From b66a70d5e9929f3b0df5a7177bba75652d2f4c3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 18:04:11 +0000 Subject: ASoC: Sync initial widget state with hardware ASoC generally uses the register defaults for everything, but in some cases the hardware will default to enabling some of the DAPM widgets (clocks for example). Ensure that DAPM knows about the actual widget state at initialisation by reading the enable bits after instantiating the widgets so they don't get left enabled needlessly. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8194f150bab7..4df96ec9a813 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1627,6 +1627,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; + unsigned int val; list_for_each_entry(w, &dapm->card->widgets, list) { @@ -1675,6 +1676,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_post: break; } + + /* Read the initial power state from the device */ + if (w->reg >= 0) { + val = snd_soc_read(w->codec, w->reg); + val &= 1 << w->shift; + if (w->invert) + val = !val; + + if (val) + w->power = 1; + } + w->new = 1; } -- cgit v1.2.1 From 8e6bfb9b1f79e07c18b0ae406c7c678fc54e4d8e Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Thu, 10 Feb 2011 13:24:32 +0100 Subject: ASoC: CX20442: fix wrong reg_cache_default content Content of the CX20442's snd_soc_codec_driver.reg_cache_default pointed area, introduced with my recent NULL pointer dereferece fix (commit f019ee5feb344ff0b22b58df4568676295aae14f), occured wrong after further testing, more thorough than just booting successfully. There are two problems with it: 1) It should read (1 << CX20442_TELOUT) | (1 << CX20442_MIC), not CX20442_TELOUT | CX20442_MIC. 2) While correctly matching actual codec hardware state on boot when fixed per 1), a few more code modifications would still be required to reflect that state not only into register cache, but also force them into DAPM pins state, otherwise an inconsitency occures which may prevent further codec state changes from being applied correctly. As a result, the phone stops ringing after reboot, until someone picks up the handset for the first time. Revert that reg_cache_default content to a working, previous de facto default value of 0, in hope this change can still be accepted as an rc cycle fix. Created and tested against linux-2.6.38-rc4 Signed-off-by: Janusz Krzysztofik Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bb4bf65b9e7e..0bb424af956f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -367,7 +367,7 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } -static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; +static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, -- cgit v1.2.1 From 3088e3b4963d26d6f6f54987f595b974ed6d48d8 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 10 Feb 2011 15:37:14 -0700 Subject: ASoC: WM8903: Fix mic detection enable logic The mic detection HW should be enabled when either mic or short detection is required, not when only both are required. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8903.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 987476a5895f..017d99ceb42e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1482,7 +1482,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, irq_mask); - if (det && shrt) { + if (det || shrt) { /* Enable mic detection, this may not have been set through * platform data (eg, if the defaults are OK). */ snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, -- cgit v1.2.1 From 173efa09e4c807a2a764509ddd593ad13a44d1df Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 11 Feb 2011 16:32:11 +0000 Subject: ASoC: WM8994: Improve robustness in some use cases Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 142 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 133 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 37b8aa8a680f..bd0cfdd1386f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- cgit v1.2.1 From c52fd021bc027a90a10782c0dcf667ac0135e478 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 11 Feb 2011 16:32:12 +0000 Subject: ASoC: WM8994: Improve playback robustness On WM8994 revision D and earlier ensure proper playback robustness as some rare use cases can trigger issues. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 59 +++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 52 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bd0cfdd1386f..a60b5dbf0154 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1083,6 +1083,17 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int dac_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int mask = 1 << w->shift; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, mask); + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1374,6 +1385,24 @@ SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) }; +static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { +SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0, + dac_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0, + dac_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0, + dac_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0, + dac_ev, SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = { +SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), +SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), +SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1471,11 +1500,6 @@ SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), -SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), -SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), -SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), -SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), - SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), @@ -2627,6 +2651,22 @@ static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int i, ret; + unsigned int val, mask; + + if (wm8994->revision < 4) { + /* force a HW read */ + val = wm8994_reg_read(codec->control_data, + WM8994_POWER_MANAGEMENT_5); + + /* modify the cache only */ + codec->cache_only = 1; + mask = WM8994_DAC1R_ENA | WM8994_DAC1L_ENA | + WM8994_DAC2R_ENA | WM8994_DAC2L_ENA; + val &= mask; + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, val); + codec->cache_only = 0; + } /* Restore the registers */ ret = snd_soc_cache_sync(codec); @@ -3238,12 +3278,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); - else + snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, + ARRAY_SIZE(wm8994_dac_revd_widgets)); + } else { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + } break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, -- cgit v1.2.1 From 3017358a75917b5ed5ad361c02ba2a7e257d3b2a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Feb 2011 11:42:19 +0000 Subject: ASoC: Ensure supplies are maintained for force enabled widgets If a widget has been force enabled then not only do we need to keep the widget itself enabled, we also need to keep any supplies the widget requires enabled. The user could force all the individual widgets on but this requires too much knowledge of device internals. Signed-off-by: Mark Brown Tested-by: Stephen Warren Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4df96ec9a813..25e54230cc6a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -712,7 +712,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) !path->connected(path->source, path->sink)) continue; - if (path->sink && path->sink->power_check && + if (!path->sink) + continue; + + if (path->sink->force) { + power = 1; + break; + } + + if (path->sink->power_check && path->sink->power_check(path->sink)) { power = 1; break; -- cgit v1.2.1 From 5e5677f239ba69fc718ec9a87ac4ba035dafe2c0 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Mon, 14 Feb 2011 07:33:24 +0800 Subject: ALSA: au88x0 - Modify pointer callback to give accurate playback position Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 23f49f356e0f..16c0bdfbb164 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; - int temp; + int temp, page, delta; temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2)); - temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1)); - return temp; + page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT; + if (dma->nr_periods >= 4) + delta = (page - dma->period_real) & 3; + else { + delta = (page - dma->period_real); + if (delta < 0) + delta += dma->nr_periods; + } + return (dma->period_virt + delta) * dma->period_bytes + + (temp & (dma->period_bytes - 1)); } static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma) -- cgit v1.2.1 From eaae55dac6b64c0616046436b294e69fc5311581 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2011 22:45:59 +0100 Subject: ALSA: caiaq - Fix possible string-buffer overflow Use strlcpy() to assure not to overflow the string array sizes by too long USB device name string. Reported-by: Rafa Cc: stable Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/midi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 68b97477577b..66eabafb1c24 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) } dev->pcm->private_data = dev; - strcpy(dev->pcm->name, dev->product_name); + strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name)); memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 2f218c77fff2..a1a47088fd0c 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (ret < 0) return ret; - strcpy(rmidi->name, device->product_name); + strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name)); rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = device; -- cgit v1.2.1 From b540afc2b3d6e4cd1d1f137ef6d9e9c78d67fecd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 14 Feb 2011 20:27:44 +0100 Subject: ALSA: HDA: Add position_fix quirk for an Asus device The bug reporter claims that position_fix=1 is needed for his microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40). Reported-by: Kjell L. BugLink: http://bugs.launchpad.net/bugs/718402 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0baffcdee8f9..fcedad9a5fef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), -- cgit v1.2.1 From 983345e51e0de144775c7449e5cb01ce6cdd1346 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 15 Feb 2011 19:57:09 +0100 Subject: ALSA: HDA: Conexant auto: Handle multiple connections to ADC node Conexant 20641 has several inputs to its ADC node, with one selector and individual amps for all inputs. This patch adds support in the Conexant auto parser to handle that case. It also means that the pin node's volume is being renamed to "Boost" to avoid name clash with the new volume controls on the ADC node. BugLink: http://bugs.launchpad.net/bugs/719524 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 61 +++++++++++++++++++++++++++++++++--------- 1 file changed, 48 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fbe97d32140d..cd29eafdc0ed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3729,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, +static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir) + hda_nid_t nid, int hda_dir, int amp_idx) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -3743,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir); + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]); @@ -3759,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, return 0; } +#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -3808,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; static const char *prev_label; - int i, err, cidx; + int i, err, cidx, conn_len; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + + int multi_adc_volume = 0; /* If the ADC nid has several input volumes */ + int adc_nid = spec->adc_nids[0]; + + conn_len = snd_hda_get_connections(codec, adc_nid, conn, + HDA_MAX_CONNECTIONS); + if (conn_len < 0) + return conn_len; + + multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1; + if (!multi_adc_volume) { + err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid, + HDA_INPUT); + if (err < 0) + return err; + } - err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0], - HDA_INPUT); - if (err < 0) - return err; prev_label = NULL; cidx = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; const char *label; - if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + int j; + int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; + if (!pin_amp && !multi_adc_volume) continue; + label = hda_get_autocfg_input_label(codec, cfg, i); if (label == prev_label) cidx++; else cidx = 0; prev_label = label; - err = cx_auto_add_volume(codec, label, " Capture", cidx, - nid, HDA_INPUT); - if (err < 0) - return err; + + if (pin_amp) { + err = cx_auto_add_volume(codec, label, " Boost", cidx, + nid, HDA_INPUT); + if (err < 0) + return err; + } + + if (!multi_adc_volume) + continue; + for (j = 0; j < conn_len; j++) { + if (conn[j] == nid) { + err = cx_auto_add_volume_idx(codec, label, + " Capture", cidx, adc_nid, HDA_INPUT, j); + if (err < 0) + return err; + break; + } + } } return 0; } -- cgit v1.2.1 From 89724958e5d596bb91328644c97dd80399443e87 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Feb 2011 21:34:04 +0100 Subject: ALSA: HDA: Do not announce false surround in Conexant auto Without this patch, one line-out and one speaker and Conexant's auto parser would announce (non-working) surround capabilities. BugLink: http://bugs.launchpad.net/bugs/721126 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cd29eafdc0ed..dd7c5c12225d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3410,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec) } } spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.max_channels = nums * 2; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (cfg->hp_outs > 0) spec->auto_mute = 1; -- cgit v1.2.1 From eeda276bef36026fce3029e6423e1a09a50c359e Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Mon, 21 Feb 2011 13:45:04 +0800 Subject: ALSA: fix one memory leak in sound jack Signed-off-by: Lu Guanqun Reviewed-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/core/jack.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index 4902ae568730..53b53e97c896 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, fail_input: input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return err; } -- cgit v1.2.1 From 306496761745942d8167e9193a738b559a7fb0b3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 21 Feb 2011 10:23:18 +0100 Subject: ALSA: HDA: Fix mic initialization in VIA auto parser This typo caused some microphone inputs not to be correctly initialized on VIA codecs. Reported-By: Mark Goldstein Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a76c3260d941..63b0054200a8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(spec, nid)) ctl = PIN_OUT; - else if (i == AUTO_PIN_MIC) + else if (cfg->inputs[i].type == AUTO_PIN_MIC) ctl = PIN_VREF50; else ctl = PIN_IN; -- cgit v1.2.1 From 406e56c9dfa0e654870631cd4d9ea20391a527eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 20:41:25 -0800 Subject: ASoC: Fix WM8958 default microphone detection argument ordering Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a60b5dbf0154..ebaee5ca7434 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3000,11 +3000,10 @@ static void wm8958_default_micdet(u16 status, void *data) report |= SND_JACK_BTN_5; done: - snd_soc_jack_report(wm8994->micdet[0].jack, + snd_soc_jack_report(wm8994->micdet[0].jack, report, 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_JACK_MICROPHONE | SND_JACK_VIDEOOUT, - report); + SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT); } /** -- cgit v1.2.1 From 8ceed344afab2d89516e6d52634ad81920762993 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 10:44:42 -0800 Subject: ASoC: Correct definition of WM8903_VMID_RES_5K Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index e8490f3edd03..e3ec2433b215 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -165,7 +165,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_VMID_RES_50K 2 #define WM8903_VMID_RES_250K 3 -#define WM8903_VMID_RES_5K 4 +#define WM8903_VMID_RES_5K 6 /* * R8 (0x08) - Analogue DAC 0 -- cgit v1.2.1 From cea2bc50a3dd88e43be2e926a9ae31ab7816bf2d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 15:05:53 -0800 Subject: ASoC: Hook wm_hubs micbiases up to CLK_SYS The microphone detection functionality requires a clock to work. In any non-detection case where the MICBIAS is enabled CLK_SYS will be needed anyway so there is no negative impact on power consumption. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 613df5db0b32..516892706063 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -674,6 +674,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"), }; static const struct snd_soc_dapm_route analogue_routes[] = { + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, -- cgit v1.2.1 From 382225e62bdb8059b7f915b133426425516dd300 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Feb 2011 10:21:18 +0100 Subject: ALSA: usb-audio: fix oops due to cleanup race when disconnecting When a USB audio device is disconnected, snd_usb_audio_disconnect() kills all audio URBs. At the same time, the application, after being notified of the disconnection, might close the device, in which case ALSA calls the .hw_free callback, which should free the URBs too. Commit de1b8b93a0ba "[ALSA] Fix hang-up at disconnection of usb-audio" prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that resulted from this race, but this introduced another race because the URB callbacks could now be executed after snd_usb_hw_free() has returned, and try to access already freed data. Fix the first race by introducing a mutex to serialize the disconnect callback and all PCM callbacks that manage URBs (hw_free and hw_params). Reported-and-tested-by: Pierre-Louis Bossart Cc: [CL: also serialize hw_params callback] Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.c | 4 ++++ sound/usb/pcm.c | 7 +++++-- sound/usb/usbaudio.h | 1 + 3 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 800f7cb4f251..c0f8270bc199 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENOMEM; } + mutex_init(&chip->shutdown_mutex); chip->index = idx; chip->dev = dev; chip->card = card; @@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) chip = ptr; card = chip->card; mutex_lock(®ister_mutex); + mutex_lock(&chip->shutdown_mutex); chip->shutdown = 1; chip->num_interfaces--; if (chip->num_interfaces <= 0) { @@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; + mutex_unlock(&chip->shutdown_mutex); mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { + mutex_unlock(&chip->shutdown_mutex); mutex_unlock(®ister_mutex); } } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4132522ac90f..e3f680526cb5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, } if (changed) { + mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ snd_usb_release_substream_urbs(subs, 0); /* influenced: period_bytes, channels, rate, format, */ @@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, params_rate(hw_params), snd_pcm_format_physical_width(params_format(hw_params)) * params_channels(hw_params)); + mutex_unlock(&subs->stream->chip->shutdown_mutex); } return ret; @@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; - if (!subs->stream->chip->shutdown) - snd_usb_release_substream_urbs(subs, 0); + mutex_lock(&subs->stream->chip->shutdown_mutex); + snd_usb_release_substream_urbs(subs, 0); + mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index db3eb21627ee..6e66fffe87f5 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,6 +36,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + struct mutex shutdown_mutex; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- cgit v1.2.1 From 6da8b51657a9cd5a87b4e6e4c7bc76b598a95175 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 8 Feb 2011 07:16:06 +0100 Subject: ALSA: HDA: Add a new Conexant codec 506e (20590) Conexant 506e/20590 has the same graph as the rest of the 5066 family. BugLink: http://bugs.launchpad.net/bugs/723672 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd7c5c12225d..909ce9e09441 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3937,6 +3937,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506e, .name = "CX20590", + .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", @@ -3963,6 +3965,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); MODULE_ALIAS("snd-hda-codec-id:14f150a1"); -- cgit v1.2.1 From ebbd224c22a00dbbee95031a0d6d595460f6f2b3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 23 Feb 2011 13:15:56 +0100 Subject: ALSA: HDA: Add ideapad quirk for two Dell machines These two Dell machines have been reported working well with the ideapad model. BugLink: http://bugs.launchpad.net/bugs/723676 Cc: stable@kernel.org Tested-by: David Chen Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 909ce9e09441..4d5004e693f0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3114,6 +3114,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), -- cgit v1.2.1 From 4dfb8a45d533808e78d67ef27e0a47d456c12a92 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Tue, 22 Feb 2011 17:32:19 -0600 Subject: ALSA: hda - Add support for new IDT 92HD98 and 92HD99 codecs Also fix number of 92HD87 pins to exclude invalid pins. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9ea48b425d0b..bd7b123f6440 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -586,7 +586,12 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x1f, 0x20, }; -static hda_nid_t stac92hd88xxx_pin_nids[10] = { +static hda_nid_t stac92hd87xxx_pin_nids[6] = { + 0x0a, 0x0b, 0x0c, 0x0d, + 0x0f, 0x11, +}; + +static hda_nid_t stac92hd88xxx_pin_nids[8] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0f, 0x11, 0x1f, 0x20, }; @@ -5430,12 +5435,13 @@ again: switch (codec->vendor_id) { case 0x111d76d1: case 0x111d76d9: + case 0x111d76e5: spec->dmic_nids = stac92hd87b_dmic_nids; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd87b_dmic_nids, STAC92HD87B_NUM_DMICS); - spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); - spec->pin_nids = stac92hd88xxx_pin_nids; + spec->num_pins = ARRAY_SIZE(stac92hd87xxx_pin_nids); + spec->pin_nids = stac92hd87xxx_pin_nids; spec->mono_nid = 0; spec->num_pwrs = 0; break; @@ -5443,6 +5449,7 @@ again: case 0x111d7667: case 0x111d7668: case 0x111d7669: + case 0x111d76e3: spec->num_dmics = stac92xx_connected_ports(codec, stac92hd88xxx_dmic_nids, STAC92HD88XXX_NUM_DMICS); @@ -6387,6 +6394,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx}, {} /* terminator */ }; -- cgit v1.2.1 From 4bfc4e2508234f9149fd33fae853e99fb9e4a75b Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Wed, 23 Feb 2011 02:29:11 +0300 Subject: ASoC: correct pxa AC97 DAI names Correct names for pxa AC97 DAI are pxa2xx-ac97 and pxa2xx-ac97-aux. Fix that for all PXA platforms. Signed-off-by: Dmitry Eremin-Solenikov Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/e740_wm9705.c | 4 ++-- sound/soc/pxa/e750_wm9705.c | 4 ++-- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/pxa/zylonite.c | 4 ++-- 8 files changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 28333e7d9c50..dc65650a6fa1 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -117,7 +117,7 @@ static struct snd_soc_dai_link e740_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", @@ -126,7 +126,7 @@ static struct snd_soc_dai_link e740_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 01bf31675c55..51897fcd911b 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -99,7 +99,7 @@ static struct snd_soc_dai_link e750_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", @@ -109,7 +109,7 @@ static struct snd_soc_dai_link e750_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name ="wm9705-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index c6a37c6ef23b..053ed208e59f 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -89,7 +89,7 @@ static struct snd_soc_dai_link e800_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", @@ -98,7 +98,7 @@ static struct snd_soc_dai_link e800_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name ="wm9712-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index fc22e6eefc98..b13a4252812d 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", @@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name ="wm9712-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0d70fc8c12bd..38ca6759907e 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -162,7 +162,7 @@ static struct snd_soc_dai_link mioa701_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", .init = mioa701_wm9713_init, @@ -172,7 +172,7 @@ static struct snd_soc_dai_link mioa701_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name ="wm9713-aux", .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 857db96d4a4f..504e4004f004 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -132,7 +132,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .platform_name = "pxa-pcm-audio", @@ -141,7 +141,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .codec_name = "wm9712-codec", .platform_name = "pxa-pcm-audio", diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f75804ef0897..4b6e5d608b42 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -219,7 +219,7 @@ static struct snd_soc_dai_link tosa_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", @@ -229,7 +229,7 @@ static struct snd_soc_dai_link tosa_dai[] = { { .name = "AC97 Aux", .stream_name = "AC97 Aux", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b222a7d72027..25bba108fea3 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .stream_name = "AC97 HiFi", .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", - .cpu_dai_name = "pxa-ac97.0", + .cpu_dai_name = "pxa2xx-ac97", .codec_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, @@ -175,7 +175,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .stream_name = "AC97 Aux", .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", - .cpu_dai_name = "pxa-ac97.1", + .cpu_dai_name = "pxa2xx-ac97-aux", .codec_name = "wm9713-aux", }, { -- cgit v1.2.1 From 43c63188821dc21b2af23a40a18faea6e386e90a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Eric=20B=C3=A9nard?= Date: Fri, 25 Feb 2011 13:47:46 +0100 Subject: eukrea-tlv320: fix platform_name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit f0fba2ad1b6b53d5360125c41953b7afcd6deff0 included a mistake on the name of the platform in the snd_soc_dai_link structure. Signed-off-by: Eric Bénard Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/imx/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index e20c9e1457c0..1e9bccae4e80 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -79,7 +79,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-fiq-pcm-audio.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, -- cgit v1.2.1 From f0ce27996217d06207c8bfda1b1bbec2fbab48c6 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 28 Feb 2011 15:58:07 +0100 Subject: ALSA: HDA: Realtek: Fixup jack detection to input subsystem This patch fixes an error in the jack detection reporting, causing the jack detection sometimes not to be reported correctly to the input subsystem. It should apply to several Realtek codecs. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3328a259a242..c052fc5ad0c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1133,11 +1133,8 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) nid = spec->autocfg.hp_pins[i]; if (!nid) break; - if (snd_hda_jack_detect(codec, nid)) { - spec->jack_present = 1; - break; - } - alc_report_jack(codec, spec->autocfg.hp_pins[i]); + alc_report_jack(codec, nid); + spec->jack_present |= snd_hda_jack_detect(codec, nid); } mute = spec->jack_present ? HDA_AMP_MUTE : 0; -- cgit v1.2.1 From f07eb223a081b278be02a58394cb5fd66f1a1bbd Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Tue, 22 Feb 2011 21:05:04 -0700 Subject: dt/sound: Eliminate users of of_platform_{,un}register_driver Get rid of users of of_platform_driver in drivers/sound. The of_platform_{,un}register_driver functions are going away, so the users need to be converted to using the platform_bus_type directly. Signed-off-by: Grant Likely --- sound/soc/fsl/fsl_dma.c | 9 ++++----- sound/soc/fsl/fsl_ssi.c | 9 ++++----- sound/soc/fsl/mpc5200_dma.c | 24 +++++++++++------------- sound/soc/fsl/mpc5200_psc_ac97.c | 9 ++++----- sound/soc/fsl/mpc5200_psc_i2s.c | 9 ++++----- sound/sparc/amd7930.c | 8 ++++---- sound/sparc/cs4231.c | 16 ++++++++-------- sound/sparc/dbri.c | 8 ++++---- 8 files changed, 43 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4cf98c03af22..15dac0f20cd8 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -896,8 +896,7 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -static int __devinit fsl_soc_dma_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) { struct dma_object *dma; struct device_node *np = pdev->dev.of_node; @@ -979,7 +978,7 @@ static const struct of_device_id fsl_soc_dma_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); -static struct of_platform_driver fsl_soc_dma_driver = { +static struct platform_driver fsl_soc_dma_driver = { .driver = { .name = "fsl-pcm-audio", .owner = THIS_MODULE, @@ -993,12 +992,12 @@ static int __init fsl_soc_dma_init(void) { pr_info("Freescale Elo DMA ASoC PCM Driver\n"); - return of_register_platform_driver(&fsl_soc_dma_driver); + return platform_driver_register(&fsl_soc_dma_driver); } static void __exit fsl_soc_dma_exit(void) { - of_unregister_platform_driver(&fsl_soc_dma_driver); + platform_driver_unregister(&fsl_soc_dma_driver); } module_init(fsl_soc_dma_init); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4cc167a7aeb8..313e0ccedd5b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -624,8 +624,7 @@ static void make_lowercase(char *s) } } -static int __devinit fsl_ssi_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_ssi_probe(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private; int ret = 0; @@ -774,7 +773,7 @@ static const struct of_device_id fsl_ssi_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); -static struct of_platform_driver fsl_ssi_driver = { +static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", .owner = THIS_MODULE, @@ -788,12 +787,12 @@ static int __init fsl_ssi_init(void) { printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - return of_register_platform_driver(&fsl_ssi_driver); + return platform_driver_register(&fsl_ssi_driver); } static void __exit fsl_ssi_exit(void) { - of_unregister_platform_driver(&fsl_ssi_driver); + platform_driver_unregister(&fsl_ssi_driver); } module_init(fsl_ssi_init); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f92dca07cd35..fff695ccdd3e 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -368,8 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op, - const struct of_device_id *match) +static int mpc5200_hpcd_probe(struct of_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -511,32 +510,31 @@ static int mpc5200_hpcd_remove(struct of_device *op) } static struct of_device_id mpc5200_hpcd_match[] = { - { - .compatible = "fsl,mpc5200-pcm", - }, + { .compatible = "fsl,mpc5200-pcm", }, {} }; MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); -static struct of_platform_driver mpc5200_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc5200-pcm-audio", - .match_table = mpc5200_hpcd_match, +static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, + .dev = { + .owner = THIS_MODULE, + .name = "mpc5200-pcm-audio", + .of_match_table = mpc5200_hpcd_match, + } }; static int __init mpc5200_hpcd_init(void) { - return of_register_platform_driver(&mpc5200_hpcd_of_driver); + return platform_driver_register(&mpc5200_hpcd_of_driver); } +module_init(mpc5200_hpcd_init); static void __exit mpc5200_hpcd_exit(void) { - of_unregister_platform_driver(&mpc5200_hpcd_of_driver); + platform_driver_unregister(&mpc5200_hpcd_of_driver); } - -module_init(mpc5200_hpcd_init); module_exit(mpc5200_hpcd_exit); MODULE_AUTHOR("Grant Likely "); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 40acc8e2b1ca..ad36b095bb79 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -272,8 +272,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { * - Probe/remove operations * - OF device match table */ -static int __devinit psc_ac97_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_ac97_of_probe(struct platform_device *op) { int rc; struct snd_ac97 ac97; @@ -316,7 +315,7 @@ static struct of_device_id psc_ac97_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_ac97_match); -static struct of_platform_driver psc_ac97_driver = { +static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, .remove = __devexit_p(psc_ac97_of_remove), .driver = { @@ -332,13 +331,13 @@ static struct of_platform_driver psc_ac97_driver = { */ static int __init psc_ac97_init(void) { - return of_register_platform_driver(&psc_ac97_driver); + return platform_driver_register(&psc_ac97_driver); } module_init(psc_ac97_init); static void __exit psc_ac97_exit(void) { - of_unregister_platform_driver(&psc_ac97_driver); + platform_driver_unregister(&psc_ac97_driver); } module_exit(psc_ac97_exit); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9018fa5bf0db..87cf2a5c2b2c 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -150,8 +150,7 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{ * - Probe/remove operations * - OF device match table */ -static int __devinit psc_i2s_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_i2s_of_probe(struct platform_device *op) { int rc; struct psc_dma *psc_dma; @@ -213,7 +212,7 @@ static struct of_device_id psc_i2s_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_i2s_match); -static struct of_platform_driver psc_i2s_driver = { +static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, .remove = __devexit_p(psc_i2s_of_remove), .driver = { @@ -229,13 +228,13 @@ static struct of_platform_driver psc_i2s_driver = { */ static int __init psc_i2s_init(void) { - return of_register_platform_driver(&psc_i2s_driver); + return platform_driver_register(&psc_i2s_driver); } module_init(psc_i2s_init); static void __exit psc_i2s_exit(void) { - of_unregister_platform_driver(&psc_i2s_driver); + platform_driver_unregister(&psc_i2s_driver); } module_exit(psc_i2s_exit); diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index f8bcfc30f800..ad7d4d7d9237 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1002,7 +1002,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit amd7930_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; static int dev_num; @@ -1064,7 +1064,7 @@ static const struct of_device_id amd7930_match[] = { {}, }; -static struct of_platform_driver amd7930_sbus_driver = { +static struct platform_driver amd7930_sbus_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -1075,7 +1075,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - return of_register_platform_driver(&amd7930_sbus_driver); + return platform_driver_register(&amd7930_sbus_driver); } static void __exit amd7930_exit(void) @@ -1092,7 +1092,7 @@ static void __exit amd7930_exit(void) amd7930_list = NULL; - of_unregister_platform_driver(&amd7930_sbus_driver); + platform_driver_unregister(&amd7930_sbus_driver); } module_init(amd7930_init); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index c276086c3b57..0e618f82808c 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1856,7 +1856,7 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -2048,7 +2048,7 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_ebus_probe(struct platform_device *op) { struct snd_card *card; int err; @@ -2072,16 +2072,16 @@ static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct } #endif -static int __devinit cs4231_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_probe(struct platform_device *op) { #ifdef EBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "ebus")) - return cs4231_ebus_probe(op, match); + return cs4231_ebus_probe(op); #endif #ifdef SBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "sbus") || !strcmp(op->dev.of_node->parent->name, "sbi")) - return cs4231_sbus_probe(op, match); + return cs4231_sbus_probe(op); #endif return -ENODEV; } @@ -2108,7 +2108,7 @@ static const struct of_device_id cs4231_match[] = { MODULE_DEVICE_TABLE(of, cs4231_match); -static struct of_platform_driver cs4231_driver = { +static struct platform_driver cs4231_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -2120,12 +2120,12 @@ static struct of_platform_driver cs4231_driver = { static int __init cs4231_init(void) { - return of_register_platform_driver(&cs4231_driver); + return platform_driver_register(&cs4231_driver); } static void __exit cs4231_exit(void) { - of_unregister_platform_driver(&cs4231_driver); + platform_driver_unregister(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 39cd5d69d051..73f9cbacc077 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2592,7 +2592,7 @@ static void snd_dbri_free(struct snd_dbri *dbri) (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit dbri_probe(struct platform_device *op) { struct snd_dbri *dbri; struct resource *rp; @@ -2686,7 +2686,7 @@ static const struct of_device_id dbri_match[] = { MODULE_DEVICE_TABLE(of, dbri_match); -static struct of_platform_driver dbri_sbus_driver = { +static struct platform_driver dbri_sbus_driver = { .driver = { .name = "dbri", .owner = THIS_MODULE, @@ -2699,12 +2699,12 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_platform_driver(&dbri_sbus_driver); + return platform_driver_register(&dbri_sbus_driver); } static void __exit dbri_exit(void) { - of_unregister_platform_driver(&dbri_sbus_driver); + platform_driver_unregister(&dbri_sbus_driver); } module_init(dbri_init); -- cgit v1.2.1 From c790ad31a28671b9b478f5d4db2f8b05dabaae4e Mon Sep 17 00:00:00 2001 From: Chih-Wei Huang Date: Fri, 25 Feb 2011 11:14:31 +0800 Subject: ALSA: hda - Fix unable to record issue on ASUS N82JV The codec of N82JV is ALC269VB. Signed-off-by: Chih-Wei Huang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c052fc5ad0c9..4261bb8eec1d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15012,7 +15012,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC), SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC), SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC), SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC), -- cgit v1.2.1 From 3ee845acba58549578d03a46ed307c0a56c7f777 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Mar 2011 20:05:23 +0000 Subject: ASoC: Fix WM9081 platform data initialisation It went AWOL in the multi-component conversion. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm9081.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 43825b2102a5..cce704c275c6 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -1341,6 +1342,10 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_type = SND_SOC_I2C; wm9081->control_data = i2c; + if (dev_get_platdata(&i2c->dev)) + memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), + sizeof(wm9081->retune)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) -- cgit v1.2.1 From a3cff81ac19ace1ce5ba3fc88e46aea2cb4ebe1a Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 28 Feb 2011 17:24:11 +0000 Subject: ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly Since we began using the late clock disable functionality, ensure that we don't disable the clock if any of the ADC or DAC paths are still enabled. This happens when we have simultaneous playback and recording. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ebaee5ca7434..9e91525eddaa 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -110,6 +110,9 @@ struct wm8994_priv { unsigned int aif1clk_enable:1; unsigned int aif2clk_enable:1; + + unsigned int aif1clk_disable:1; + unsigned int aif2clk_disable:1; }; static int wm8994_readable(unsigned int reg) @@ -1015,14 +1018,18 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) + if (wm8994->aif1clk_enable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); - if (wm8994->aif2clk_enable) + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } break; } @@ -1037,15 +1044,15 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_enable) { + if (wm8994->aif1clk_disable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_enable = 0; + wm8994->aif1clk_disable = 0; } - if (wm8994->aif2clk_enable) { + if (wm8994->aif2clk_disable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_enable = 0; + wm8994->aif2clk_disable = 0; } break; } @@ -1063,6 +1070,9 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif1clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif1clk_disable = 1; + break; } return 0; @@ -1078,6 +1088,9 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif2clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif2clk_disable = 1; + break; } return 0; -- cgit v1.2.1 From 04d286819ba499839d04cbf847f2ea28d5cf4296 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Mar 2011 11:47:10 +0000 Subject: ASoC: WM8994: Ensure late enable events are processed for the ADCs Ensure that the ADCs are provided with a clock as the previous patch "ASoC: WM8994: Improve playback robustness" did not handle this case properly. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9e91525eddaa..4afbe3b2e443 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1096,6 +1096,13 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int adc_mux_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1416,6 +1423,18 @@ SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; +static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = { +SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = { +SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1510,9 +1529,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), -SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), - SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), @@ -3293,11 +3309,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, ARRAY_SIZE(wm8994_dac_revd_widgets)); } else { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, ARRAY_SIZE(wm8994_dac_widgets)); } -- cgit v1.2.1 From c8900a0fad5ae9f4823451de17ba5dec6653ac74 Mon Sep 17 00:00:00 2001 From: Richard Samson Date: Thu, 3 Mar 2011 12:46:13 +0100 Subject: ALSA: hda - add new Fermi 5xx codec IDs to snd-hda Added the missing HDMI codec IDs for new Nvidia stuff. Note that ID 0x17 isn't assigned to anything so far, as suggested by Stephen. [Modified to get rid of 0x17 by tiwai] Signed-off-by: Richard Samson Acked-by: Acked-By: Stephen Warren Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a58767736727..ec0fa2dd0a27 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1634,6 +1634,9 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +/* 17 is known to be absent */ { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, @@ -1676,6 +1679,8 @@ MODULE_ALIAS("snd-hda-codec-id:10de0011"); MODULE_ALIAS("snd-hda-codec-id:10de0012"); MODULE_ALIAS("snd-hda-codec-id:10de0013"); MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0015"); +MODULE_ALIAS("snd-hda-codec-id:10de0016"); MODULE_ALIAS("snd-hda-codec-id:10de0018"); MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); -- cgit v1.2.1 From 38c07641905c0db58e800ea974cd9158717c6610 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 3 Mar 2011 14:54:19 +0100 Subject: ALSA: hda - Don't set to D3 in Cirrus errata init verbs The errata init verbs for CS42xx codecs contain the verbs to set the power-state of SPDIF nodes to D3, which seem to break the SPDIF output on some MacBooks. Since this is executed during the power-up initialization, we shouldn't turn them down there. Reported-by: Arun Raghavan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a07b031090d8..067982f4f182 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1039,9 +1039,11 @@ static struct hda_verb cs_errata_init_verbs[] = { {0x11, AC_VERB_SET_PROC_COEF, 0x0008}, {0x11, AC_VERB_SET_PROC_STATE, 0x00}, +#if 0 /* Don't to set to D3 as we are in power-up sequence */ {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */ {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */ /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */ +#endif {} /* terminator */ }; -- cgit v1.2.1 From ffd6eae2a0d18ca4a741615292a9c9ce904307fb Mon Sep 17 00:00:00 2001 From: Abhilash K V Date: Tue, 8 Mar 2011 21:02:43 +0530 Subject: ASoC: AM3517: Update codec name after multi-component update The i2c client device name (".2-001a" in this case, including the separator period) for the AIC23 codec on the TI AM3517-EVM was appended to the codec_name member of am3517evm_dai to resolve the names mismatch happening in soc_bind_dai_link(), due to which the card was not getting registered. Signed-off-by: Abhilash K V Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/am3517evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 161750443ebc..73dde4a1adc3 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = { .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", - .codec_name = "tlv320aic23-codec", + .codec_name = "tlv320aic23-codec.2-001a", .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; -- cgit v1.2.1 From 823dba5191220fc94b83dc0b3f2178ff0842e294 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:01:18 +0000 Subject: ASoC: Fix broken bitfield definitions in WM8978 Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8978.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703f..8dfb0a0da673 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 15, 0), + WM8978_DAC_LIMITER_2, 0, 12, 0), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), - SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0), SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", alc3), - SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), - SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", @@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), /* DAC / ADC oversampling */ - SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), - SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, + 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, + 5, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ -- cgit v1.2.1 From 28e868081086c495c897a48c50d2d5187ef677d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 19:29:53 +0000 Subject: ASoC: Use the correct DAPM context when cleaning up final widget set Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25e54230cc6a..1790f83ee665 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -941,7 +941,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(dapm, &pending); + dapm_seq_run_coalesced(cur_dapm, &pending); } static void dapm_widget_update(struct snd_soc_dapm_context *dapm) -- cgit v1.2.1 From 0627bd2575a30a83901b79d7bcf2ca1fa09fbb8b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 19:09:17 +0000 Subject: ASoC: Fix typo in late revision WM8994 DAC2R name Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4afbe3b2e443..d92673314f43 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1418,7 +1418,7 @@ SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0, static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = { SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), -SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; -- cgit v1.2.1 From 7c2de633863fcd46537d9ddbf5a9701f48225268 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 19:10:15 +0000 Subject: ASoC: Ensure WM8958 gets all WM8994 late revision widgets Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d92673314f43..c6c958ee5d59 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3325,6 +3325,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); break; @@ -3350,6 +3356,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); snd_soc_dapm_add_routes(dapm, wm8958_intercon, ARRAY_SIZE(wm8958_intercon)); break; -- cgit v1.2.1