon Sep 17 00:00:00 2001 From: Sangsu Park Date: Mon, 2 Jan 2012 17:15:10 +0900 Subject: ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 38 ++++++++++++++++++-------------------- 1 file changed, 18 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec6eab2..cdc860a5ff37 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +641,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- cgit v1.2.1 From 7a748e4318909e680b3900e3b97ea42a92724c68 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 1 Jan 2012 18:36:14 +0800 Subject: ASoC: sta32x: Optimize the array work to find rate_min and rate_max For a given ir and fs, there is at most one possible match for the case mclk_ratios[ir][j].ratio * fs == freq. Thus we can break from the inner loop once a match is found. Signed-off-by: Axel Lin Acked-by: Johannes Stezenbach Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd145091356..7db6fa515028 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -522,6 +522,7 @@ static int sta32x_set_dai_sysclk(struct snd_soc_dai *codec_dai, rate_min = fs; if (fs > rate_max) rate_max = fs; + break; } } } -- cgit v1.2.1 From 739be96ab83755e10fd0c2b6a34c8a73254527f7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 6 Jan 2012 14:54:24 +0800 Subject: ASoC: Fix build dependency for SND_ATMEL_SOC_SSC Make SND_ATMEL_SOC_SSC select ATMEL_SSC to fix below build errors: LD .tmp_vmlinux1 sound/built-in.o: In function `atmel_ssc_remove': sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free' sound/built-in.o: In function `atmel_ssc_probe': sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request' sound/built-in.o: In function `atmel_ssc_set_audio': sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request' sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1fcc816ce97..a4d6742d61e3 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,6 +9,7 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC + select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual -- cgit v1.2.1 From 25e9e7565f9aa9e4b976387a3fab60bfaa4efac8 Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Sun, 1 Jan 2012 01:58:44 +0100 Subject: ASoC: check for substream not channels_min in pcm engines This is a follow up on 53dea36c70c1857 which fixes the other affected pcm engines. Description from 53dea36c70c1857: Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Without this patch I was seeing null-pointer dereferenc in atmel-pcm. Signed-off-by: Joachim Eastwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 5 ++--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 5 ++--- sound/soc/blackfin/bf5xx-i2s-pcm.c | 5 ++--- sound/soc/blackfin/bf5xx-tdm-pcm.c | 5 ++--- sound/soc/davinci/davinci-pcm.c | 5 ++--- sound/soc/ep93xx/ep93xx-pcm.c | 5 ++--- sound/soc/jz4740/jz4740-pcm.c | 5 ++--- sound/soc/kirkwood/kirkwood-dma.c | 5 ++--- sound/soc/mid-x86/sst_platform.c | 5 ++--- sound/soc/omap/omap-pcm.c | 5 ++--- sound/soc/samsung/dma.c | 5 ++--- sound/soc/samsung/idma.c | 3 +-- sound/soc/tegra/tegra_pcm.c | 5 ++--- 13 files changed, 25 insertions(+), 38 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 60de05525c06..a21ff459e5d3 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -367,7 +367,6 @@ static u64 atmel_pcm_dmamask = 0xffffffff; static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -376,14 +375,14 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = atmel_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { pr_debug("atmel-pcm:" "Allocating PCM capture DMA buffer\n"); ret = atmel_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index fcff58390848..d7dc9bde0976 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -421,7 +421,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -431,14 +430,14 @@ static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 6ec3d41b9b6d..63205d723eab 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -260,7 +260,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -270,14 +269,14 @@ static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 4406f9a865ae..254490cf1876 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -286,7 +286,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -295,14 +294,14 @@ static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 65bff3d30dd7..b26401f87b85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -831,7 +831,6 @@ static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; @@ -840,7 +839,7 @@ static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = davinci_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK, pcm_hardware_playback.buffer_bytes_max); @@ -848,7 +847,7 @@ static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = davinci_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE, pcm_hardware_capture.buffer_bytes_max); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index a2de9c42b702..3fc96130d1a6 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -286,7 +286,6 @@ static u64 ep93xx_pcm_dmamask = 0xffffffff; static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -295,14 +294,14 @@ static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = ep93xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) return ret; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = ep93xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index 50cda9ea9156..9b8cf256847d 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -302,7 +302,6 @@ static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -312,14 +311,14 @@ static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = jz4740_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto err; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = jz4740_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 210438261a49..d4a17780cef4 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -315,7 +315,6 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; @@ -324,14 +323,14 @@ static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = kirkwood_dma_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) return ret; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = kirkwood_dma_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index c2bf172a196e..d34563b12c3b 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -446,13 +446,12 @@ static void sst_pcm_free(struct snd_pcm *pcm) static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; pr_debug("sst_pcm_new called\n"); - if (dai->driver->playback.channels_min || - dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 52a0f634948e..a59bd352d342 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -378,7 +378,6 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -387,14 +386,14 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(64); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = omap_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = omap_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 797c3d5e79e5..427ae0d9817b 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -403,7 +403,6 @@ static u64 dma_mask = DMA_BIT_MASK(32); static int dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -414,14 +413,14 @@ static int dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 2bcf75815624..3ba6aba8e2b9 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -387,7 +387,6 @@ static u64 idma_mask = DMA_BIT_MASK(32); static int idma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -396,7 +395,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_idma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 90345ee138f3..c22431516ab2 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -330,7 +330,6 @@ static u64 tegra_dma_mask = DMA_BIT_MASK(32); static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -339,14 +338,14 @@ static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = tegra_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto err; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = tegra_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) -- cgit v1.2.1 From b2ed1b0bc69e53d68aa01b79ca0944311b553fc1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 8 Jan 2012 22:50:00 -0800 Subject: ASoC: Fix idma build after update for channel count check Signed-off-by: Mark Brown --- sound/soc/samsung/idma.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 3ba6aba8e2b9..c227c3163cae 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -398,6 +398,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_idma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); + } return ret; } -- cgit v1.2.1 From f75a8ff67d161b5166a2c2360bb2ffaefd5eb853 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Fri, 30 Dec 2011 04:04:54 +0100 Subject: ASoC: cx20442: add bias control over a platform provided regulator Now that a regulator device for controlling the codec chip reset state over a platform agnostic regulator API is available on the only board using this driver so far, extend the driver with a bias control function which will request virtual power to the codec chip from that virtual regulator, and will supersede the present implementation existing at the sound card level. Thanks to the regulator sharing mechanism, both the old (the sound card) and the new (the codec) implementations should coexist smoothly until the sound card file is updated. For this to work as expected, update the sound card .set_bias_level callback to not touch codec->dapm.bias_level. While extending the cx20442 structure, drop unused control_type member. Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin". Signed-off-by: Janusz Krzysztofik Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/cx20442.c | 48 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/omap/ams-delta.c | 8 +++----- 2 files changed, 49 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index ae55e31bfc72..d5fd00a64748 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -25,8 +26,8 @@ struct cx20442_priv { - enum snd_soc_control_type control_type; void *control_data; + struct regulator *por; }; #define CX20442_PM 0x0 @@ -324,6 +325,38 @@ static struct snd_soc_dai_driver cx20442_dai = { }, }; +static int cx20442_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); + int err = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_enable(cx20442->por); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_disable(cx20442->por); + break; + default: + break; + } + if (!err) + codec->dapm.bias_level = level; + + return err; +} + static int cx20442_codec_probe(struct snd_soc_codec *codec) { struct cx20442_priv *cx20442; @@ -331,9 +364,13 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); if (cx20442 == NULL) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cx20442); + cx20442->por = regulator_get(codec->dev, "POR"); + if (IS_ERR(cx20442->por)) + dev_warn(codec->dev, "failed to get the regulator"); cx20442->control_data = NULL; + + snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; codec->card->pop_time = 0; @@ -350,6 +387,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) tty_hangup(tty); } + if (!IS_ERR(cx20442->por)) { + /* should be already in STANDBY, hence disabled */ + regulator_put(cx20442->por); + } + + snd_soc_codec_set_drvdata(codec, NULL); kfree(cx20442); return 0; } @@ -359,6 +402,7 @@ static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .set_bias_level = cx20442_set_bias_level, .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 3e523a7a9efb..a67f4370bc9f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -431,22 +431,20 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = card->rtd->codec; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, AMS_DELTA_LATCH2_MODEM_NRESET); break; case SND_SOC_BIAS_OFF: - if (codec->dapm.bias_level != SND_SOC_BIAS_OFF) + if (card->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } - codec->dapm.bias_level = level; + card->dapm.bias_level = level; return 0; } -- cgit v1.2.1 From e4e9e05409280b50003280afffe27ade21480dd7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 10 Jan 2012 14:19:12 +0800 Subject: ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC" introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation. This patch remove the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC need to depend on ATMEL_SSC. Reported-by: Stephen Rothwell Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742d61e3..72b09cfd3dc3 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,7 +9,6 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC - select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual @@ -27,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 help -- cgit v1.2.1 From 36ae1a96c4dcb0f6581d595cc5d43cf3a7e648c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Jan 2012 17:12:45 -0800 Subject: ASoC: Dynamically allocate the rtd device for a non-empty release() The device model needs a release() function so it can free devices when they become dereferenced. Do that for rtds. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 44 +++++++++++++++++++++++++------------------- sound/soc/soc-dapm.c | 3 +-- 2 files changed, 26 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index acbb96005a69..3986520b4677 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -169,8 +169,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_pcm_runtime *rtd = - container_of(dev, struct snd_soc_pcm_runtime, dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0); } @@ -180,8 +179,7 @@ static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_pcm_runtime *rtd = - container_of(dev, struct snd_soc_pcm_runtime, dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); return sprintf(buf, "%ld\n", rtd->pmdown_time); } @@ -190,8 +188,7 @@ static ssize_t pmdown_time_set(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_soc_pcm_runtime *rtd = - container_of(dev, struct snd_soc_pcm_runtime, dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; ret = strict_strtol(buf, 10, &rtd->pmdown_time); @@ -884,9 +881,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(&rtd->dev, &dev_attr_pmdown_time); - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - device_unregister(&rtd->dev); + device_remove_file(rtd->dev, &dev_attr_pmdown_time); + device_remove_file(rtd->dev, &dev_attr_codec_reg); + device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1061,7 +1058,10 @@ err_probe: return ret; } -static void rtd_release(struct device *dev) {} +static void rtd_release(struct device *dev) +{ + kfree(dev); +} static int soc_post_component_init(struct snd_soc_card *card, struct snd_soc_codec *codec, @@ -1104,11 +1104,17 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->dev.parent = card->dev; - rtd->dev.release = rtd_release; - rtd->dev.init_name = name; + + rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); + if (!rtd->dev) + return -ENOMEM; + device_initialize(rtd->dev); + rtd->dev->parent = card->dev; + rtd->dev->release = rtd_release; + rtd->dev->init_name = name; + dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); - ret = device_register(&rtd->dev); + ret = device_add(rtd->dev); if (ret < 0) { dev_err(card->dev, "asoc: failed to register runtime device: %d\n", ret); @@ -1117,14 +1123,14 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev_registered = 1; /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(&rtd->dev); + ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) dev_err(codec->dev, "asoc: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ - ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) dev_err(codec->dev, "asoc: failed to add codec sysfs files: %d\n", ret); @@ -1213,7 +1219,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (ret) return ret; - ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time); + ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); @@ -1311,8 +1317,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - device_unregister(&rtd->dev); + device_remove_file(rtd->dev, &dev_attr_codec_reg); + device_del(rtd->dev); rtd->dev_registered = 0; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e174d0811dae..3ad1f59b8028 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1738,8 +1738,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_pcm_runtime *rtd = - container_of(dev, struct snd_soc_pcm_runtime, dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); struct snd_soc_codec *codec =rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; -- cgit v1.2.1 From e48b46ba169181dc88ea48e31dcb4afcf8778397 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 11 Jan 2012 12:43:24 +0000 Subject: ASoC: twl6040 - Add method to query optimum PDM_DL1 gain The DL1 PDM interface adds a little gain depending on the output device. Add a method to retrieve the gain value for machine driver usage. Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 1 + 2 files changed, 24 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3376e6fad2a2..5b9c79b6f65e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include @@ -1012,6 +1013,28 @@ static int twl6040_pll_put_enum(struct snd_kcontrol *kcontrol, return 0; } +int twl6040_get_dl1_gain(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + if (snd_soc_dapm_get_pin_status(dapm, "EP")) + return -1; /* -1dB */ + + if (snd_soc_dapm_get_pin_status(dapm, "HSOR") || + snd_soc_dapm_get_pin_status(dapm, "HSOL")) { + + u8 val = snd_soc_read(codec, TWL6040_REG_HSLCTL); + if (val & TWL6040_HSDACMODE) + /* HSDACL in LP mode */ + return -8; /* -8dB */ + else + /* HSDACL in HP mode */ + return -1; /* -1dB */ + } + return 0; /* 0dB */ +} +EXPORT_SYMBOL_GPL(twl6040_get_dl1_gain); + int twl6040_get_clk_id(struct snd_soc_codec *codec) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index a83277bdb851..ef273f1fac2f 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -34,6 +34,7 @@ enum twl6040_trim { #define TWL6040_HSF_TRIM_LEFT(x) (x & 0x0f) #define TWL6040_HSF_TRIM_RIGHT(x) ((x >> 4) & 0x0f) +int twl6040_get_dl1_gain(struct snd_soc_codec *codec); void twl6040_hs_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int report); int twl6040_get_clk_id(struct snd_soc_codec *codec); -- cgit v1.2.1