From 7e1f7c8a6e517900cd84da1b8ae020f08f286c3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 17:29:36 +0100 Subject: [PATCH 01/20] ASoC: dapm: Ensure power gets managed for line widgets Line widgets had not been included in either the power up or power down sequences so if a widget had an event associated with it that event would never be run. Fix this minimally by adding them to the sequences, we should probably be doing away with the specific widget types as they all have the same priority anyway. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6241490fff30..dc7dbfe61cd0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -67,6 +67,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_line] = 10, [snd_soc_dapm_post] = 11, }; @@ -75,6 +76,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, From 86fc49982369f6918dd9c6eeb70b38ab2303ed0a Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 12 Apr 2012 21:54:34 +0200 Subject: [PATCH 02/20] ASoC: cs42l73: don't use negative array index If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using the (negative) return value as array index on the very next line of code - that's bad. Catch the negative return value and propagate it to the caller (which checks for it) and things are a bit more sane :-) Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3e0e95..07c44b71f096 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) /* MCLKX -> MCLK */ mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / cs42l73_mclkx_coeffs[mclkx_coeff].ratio; From a7dbb603423d499acacefb5fad65d2b406f16370 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 17 Apr 2012 18:00:11 +0100 Subject: [PATCH 03/20] ASoC: core: Fix card RTD count for deferred probe. Currently we increment the number of RTD's per card during the DAI link bind. This can cause an incorrect RTD count when we cannot find a component and defer the probe (and hence perform the DAI link bind for the card again). Fix the count so that it is cleared before every card registration and bind attempt. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8d2ebf502df4..3a4e93e52b6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3119,6 +3119,7 @@ int snd_soc_register_card(struct snd_soc_card *card) GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) From cdf27f373781d8740b874b0b5c18142df32ebb52 Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Tue, 17 Apr 2012 19:13:04 -0700 Subject: [PATCH 04/20] ASoC: fsi: update for dmaengine prep_slave_sg fallout. Leading up to the ->device_prep_slave_sg change in 185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was added in place to guard against the API change, though the fsi driver wasn't updated in the process (presumably its dmaengine support hadn't been merged yet at the time). This trivially switches over to the new wrapper and gets it building again. Signed-off-by: Paul Mundt Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 378cc5b056d7..74ed2dffbffd 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data) sg_dma_address(&sg) = buf; sg_dma_len(&sg) = len; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { - dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; } From 5ac57550f279c3d991ef0b398681bcaca18169f7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 20 Apr 2012 10:01:46 +0200 Subject: [PATCH 05/20] ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E According to the reporter, external mic starts to work if the laptop-dmic model is used. According to BIOS pin config, all pins are consistent with the alc269vb_laptop_dmic fixup, except for the external mic, which is not present. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/950490 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e65e35433055..818f90bc7d57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6109,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), From 1a38336b8611a04f0a624330c1f815421f4bf5f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 19:47:11 +0100 Subject: [PATCH 06/20] ASoC: wm8994: Improve sequencing of AIF channel enables This ensures a clean startup of the channels, without this change some use cases could result in issues in a small proportion of cases. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 278 ++++++++++++++++++++++++++++++-------- 1 file changed, 223 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642af052..6c1fe3afd4b5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1000,6 +1000,204 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +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 *control = codec->control_data; + int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int dac; + int adc; + int val; + + switch (control->type) { + case WM8994: + case WM8958: + mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA; + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); + if ((val & WM8994_AIF1ADCL_SRC) && + (val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA; + else if (!(val & WM8994_AIF1ADCL_SRC) && + !(val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + else + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2); + if ((val & WM8994_AIF1DACL_SRC) && + (val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA; + else if (!(val & WM8994_AIF1DACL_SRC) && + !(val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + else + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask, + WM8994_AIF1ADC1R_ENA | + WM8994_AIF1ADC1L_ENA | + WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC2L_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask, + WM8994_AIF1DAC1R_ENA | + WM8994_AIF1DAC1L_ENA | + WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC2L_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF2DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF1DSPCLK_ENA, val); + 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; + int dac; + int adc; + int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1); + if ((val & WM8994_AIF2ADCL_SRC) && + (val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCR_ENA; + else if (!(val & WM8994_AIF2ADCL_SRC) && + !(val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCL_ENA; + else + adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA; + + + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2); + if ((val & WM8994_AIF2DACL_SRC) && + (val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACR_ENA; + else if (!(val & WM8994_AIF2DACL_SRC) && + !(val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACL_ENA; + else + dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF1DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF2DSPCLK_ENA, val); + break; + } + + return 0; +} + +static int aif1clk_late_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; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif1clk_disable = 1; + break; + } + + return 0; +} + +static int aif2clk_late_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; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif2clk_disable = 1; + break; + } + + return 0; +} + static int late_enable_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1009,12 +1207,14 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (wm8994->aif1clk_enable) { + aif1clk_ev(w, kcontrol, event); snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); wm8994->aif1clk_enable = 0; } if (wm8994->aif2clk_enable) { + aif2clk_ev(w, kcontrol, event); snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); @@ -1040,11 +1240,13 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, if (wm8994->aif1clk_disable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); + aif1clk_ev(w, kcontrol, event); wm8994->aif1clk_disable = 0; } if (wm8994->aif2clk_disable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); + aif2clk_ev(w, kcontrol, event); wm8994->aif2clk_disable = 0; } break; @@ -1053,42 +1255,6 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, 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; - case SND_SOC_DAPM_POST_PMD: - wm8994->aif1clk_disable = 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; - case SND_SOC_DAPM_POST_PMD: - wm8994->aif2clk_disable = 1; - break; - } - - return 0; -} - static int adc_mux_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1385,9 +1551,9 @@ 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_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_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_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_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, @@ -1416,8 +1582,10 @@ 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), +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -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("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 9, 0), + 0, SND_SOC_NOPM, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 8, 0), + 0, SND_SOC_NOPM, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, + SND_SOC_NOPM, 9, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, + SND_SOC_NOPM, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 11, 0), + 0, SND_SOC_NOPM, 11, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 10, 0), + 0, SND_SOC_NOPM, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, + SND_SOC_NOPM, 11, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, + SND_SOC_NOPM, 10, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, @@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, dac1r_mix, ARRAY_SIZE(dac1r_mix)), SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 13, 0), + SND_SOC_NOPM, 13, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 12, 0), + SND_SOC_NOPM, 12, 0), SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev, + SND_SOC_NOPM, 13, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, + SND_SOC_NOPM, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), From de050acaa1fdba4852cb195baf2bfed75368e0be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Apr 2012 20:28:10 +0100 Subject: [PATCH 07/20] ASoC: wm_hubs: Make sure we don't disable differential line outputs While we need to clean up unused single ended line outputs we don't want to do this if the outputs are in differential mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f2886339c..6c028c470601 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); - int val; + int mask, val; switch (level) { case SND_SOC_BIAS_STANDBY: @@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; + mask = 0; + + if (hubs->lineout1_se) + mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; if (hubs->lineout1_se && hubs->lineout1n_ena) val |= WM8993_LINEOUT1N_ENA; @@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, val |= WM8993_LINEOUT2P_ENA; snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, - WM8993_LINEOUT1N_ENA | - WM8993_LINEOUT1P_ENA | - WM8993_LINEOUT2N_ENA | - WM8993_LINEOUT2P_ENA, - val); + mask, val); /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, From ddb6706af3cd372194cecd2cc61950519df620d7 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Apr 2012 01:11:09 -0300 Subject: [PATCH 08/20] ASoC: dt: sgtl5000.txt: Add description for 'reg' field Add description for 'reg' field. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 2c3cd413f042..9cc44449508d 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -3,6 +3,8 @@ Required properties: - compatible : "fsl,sgtl5000". +- reg : the I2C address of the device + Example: codec: sgtl5000@0a { From c34ce320d9fe328e3272def20b152f39ccfa045e Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 24 Apr 2012 15:24:43 +0800 Subject: [PATCH 09/20] ASoC: core: check of_property_count_strings failure Signed-off-by: Richard Zhao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4e93e52b6d..b390f00b4e99 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3631,10 +3631,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; From a3a53fe1545a87337cc539f415810128bbdad465 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 25 Apr 2012 11:29:47 +0200 Subject: [PATCH 10/20] ASoC: bf5xx-ssm2602: Set DAI format Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC machine drivers to use the dai_links dai_fmt field to setup their DAI format. For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt, but missed to set the dai_links dai_fmt field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73f8778..b39ad356b92b 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; From e875c1e3e758447ba81ca450d89434b3b0496d37 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Eric=20B=C3=A9nard?= Date: Sun, 29 Apr 2012 17:37:57 +0200 Subject: [PATCH 11/20] ASoC: tlv312aic23: unbreak resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit * commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to a bug preventing resumeof the codec as regmap expects a 9 bits data register but 0xFFFF is passed in tlv320aic23_set_bias_level and this values gets cached preventing any write to the TLV320AIC23_PWR register as the final value produced by regmap is (register << 9) | value * this patch solves the problem by only working on the 9 bits the register contains. Signed-off-by: Eric Bénard Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f91a653..df1e07ffac32 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; From 30facd4d51d630b6cba386badd7f42456962089b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Apr 2012 20:11:55 +0100 Subject: [PATCH 12/20] ASoC: wm8350: Don't use locally allocated codec struct The core allocates the live copies, we shouldn't try to duplicate it and were buggy trying to do so as we were using uninitialised data for the control data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c9591ec05..aa12c6b6beeb 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ From 06412088ce98f745405b8f65cfc51ddd6b842bbf Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Heiko=20St=C3=BCbner?= Date: Mon, 30 Apr 2012 13:17:21 +0200 Subject: [PATCH 13/20] ASoC: s3c2412-i2s: Fix dai registration As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai. Without this call the snd_soc_dai_ops structure isn't initialised correctly. Signed-off-by: Heiko Stuebner Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 72185078ddf8..79fbeea99d46 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) From fad9365bcc2a69ae16adc092e8ac192354980665 Mon Sep 17 00:00:00 2001 From: Oleg Matcovschi Date: Tue, 24 Apr 2012 19:02:02 -0700 Subject: [PATCH 14/20] ASoC: omap-pcm: Free dma buffers in case of error. Signed-off-by: Oleg Matcovschi Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd352d342..5a649da9122a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } From c914f55f7cdfafe9d7d5b248751902c7ab57691e Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Mon, 30 Apr 2012 19:39:22 +0100 Subject: [PATCH 15/20] ALSA: echoaudio: Remove incorrect part of assertion This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills Cc: Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio_dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 64417a733220..d8c670c9d62c 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip) const struct firmware *fw; int box_type, err; - if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page)) + if (snd_BUG_ON(!chip->comm_page)) return -EPERM; /* See if the ASIC is present and working - only if the DSP is already loaded */ From f5c53d898cc34079373c63a290528963db31d681 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 10:07:33 +0200 Subject: [PATCH 16/20] ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G Acer Aspire 5739G requires the same fix-up for 4930G to support the surround / bass speakers. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 818f90bc7d57..27d0f637864a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5405,6 +5405,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G", + ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), From bca40138558f0b39357fd1ca477868e4f52f4b1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 11:13:14 +0200 Subject: [PATCH 17/20] ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo Reported-by: Philipp Matthias Hahn Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 27d0f637864a..8ea613eb73f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5440,6 +5440,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), From 42eb92380f73f28e3a5a51973af1183fdbac82f2 Mon Sep 17 00:00:00 2001 From: Andre Schramm Date: Mon, 7 May 2012 18:52:51 +0200 Subject: [PATCH 18/20] ALSA: hdsp - Provide ioctl_compat snd_hdsp uses its own ioctls to acquire config- and status information. Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels. Signed-off-by: Andre Schramm Reviewed-by: Adrian Knoth Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b68cdec03b9e..0b2aea2ce172 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) strcpy(hw->name, "HDSP hwdep interface"); hw->ops.ioctl = snd_hdsp_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl; return 0; } From af741c150f66db8d1da6f82ac75e2571f7f1dd38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 18:09:48 +0200 Subject: [PATCH 19/20] ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup The call for alc_auto_parse_customize_define() must be done after the fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work properly (e.g. HP RP5800 with ALC662 codec). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ea613eb73f0..7810913d07a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5641,13 +5641,13 @@ static int patch_alc262(struct hda_codec *codec) snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80); } #endif - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x0a, 10); alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); if (err < 0) @@ -6252,8 +6252,6 @@ static int patch_alc269(struct hda_codec *codec) spec->mixer_nid = 0x0b; - alc_auto_parse_customize_define(codec); - err = alc_codec_rename_from_preset(codec); if (err < 0) goto error; @@ -6286,6 +6284,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_fixup_tbl, alc269_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) @@ -6862,8 +6862,6 @@ static int patch_alc662(struct hda_codec *codec) /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x04, 15); err = alc_codec_rename_from_preset(codec); @@ -6880,6 +6878,9 @@ static int patch_alc662(struct hda_codec *codec) alc_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) From 619a341b78f17fb86d92e89c04612676cd05e26f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 May 2012 16:30:59 +0200 Subject: [PATCH 20/20] Revert "ALSA: hda - Set codec to D3 forcibly even if not used" This reverts commit 785f857d1cb0856b612b46a0545b74aa2596e44a. The commit causes a problem with the wrong D3 state after suspend because the call of hda_set_power_state() involves with the power-up sequence, which changes the power_count, and this confuses the resume sequence that checks the power_count as well. Originally, this go-to-D3 sequence should be a simple task without the power-up sequence. But, it'd need some proper sanity checks in the case of power-saved state, so it's not too easy to write now in the 3.4-rc cycle. In short, the safest option now is to revert this affecting commit. Of course, we need to clean up and robustify the power-saving code better for 3.5 kernel. Reported-by: Konstantin Khlebnikov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ---- sound/pci/hda/hda_intel.c | 14 +++++++++++++- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a8fcc4c15f8..841475cc13b6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5444,10 +5444,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - else /* forcibly change the power to D3 even if not used */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D3); if (codec->patch_ops.post_suspend) codec->patch_ops.post_suspend(codec); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c19e71a94e1b..6e958bf94191 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2351,6 +2351,17 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ +static int snd_hda_codecs_inuse(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + if (snd_hda_codec_needs_resume(codec)) + return 1; + } + return 0; +} + static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2397,7 +2408,8 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + if (snd_hda_codecs_inuse(chip->bus)) + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0);