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Message-ID: <s5hvcnhyqqc.wl%tiwai@suse.de>
Date: Wed, 08 Feb 2012 22:12:11 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...com>, linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes #2
Linus,
The following changes since commit b5bcc189401c815988b7dd37611fc56f40c9139d:
ALSA: hda - Disable dynamic-power control for VIA as default (2012-02-02 10:34:06 +0100)
are available in the git repository at:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound-fixes
for you to fetch changes up to 982d411c303a475424c67966990b5803cc536319:
Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus (2012-02-08 21:29:38 +0100)
----------------------------------------------------------------
sound fixes #2 for 3.3-rc3
A collection of small fixes, mostly for regressions.
In addition, a few ASoC wm8994 updates are included, too.
----------------------------------------------------------------
Axel Lin (1):
ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode
Clemens Ladisch (2):
ALSA: usb-audio: add Edirol UM-3G support
ALSA: oxygen, virtuoso: fix exchanged L/R volumes of aux and CD inputs
Jaroslav Kysela (1):
ALSA: hda - add support for Uniwill ECS M31EI notebook
Jesper Juhl (1):
ALSA: emu8000: Remove duplicate linux/moduleparam.h include from emu8000_patch.c
Mark Brown (7):
ASoC: wm_hubs: Fix routing of input PGAs to line output mixer
ASoC: wm8994: Remove ASoC level register cache sync
ASoC: core: Better support for idle_bias_off suspend ignores
ASoC: wm_hubs: Correct line input to line output 2 paths
ASoC: wm8994: Enabling VMID should take a runtime PM reference
ASoC: wm8994: Fix typo in VMID ramp setting
ASoC: wm8994: Disable line output discharge prior to ramping VMID
Susan Gao (1):
ASoC: wm8962: Fix word length configuration
Takashi Iwai (5):
ALSA: hda/realtek - Add missing Bass and CLFE as vmaster slaves
ALSA: hda/realtek - Fix a wrong condition
Merge tag 'asoc-3.3' of git://git.kernel.org/.../broonie/sound into for-linus
ALSA: hda - Fix error handling in patch_ca0132.c
Merge tag 'asoc-3.3' of git://git.kernel.org/.../broonie/sound into for-linus
sound/isa/sb/emu8000_patch.c | 1 -
sound/pci/hda/patch_ca0132.c | 33 +++++++++++++++++++--------------
sound/pci/hda/patch_realtek.c | 7 ++++++-
sound/pci/oxygen/oxygen_mixer.c | 25 ++++++++++++++-----------
sound/soc/codecs/cs42l73.c | 2 +-
sound/soc/codecs/wm8962.c | 6 +++---
sound/soc/codecs/wm8994.c | 16 ++++++++++------
sound/soc/codecs/wm_hubs.c | 8 ++++----
sound/soc/soc-core.c | 11 +++++++++++
sound/usb/quirks-table.h | 8 ++++++++
10 files changed, 76 insertions(+), 41 deletions(-)
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index e09f144..c99c607 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -22,7 +22,6 @@
#include "emu8000_local.h"
#include <asm/uaccess.h>
#include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
static int emu8000_reset_addr;
module_param(emu8000_reset_addr, int, 0444);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 35abe3c..21d91d5 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;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a8e82be..9350f3c 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,
};
@@ -2318,7 +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;
@@ -5623,6 +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),
{}
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 26c7e8b..c0dbb52 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);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 9d38db8..78979b3 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;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index bda3da8..29c4b02 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;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 93d27b6..ec69a6c 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",
@@ -783,7 +785,12 @@ 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));
+
+ /* 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,
@@ -837,6 +844,8 @@ 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,
@@ -2753,11 +2762,6 @@ 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++) {
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index ea26724..8a68cea 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("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),
};
@@ -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" },
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b5ecf6d..92cee24 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;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 8edc503..d89ab4c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1618,6 +1618,14 @@ 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),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
--
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