From ee76744c51ec342df9822b4a85dbbfc3887b6d60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 11:55:32 +0000 Subject: [PATCH 01/16] ASoC: wm_hubs: Fix routing of input PGAs to line output mixer IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1 and IN2 to LINEOUT2. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ea2672455d07..c1a3f8c39691 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; @@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, From 67f97f5c3edad35c4d37a94f994c76111a177fb6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 14:51:29 +0000 Subject: [PATCH 02/16] ASoC: wm8994: Remove ASoC level register cache sync Now we've switched over to regmap the ASoC level cache sync will be ineffectual and potentially harmful as there is no longer an ASoC level cache. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b660257..8623950d55f8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2753,11 +2753,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { From 125a25da5729740b7d1dc417a3d5549321baae17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 15:49:10 +0000 Subject: [PATCH 03/16] ASoC: core: Better support for idle_bias_off suspend ignores If an idle_bias_off device is in any state other than off then it is still active for some reason (typically a low power function such as accessory detection). This wasn't an issue when the feature was implemented as we always went to _ON for any active function, subsequent power improvements have changed things. With the modern way of doing things we should overhaul the infrastructure to allow devices to explicitly take references for these functions but that's a much more invasive change and will require driver updates to deploy, this will bring the framework into line with the existing driver set before we do that work. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5ecf6d23214..92cee24ed2dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev) if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: + /* + * If the CODEC is capable of idle + * bias off then being in STANDBY + * means it's doing something, + * otherwise fall through. + */ + if (codec->dapm.idle_bias_off) { + dev_dbg(codec->dev, + "idle_bias_off CODEC on over suspend\n"); + break; + } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; From 2b6712b19531e22455e7fa18371c5ba9eec76699 Mon Sep 17 00:00:00 2001 From: Susan Gao Date: Mon, 30 Jan 2012 13:57:04 -0800 Subject: [PATCH 04/16] ASoC: wm8962: Fix word length configuration Signed-off-by: Susan Gao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bda3da887d7e..29c4b02c4790 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; From 44bed4838dc191988fd1d03deccc3a845705d2de Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 31 Jan 2012 09:49:04 +0800 Subject: [PATCH 05/16] ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode. Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9d38db8f1919..78979b3e0e95 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc &= MCK_SCLK_64FS; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; From 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 23:46:58 +0000 Subject: [PATCH 06/16] ASoC: wm_hubs: Correct line input to line output 2 paths The second line output mixer has the controls for the line input bypasses in the opposite order. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c1a3f8c39691..8a68cea4a3ee 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; From b544d1e0e233f83a2e6d20ee96b54ea272d5d5ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 11:56:35 +0100 Subject: [PATCH 07/16] ALSA: hda/realtek - Add missing Bass and CLFE as vmaster slaves The recent changes in Realtek auto-parser added the new "Bass Speaker" and "CLFE" mixer elements which should be tracked as vmaster slaves, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42720 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be3d2fc..33b6077fcdb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1855,6 +1855,8 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", "PCM Playback Volume", NULL, }; @@ -1870,6 +1872,8 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", "PCM Playback Switch", NULL, }; From 226e01ef0da0b1a4c2c3922fb83ff3f9e4dfb508 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Sun, 5 Feb 2012 01:27:44 +0100 Subject: [PATCH 08/16] ALSA: emu8000: Remove duplicate linux/moduleparam.h include from emu8000_patch.c The header 'linux/moduleparam.h' is included twice in 'sound/isa/sb/emu8000_patch.c'. Once is enough. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_patch.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index e09f144177f5..c99c6078be33 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -22,7 +22,6 @@ #include "emu8000_local.h" #include #include -#include static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); From eedec3d3854a390fc14008f265930f8c22b0373f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Feb 2012 10:24:04 +0100 Subject: [PATCH 09/16] ALSA: hda/realtek - Fix a wrong condition sparse complains that "spec->multiout.dac_nids" is a pointer. sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>) sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids sound/pci/hda/patch_realtek.c:2321:37: right side has type int It was meant to be num_dacs instead of dac_nids. Although the current code still works as expected (when num_dacs is zero, dac_nids should be NULL, too), better to fix now, of course. Reported-by: Dan Carpenter Cc: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33b6077fcdb8..485a83746e5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2322,7 +2322,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback; From db966f8abb9ba74f7d5a7230f51572f52c31c4e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 12:07:08 +0000 Subject: [PATCH 10/16] ASoC: wm8994: Enabling VMID should take a runtime PM reference We can enable VMID independently of the bias in some use cases so we need to ensure that the core device is powered up. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8623950d55f8..81795ebcab17 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + pm_runtime_get_sync(codec->dev); + wm8994->vmid_refcount++; dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", @@ -837,6 +839,8 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); } + + pm_runtime_put(codec->dev); } static int vmid_event(struct snd_soc_dapm_widget *w, From b97f6bfdd1af95681de5a9f652da644a6525e376 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Feb 2012 11:00:53 +0100 Subject: [PATCH 11/16] ALSA: hda - Fix error handling in patch_ca0132.c In patch_ca0132.c, the error returned from chipio_write() isn't checked always. Also, the power-up/down sequence isn't tracked properly in some error paths. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c62908..21d91d580da8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_hp_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, @@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_speaker_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, @@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); if (err < 0) - return err; + goto exit; val = 31 - left_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_L, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_L, data); if (err < 0) - return err; + goto exit; val = 31 - right_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_R, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_R, data); if (err < 0) - return err; + goto exit; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) @@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; err = add_in_volume(codec, spec->dig_in, "IEC958"); + if (err < 0) + return err; } return 0; } From 416846d2b31fc740ed9d5a5ec116964fb43c4358 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 7 Feb 2012 14:18:14 +0100 Subject: [PATCH 12/16] ALSA: hda - add support for Uniwill ECS M31EI notebook This hardware requires same fixup for the node 0x0f like Asus A6Rp. More information: https://bugzilla.redhat.com/show_bug.cgi?id=785417 Signed-off-by: Jaroslav Kysela 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 485a83746e5a..9350f3c3bdf8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5627,6 +5627,7 @@ static const struct alc_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} From 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 4 Feb 2012 20:51:43 +0100 Subject: [PATCH 13/16] ALSA: usb-audio: add Edirol UM-3G support Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8edc5035fc8f..d89ab4c7d44b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1617,6 +1617,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Edirol UM-3G */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = 0, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, { /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), From 2492250e4412c6411324c14ab289629360640b0a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 4 Feb 2012 20:56:47 +0100 Subject: [PATCH 14/16] ALSA: oxygen, virtuoso: fix exchanged L/R volumes of aux and CD inputs The driver accidentally exchanged the left/right fields for stereo AC'97 mixer registers. This affected only the aux and CD inputs because the line input bypasses the AC'97 codec and the mic input is mono; cards without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected. Reported-and-tested-by: Abby Cedar Signed-off-by: Clemens Ladisch Cc: 2.6.31+ Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 26c7e8bcb229..c0dbb52d45be 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); - value->value.integer.value[0] = 31 - (reg & 0x1f); - if (stereo) - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (!stereo) { + value->value.integer.value[0] = 31 - (reg & 0x1f); + } else { + value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f); + value->value.integer.value[1] = 31 - (reg & 0x1f); + } return 0; } @@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); oldreg = oxygen_read_ac97(chip, codec, index); - newreg = oldreg; - newreg = (newreg & ~0x1f) | - (31 - (value->value.integer.value[0] & 0x1f)); - if (stereo) - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[1] & 0x1f)) << 8); - else - newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); + if (!stereo) { + newreg = oldreg & ~0x1f; + newreg |= 31 - (value->value.integer.value[0] & 0x1f); + } else { + newreg = oldreg & ~0x1f1f; + newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8; + newreg |= 31 - (value->value.integer.value[1] & 0x1f); + } change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); From f647e1526fd6c7c8ab720781c40d11e11f930e93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 17:24:19 +0000 Subject: [PATCH 15/16] ASoC: wm8994: Fix typo in VMID ramp setting The VMID ramp rate is supposed to be 0x3, not 11b. Fix that. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 81795ebcab17..b75a65273dab 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -785,7 +785,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, From a7c4183be2d6a7da8c97a9b671b5f3aed321127e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 14:18:29 +0000 Subject: [PATCH 16/16] ASoC: wm8994: Disable line output discharge prior to ramping VMID Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b75a65273dab..ec69a6c152fe 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -787,6 +787,11 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); + /* Remove discharge for line out */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, 0); + /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, WM8994_BIAS_ENA |