From b90bf1de7cb65e7f61798fcfbcf74ae72207b0dc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Jan 2012 11:42:55 +0100
Subject: ALSA: hda/realtek - Avoid multi-ios conflicting with multi-speakers
When a machine has multiple speakers, we don't need to create the
controls for multi-ios. Check the number of primary outputs beforehand.
Note that this workaround might not work always with new codecs in
future; this assumes that both speakers and multi-io jacks share the
same mixers/DACs. If they are routed with different mixers, the
individual mixer controls should be needed. But, so far, this doesn't
happen with the existing ALC codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 5e82acf77c5a..61ccbe832b75 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3233,7 +3233,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
int i, err, noutputs;
noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
+ if (spec->multi_ios > 0 && cfg->line_outs < 3)
noutputs += spec->multi_ios;
for (i = 0; i < noutputs; i++) {
--
cgit v1.2.3-70-g09d2
From f21d78e2698b6380a5387461e3b126bb2dee23aa Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Jan 2012 12:10:29 +0100
Subject: ALSA: hda/realtek - Avoid conflict of unsol-events with static quirks
The recently added jack-kctl support sets the unsol event tags
dynamically, while static quirks usually set the fixed tags in the
init_verbs array. Due to this conflict, the own unsol event handler
can't retrieve the tag and handle it properly any more.
For fixing this, avoid calling snd_hda_jack_add_kctls() for static
quirks, and always let them use own handlers instead of the standard
one for the auto-pareser.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/alc880_quirks.c | 17 +++++++++++-----
sound/pci/hda/alc882_quirks.c | 15 ++++++++++-----
sound/pci/hda/patch_realtek.c | 45 +++++++++++++++++++++++++++++--------------
3 files changed, 53 insertions(+), 24 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index 5b68435d195b..501501ef36a9 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
- switch (res >> 28) {
+ res >>= 28;
+ switch (res) {
case ALC_MIC_EVENT:
alc88x_simple_mic_automute(codec);
break;
default:
- alc_sku_unsol_event(codec, res);
+ alc_exec_unsol_event(codec, res);
break;
}
}
+static void alc880_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ alc_exec_unsol_event(codec, res >> 28);
+}
+
static void alc880_uniwill_p53_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
- if ((res >> 28) == ALC_DCVOL_EVENT)
+ res >>= 28;
+ if (res == ALC_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
else
- alc_sku_unsol_event(codec, res);
+ alc_exec_unsol_event(codec, res);
}
/*
@@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_lg_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc880_unsol_event,
.setup = alc880_lg_setup,
.init_hook = alc_hp_automute,
#ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index bdf0ed4ab3e2..bb364a53f546 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
alc889A_mb31_automute(codec);
}
+static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ alc_exec_unsol_event(codec, res >> 26);
+}
+
/*
* configuration and preset
*/
@@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = {
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mba21_setup,
.init_hook = alc_hp_automute,
},
@@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mbp3_setup,
.init_hook = alc_hp_automute,
},
@@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mb5_setup,
.init_hook = alc_hp_automute,
},
@@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &macmini3_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_macmini3_setup,
.init_hook = alc_hp_automute,
},
@@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &alc889A_imac91_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_imac91_setup,
.init_hook = alc_hp_automute,
},
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 61ccbe832b75..2326bf379525 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -621,17 +621,10 @@ static void alc_mic_automute(struct hda_codec *codec)
alc_mux_select(codec, 0, spec->int_mic_idx, false);
}
-/* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+/* handle the specified unsol action (ALC_XXX_EVENT) */
+static void alc_exec_unsol_event(struct hda_codec *codec, int action)
{
- struct alc_spec *spec = codec->spec;
- if (codec->vendor_id == 0x10ec0880)
- res >>= 28;
- else
- res >>= 26;
- if (spec->use_jack_tbl)
- res = snd_hda_jack_get_action(codec, res);
- switch (res) {
+ switch (action) {
case ALC_HP_EVENT:
alc_hp_automute(codec);
break;
@@ -645,6 +638,19 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
snd_hda_jack_report_sync(codec);
}
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ struct alc_spec *spec = codec->spec;
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ if (spec->use_jack_tbl)
+ res = snd_hda_jack_get_action(codec, res);
+ alc_exec_unsol_event(codec, res);
+}
+
/* call init functions of standard auto-mute helpers */
static void alc_inithook(struct hda_codec *codec)
{
@@ -1883,7 +1889,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = {
};
#endif
-static int alc_build_controls(struct hda_codec *codec)
+static int __alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct snd_kcontrol *kctl = NULL;
@@ -2029,11 +2035,16 @@ static int alc_build_controls(struct hda_codec *codec)
alc_free_kctls(codec); /* no longer needed */
- err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
+ return 0;
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err = __alc_build_controls(codec);
if (err < 0)
return err;
-
- return 0;
+ return snd_hda_jack_add_kctls(codec, &spec->autocfg);
}
@@ -4168,6 +4179,8 @@ static int patch_alc880(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc880_loopbacks;
@@ -4297,6 +4310,8 @@ static int patch_alc260(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
spec->shutup = alc_eapd_shutup;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -4691,6 +4706,8 @@ static int patch_alc882(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
--
cgit v1.2.3-70-g09d2
From a7309792c4e313d4e4c30084dc0ecbc834082433 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Jan 2012 15:03:48 +0100
Subject: ALSA: hda/realtek - Remove use_jack_tbl field
Now that all quirks have the own unsol handlers, we don't need to check
use_jack_tbl flag any more. Let's kill it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 5 +----
1 file changed, 1 insertion(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2326bf379525..ddbed9705ef4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -185,7 +185,6 @@ struct alc_spec {
unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
- unsigned int use_jack_tbl:1; /* 1 for model=auto */
/* auto-mute control */
int automute_mode;
@@ -646,8 +645,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
res >>= 28;
else
res >>= 26;
- if (spec->use_jack_tbl)
- res = snd_hda_jack_get_action(codec, res);
+ res = snd_hda_jack_get_action(codec, res);
alc_exec_unsol_event(codec, res);
}
@@ -3915,7 +3913,6 @@ static void set_capture_mixer(struct hda_codec *codec)
static void alc_auto_init_std(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- spec->use_jack_tbl = 1;
alc_auto_init_multi_out(codec);
alc_auto_init_extra_out(codec);
alc_auto_init_analog_input(codec);
--
cgit v1.2.3-70-g09d2
From b9ecc4ee28a5ff5b3997da247cd9df1320c602a9 Mon Sep 17 00:00:00 2001
From: Albert Pool <albertpool@solcon.nl>
Date: Thu, 19 Jan 2012 22:08:50 +0100
Subject: snd-hda-intel: better Alienware M17x R3 quirk
I have been told that this way the rear headphone connector is
working as well; with model=alienware only laptop speakers work.
The subsystem of both controller and codec is 1028:0490.
Signed-off-by: Albert Pool <albertpool@solcon.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3556408d6ece..1a26dbca9483 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
"Alienware M17x", STAC_ALIENWARE_M17X),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
- "Alienware M17x", STAC_ALIENWARE_M17X),
+ "Alienware M17x R3", STAC_DELL_EQ),
{} /* terminator */
};
--
cgit v1.2.3-70-g09d2
From cb0cdebbf8b834110ef67ed9335d5bafed7835df Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Jan 2012 12:14:12 +0100
Subject: ALSA: hda - Fix a unused variable warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Just overlooked.
sound/pci/hda/patch_realtek.c: In function ‘alc_sku_unsol_event’:
sound/pci/hda/patch_realtek.c:643:19: warning: unused variable ‘spec’ [-Wunused-variable]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 -
1 file changed, 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ddbed9705ef4..c95c8bde12d0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -640,7 +640,6 @@ static void alc_exec_unsol_event(struct hda_codec *codec, int action)
/* unsolicited event for HP jack sensing */
static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
{
- struct alc_spec *spec = codec->spec;
if (codec->vendor_id == 0x10ec0880)
res >>= 28;
else
--
cgit v1.2.3-70-g09d2
From bb362e2e4f4874f3fd4cbc2497385b9bceb3a08a Mon Sep 17 00:00:00 2001
From: Zeng Zhaoming <zengzm.kernel@gmail.com>
Date: Wed, 18 Jan 2012 13:58:07 +0800
Subject: ASoC: sgtl5000: Fix wrong register name in restore
Correct SGTL5000_CHIP_CLK_CTRL to SGTL5000_CHIP_REF_CTRL in
sgtl5000_restore_regs(), and add comment to explain the
restore order.
Reported-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/sgtl5000.c | 17 +++++++++++++----
1 file changed, 13 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f8863ebb4304..7f4ba819a9f6 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
/* restore regular registers */
for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
- /* this regs depends on the others */
+ /* These regs should restore in particular order */
if (reg == SGTL5000_CHIP_ANA_POWER ||
reg == SGTL5000_CHIP_CLK_CTRL ||
reg == SGTL5000_CHIP_LINREG_CTRL ||
reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
- reg == SGTL5000_CHIP_CLK_CTRL)
+ reg == SGTL5000_CHIP_REF_CTRL)
continue;
snd_soc_write(codec, reg, cache[reg]);
@@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
snd_soc_write(codec, reg, cache[reg]);
/*
- * restore power and other regs according
- * to set_power() and set_clock()
+ * restore these regs according to the power setting sequence in
+ * sgtl5000_set_power_regs() and clock setting sequence in
+ * sgtl5000_set_clock().
+ *
+ * The order of restore is:
+ * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after
+ * SGTL5000_CHIP_ANA_POWER PLL bits set
+ * 2. SGTL5000_CHIP_LINREG_CTRL should be set before
+ * SGTL5000_CHIP_ANA_POWER LINREG_D restored
+ * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage,
+ * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored
*/
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL]);
--
cgit v1.2.3-70-g09d2
From 01b37e94c04bc6dae2c4837a2eb6fff6819ea82a Mon Sep 17 00:00:00 2001
From: Wolfram Sang <w.sang@pengutronix.de>
Date: Wed, 18 Jan 2012 11:48:58 +0100
Subject: ASoC: tlv320aic32x4: always enable dividers
Dividers (such as MDAC) are always needed, independent of the codec
being I2S master or slave. Needed on a custom board where the codec has
to be slave.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/tlv320aic32x4.c | 102 ++++++++++++++++++---------------------
1 file changed, 46 insertions(+), 56 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index eb401ef021fb..3806cb6d9d4d 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -60,7 +60,6 @@ struct aic32x4_rate_divs {
struct aic32x4_priv {
u32 sysclk;
- s32 master;
u8 page_no;
void *control_data;
u32 power_cfg;
@@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u8 iface_reg_1;
u8 iface_reg_2;
u8 iface_reg_3;
@@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- aic32x4->master = 1;
iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- aic32x4->master = 0;
break;
default:
printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
@@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-
switch (level) {
case SND_SOC_BIAS_ON:
- if (aic32x4->master) {
- /* Switch on PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, AIC32X4_PLLEN);
-
- /* Switch on NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, AIC32X4_NDACEN);
-
- /* Switch on MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, AIC32X4_MDACEN);
-
- /* Switch on NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, AIC32X4_NADCEN);
-
- /* Switch on MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, AIC32X4_MADCEN);
-
- /* Switch on BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, AIC32X4_BCLKEN);
- }
+ /* Switch on PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, AIC32X4_PLLEN);
+
+ /* Switch on NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, AIC32X4_NDACEN);
+
+ /* Switch on MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, AIC32X4_MDACEN);
+
+ /* Switch on NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, AIC32X4_NADCEN);
+
+ /* Switch on MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, AIC32X4_MADCEN);
+
+ /* Switch on BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, AIC32X4_BCLKEN);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (aic32x4->master) {
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
-
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
-
- /* Switch off MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, 0);
-
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
-
- /* Switch off MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, 0);
-
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
- }
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
+
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
+
+ /* Switch off MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, 0);
+
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
+
+ /* Switch off MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, 0);
+
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
break;
case SND_SOC_BIAS_OFF:
break;
--
cgit v1.2.3-70-g09d2
From 0c93a167a6b3fa510c74e88477852c41defda075 Mon Sep 17 00:00:00 2001
From: Wolfram Sang <w.sang@pengutronix.de>
Date: Wed, 18 Jan 2012 11:48:59 +0100
Subject: ASoC: tlv320aic32x4: always enable analouge block
Register LDOCTLEN must always be initialized to clear the analog power
control bit, otherwise the analog block will stay deactivated.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/tlv320aic32x4.c | 8 +++++---
1 file changed, 5 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 3806cb6d9d4d..372b0b83bd9f 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -641,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
}
- if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
- snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
- }
+
+ tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
+ AIC32X4_LDOCTLEN : 0;
+ snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
+
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
tmp_reg |= AIC32X4_LDOIN_18_36;
--
cgit v1.2.3-70-g09d2
From e53e417331c57b9b97e3f8be870214a02c99265c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Jan 2012 20:02:38 +0000
Subject: ASoC: Mark WM5100 register map cache only when going into BIAS_OFF
Writing to the registers won't work if we do actually manage to hit a fully
powered off state.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm5100.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 8b24323d6b2c..3fd9cfe6dcd7 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1402,6 +1402,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
+ regcache_cache_only(wm5100->regmap, true);
if (wm5100->pdata.ldo_ena)
gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
--
cgit v1.2.3-70-g09d2
From 495174a8ffbaa0d15153d855cf206cdc46d51cf4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 19 Jan 2012 11:16:37 +0000
Subject: ASoC: Don't go through cache when applying WM5100 rev A updates
These are all to either uncached registers or fixes to register defaults,
in the former case the cache won't do anything and in the latter case
we're fixing things so the cache sync will do the right thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm5100.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 3fd9cfe6dcd7..66f0611e68b6 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1377,6 +1377,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
switch (wm5100->rev) {
case 0:
+ regcache_cache_bypass(wm5100->regmap, true);
snd_soc_write(codec, 0x11, 0x3);
snd_soc_write(codec, 0x203, 0xc);
snd_soc_write(codec, 0x206, 0);
@@ -1392,6 +1393,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec,
wm5100_reva_patches[i].reg,
wm5100_reva_patches[i].val);
+ regcache_cache_bypass(wm5100->regmap, false);
break;
default:
break;
--
cgit v1.2.3-70-g09d2
From fed22007113cb857e917913ce016d9b539dc3a80 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Jan 2012 19:17:06 +0000
Subject: ASoC: Disable register synchronisation for low frequency WM8996
SYSCLK
With a low frequency SYSCLK and a fast I2C clock register synchronisation
may occasionally take too long to take effect, causing I/O issues. Disable
synchronisation in order to avoid any issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm8996.c | 4 ++++
sound/soc/codecs/wm8996.h | 4 ++++
2 files changed, 8 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d8da10fe5b52..86f5b6bd7af2 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2007,6 +2007,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
int lfclk = 0;
int ratediv = 0;
+ int sync = WM8996_REG_SYNC;
int src;
int old;
@@ -2051,6 +2052,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
case 32000:
case 32768:
lfclk = WM8996_LFCLK_ENA;
+ sync = 0;
break;
default:
dev_warn(codec->dev, "Unsupported clock rate %dHz\n",
@@ -2064,6 +2066,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK,
src << WM8996_SYSCLK_SRC_SHIFT | ratediv);
snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk);
+ snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1,
+ WM8996_REG_SYNC, sync);
snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1,
WM8996_SYSCLK_ENA, old);
diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h
index 0fde643194ce..de9ac3e44aec 100644
--- a/sound/soc/codecs/wm8996.h
+++ b/sound/soc/codecs/wm8996.h
@@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
/*
* R257 (0x101) - Control Interface (1)
*/
+#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */
+#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */
#define WM8996_AUTO_INC 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */
--
cgit v1.2.3-70-g09d2
From 6b35f924b80a0e6d71711e66f5b3c16f427f3d2a Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 19 Jan 2012 10:23:22 -0200
Subject: ASoC: mxs: Fix mxs-saif timeout
On a mx28evk board the following errors happens on mxs-sgtl5000 probe:
[ 0.660000] saif0_clk_set_rate: divider writing timeout
[ 0.670000] mxs-sgtl5000: probe of mxs-sgtl5000.0 failed with error -110
[ 0.670000] ALSA device list:
[ 0.680000] No soundcards found.
This timeout happens because clk_set_rate will result in writing to the DIV bits
of register HW_CLKCTRL_SAIF0 with the saif clock gated (CLKGATE bit set to one).
MX28 Reference states the following about CLKGATE:
"The DIV field can change ONLY when this clock gate bit field is low."
So call clk_prepare_enable prior to clk_set_rate to fix this problem.
After this change the mxs-saif driver can be correctly probed and audio is functional.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/mxs/mxs-saif.c | 5 +++++
1 file changed, 5 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index dccfb37a9626..f204dbac11d4 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -124,6 +124,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
*
* If MCLK is not used, we just set saif clk to 512*fs.
*/
+ clk_prepare_enable(master_saif->clk);
+
if (master_saif->mclk_in_use) {
if (mclk % 32 == 0) {
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
@@ -133,6 +135,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
ret = clk_set_rate(master_saif->clk, 384 * rate);
} else {
/* SAIF MCLK should be either 32x or 48x */
+ clk_disable_unprepare(master_saif->clk);
return -EINVAL;
}
} else {
@@ -140,6 +143,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
}
+ clk_disable_unprepare(master_saif->clk);
+
if (ret)
return ret;
--
cgit v1.2.3-70-g09d2
From a14304edcd5e8323205db34b08f709feb5357e64 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 21 Jan 2012 21:48:53 +0000
Subject: ASoC: wm8996: Call _POST_PMU callback for CPVDD
We should be allowing a 5ms delay after the charge pump is started in
order to ensure it has finished ramping.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/codecs/wm8996.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 86f5b6bd7af2..13aa2bdaa7d7 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0),
--
cgit v1.2.3-70-g09d2
From 52409aa6a0e96337da137c069856298f4dd825a0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Jan 2012 17:10:24 +0100
Subject: ALSA: hda - Fix buffer-alignment regression with Nvidia HDMI
The commit 2ae66c26550cd94b0e2606a9275eb0ab7070ad0e
ALSA: hda: option to enable arbitrary buffer/period sizes
introduced a regression on machines with Intel controller and Nvidia
HDMI. The reason is that the driver modifies the global variable
align_buffer_size when an Intel controller is found, and the Nvidia
HDMI controller is probed after Intel although Nvidia chips require
the aligned buffers.
This patch fixes the problem by moving the flag into the local struct
so that it's not affected by other controllers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42567
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fb35474c1203..95dfb6874941 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -469,6 +469,7 @@ struct azx {
unsigned int irq_pending_warned :1;
unsigned int probing :1; /* codec probing phase */
unsigned int snoop:1;
+ unsigned int align_buffer_size:1;
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
@@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rates = hinfo->rates;
snd_pcm_limit_hw_rates(runtime);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- if (align_buffer_size)
+ if (chip->align_buffer_size)
/* constrain buffer sizes to be multiple of 128
bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and
@@ -2773,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
+ chip->align_buffer_size = align_buffer_size;
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
- align_buffer_size = 0;
+ chip->align_buffer_size = 0;
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
--
cgit v1.2.3-70-g09d2
From 29c5fbbcfefba5225a6783683c46c39e10877703 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 23 Jan 2012 16:39:55 +0100
Subject: ALSA: HDA: Use model=auto for Thinkpad T510
The user reports that model=auto works fine for him. Using
model=auto bring in new features such as jack detection notification
to userspace.
Alsa info is available at http://paste.ubuntu.com/805351/
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 8a32a69c83c3..a7a5733aa4d2 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
--
cgit v1.2.3-70-g09d2
From b4ead019afc201f71c39cd0dfcaafed4a97b3dd2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Jan 2012 18:23:36 +0100
Subject: ALSA: hda - Fix silent outputs from docking-station jacks of Dell
laptops
The recent change of the power-widget handling for IDT codecs caused
the silent output from the docking-station line-out jack. This was
partially fixed by the commit f2cbba7602383cd9cdd21f0a5d0b8bd1aad47b33
"ALSA: hda - Fix the lost power-setup of seconary pins after PM resume".
But the line-out on the docking-station is still silent when booted
with the jack plugged even by this fix.
The remainig bug is that the power-widget is set off in stac92xx_init()
because the pins in cfg->line_out_pins[] aren't checked there properly
but only hp_pins[] are checked in is_nid_hp_pin().
This patch fixes the problem by checking both HP and line-out pins
and leaving the power-map correctly.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42637
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 8 +++++---
1 file changed, 5 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1a26dbca9483..336cfcd324f9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
return 1;
}
-static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
for (i = 0; i < cfg->hp_outs; i++)
if (cfg->hp_pins[i] == nid)
return 1; /* nid is a HP-Out */
-
+ for (i = 0; i < cfg->line_outs; i++)
+ if (cfg->line_out_pins[i] == nid)
+ return 1; /* nid is a line-Out */
return 0; /* nid is not a HP-Out */
};
@@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec)
continue;
}
- if (is_nid_hp_pin(cfg, nid))
+ if (is_nid_out_jack_pin(cfg, nid))
continue; /* already has an unsol event */
pinctl = snd_hda_codec_read(codec, nid, 0,
--
cgit v1.2.3-70-g09d2
From 7edf1a4f27f44588d69cbde955651990090eb25d Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Mon, 23 Jan 2012 21:15:48 +0100
Subject: ASoC: wm8958: Use correct format string in dev_err() call
To print a value of type size_t one should use %zd, not %d.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8958-dsp2.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 8d4ea43d40a3..40ac888faf3d 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
return 0;
if (fw->size < 32) {
- dev_err(codec->dev, "%s: firmware too short (%d bytes)\n",
+ dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n",
name, fw->size);
goto err;
}
--
cgit v1.2.3-70-g09d2
From c83f1d7e71625801c72f4013291194e09b6f0a6e Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Mon, 23 Jan 2012 22:28:44 +0100
Subject: ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on
error
In wm2000_i2c_probe(), if we take the true branch in
"
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
NULL, 0);
if (ret != 0)
goto err_fw;
"
then we'll release_firmware(fw) at the 'err_fw' label. But we've already
done that just a few lines above. That's a use-after-free bug.
This patch restructures the code so that we always call
release_firmware(fw) before leaving the function, but only ever call
it once.
This means that we have to initialize 'fw' to NULL since some paths
may now end up calling it without having called request_firmware(),
but since request_firmware() deals gracefully with NULL pointers, we
are fine if we just NULL initialize it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm2000.c | 31 +++++++++++++------------------
1 file changed, 13 insertions(+), 18 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c2880907fced..a75c3766aede 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
struct wm2000_priv *wm2000;
struct wm2000_platform_data *pdata;
const char *filename;
- const struct firmware *fw;
- int reg, ret;
+ const struct firmware *fw = NULL;
+ int ret;
+ int reg;
u16 id;
wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv),
@@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = PTR_ERR(wm2000->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ goto out;
}
/* Verify that this is a WM2000 */
@@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (id != 0x2000) {
dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id);
ret = -ENODEV;
- goto err_regmap;
+ goto out_regmap_exit;
}
reg = wm2000_read(i2c, WM2000_REG_REVISON);
@@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = request_firmware(&fw, filename, &i2c->dev);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret);
- goto err_regmap;
+ goto out_regmap_exit;
}
/* Pre-cook the concatenation of the register address onto the image */
@@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (wm2000->anc_download == NULL) {
dev_err(&i2c->dev, "Out of memory\n");
ret = -ENOMEM;
- goto err_fw;
+ goto out_regmap_exit;
}
wm2000->anc_download[0] = 0x80;
wm2000->anc_download[1] = 0x00;
memcpy(wm2000->anc_download + 2, fw->data, fw->size);
- release_firmware(fw);
-
wm2000->anc_eng_ena = 1;
wm2000->anc_active = 1;
wm2000->spk_ena = 1;
@@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
wm2000_reset(wm2000);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
- NULL, 0);
- if (ret != 0)
- goto err_fw;
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0);
+ if (!ret)
+ goto out;
- return 0;
-
-err_fw:
- release_firmware(fw);
-err_regmap:
+out_regmap_exit:
regmap_exit(wm2000->regmap);
-err:
+out:
+ release_firmware(fw);
return ret;
}
--
cgit v1.2.3-70-g09d2
From 4d20bb1d5fe1afbdbff951c06cd3d3654fa5ceed Mon Sep 17 00:00:00 2001
From: Raymond Yau <superquad.vortex2@gmail.com>
Date: Tue, 17 Jan 2012 11:41:47 +0800
Subject: ALSA: ymfpci - Don't create invalid PCM & mixers when AC97 doesn't
support
- check SDAC bit of AC97 primary codec when create "rear" device 3,
"4ch" device 2 and "4ch Duplication" switch as the card need a four channels
AC97 codec to support surround40.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/ymfpci/ymfpci.c | 21 +++++++++++++--------
sound/pci/ymfpci/ymfpci_main.c | 21 ++++++++++++++-------
2 files changed, 27 insertions(+), 15 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index e57b89e8aa89..94ab728f5ca8 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) {
+ err = snd_ymfpci_mixer(chip, rear_switch[dev]);
+ if (err < 0) {
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) {
- snd_card_free(card);
- return err;
- }
- if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) {
- snd_card_free(card);
- return err;
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ err = snd_ymfpci_pcm_4ch(chip, 2, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_ymfpci_pcm2(chip, 3, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
}
if ((err = snd_ymfpci_timer(chip, 0)) < 0) {
snd_card_free(card);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 03ee4e365311..12a9a2b03387 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "4ch Duplication",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_ymfpci_info_dup4ch,
+ .get = snd_ymfpci_get_dup4ch,
+ .put = snd_ymfpci_put_dup4ch,
+};
static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = {
{
@@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "4ch Duplication",
- .info = snd_ymfpci_info_dup4ch,
- .get = snd_ymfpci_get_dup4ch,
- .put = snd_ymfpci_put_dup4ch,
-},
};
@@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch)
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0)
return err;
}
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip);
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ }
/* add S/PDIF control */
if (snd_BUG_ON(!chip->pcm_spdif))
--
cgit v1.2.3-70-g09d2
From 769fab2a41da4bd3c59eee38f47d6d5405738fe0 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Mon, 23 Jan 2012 21:02:57 +0100
Subject: ALSA: Fix memory leak on error in snd_compr_set_params()
If copy_from_user() does not return 0 we'll leak the memory we
allocated for 'params' when that variable goes out of scope.
Also a small CodingStyle cleanup: Use braces on both branches of
if/else when one branch needs it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/compress_offload.c | 13 ++++++++-----
1 file changed, 8 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index dac3633507c9..a68aed7fce02 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
- if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
- return -EFAULT;
+ if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
+ retval = -EFAULT;
+ goto out;
+ }
retval = snd_compr_allocate_buffer(stream, params);
if (retval) {
- kfree(params);
- return -ENOMEM;
+ retval = -ENOMEM;
+ goto out;
}
retval = stream->ops->set_params(stream, params);
if (retval)
goto out;
stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- } else
+ } else {
return -EPERM;
+ }
out:
kfree(params);
return retval;
--
cgit v1.2.3-70-g09d2
From 3b25eb690e8c7424eecffe1458c02b87b32aa001 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 25 Jan 2012 09:55:46 +0100
Subject: ALSA: hda - Fix silent output on ASUS A6Rp
The refactoring of Realtek codec driver in 3.2 kernel caused a
regression for ASUS A6Rp laptop; it doesn't give any output.
The reason was that this machine has a secret master mute (or EAPD)
control via NID 0x0f VREF. Setting VREF50 on this node makes the
sound working again.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 10 ++++++++++
1 file changed, 10 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c95c8bde12d0..a23479926f89 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5586,6 +5586,7 @@ static const struct hda_amp_list alc861_loopbacks[] = {
/* Pin config fixes */
enum {
PINFIX_FSC_AMILO_PI1505,
+ PINFIX_ASUS_A6RP,
};
static const struct alc_fixup alc861_fixups[] = {
@@ -5597,9 +5598,18 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
+ [PINFIX_ASUS_A6RP] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* node 0x0f VREF seems controlling the master output */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
+ { }
+ },
+ },
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
{}
};
--
cgit v1.2.3-70-g09d2
From a6a600d10aaddf1da38053c4c6b64f50f56176e6 Mon Sep 17 00:00:00 2001
From: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Date: Tue, 24 Jan 2012 13:27:56 -0200
Subject: ALSA: hda: set mute led polarity for laptops with buggy BIOS based on
SSID
HP laptop models with buggy BIOS are apparently frequent, including
machines with different codecs. Set the polarity of the mute led based
on the SSID and include an entry for the HP Mini 110-3100.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: Predrag Ivanovic <predivan@open.telekom.rs>
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 336cfcd324f9..948f0be2f4f3 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4870,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
/* BIOS bug: unfilled OEM string */
if (strstr(dev->name, "HP_Mute_LED_P_G")) {
set_hp_led_gpio(codec);
- spec->gpio_led_polarity = 1;
+ switch (codec->subsystem_id) {
+ case 0x103c148a:
+ spec->gpio_led_polarity = 0;
+ break;
+ default:
+ spec->gpio_led_polarity = 1;
+ break;
+ }
return 1;
}
}
--
cgit v1.2.3-70-g09d2
From 9fb83526a898f14adbd3f6f52fa7126f528f15ac Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 25 Jan 2012 15:19:20 +0000
Subject: ASoC: wm5100: Make sure we switch to button reporting mode
When we have identified an accessory make sure we've flagged that we've
done so in order to make sure we always report buttons and don't continue
to polarity flip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm5100.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 66f0611e68b6..3f8fd3ca9454 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2183,6 +2183,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Microphone detected\n");
wm5100->jack_mic = true;
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack,
SND_JACK_HEADSET,
SND_JACK_HEADSET | SND_JACK_BTN_0);
@@ -2221,6 +2222,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
SND_JACK_BTN_0);
} else if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Headphone detected\n");
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE,
SND_JACK_HEADPHONE);
--
cgit v1.2.3-70-g09d2
From a188fcba73837f83a78dc90a44998a978f50ac83 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 25 Jan 2012 17:57:16 +0000
Subject: ASoC: wm5100: Fix microphone configuration
We need to write the configuration for each microphone to a different
register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm5100.c | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 3f8fd3ca9454..fb757af19363 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2612,6 +2612,13 @@ static const struct regmap_config wm5100_regmap = {
.cache_type = REGCACHE_RBTREE,
};
+static const unsigned int wm5100_mic_ctrl_reg[] = {
+ WM5100_IN1L_CONTROL,
+ WM5100_IN2L_CONTROL,
+ WM5100_IN3L_CONTROL,
+ WM5100_IN4L_CONTROL,
+};
+
static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2744,7 +2751,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
}
for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) {
- regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL,
+ regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i],
WM5100_IN1_MODE_MASK |
WM5100_IN1_DMIC_SUP_MASK,
(wm5100->pdata.in_mode[i] <<
--
cgit v1.2.3-70-g09d2
From 1b76d2ee4012f325ae14e0e71dad1a0835195906 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 25 Jan 2012 21:10:07 +0000
Subject: ASoC: wm8996: Mark register cache as dirty when regulators are
disabled
Otherwise we won't resync later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8996.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 13aa2bdaa7d7..61f7daa4d0e6 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -108,7 +108,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \
struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8996->regmap, true); \
+ regcache_mark_dirty(wm8996->regmap); \
} \
return 0; \
}
--
cgit v1.2.3-70-g09d2
From 5539a102882d5ddd1bb95ea9f6f43130a789cb7f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 25 Jan 2012 21:10:21 +0000
Subject: ASoC: wm8962: Mark register cache as dirty when regulators are
disabled
Otherwise we won't resync later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8962.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 296de4e30d26..bda3da887d7e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -96,7 +96,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \
struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8962->regmap, true); \
+ regcache_mark_dirty(wm8962->regmap); \
} \
return 0; \
}
--
cgit v1.2.3-70-g09d2
From 5c1b136b7bf702e550039cb0039ec9c790c48f99 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 25 Jan 2012 21:10:33 +0000
Subject: ASoC: wm5100: Mark register cache as dirty when regulators are
disabled
Otherwise we won't resync later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm5100.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index fb757af19363..89f2af77b1c3 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1405,6 +1405,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_OFF:
regcache_cache_only(wm5100->regmap, true);
+ regcache_mark_dirty(wm5100->regmap);
if (wm5100->pdata.ldo_ena)
gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
--
cgit v1.2.3-70-g09d2
From b3a81520bd37a28f77cb0f7002086fb14061824d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 26 Jan 2012 15:56:16 +0100
Subject: ALSA: hda - Fix silent output on Haier W18 laptop
The very same problem is seen on Haier W18 laptop with ALC861 as seen
on ASUS A6Rp, which was fixed by the commit 3b25eb69.
Now we just need to add a new SSID entry pointing to the same fixup.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42656
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a23479926f89..0db1dc49382b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5610,6 +5610,7 @@ static const struct alc_fixup alc861_fixups[] = {
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", 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),
{}
};
--
cgit v1.2.3-70-g09d2
From 77231abe55433aa17eca712718745275853fa66d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 20 Jan 2012 12:19:43 +0000
Subject: ASoC: wm_hubs: Enable line out VMID buffer for single ended line
outputs
For optimal performance the single ended line outputs require that the
line output VMID buffer be enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm_hubs.c | 6 ++++++
1 file changed, 6 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2a61094075f8..9ccc416d8cbc 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
--
cgit v1.2.3-70-g09d2
From a389d67cf9849aff1722ed73186a584e2196a873 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 27 Jan 2012 14:31:19 +0100
Subject: ALSA: HDA: Remove quirk for Asus N53Jq
The user reports that he needs to add model=auto for audio to
work properly. In fact, since node 0x15 is not even a pin node,
the existing fixup is definitely wrong. Relevant information can
be found in the buglink below.
Cc: stable@kernel.org (3.2+)
BugLink: https://bugs.launchpad.net/bugs/918254
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 -
1 file changed, 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0db1dc49382b..a7f17becbd7c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5377,7 +5377,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC),
--
cgit v1.2.3-70-g09d2
From 114395c61ad2eb5a7a5cd163fcadb2414e48245a Mon Sep 17 00:00:00 2001
From: UK KIM <w0806.kim@samsung.com>
Date: Sat, 28 Jan 2012 01:52:22 +0900
Subject: ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixer
Signed-off-by: UK KIM <w0806.kim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm_hubs.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 9ccc416d8cbc..ea2672455d07 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -592,8 +592,8 @@ SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new line2n_mix[] = {
-SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
};
static const struct snd_kcontrol_new line2p_mix[] = {
--
cgit v1.2.3-70-g09d2
From 1ae5cbc52e7c6619a3f44b87809fd25370df31bb Mon Sep 17 00:00:00 2001
From: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Date: Mon, 30 Jan 2012 00:31:47 +0100
Subject: ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The Openmoko GTA01 machine has been removed from the machine ID database,
so we need to remove references to it as well.
Without that fix we have:
sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’:
sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’
Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/samsung/neo1973_wm8753.c | 65 +-------------------------------------
1 file changed, 1 insertion(+), 64 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 7ac0ba2025c3..c6012ff5bd3e 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
/* GTA02 specific routes and controls */
-#ifdef CONFIG_MACH_NEO1973_GTA02
-
static int gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
return 0;
}
-#else
-static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
-#endif
-
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
int ret;
/* set up NC codec pins */
- if (machine_is_neo1973_gta01()) {
- snd_soc_dapm_nc_pin(dapm, "LOUT2");
- snd_soc_dapm_nc_pin(dapm, "ROUT2");
- }
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "OUT4");
snd_soc_dapm_nc_pin(dapm, "LINE1");
@@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-/* GTA01 specific controls */
-
-#ifdef CONFIG_MACH_NEO1973_GTA01
-
-static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
- {"Amp IN", NULL, "ROUT1"},
- {"Amp IN", NULL, "LOUT1"},
-
- {"Handset Spk", NULL, "Amp EP"},
- {"Stereo Out", NULL, "Amp LS"},
- {"Headphone", NULL, "Amp HP"},
-};
-
-static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
- SND_SOC_DAPM_SPK("Stereo Out", NULL),
- SND_SOC_DAPM_HP("Headphone", NULL),
-};
-
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
-{
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
- ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
- ARRAY_SIZE(neo1973_lm4857_routes));
- if (ret)
- return ret;
-
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
- snd_soc_dapm_ignore_suspend(dapm, "Headphone");
-
- return 0;
-}
-
-#else
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
-#endif
-
static struct snd_soc_dai_link neo1973_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
@@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = {
.name = "dfbmcs320",
.codec_name = "dfbmcs320.0",
},
- {
- .name = "lm4857",
- .codec_name = "lm4857.0-007c",
- .init = neo1973_lm4857_init,
- },
};
static struct snd_soc_codec_conf neo1973_codec_conf[] = {
@@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = {
},
};
-#ifdef CONFIG_MACH_NEO1973_GTA02
static const struct gpio neo1973_gta02_gpios[] = {
{ GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
{ GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
};
-#else
-static const struct gpio neo1973_gta02_gpios[] = {};
-#endif
static struct snd_soc_card neo1973 = {
.name = "neo1973",
@@ -480,7 +417,7 @@ static int __init neo1973_init(void)
{
int ret;
- if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
+ if (!machine_is_neo1973_gta02())
return -ENODEV;
if (machine_is_neo1973_gta02()) {
--
cgit v1.2.3-70-g09d2
From 31150f2327cbb66363f38e13ca1be973d2f9203a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 30 Jan 2012 10:54:08 +0100
Subject: ALSA: hda - Apply 0x0f-VREF fix to all ASUS laptops with ALC861/660
It turned out that other ASUS laptops require the similar fix to
enable the VREF on the pin 0x0f for the secret output amp, not only
ASUS A6Rp. Moreover, it's required even when the pin is being used
as the output. Thus, writing a fixed value doesn't work always.
This patch applies the VREF-fix for all ASUS laptops with ALC861/660
in a fixup function that checks the current value and turns on only
the VREF value no matter whether input or output direction is set.
The automute function is modified as well to keep the pin VREF upon
muting/unmuting via pin-control; otherwise the pin VREF is reset at
plugging/unplugging a jack.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++--------
1 file changed, 35 insertions(+), 8 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a7f17becbd7c..42b6a01e17db 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -177,6 +177,7 @@ struct alc_spec {
unsigned int detect_lo:1; /* Line-out detection enabled */
unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
unsigned int automute_lo_possible:1; /* there are line outs and HP */
+ unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
@@ -495,13 +496,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
+ unsigned int val;
if (!nid)
break;
switch (spec->automute_mode) {
case ALC_AUTOMUTE_PIN:
+ /* don't reset VREF value in case it's controlling
+ * the amp (see alc861_fixup_asus_amp_vref_0f())
+ */
+ if (spec->keep_vref_in_automute) {
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ val &= ~PIN_HP;
+ } else
+ val = 0;
+ val |= pin_bits;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_bits);
+ val);
break;
case ALC_AUTOMUTE_AMP:
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -5588,6 +5600,25 @@ enum {
PINFIX_ASUS_A6RP,
};
+/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
+static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ val = snd_hda_codec_read(codec, 0x0f, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
+ val |= AC_PINCTL_IN_EN;
+ val |= AC_PINCTL_VREF_50;
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ spec->keep_vref_in_automute = 1;
+}
+
static const struct alc_fixup alc861_fixups[] = {
[PINFIX_FSC_AMILO_PI1505] = {
.type = ALC_FIXUP_PINS,
@@ -5598,17 +5629,13 @@ static const struct alc_fixup alc861_fixups[] = {
}
},
[PINFIX_ASUS_A6RP] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- /* node 0x0f VREF seems controlling the master output */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
- { }
- },
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc861_fixup_asus_amp_vref_0f,
},
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP),
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", 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),
{}
--
cgit v1.2.3-70-g09d2
From ee76744c51ec342df9822b4a85dbbfc3887b6d60 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 31 Jan 2012 11:55:32 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm_hubs.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
(limited to 'sound')
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" },
--
cgit v1.2.3-70-g09d2
From 05c3b36e539627b7aed67d038381d0d9fa9d61e7 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 31 Jan 2012 09:04:15 +0100
Subject: ALSA: HDA: Fix jack creation for codecs with front and rear Line In
If a codec has both a front and a rear Line In, two controls both
named "Line Jack" will be created, which causes parsing to fail.
While a long term solution might be to name the jacks differently,
this extra check is consistent with what is already being done in many
auto-parsers, and will also protect against other cases when two
inputs have the same label.
BugLink: https://bugs.launchpad.net/bugs/923409
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_jack.c | 24 +++++++++++++++---------
1 file changed, 15 insertions(+), 9 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d8a35da0803f..9d819c4b4923 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg)
+ const struct auto_pin_cfg *cfg,
+ char *lastname, int *lastidx)
{
unsigned int def_conf, conn;
char name[44];
@@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
return 0;
snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+ if (!strcmp(name, lastname) && idx == *lastidx)
+ idx++;
+ strncpy(lastname, name, 44);
+ *lastidx = idx;
err = snd_hda_jack_add_kctl(codec, nid, name, idx);
if (err < 0)
return err;
@@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
const hda_nid_t *p;
- int i, err;
+ int i, err, lastidx = 0;
+ char lastname[44] = "";
for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0; i < cfg->num_inputs; i++) {
- err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
+ err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
- err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
+ err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
- err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
+ err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
return 0;
--
cgit v1.2.3-70-g09d2
From 3422a47041b8cb8f14ac1e3926bcf711121df6dc Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 31 Jan 2012 10:31:49 +0100
Subject: ALSA: HDA: Remove quirk for Toshiba Qosmio G50
The user reports that model=auto works better than current handling
on a 3.2 based kernel (with jack detection patches backported).
Since model=auto is what we prefer these days anyway, the quirk
should be removed.
Alsa-info for the relevant machine:
https://bugs.launchpad.net/ubuntu/+source/linux/+bug/923316/+attachment/2702812/+files/alsa-info.txt.Pbfno2x7bp
BugLink: https://bugs.launchpad.net/bugs/923316
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 13 -------------
1 file changed, 13 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 42b6a01e17db..a8e82be3d2fc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4747,7 +4747,6 @@ enum {
ALC262_FIXUP_FSC_H270,
ALC262_FIXUP_HP_Z200,
ALC262_FIXUP_TYAN,
- ALC262_FIXUP_TOSHIBA_RX1,
ALC262_FIXUP_LENOVO_3000,
ALC262_FIXUP_BENQ,
ALC262_FIXUP_BENQ_T31,
@@ -4777,16 +4776,6 @@ static const struct alc_fixup alc262_fixups[] = {
{ }
}
},
- [ALC262_FIXUP_TOSHIBA_RX1] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x90170110 }, /* speaker */
- { 0x15, 0x0421101f }, /* HP */
- { 0x1a, 0x40f000f0 }, /* N/A */
- { 0x1b, 0x40f000f0 }, /* N/A */
- { 0x1e, 0x40f000f0 }, /* N/A */
- }
- },
[ALC262_FIXUP_LENOVO_3000] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -4819,8 +4808,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN),
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
- ALC262_FIXUP_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270),
SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ),
--
cgit v1.2.3-70-g09d2
From 67f97f5c3edad35c4d37a94f994c76111a177fb6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 31 Jan 2012 14:51:29 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8994.c | 5 -----
1 file changed, 5 deletions(-)
(limited to 'sound')
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++) {
--
cgit v1.2.3-70-g09d2
From 125a25da5729740b7d1dc417a3d5549321baae17 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 31 Jan 2012 15:49:10 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
sound/soc/soc-core.c | 11 +++++++++++
1 file changed, 11 insertions(+)
(limited to 'sound')
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;
--
cgit v1.2.3-70-g09d2
From f70eecde3bca92630d3886496e73316ff353f185 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Tue, 31 Jan 2012 13:04:41 -0800
Subject: ALSA: hda - Fix calling cs_automic twice for Cirrus codecs.
If cs_automic is called twice (like it is during init) while the mic
is present, it will over-write the last_input with the new one,
causing it to switch back to the automic input when the mic is
unplugged. This leaves the driver in a state (cur_input, last_input,
and automix_idx the same) where the internal mic can not be selected
until it is rebooted without the mic attached.
Check that the mic hasn't already been switched to before setting
last_input.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_cirrus.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 0e99357e822c..bc5a993d1146 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec)
change_cur_input(codec, !spec->automic_idx, 0);
} else {
if (present) {
- spec->last_input = spec->cur_input;
- spec->cur_input = spec->automic_idx;
+ if (spec->cur_input != spec->automic_idx) {
+ spec->last_input = spec->cur_input;
+ spec->cur_input = spec->automic_idx;
+ }
} else {
spec->cur_input = spec->last_input;
}
--
cgit v1.2.3-70-g09d2
From 2b6712b19531e22455e7fa18371c5ba9eec76699 Mon Sep 17 00:00:00 2001
From: Susan Gao <sgao@opensource.wolfsonmicro.com>
Date: Mon, 30 Jan 2012 13:57:04 -0800
Subject: ASoC: wm8962: Fix word length configuration
Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm8962.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
(limited to 'sound')
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;
--
cgit v1.2.3-70-g09d2
From 44bed4838dc191988fd1d03deccc3a845705d2de Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@gmail.com>
Date: Tue, 31 Jan 2012 09:49:04 +0800
Subject: 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 <axel.lin@gmail.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/cs42l73.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
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;
--
cgit v1.2.3-70-g09d2
From 54c2a89f60fd71b924d0f848ac892442951401a6 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 1 Feb 2012 12:05:41 +0100
Subject: ALSA: HDA: Fix duplicated output to more than one codec
This typo caused the wrong codec's nid to be checked for wcaps type.
As a result, sometimes speakers would duplicate the output sent to
HDMI output.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/924320
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4df72c0e8c37..c2c65f63bf06 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
for (i = 0; i < c->cvt_setups.used; i++) {
p = snd_array_elem(&c->cvt_setups, i);
if (!p->active && p->stream_tag == stream_tag &&
- get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+ get_wcaps_type(get_wcaps(c, p->nid)) == type)
p->dirty = 1;
}
}
--
cgit v1.2.3-70-g09d2
From 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 1 Feb 2012 23:46:58 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm_hubs.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
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),
};
--
cgit v1.2.3-70-g09d2
From 054d867e032daf55c3343fc6d36c5c5f1e7954db Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Jan 2012 12:25:50 +0100
Subject: ALSA: hda - Check power-state before changing in patch_via.c
Instead of always writing AC_VERB_SET_POWER_STATE, check the current
power-state and don't write again if the value is already set.
This may reduce the click noise upon the dynamic power-state change
(e.g. in analog-input mixer).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 256 +++++++++++++++++++---------------------------
1 file changed, 107 insertions(+), 149 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 03e63fed9caf..fb1f0ffc556b 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -687,6 +687,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
}
}
+static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int parm)
+{
+ if (snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0) == parm)
+ return;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+}
+
static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int *affected_parm)
{
@@ -709,7 +718,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
} else
parm = AC_PWRST_D3;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, nid, parm);
}
static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol,
@@ -2295,10 +2304,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
if (mux) {
/* switch to D0 beofre change index */
- if (snd_hda_codec_read(codec, mux, 0,
- AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
- snd_hda_codec_write(codec, mux, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, mux, AC_PWRST_D0);
snd_hda_codec_write(codec, mux, 0,
AC_VERB_SET_CONNECT_SEL,
spec->inputs[cur].mux_idx);
@@ -2922,9 +2928,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW 0/1 (13h/14h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -2932,8 +2938,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x19, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW6 (22h), SW2 (26h), AOW2 (24h) */
if (is_8ch) {
@@ -2941,20 +2947,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x22, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x26, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x24, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x24, parm);
} else if (codec->vendor_id == 0x11064397) {
/* PW7(23h), SW2(27h), AOW2(25h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x23, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
+ update_power_state(codec, 0x25, parm);
}
/* PW 3/4/7 (1ch/1dh/23h) */
@@ -2966,17 +2968,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x23, &parm);
/* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
if (is_8ch) {
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
+ update_power_state(codec, 0x27, parm);
} else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
}
static int patch_vt1708S(struct hda_codec *codec);
@@ -3149,10 +3147,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */
/* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x12, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x20, parm);
/* outputs */
/* PW 3/4 (16h/17h) */
@@ -3160,10 +3158,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
set_pin_power_state(codec, 0x17, &parm);
set_pin_power_state(codec, 0x16, &parm);
/* MW0 (1ah), AOW 0/1 (10h/1dh) */
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x1d, parm);
}
static int patch_vt1702(struct hda_codec *codec)
@@ -3228,52 +3225,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* PW3 (27h), MW2 (1ah), AOW3 (bh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x27, &parm);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, parm);
+ update_power_state(codec, 0xb, parm);
/* PW2 (26h), AOW2 (ah) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2b, &parm);
- snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0xa, parm);
/* PW0 (24h), AOW0 (8h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (!spec->hp_independent_mode) /* check for redirected HP */
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
- snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm);
/* PW1 (25h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2a, &parm);
- snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x9, parm);
if (spec->hp_independent_mode) {
/* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xc, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1b, parm);
+ update_power_state(codec, 0x34, parm);
+ update_power_state(codec, 0xc, parm);
}
}
@@ -3433,8 +3426,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW0(13h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x1e, &parm);
@@ -3442,12 +3435,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (spec->dmic_enabled)
set_pin_power_state(codec, 0x22, &parm);
else
- snd_hda_codec_write(codec, 0x22, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x22, AC_PWRST_D3);
/* SW2(26h), AIW1(14h) */
- snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -3456,8 +3448,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW2(1bh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW7 (23h), SW3 (27h), AOW3 (25h) */
parm = AC_PWRST_D3;
@@ -3465,12 +3457,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW1(1ah) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
/* Smart 5.1 PW5(1eh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1e, &parm);
- snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* Mono out */
/* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
@@ -3486,9 +3478,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
mono_out = 1;
}
parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
- snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x28, parm);
+ update_power_state(codec, 0x29, parm);
+ update_power_state(codec, 0x2a, parm);
/* PW 3/4 (1ch/1dh) */
parm = AC_PWRST_D3;
@@ -3496,15 +3488,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
set_pin_power_state(codec, 0x1d, &parm);
/* HP Independent Mode, power on AOW3 */
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* force to D0 for internal Speaker */
/* MW0 (16h), AOW0 (10h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
- mono_out ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm);
}
static int patch_vt1716S(struct hda_codec *codec)
@@ -3580,54 +3569,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
if (spec->codec_type == VT1802) {
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
} else {
/* PW4 (26h), MW4 (1ch), MUX4(37h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x37, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x37, parm);
}
if (spec->codec_type == VT1802) {
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
} else {
/* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x19, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x19, parm);
+ update_power_state(codec, 0x35, parm);
}
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Class-D */
/* PW0 (24h), MW0(18h/14h), MUX0(34h) */
@@ -3637,12 +3617,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x24, &parm);
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802)
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x14, parm);
else
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x34, parm);
/* Mono Out */
present = snd_hda_jack_detect(codec, 0x26);
@@ -3650,28 +3628,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802) {
/* PW15 (33h), MW8(1ch), MUX8(3ch) */
- snd_hda_codec_write(codec, 0x33, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x33, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x3c, parm);
} else {
/* PW15 (31h), MW8(17h), MUX8(3bh) */
- snd_hda_codec_write(codec, 0x31, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x17, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3b, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x31, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x3b, parm);
}
/* MW9 (21h) */
if (imux_is_smixer || !is_aa_path_mute(codec))
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x21, AC_PWRST_D0);
else
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x21, AC_PWRST_D3);
}
/* patch for vt2002P */
@@ -3731,30 +3701,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x8, AC_PWRST_D0);
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Internal Speaker */
/* PW0 (24h), MW0(14h), MUX0(34h) */
@@ -3763,15 +3731,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x14, AC_PWRST_D3);
+ update_power_state(codec, 0x34, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x14, AC_PWRST_D0);
+ update_power_state(codec, 0x34, AC_PWRST_D0);
}
@@ -3782,26 +3746,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x31, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x1c, AC_PWRST_D3);
+ update_power_state(codec, 0x3c, AC_PWRST_D3);
+ update_power_state(codec, 0x3e, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x1c, AC_PWRST_D0);
+ update_power_state(codec, 0x3c, AC_PWRST_D0);
+ update_power_state(codec, 0x3e, AC_PWRST_D0);
}
/* PW15 (33h), MW15 (1dh), MUX15(3dh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x33, &parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1d, parm);
+ update_power_state(codec, 0x3d, parm);
}
--
cgit v1.2.3-70-g09d2
From 924339239fd5ba3e505f9420d41f0939196f3530 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Jan 2012 13:58:36 +0100
Subject: ALSA: hda - Fix the logic to detect VIA analog low-current mode
The analog low-current mode must be enabled when the no stream is
running but the current detection checks it in a wrong way.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 fb1f0ffc556b..de43cd92b0a5 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1051,7 +1051,7 @@ static void analog_low_current_mode(struct hda_codec *codec)
bool enable;
unsigned int verb, parm;
- enable = is_aa_path_mute(codec) && (spec->opened_streams != 0);
+ enable = is_aa_path_mute(codec) && !spec->opened_streams;
/* decide low current mode's verb & parameter */
switch (spec->codec_type) {
--
cgit v1.2.3-70-g09d2
From e9d010c2e8f03952e67a6fd8aed0f0dc92084ccc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Feb 2012 10:33:23 +0100
Subject: ALSA: hda - Allow analog low-current mode when dynamic power-control
is on
VIA codecs have several different power-saving features, and one of
them is the analog low-current mode. But it turned out that the ALC
mode causes pop-noises at each on/off time on some machines. As a
quick workaround, disable the ALC when another power-saving feature,
the dynamic pin power-control, is turned off, too, since the dynamic
power-control is already exposed as a mixer enum element so that user
can turn it on/off freely.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 27 +++++++++++++++++++++------
1 file changed, 21 insertions(+), 6 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index de43cd92b0a5..79166fb8b074 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -199,6 +199,9 @@ struct via_spec {
unsigned int no_pin_power_ctl;
enum VIA_HDA_CODEC codec_type;
+ /* analog low-power control */
+ bool alc_mode;
+
/* smart51 setup */
unsigned int smart51_nums;
hda_nid_t smart51_pins[2];
@@ -758,6 +761,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol,
return 0;
spec->no_pin_power_ctl = val;
set_widgets_power_state(codec);
+ analog_low_current_mode(codec);
return 1;
}
@@ -1045,13 +1049,19 @@ static bool is_aa_path_mute(struct hda_codec *codec)
}
/* enter/exit analog low-current mode */
-static void analog_low_current_mode(struct hda_codec *codec)
+static void __analog_low_current_mode(struct hda_codec *codec, bool force)
{
struct via_spec *spec = codec->spec;
bool enable;
unsigned int verb, parm;
- enable = is_aa_path_mute(codec) && !spec->opened_streams;
+ if (spec->no_pin_power_ctl)
+ enable = false;
+ else
+ enable = is_aa_path_mute(codec) && !spec->opened_streams;
+ if (enable == spec->alc_mode && !force)
+ return;
+ spec->alc_mode = enable;
/* decide low current mode's verb & parameter */
switch (spec->codec_type) {
@@ -1083,6 +1093,11 @@ static void analog_low_current_mode(struct hda_codec *codec)
snd_hda_codec_write(codec, codec->afg, 0, verb, parm);
}
+static void analog_low_current_mode(struct hda_codec *codec)
+{
+ return __analog_low_current_mode(codec, false);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -1508,10 +1523,6 @@ static int via_build_controls(struct hda_codec *codec)
return err;
}
- /* init power states */
- set_widgets_power_state(codec);
- analog_low_current_mode(codec);
-
via_free_kctls(codec); /* no longer needed */
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
@@ -2782,6 +2793,10 @@ static int via_init(struct hda_codec *codec)
for (i = 0; i < spec->num_iverbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ /* init power states */
+ set_widgets_power_state(codec);
+ __analog_low_current_mode(codec, true);
+
via_auto_init_multi_out(codec);
via_auto_init_hp_out(codec);
via_auto_init_speaker_out(codec);
--
cgit v1.2.3-70-g09d2
From b5bcc189401c815988b7dd37611fc56f40c9139d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 2 Feb 2012 10:30:17 +0100
Subject: ALSA: hda - Disable dynamic-power control for VIA as default
Since the dynamic pin power-control and the analog low-current mode
may lead to pop-noise, it's safer to set it off as default.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 79166fb8b074..284e311040fe 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1470,6 +1470,7 @@ static int via_build_controls(struct hda_codec *codec)
struct snd_kcontrol *kctl;
int err, i;
+ spec->no_pin_power_ctl = 1;
if (spec->set_widgets_power_state)
if (!via_clone_control(spec, &via_pin_power_ctl_enum))
return -ENOMEM;
--
cgit v1.2.3-70-g09d2
From b544d1e0e233f83a2e6d20ee96b54ea272d5d5ba Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Feb 2012 11:56:35 +0100
Subject: 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 <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
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,
};
--
cgit v1.2.3-70-g09d2
From 226e01ef0da0b1a4c2c3922fb83ff3f9e4dfb508 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Sun, 5 Feb 2012 01:27:44 +0100
Subject: 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 <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/isa/sb/emu8000_patch.c | 1 -
1 file changed, 1 deletion(-)
(limited to 'sound')
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 <asm/uaccess.h>
#include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
static int emu8000_reset_addr;
module_param(emu8000_reset_addr, int, 0444);
--
cgit v1.2.3-70-g09d2
From eedec3d3854a390fc14008f265930f8c22b0373f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Feb 2012 10:24:04 +0100
Subject: 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 <dan.carpenter@oracle.com>
Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 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;
--
cgit v1.2.3-70-g09d2
From db966f8abb9ba74f7d5a7230f51572f52c31c4e5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 6 Feb 2012 12:07:08 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm8994.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
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,
--
cgit v1.2.3-70-g09d2
From b97f6bfdd1af95681de5a9f652da644a6525e376 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Feb 2012 11:00:53 +0100
Subject: 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 <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_ca0132.c | 33 +++++++++++++++++++--------------
1 file changed, 19 insertions(+), 14 deletions(-)
(limited to 'sound')
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;
}
--
cgit v1.2.3-70-g09d2
From 416846d2b31fc740ed9d5a5ec116964fb43c4358 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Tue, 7 Feb 2012 14:18:14 +0100
Subject: 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 <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
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),
{}
--
cgit v1.2.3-70-g09d2
From 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Sat, 4 Feb 2012 20:51:43 +0100
Subject: ALSA: usb-audio: add Edirol UM-3G support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/quirks-table.h | 8 ++++++++
1 file changed, 8 insertions(+)
(limited to 'sound')
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),
--
cgit v1.2.3-70-g09d2
From 2492250e4412c6411324c14ab289629360640b0a Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Sat, 4 Feb 2012 20:56:47 +0100
Subject: 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 <abbycedar@yahoo.com.au>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.31+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/oxygen/oxygen_mixer.c | 25 ++++++++++++++-----------
1 file changed, 14 insertions(+), 11 deletions(-)
(limited to 'sound')
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);
--
cgit v1.2.3-70-g09d2
From f647e1526fd6c7c8ab720781c40d11e11f930e93 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 7 Feb 2012 17:24:19 +0000
Subject: 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 <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.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 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,
--
cgit v1.2.3-70-g09d2
From a7c4183be2d6a7da8c97a9b671b5f3aed321127e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 7 Feb 2012 14:18:29 +0000
Subject: ASoC: wm8994: Disable line output discharge prior to ramping VMID
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8994.c | 5 +++++
1 file changed, 5 insertions(+)
(limited to 'sound')
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 |
--
cgit v1.2.3-70-g09d2
From a1e0c3cf7fb07227fe1f26161d969101dba78287 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 9 Feb 2012 09:32:19 +0100
Subject: ALSA: hda - Fix mute-LED VREF value for new HP laptops
The new HP laptops turns off the mute LED with VREF50 or VREF80, but
not in HIZ unlike the previous models. Since VREF50 (also 80) works
with the previous models, let's use VREF50 for all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 948f0be2f4f3..6345df131a00 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
spec->gpio_dir, spec->gpio_data);
} else {
notmtd_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD;
+ AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
muted_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
+ AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
--
cgit v1.2.3-70-g09d2
From 1987877d869027ab63dc9df515e11f19279a8091 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 8 Feb 2012 16:57:29 -0800
Subject: ASoC: fsi: fixup fsi_pointer() calculation method
current fsi_pointer() calculation was not correct for FSI driver.
This patch fix it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/sh/fsi.c | 6 +-----
1 file changed, 1 insertion(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db6c89a28bda..ea4a82d01160 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
- int samples_pos = io->buff_sample_pos - 1;
- if (samples_pos < 0)
- samples_pos = 0;
-
- return fsi_sample2frame(fsi, samples_pos);
+ return fsi_sample2frame(fsi, io->buff_sample_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
--
cgit v1.2.3-70-g09d2
From fc1156c0b0f7ad45ec03d919866349eeca2bf18c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Feb 2012 15:04:06 +0100
Subject: ALSA: hda - Fix initialization of secondary capture source on VT1705
VT1705 codec has two ADCs where the secondary ADC has no MUX but only
a fixed connection to the mic pin. This confused the driver and it
tries always overriding the input-source selection by assumption of
the existing MUX for the secondary ADC, resulted in resetting the
input-source at each time PM (including power-saving) occurs.
The fix is simply to check the existence of MUX for secondary ADCs in
the initialization code.
Tested-by: Anisse Astier <anisse@astier.eu>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 3 +++
1 file changed, 3 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 284e311040fe..dff9a00ee8fb 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init input-src */
for (i = 0; i < spec->num_adc_nids; i++) {
int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+ /* secondary ADCs must have the unique MUX */
+ if (i > 0 && !spec->mux_nids[i])
+ break;
if (spec->mux_nids[adc_idx]) {
int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
--
cgit v1.2.3-70-g09d2
From 02a237b24d57e2e2d5402c92549e9e792aa24359 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Feb 2012 15:25:07 +0100
Subject: ALSA: hda - Fix silent speaker output on Acer Aspire 6935
Since 3.2 kernel, the driver starts trying to assign the multi-io DACs
before the speaker, thus it assigns DAC2/3 for multi-io and DAC4 for
the speaker for a standard laptop setup like a HP, a speaker, a mic-in
and a line-in. However, on Acer Aspire 6935, it seems that the
speaker pin 0x14 must be connected with either DAC1 or 2; otherwise it
results in silence by some reason, although the codec itself allows
the routing to DAC3/4.
As a workaround, the connection list of each pin is reduced to be
mapped to either only DAC1/2 or DAC3/4, so that the compatible
assignment as in kernel 3.1 is achieved.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++
1 file changed, 23 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1d07e8fa2433..c4bde7108328 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4201,8 +4201,26 @@ enum {
PINFIX_PB_M5210,
PINFIX_ACER_ASPIRE_7736,
PINFIX_ASUS_W90V,
+ ALC889_FIXUP_DAC_ROUTE,
};
+/* Fix the connection of some pins for ALC889:
+ * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't
+ * work correctly (bko#42740)
+ */
+static void alc889_fixup_dac_route(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+ hda_nid_t conn1[2] = { 0x0c, 0x0d };
+ hda_nid_t conn2[2] = { 0x0e, 0x0f };
+ snd_hda_override_conn_list(codec, 0x14, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x15, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x18, 2, conn2);
+ snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+ }
+}
+
static const struct alc_fixup alc882_fixups[] = {
[PINFIX_ABIT_AW9D_MAX] = {
.type = ALC_FIXUP_PINS,
@@ -4239,10 +4257,15 @@ static const struct alc_fixup alc882_fixups[] = {
{ }
}
},
+ [ALC889_FIXUP_DAC_ROUTE] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc889_fixup_dac_route,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
+ SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
--
cgit v1.2.3-70-g09d2
From 27c3afe6e1cf129faac90405121203962da08ff4 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Mon, 13 Feb 2012 23:44:22 -0500
Subject: ALSA: intel8x0: Fix default inaudible sound on Gateway M520
BugLink: https://bugs.launchpad.net/bugs/930842
The reporter states that audio is inaudible by default without muting
'External Amplifier'. Add a quirk to handle his SSID so that changing
the control is not necessary.
Reported-and-tested-by: Benjamin Carlson <elderbubba0810@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/intel8x0.c | 6 ++++++
1 file changed, 6 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 9f3b01bb72c8..e0a4263baa20 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2100,6 +2100,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
.name = "MSI P4 ATX 645 Ultra",
.type = AC97_TUNE_HP_ONLY
},
+ {
+ .subvendor = 0x161f,
+ .subdevice = 0x202f,
+ .name = "Gateway M520",
+ .type = AC97_TUNE_INV_EAPD
+ },
{
.subvendor = 0x161f,
.subdevice = 0x203a,
--
cgit v1.2.3-70-g09d2
From 31794bc37bf2db84f085da52b72bfba65739b2d2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 13 Feb 2012 22:00:47 -0800
Subject: ASoC: wm8962: Fix sidetone enumeration texts
The sidetone enumeration texts have left and right swapped.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/codecs/wm8962.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 29c4b02c4790..0ac228b7dc04 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
--
cgit v1.2.3-70-g09d2
From 8866f405efd4171f9d9c91901d2dd02f01bacb60 Mon Sep 17 00:00:00 2001
From: Xi Wang <xi.wang@gmail.com>
Date: Tue, 14 Feb 2012 05:18:48 -0500
Subject: ALSA: usb-audio: avoid integer overflow in
create_fixed_stream_quirk()
A malicious USB device could feed in a large nr_rates value. This would
cause the subsequent call to kmemdup() to allocate a smaller buffer than
expected, leading to out-of-bounds access.
This patch validates the nr_rates value and reuses the limit introduced
in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow
in parse_uac2_sample_rate_range()").
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/card.h | 1 +
sound/usb/format.c | 4 +---
sound/usb/quirks.c | 6 +++++-
3 files changed, 7 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/usb/card.h b/sound/usb/card.h
index a39edcc32a93..da5fa1ac4eda 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -1,6 +1,7 @@
#ifndef __USBAUDIO_CARD_H
#define __USBAUDIO_CARD_H
+#define MAX_NR_RATES 1024
#define MAX_PACKS 20
#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */
#define MAX_URBS 8
diff --git a/sound/usb/format.c b/sound/usb/format.c
index e09aba19375c..ddfef57c4c9f 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
return 0;
}
-#define MAX_UAC2_NR_RATES 1024
-
/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
@@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
nr_rates++;
- if (nr_rates >= MAX_UAC2_NR_RATES) {
+ if (nr_rates >= MAX_NR_RATES) {
snd_printk(KERN_ERR "invalid uac2 rates\n");
break;
}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a3ddac0deffd..27817266867a 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
unsigned *rate_table = NULL;
fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
- if (! fp) {
+ if (!fp) {
snd_printk(KERN_ERR "cannot memdup\n");
return -ENOMEM;
}
+ if (fp->nr_rates > MAX_NR_RATES) {
+ kfree(fp);
+ return -EINVAL;
+ }
if (fp->nr_rates > 0) {
rate_table = kmemdup(fp->rate_table,
sizeof(int) * fp->nr_rates, GFP_KERNEL);
--
cgit v1.2.3-70-g09d2
From c14c95f62ecb8710af14ae0d48e01991b70bb6f4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Feb 2012 16:38:07 +0100
Subject: ALSA: hda/realtek - Fix overflow of vol/sw check bitmap
The bitmap introduced in the commit [527e4d73: ALSA: hda/realtek - Fix
missing volume controls with ALC260] is too narrow for some codecs,
which may have more NIDs than 0x20, thus it may overflow the bitmap
array on them.
Just double the number to cover all and also add a sanity-check code
to be safer.
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 11 ++++++++---
1 file changed, 8 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1358987c49d8..389a28a21fa9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */
};
+#define MAX_VOL_NIDS 0x40
+
struct alc_spec {
/* codec parameterization */
const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
hda_nid_t mixer_nid; /* analog-mixer NID */
- DECLARE_BITMAP(vol_ctls, 0x20 << 1);
- DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+ DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+ DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
/* capture setup for dynamic dual-adc switch */
hda_nid_t cur_adc;
@@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
static inline unsigned int get_ctl_pos(unsigned int data)
{
hda_nid_t nid = get_amp_nid_(data);
- unsigned int dir = get_amp_direction_(data);
+ unsigned int dir;
+ if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+ return 0;
+ dir = get_amp_direction_(data);
return (nid << 1) | dir;
}
--
cgit v1.2.3-70-g09d2
From ef8d60fb79614a86a82720dc2402631dbcafb315 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 17 Feb 2012 10:12:38 +0100
Subject: ALSA: hda/realtek - Fix surround output regression on Acer Aspire
5935
The previous fix for the speaker on Acer Aspire 59135 introduced
another problem for surround outputs. It changed the connections on
the line-in/mic pins for limiting the routes, but it left the modified
connections. Thus wrong connection indices were written when set to
4ch or 6ch mode.
This patch fixes it by restoring the right connections just after
parsing the tree but before the initialization.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 8 ++++++++
1 file changed, 8 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 389a28a21fa9..3647baa9bfed 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4441,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec,
const struct alc_fixup *fix, int action)
{
if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+ /* fake the connections during parsing the tree */
hda_nid_t conn1[2] = { 0x0c, 0x0d };
hda_nid_t conn2[2] = { 0x0e, 0x0f };
snd_hda_override_conn_list(codec, 0x14, 2, conn1);
snd_hda_override_conn_list(codec, 0x15, 2, conn1);
snd_hda_override_conn_list(codec, 0x18, 2, conn2);
snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+ } else if (action == ALC_FIXUP_ACT_PROBE) {
+ /* restore the connections */
+ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+ snd_hda_override_conn_list(codec, 0x14, 5, conn);
+ snd_hda_override_conn_list(codec, 0x15, 5, conn);
+ snd_hda_override_conn_list(codec, 0x18, 5, conn);
+ snd_hda_override_conn_list(codec, 0x1a, 5, conn);
}
}
--
cgit v1.2.3-70-g09d2
From e555cf363167f09efae96d32a363e24c4de16b7b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 20 Feb 2012 20:14:16 -0800
Subject: ASoC: ak4642: fixup HeadPhone L/R dapm settings
Current ak4642 driver had wrong dapm settings for headphone L/R.
If you select headphone L, and select R after that,
headphone L setting was removed by R settings.
This patch fixes it up.
It provides just "Headphone Enable" to user side
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/ak4642.c | 31 ++++++++++++++++---------------
1 file changed, 16 insertions(+), 15 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..278c0a0575f5 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
--
cgit v1.2.3-70-g09d2
From cb74eb15ac88d6aacf7e58db1d8f8dadee710fd9 Mon Sep 17 00:00:00 2001
From: Mark Hills <mark@pogo.org.uk>
Date: Tue, 21 Feb 2012 21:26:31 +0000
Subject: ALSA: snd-usb-caiaq: Fix the return of XRUN
Commit 3702b08 added a lock, but did not account for the case of
SNDRV_PCM_POS_XRUN, which would get immediately overwritten.
This could be bundled into one if-else-if statement, but the goto
helps to clarify the 'exceptional' case.
Thanks to Andreas Pape for spotting this.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/caiaq/audio.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed4..fde9a7a29cb6 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
spin_lock(&dev->spinlock);
- if (dev->input_panic || dev->output_panic)
+ if (dev->input_panic || dev->output_panic) {
ptr = SNDRV_PCM_POS_XRUN;
+ goto unlock;
+ }
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
ptr = bytes_to_frames(sub->runtime,
dev->audio_in_buf_pos[index]);
+unlock:
spin_unlock(&dev->spinlock);
return ptr;
}
--
cgit v1.2.3-70-g09d2
From 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 22 Feb 2012 15:52:56 +0000
Subject: ASoC: dapm: Check for bias level when powering down
Recent enhancements in the bias management means that we might not be
in standby when the CODEC is idle and can have active widgets without
being in full power mode but the shutdown functionality assumes these
things. Add checks for the bias level at each stage so that we don't
do transitions other than the ON->PREPARE->STANDBY->OFF ones that the
drivers are expecting.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
sound/soc/soc-dapm.c | 12 +++++++++---
1 file changed, 9 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1f55ded4047f..1315663c1c09 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+ if (dapm->bias_level == SND_SOC_BIAS_ON)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_STANDBY);
}
}
@@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&codec->dapm,
+ SND_SOC_BIAS_OFF);
}
}
--
cgit v1.2.3-70-g09d2
From 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 Mon Sep 17 00:00:00 2001
From: Javier Martin <javier.martin@vista-silicon.com>
Date: Thu, 23 Feb 2012 15:43:18 +0100
Subject: ASoC: i.MX SSI: Fix DSP_A format.
According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects
whether the most significant or the less significant part of the
data word written to the FIFO is transmitted.
As DSP_A is the same as DSP_B with a data offset of 1 bit, it
doesn't make any sense to remove TXBIT0 bit here.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/imx/imx-ssi.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf02..b6adbed6e506 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
break;
}
--
cgit v1.2.3-70-g09d2
From 068b939431486f524438330b0848a8222e33d421 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 25 Feb 2012 11:13:16 +0100
Subject: ALSA: hda/realtek - Fix resume of multiple input sources
When there are multiple input sources, the driver wrongly overwrites with
the value of the last input source on other slots at resume. Thus the
primary input source may be shown wrongly.
Reported-and-tested-by: Julian Sikorski <belegdol@gmail.com>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 3647baa9bfed..4fe2d5960a04 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3797,7 +3797,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
else
nums = spec->num_adc_nids;
for (c = 0; c < nums; c++)
- alc_mux_select(codec, 0, spec->cur_mux[c], true);
+ alc_mux_select(codec, c, spec->cur_mux[c], true);
}
/* add mic boosts if needed */
--
cgit v1.2.3-70-g09d2
From 87c9e7d7027643bf248b396c15c804456e967fcd Mon Sep 17 00:00:00 2001
From: Alban Bedel <albeu@free.fr>
Date: Sat, 25 Feb 2012 16:15:57 +0100
Subject: ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3
opl3->private_data was set even if opl3 could not be created.
Signed-off-by: Alban Bedel <albeu@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/azt3328.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e..496f14c1a731 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
if (err < 0)
goto out_err;
+ opl3->private_data = chip;
}
- opl3->private_data = chip;
-
sprintf(card->longname, "%s at 0x%lx, irq %i",
card->shortname, chip->ctrl_io, chip->irq);
--
cgit v1.2.3-70-g09d2
From 7bff172a352a2fbe9856bba517d71a2072aab041 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Feb 2012 09:41:17 +0100
Subject: ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs
A bug report with an old Sony laptop showed that we can't rely on BIOS
setting the pins of headphones but the driver should set always by
itself.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a00..9dbb5735d778 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4629,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
if (no_hp_sensing(spec, i))
continue;
- if (presence)
+ if (1 /*presence*/)
stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
#if 0 /* FIXME */
/* Resetting the pinctl like below may lead to (a sort of) regressions
--
cgit v1.2.3-70-g09d2
From 3868137ea41866773e75d9ac4b9988dcc361ff1d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Feb 2012 15:00:58 +0100
Subject: ALSA: hda - Add a fake mute feature
Some codecs don't supply the mute amp-capabilities although the lowest
volume gives the mute. It'd be handy if the parser provides the mute
mixers in such a case.
This patch adds an extension amp-cap bit (which is used only in the
driver) to represent the min volume = mute state. Also modified the
amp cache code to support the fake mute feature when this bit is set
but the real mute bit is unset.
In addition, conexant cx5051 parser uses this new feature to implement
the missing mute controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 8 ++++++--
sound/pci/hda/hda_codec.h | 3 +++
sound/pci/hda/patch_conexant.c | 22 +++++++++++++++++++++-
3 files changed, 30 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f63bf06..0ae6eb20b13b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
parm |= index << AC_AMP_SET_INDEX_SHIFT;
- parm |= val;
+ if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+ (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+ ; /* set the zero value as a fake mute */
+ else
+ parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val;
}
@@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
val1 += ofs;
val1 = ((int)val1) * ((int)val2);
- if (min_mute)
+ if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
val2 |= TLV_DB_SCALE_MUTE;
if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
return -EFAULT;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e9f71dc0d464..f0f1943a4b2c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -298,6 +298,9 @@ enum {
#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
#define AC_AMPCAP_MUTE_SHIFT 31
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */
+
/* Connection list */
#define AC_CLIST_LENGTH (0x7f<<0)
#define AC_CLIST_LONG (1<<7)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733aa4d2..ca117cf5132f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
- if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+ if (!(query_amp_caps(codec, nid, hda_dir) &
+ (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
break;
}
return 0;
@@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = {
{}
};
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+ static hda_nid_t out_nids[] = {
+ 0x10, 0x11, 0
+ };
+ hda_nid_t *p;
+
+ for (p = out_nids; *p; p++)
+ snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+ AC_AMPCAP_MIN_MUTE |
+ query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
static int patch_conexant_auto(struct hda_codec *codec)
{
struct conexant_spec *spec;
@@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
case 0x14f15045:
spec->single_adc_amp = 1;
break;
+ case 0x14f15051:
+ add_cx5051_fake_mutes(codec);
+ break;
}
apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
--
cgit v1.2.3-70-g09d2
From e49a3434f1bc64dc49ff3a56e416bb5894868dde Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 1 Mar 2012 18:14:41 +0100
Subject: ALSA: hda - Kill hyphenated names
Kill hyphens from "Line-Out" name strings, as suggested by Mark Brown.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 4 ++--
sound/pci/hda/patch_cirrus.c | 4 ++--
sound/pci/hda/patch_conexant.c | 2 +-
sound/pci/hda/patch_realtek.c | 6 +++---
4 files changed, 8 insertions(+), 8 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0ae6eb20b13b..684307372d73 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -5118,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
const char *pfx = "", *sfx = "";
/* handle as a speaker if it's a fixed line-out */
- if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT)
+ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
name = "Speaker";
/* check the location */
switch (attr) {
@@ -5177,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
switch (get_defcfg_device(def_conf)) {
case AC_JACK_LINE_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Line-Out",
+ return fill_audio_out_name(codec, nid, cfg, "Line Out",
label, maxlen, indexp);
case AC_JACK_SPEAKER:
return fill_audio_out_name(codec, nid, cfg, "Speaker",
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bc5a993d1146..c83ccdba1e5a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
"Front Speaker", "Surround Speaker", "Bass Speaker"
};
static const char * const line_outs[] = {
- "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+ "Front Line Out", "Surround Line Out", "Bass Line Out"
};
fix_volume_caps(codec, dac);
@@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
if (num_ctls > 1)
name = line_outs[idx];
else
- name = "Line-Out";
+ name = "Line Out";
break;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ca117cf5132f..d29d6d377904 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4fe2d5960a04..f286bb8fda13 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -802,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
@@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = {
"Headphone Playback Volume",
"Speaker Playback Volume",
"Mono Playback Volume",
- "Line-Out Playback Volume",
+ "Line Out Playback Volume",
"CLFE Playback Volume",
"Bass Speaker Playback Volume",
"PCM Playback Volume",
@@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = {
"Speaker Playback Switch",
"Mono Playback Switch",
"IEC958 Playback Switch",
- "Line-Out Playback Switch",
+ "Line Out Playback Switch",
"CLFE Playback Switch",
"Bass Speaker Playback Switch",
"PCM Playback Switch",
--
cgit v1.2.3-70-g09d2
From b2ccf065f7b23147ed135a41b01d05a332ca6b7e Mon Sep 17 00:00:00 2001
From: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Date: Sun, 26 Feb 2012 19:21:54 +0100
Subject: ASoC: neo1973: fix neo1973 wm8753 initialization
The neo1973 driver had wrong codec name which prevented the "sound card"
from appearing.
Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
sound/soc/samsung/neo1973_wm8753.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c6012ff5bd3e..d23b19a59d83 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.stream_name = "Voice",
.cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.ops = &neo1973_voice_ops,
},
};
--
cgit v1.2.3-70-g09d2
From 8f2392142346f2754c8292a94cc62a157ed1e093 Mon Sep 17 00:00:00 2001
From: Marton Balint <cus@passwd.hu>
Date: Mon, 5 Mar 2012 21:33:23 +0100
Subject: ALSA: hda - add quirk to detect CD input on Gigabyte EP45-DS3
My CD input got lost in commit 68ef0561efe494143516df38c03a16b837b8e79c.
Raymond helped me to add the necessary pin fixup to make it appear again. In
fact, this is basically his patch. It fixes alsa bug #5541.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 9 +++++++++
1 file changed, 9 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f286bb8fda13..5e530205bba4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4367,6 +4367,7 @@ enum {
ALC882_FIXUP_PB_M5210,
ALC882_FIXUP_ACER_ASPIRE_7736,
ALC882_FIXUP_ASUS_W90V,
+ ALC889_FIXUP_CD,
ALC889_FIXUP_VAIO_TT,
ALC888_FIXUP_EEE1601,
ALC882_FIXUP_EAPD,
@@ -4494,6 +4495,13 @@ static const struct alc_fixup alc882_fixups[] = {
{ }
}
},
+ [ALC889_FIXUP_CD] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1c, 0x993301f0 }, /* CD */
+ { }
+ }
+ },
[ALC889_FIXUP_VAIO_TT] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
@@ -4650,6 +4658,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
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),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
--
cgit v1.2.3-70-g09d2
From 526af6eb4dc71302f59806e2ccac7793963a7fe0 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Wed, 7 Mar 2012 08:25:20 +0100
Subject: ALSA: hda/realtek - Apply the coef-setup only to ALC269VB
The coef setup in alc269_fill_coef() was designed only for ALC269VB
model, and this has some bad effects for other ALC269 variants, such
as turning off the external mic input. Apply it only to ALC269VB.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 8 ++++++++
1 file changed, 8 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e530205bba4..22c73b78ac6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2068,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec)
*/
static void alc_init_special_input_src(struct hda_codec *codec);
+static int alc269_fill_coef(struct hda_codec *codec);
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
+ if (codec->vendor_id == 0x10ec0269)
+ alc269_fill_coef(codec);
+
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
@@ -5476,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = {
static int alc269_fill_coef(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
int val;
+ if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+ return 0;
+
if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8817);
--
cgit v1.2.3-70-g09d2
From 8de5d6f19bbe7c77676a62ab52be901aa10d6b54 Mon Sep 17 00:00:00 2001
From: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Date: Thu, 8 Mar 2012 15:38:04 +0100
Subject: ALSA: hdspm - Provide ioctl_compat
snd_hdspm 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: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/rme9652/hdspm.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cc9f6c83d661..bc030a2088da 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
hw->ops.open = snd_hdspm_hwdep_dummy_op;
hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+ hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl;
hw->ops.release = snd_hdspm_hwdep_dummy_op;
return 0;
--
cgit v1.2.3-70-g09d2