From 0fb50e5539c1525939b89c1813b60cc72f90a3e1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 7 Nov 2013 00:56:28 -0800 Subject: ASoC: rcar: select REGMAP 55e5b6fd5af04b6d8b0ac6635edf49476ff298ba (ASoC: rsnd: use regmap instead of original register mapping method) support regmap/regmap_field on Renesas sound driver. It needs CONFIG_REGMAP now. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 14011d90d70a..ff60e11ecb56 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD + select REGMAP help This option enables R-Car SUR/SCU/SSIU/SSI sound support -- cgit v1.2.1 From 2ab2b74277a86afe0dd92976db695a2bb8b93366 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 14:17:18 +0000 Subject: ASoC: wm8990: Mark the register map as dirty when powering down Otherwise we'll skip sync on resume. Signed-off-by: Mark Brown Acked-by: Charles Keepax Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8990.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 253c88bb7a4c..4f05fb88bddf 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } -- cgit v1.2.1 From 46bec25da6a41b7308adde746cbcdbbd0bf9b39c Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 27 Nov 2013 18:05:09 +0800 Subject: ASoC: atmel: sam9x5_wm8731: fix oops when unload module As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38d5a15..1b372283bd01 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } + snd_soc_card_set_drvdata(card, priv); + card->dev = &pdev->dev; card->owner = THIS_MODULE; card->dai_link = dai; -- cgit v1.2.1 From df9e3560923a54a559211629895ed9fa1e48ccc0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Nov 2013 15:22:13 +0000 Subject: ASoC: wm5110: Remove output OSR and PGA volume controls These are managed automatically in current revisions. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 25 ------------------------- 1 file changed, 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c3c7396a6181..99b359e19d35 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), -SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 1, 0), -SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, - ARIZONA_OUT4_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, - ARIZONA_OUT5_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, - ARIZONA_OUT6_OSR_SHIFT, 1, 0), - SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUTPUT_PATH_CONFIG_1R, - ARIZONA_OUT1L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUTPUT_PATH_CONFIG_2R, - ARIZONA_OUT2L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUTPUT_PATH_CONFIG_3R, - ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), - SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, -- cgit v1.2.1 From 17b6c19b34b43a6a8dd5936b3cdbc63d7d1ae186 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 27 Nov 2013 09:58:17 +0100 Subject: ASoC: pcm: Fix rate_max calculation In order to make sure that the sample rate is in the supported range of both components the maximum rate of the card should be the minimum of the maximum rate of each components. There is one special case to consider though, if max_rate is set to 0 this means there is no maximum specified, so use min_not_zero() macro which will give use the desired result. Signed-off-by: Lars-Peter Clausen Acked-by: Takashi iwai Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e413..9441e17d1147 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -153,7 +153,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, struct snd_soc_pcm_stream *cpu_stream) { hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); - hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(codec_stream->rate_max, cpu_stream->rate_max); hw->channels_min = max(codec_stream->channels_min, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, -- cgit v1.2.1 From 78e45c99f6d470e6069c8669ee533c97cc5fd296 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 27 Nov 2013 09:58:18 +0100 Subject: ASoC: pcm: Always honor DAI min and max sample rate constraints snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for the PCM stream based on the rates specified in the rates field. Since we call snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause the minimum or maximum rate to be set to a value outside the range of one of the components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To fix this first calculate the minimum and maximum rates using snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified in the snd_soc_pcm_stream structs. Signed-off-by: Lars-Peter Clausen Acked-by: Takashi iwai Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9441e17d1147..11a90cd027fa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, +static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *codec_stream, struct snd_soc_pcm_stream *cpu_stream) { - hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); - hw->rate_max = min_not_zero(codec_stream->rate_max, cpu_stream->rate_max); + struct snd_pcm_hardware *hw = &runtime->hw; + hw->channels_min = max(codec_stream->channels_min, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, @@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, if (cpu_stream->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + + snd_pcm_limit_hw_rates(runtime); + + hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); + hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); } /* @@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, &cpu_dai_drv->playback); } else { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, &cpu_dai_drv->capture); } ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", codec_dai->name, cpu_dai->name); -- cgit v1.2.1 From 1c195ddb1de14ff9a6327c47f88428200f7c8d88 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 26 Nov 2013 10:41:40 +0100 Subject: ASoC: kirkwood: Fix invalid S/PDIF format This patch removes the 32 bits format which is not supported by S/PDIF output. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d34d91743e3f..b42492d87221 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -33,6 +33,10 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define KIRKWOOD_SPDIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, @@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, @@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, -- cgit v1.2.1 From 4f6f1478c1ada4524e9ed21190b4549233a816a3 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Mon, 25 Nov 2013 20:19:05 +0100 Subject: ASoC: kirkwood: Fix erroneous double output while playing This patch fixes the setting of the register KIRKWOOD_PLAYCTL which did always streaming on both I2S and SPDIF, ignoring the DAI ID. The bug was introduced by the commit 75b9b65ee5a "ASoC: kirkwood: add S/PDIF support" Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index b42492d87221..0b18f654b413 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -248,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } - if (dai->id == 0) - ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ - else - ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); -- cgit v1.2.1 From 8f1ec93ae94e95e717283575997dd134a4c5397f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 29 Nov 2013 13:33:11 +0000 Subject: ASoC: core: Use consistent byte ordering in snd_soc_bytes_get snd_soc_bytes_put treats the data in the binary control as big endian words, however snd_soc_bytes_get uses the endian of the host machine. This causes the two functions to be inconsistant with how the mask is applied on little endian machines. This patch applies the big_endian format used in snd_soc_bytes_put to snd_soc_bytes_get. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87e881d..a66783e13a9c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, break; case 2: ((u16 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be16(~params->mask); break; case 4: ((u32 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be32(~params->mask); break; default: return -EINVAL; -- cgit v1.2.1 From d6437c14df89cb2df2bd9ef3692958d979b75d49 Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Tue, 5 Nov 2013 12:13:54 +0000 Subject: ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation Originally snd_hrtimer_callback() used iprtd->period_time for some jiffies based estimation to determine the right moment to call snd_pcm_period_elapsed(). As timer drifts may well be a problem, this was changed in commit b4e82b5b785670b6 to be based on buffer transmission progress, using iprtd->offset and runtime->buffer_size to calculate the amount of data since last period had elapsed. Unfortunately, iprtd->offset counts in bytes, while runtime->buffer_size counts frames, so adding these to find some delta is like comparing apples and oranges, and eventually results in negative delta values every now and then. This is no big harm, because it simply causes snd_pcm_period_elapsed() being called more often than necessary, as negative delta is taken for a large unsigned value by implicit conversion rule. Nonetheless, the calculation is broken, so one would replace the runtime->buffer_size by its equivalent in bytes. But then, there are chances snd_pcm_period_elapsed() is called late, because calculating the moment for the elapsed period into delta is based against the iprtd->last_offset, which is not necessarily the first byte of the period in question, but some random byte which the FIQ handler left us with in r8/r9 by accident. Again, negative impact is low, as there are plenty of periods already prefilled with data, and snd_pcm_period_elapsed() will probably be called latest when the following period is reached. However, the calculation is conceptually broken, and we are best off removing the clever stuff altogether. snd_pcm_period_elapsed() is now simply called once everytime snd_hrtimer_callback() is run, which may not be most accurate, but at least this way we are quite sure we dont miss an end of period. There is not much extra effort wasted by superfluous calls to snd_pcm_period_elapsed(), as the timer frequency closely matches the period size anyway. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 2fc872b2deff..f53b3261b171 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -39,8 +39,6 @@ struct imx_pcm_runtime_data { unsigned int period; int periods; unsigned long offset; - unsigned long last_offset; - unsigned long size; struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; @@ -55,7 +53,6 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; - unsigned long delta; if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing)) return HRTIMER_NORESTART; @@ -67,19 +64,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) else iprtd->offset = regs.ARM_r9 & 0xffff; - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } + snd_pcm_period_elapsed(substream); hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); @@ -96,11 +81,9 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; + iprtd->period = params_period_bytes(params); iprtd->offset = 0; - iprtd->last_offset = 0; iprtd->poll_time_ns = 1000000000 / params_rate(params) * params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -- cgit v1.2.1 From 23d8bb3bb65e3587ceff54e1b54dda3c48a14f28 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 8 Nov 2013 00:55:00 -0200 Subject: ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable Commit 68f9672b (ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation) introduced the following build warning: sound/soc/fsl/imx-pcm-fiq.c:53:26: warning: unused variable 'runtime' [-Wunused-variable] Remove the unused 'runtime' variable. Signed-off-by: Fabio Estevam Acked-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index f53b3261b171..f00b512dbada 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -51,7 +51,6 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct imx_pcm_runtime_data *iprtd = container_of(hrt, struct imx_pcm_runtime_data, hrt); struct snd_pcm_substream *substream = iprtd->substream; - struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing)) -- cgit v1.2.1 From 6c7ef410c986db7e57b83231427e4606a225606b Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sat, 9 Nov 2013 08:41:14 +0800 Subject: ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe() platform_set_drvdata(op, pdata) in pcm030_fabric_probe() will be overwrited when calling snd_soc_register_card(card), but cm030_fabric_remove() use drvdata as a type of struct pcm030_audio_data, so we should move platform_set_drvdata() below snd_soc_register_card() call. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index eb4373840bb6..3665f612819d 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op) return -ENOMEM; card->dev = &op->dev; - platform_set_drvdata(op, pdata); pdata->card = card; @@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_set_drvdata(op, pdata); + return ret; } -- cgit v1.2.1 From fb28a75ad4806e17512025e03ec7c8255d055478 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Sat, 30 Nov 2013 18:05:28 +0200 Subject: ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare N810 audio driver has stopped working at some point. Probably when OMAP2 was converted to common clock framework since now call to clk_enable dumps the stack trace in drivers/clk/clk.c: __clk_enable() due clk->prepare_count is zero. Fix this by converting clk_enable/_disable calls to those that take care of clock prepare/unprepare. I'm not queueing this to linux-stable since OMAP2 common clock framework conversion in commit ed1ebc4948fd ("ARM: OMAP2: clock: Convert to common clk") happened before N810 was really usable in mainline and user base for N810 is anyway small. Potential linux-stable candidates are only those after commit 3d3a6d18abc6 ("watchdog: introduce retu_wdt driver"). Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.1 From ebff65473f56e6c30de928fd6a4f1ce5ae36e8c5 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 2 Dec 2013 13:26:50 +0800 Subject: ASoC: core: fix devres parameter in devm_snd_soc_register_card() Since devm_card_release() expects parameter 'res' to be a pointer to struct snd_soc_card, devm_snd_soc_register_card() should really pass such a pointer rather than the one to struct device. This bug causes the kernel Oops below with imx-sgtl500 driver when we remove the module. It happens because with 'card' pointing to the wrong structure, card->num_rtd becomes 0 in function soc_remove_dai_links(). Consequently, soc_remove_link_components() and in turn soc_cleanup_codec[platform]_debugfs() will not be called on card removal. It results in that debugfs_card_root is being removed while its child entries debugfs_codec_root and debugfs_platform_root are still there, and thus the kernel Oops. Fix the bug by correcting the parameter 'res' to be the pointer to struct snd_soc_card. $ lsmod Module Size Used by snd_soc_imx_sgtl5000 3506 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8139 2 imx_pcm_dma 1380 1 snd_soc_fsl_ssi $ rmmod snd_soc_imx_sgtl5000 Unable to handle kernel paging request at virtual address e594025c pgd = be134000 [e594025c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma CPU: 0 PID: 1793 Comm: rmmod Not tainted 3.13.0-rc1 #1570 task: bee28900 ti: bfbec000 task.ti: bfbec000 PC is at debugfs_remove_recursive+0x28/0x154 LR is at snd_soc_unregister_card+0xa0/0xcc pc : [<80252b38>] lr : [<80496ac4>] psr: a0000013 sp : bfbede00 ip : bfbede28 fp : bfbede24 r10: 803281d4 r9 : bfbec000 r8 : 803271ac r7 : bef54440 r6 : 00000004 r5 : bf9a4010 r4 : bf9a4010 r3 : e5940224 r2 : 00000000 r1 : bef54450 r0 : 803271ac Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4e13404a DAC: 00000015 Process rmmod (pid: 1793, stack limit = 0xbfbec240) Stack: (0xbfbede00 to 0xbfbee000) de00: 00000000 bf9a4010 bf9a4010 00000004 bef54440 bec89000 bfbede44 bfbede28 de20: 80496ac4 80252b1c 804a4b60 bfbede60 bf9a4010 00000004 bfbede54 bfbede48 de40: 804a4b74 80496a30 bfbede94 bfbede58 80328728 804a4b6c bfbede94 a0000013 de60: bf1b5800 bef54440 00000002 bf9a4010 7f0169f8 bf9a4044 00000081 8000e9c4 de80: bfbec000 00000000 bfbedeac bfbede98 80328cb0 80328618 7f016000 bf9a4010 dea0: bfbedec4 bfbedeb0 8032561c 80328c84 bf9a4010 7f0169f8 bfbedee4 bfbedec8 dec0: 80325e84 803255a8 bee28900 7f0169f8 00000000 78208d30 bfbedefc bfbedee8 dee0: 80325410 80325dd4 beca8100 7f0169f8 bfbedf14 bfbedf00 803264f8 803253c8 df00: 7f01635c 7f016a3c bfbedf24 bfbedf18 80327098 803264d4 bfbedf34 bfbedf28 df20: 7f016370 80327090 bfbedfa4 bfbedf38 80085ef0 7f016368 bfbedf54 5f646e73 df40: 5f636f73 5f786d69 6c746773 30303035 00000000 78208008 bfbedf84 bfbedf68 df60: 800613b0 80061194 fffffffe 78208d00 7efc2f07 00000081 7f016a3c 00000800 df80: bfbedf84 00000000 00000000 fffffffe 78208d00 7efc2f07 00000000 bfbedfa8 dfa0: 8000e800 80085dcc fffffffe 78208d00 78208d30 00000800 a8c82400 a8c82400 dfc0: fffffffe 78208d00 7efc2f07 00000081 00000002 00000000 78208008 00000800 dfe0: 7efc2e1c 7efc2ba8 76f5ca47 76edec7c 80000010 78208d30 00000000 00000000 Backtrace: [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<80496ac4>] (snd_soc_unregister_card+0xa0/0xcc) r8:bec89000 r7:bef54440 r6:00000004 r5:bf9a4010 r4:bf9a4010 r3:00000000 [<80496a24>] (snd_soc_unregister_card+0x0/0xcc) from [<804a4b74>] (devm_card_release+0x14/0x18) r6:00000004 r5:bf9a4010 r4:bfbede60 r3:804a4b60 [<804a4b60>] (devm_card_release+0x0/0x18) from [<80328728>] (release_nodes+0x11c/0x1dc) [<8032860c>] (release_nodes+0x0/0x1dc) from [<80328cb0>] (devres_release_all+0x38/0x54) [<80328c78>] (devres_release_all+0x0/0x54) from [<8032561c>] (__device_release_driver+0x80/0xd4) r4:bf9a4010 r3:7f016000 [<8032559c>] (__device_release_driver+0x0/0xd4) from [<80325e84>] (driver_detach+0xbc/0xc0) r5:7f0169f8 r4:bf9a4010 [<80325dc8>] (driver_detach+0x0/0xc0) from [<80325410>] (bus_remove_driver+0x54/0x98) r6:78208d30 r5:00000000 r4:7f0169f8 r3:bee28900 [<803253bc>] (bus_remove_driver+0x0/0x98) from [<803264f8>] (driver_unregister+0x30/0x50) r4:7f0169f8 r3:beca8100 [<803264c8>] (driver_unregister+0x0/0x50) from [<80327098>] (platform_driver_unregister+0x14/0x18) r4:7f016a3c r3:7f01635c [<80327084>] (platform_driver_unregister+0x0/0x18) from [<7f016370>] (imx_sgtl5000_driver_exit+0x14/0x1c [snd_soc_imx_sgtl5000]) [<7f01635c>] (imx_sgtl5000_driver_exit+0x0/0x1c [snd_soc_imx_sgtl5000]) from [<80085ef0>] (SyS_delete_module+0x130/0x18c) [<80085dc0>] (SyS_delete_module+0x0/0x18c) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:7efc2f07 r5:78208d00 r4:fffffffe Code: 889da9f8 e5983020 e3530000 089da9f8 (e5933038) ---[ end trace 825e7e125251a225 ]--- Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/soc-devres.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d732255c02..3449c1e909ae 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res) */ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) { - struct device **ptr; + struct snd_soc_card **ptr; int ret; ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); @@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) ret = snd_soc_register_card(card); if (ret == 0) { - *ptr = dev; + *ptr = card; devres_add(dev, ptr); } else { devres_free(ptr); -- cgit v1.2.1 From b4af6ef99a60c5b56df137d7accd81ba1ee1254e Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Tue, 3 Dec 2013 18:04:54 +0800 Subject: ASoC: wm8731: fix dsp mode configuration According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0. So, fix LRP for DSP mode as the datesheet specification. Signed-off-by: Bo Shen Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c6d759..bc7472c968e3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- cgit v1.2.1