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Subject[GIT PULL] sound fixes for 2.6.32-rc1
Linus,

please pull ALSA fixes for v2.6.32-rc1 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

including the following a few fixes for ASoC and HD-audio.


Thanks!

Takashi

===

Huang Weiyi (1):
ASoC: remove unused #include <linux/version.h>

Jassi (1):
ASoC: S3C lrsync function made to work with IRQs disabled.

Julia Lawall (1):
ASoC: Clean up error handling in MPC5200 DMA setup

Mark Brown (1):
ASoC: Fix display of stream name in DAPM debugfs

Takashi Iwai (8):
ALSA: hda - Add missing model=auto entry for ALC269
ALSA: hda - Set default GPIO for STAC/IDT codecs
ALSA: hda - Set default GPIO for IDT92HD71bxx
ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
ALSA: hda - Add support for HP dv6
ALSA: hda - Fix IDT92HD83* codec setup
ALSA: hda - Fix Dell S14 pin setup
ALSA: hda - Fix MSI GX620 mixer

---
sound/pci/hda/patch_realtek.c | 21 ++++++++++--
sound/pci/hda/patch_sigmatel.c | 65 +++++++++++++++++++++-------------------
sound/soc/codecs/ad1836.c | 1 -
sound/soc/codecs/ad1938.c | 1 -
sound/soc/codecs/wm8974.c | 1 -
sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++--------
sound/soc/s3c24xx/s3c-i2s-v2.c | 16 ++++++----
sound/soc/soc-dapm.c | 7 ++--
8 files changed, 85 insertions(+), 60 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7ed47f6..1296058 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7927,8 +7927,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {

static struct snd_kcontrol_new alc883_targa_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
@@ -7947,8 +7948,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = {

static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -7960,6 +7962,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
{ } /* end */
};

+static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -9167,7 +9178,8 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc882_targa_automute,
},
[ALC883_TARGA_8ch_DIG] = {
- .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
+ alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
alc883_targa_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
@@ -13370,7 +13382,8 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
[ALC269_ASUS_EEEPC_P703] = "eeepc-p703",
[ALC269_ASUS_EEEPC_P901] = "eeepc-p901",
[ALC269_FUJITSU] = "fujitsu",
- [ALC269_LIFEBOOK] = "lifebook"
+ [ALC269_LIFEBOOK] = "lifebook",
+ [ALC269_AUTO] = "auto",
};

static struct snd_pci_quirk alc269_cfg_tbl[] = {
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e31e53d..826137e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -864,10 +864,6 @@ static struct hda_verb stac92hd73xx_core_init[] = {
};

static struct hda_verb stac92hd83xxx_core_init[] = {
- { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1},
- { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1},
- { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0},
-
/* power state controls amps */
{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
{}
@@ -1590,8 +1586,8 @@ static unsigned int ref92hd83xxx_pin_configs[10] = {
};

static unsigned int dell_s14_pin_configs[10] = {
- 0x02214030, 0x02211010, 0x02a19020, 0x01014050,
- 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160,
+ 0x0221403f, 0x0221101f, 0x02a19020, 0x90170110,
+ 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160,
0x40f000f0, 0x40f000f0,
};

@@ -1690,6 +1686,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"HP mini 1000", STAC_HP_M4),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
"HP HDX", STAC_HP_HDX), /* HDX16 */
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620,
+ "HP dv6", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
@@ -4166,7 +4164,10 @@ static int stac92xx_init(struct hda_codec *codec)
stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
AC_PINCTL_OUT_EN);
/* fake event to set up pins */
- stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+ if (cfg->hp_pins[0])
+ stac_issue_unsol_event(codec, cfg->hp_pins[0]);
+ else if (cfg->line_out_pins[0])
+ stac_issue_unsol_event(codec, cfg->line_out_pins[0]);
} else {
stac92xx_auto_init_multi_out(codec);
stac92xx_auto_init_hp_out(codec);
@@ -4688,8 +4689,13 @@ static int stac92xx_resume(struct hda_codec *codec)
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
/* fake event to set up pins again to override cached values */
- if (spec->hp_detect)
- stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+ if (spec->hp_detect) {
+ if (spec->autocfg.hp_pins[0])
+ stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+ else if (spec->autocfg.line_out_pins[0])
+ stac_issue_unsol_event(codec,
+ spec->autocfg.line_out_pins[0]);
+ }
return 0;
}

@@ -5016,7 +5022,7 @@ again:
spec->eapd_switch = 1;
break;
}
- if (spec->board_config > STAC_92HD73XX_REF) {
+ if (spec->board_config != STAC_92HD73XX_REF) {
/* GPIO0 High = Enable EAPD */
spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
spec->gpio_data = 0x01;
@@ -5066,7 +5072,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)

codec->spec = spec;
codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
- spec->mono_nid = 0x19;
spec->digbeep_nid = 0x21;
spec->mux_nids = stac92hd83xxx_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
@@ -5242,7 +5247,7 @@ again:
stac92xx_set_config_regs(codec,
stac92hd71bxx_brd_tbl[spec->board_config]);

- if (spec->board_config > STAC_92HD71BXX_REF) {
+ if (spec->board_config != STAC_92HD71BXX_REF) {
/* GPIO0 = EAPD */
spec->gpio_mask = 0x01;
spec->gpio_dir = 0x01;
@@ -5375,6 +5380,11 @@ again:
case STAC_HP_DV5:
snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
+ /* HP dv6 gives the headphone pin as a line-out. Thus we
+ * need to set hp_detect flag here to force to enable HP
+ * detection.
+ */
+ spec->hp_detect = 1;
break;
case STAC_HP_HDX:
spec->num_dmics = 1;
@@ -5557,14 +5567,17 @@ static int patch_stac927x(struct hda_codec *codec)
spec->dac_list = stac927x_dac_nids;
spec->multiout.dac_nids = spec->dac_nids;

+ if (spec->board_config != STAC_D965_REF) {
+ /* GPIO0 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = 0x01;
+ spec->gpio_dir = spec->gpio_data = 0x01;
+ }
+
switch (spec->board_config) {
case STAC_D965_3ST:
case STAC_D965_5ST:
/* GPIO0 High = Enable EAPD */
- spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01;
- spec->gpio_data = 0x01;
spec->num_dmics = 0;
-
spec->init = d965_core_init;
break;
case STAC_DELL_BIOS:
@@ -5583,16 +5596,11 @@ static int patch_stac927x(struct hda_codec *codec)
snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130);
/* fallthru */
case STAC_DELL_3ST:
- /* GPIO2 High = Enable EAPD */
- spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
- spec->gpio_data = 0x04;
- switch (codec->subsystem_id) {
- case 0x1028022f:
- /* correct EAPD to be GPIO0 */
- spec->eapd_mask = spec->gpio_mask = 0x01;
- spec->gpio_dir = spec->gpio_data = 0x01;
- break;
- };
+ if (codec->subsystem_id != 0x1028022f) {
+ /* GPIO2 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = 0x04;
+ spec->gpio_dir = spec->gpio_data = 0x04;
+ }
spec->dmic_nids = stac927x_dmic_nids;
spec->num_dmics = STAC927X_NUM_DMICS;

@@ -5601,14 +5609,9 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
break;
default:
- if (spec->board_config > STAC_D965_REF) {
- /* GPIO0 High = Enable EAPD */
- spec->eapd_mask = spec->gpio_mask = 0x01;
- spec->gpio_dir = spec->gpio_data = 0x01;
- }
spec->num_dmics = 0;
-
spec->init = stac927x_core_init;
+ break;
}

spec->num_caps = STAC927X_NUM_CAPS;
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 3612bb9..01343dc 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -18,7 +18,6 @@

#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index e62b277..9a049a1 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -28,7 +28,6 @@

#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index d8a013a..98d663a 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -12,7 +12,6 @@

#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 9ff62e3..6096d22 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
int size, irq, rc;
const __be32 *prop;
void __iomem *regs;
+ int ret;

/* Fetch the registers and IRQ of the PSC */
irq = irq_of_parse_and_map(op->node, 0);
@@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op)
/* Allocate and initialize the driver private data */
psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
if (!psc_dma) {
- iounmap(regs);
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto out_unmap;
}

/* Get the PSC ID */
prop = of_get_property(op->node, "cell-index", &size);
- if (!prop || size < sizeof *prop)
- return -ENODEV;
+ if (!prop || size < sizeof *prop) {
+ ret = -ENODEV;
+ goto out_free;
+ }

spin_lock_init(&psc_dma->lock);
mutex_init(&psc_dma->mutex);
@@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
if (!psc_dma->capture.bcom_task ||
!psc_dma->playback.bcom_task) {
dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
- iounmap(regs);
- kfree(psc_dma);
- return -ENODEV;
+ ret = -ENODEV;
+ goto out_free;
}

/* Disable all interrupts and reset the PSC */
@@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
&psc_dma_bcom_irq_tx, IRQF_SHARED,
"psc-dma-playback", &psc_dma->playback);
if (rc) {
- free_irq(psc_dma->irq, psc_dma);
- free_irq(psc_dma->capture.irq,
- &psc_dma->capture);
- free_irq(psc_dma->playback.irq,
- &psc_dma->playback);
- return -ENODEV;
+ ret = -ENODEV;
+ goto out_irq;
}

/* Save what we've done so it can be found again later */
@@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op)

/* Tell the ASoC OF helpers about it */
return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+out_irq:
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+ kfree(psc_dma);
+out_unmap:
+ iounmap(regs);
+ return ret;
}
EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index aa7af0b..9bc4aa3 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}

+#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
+
/*
* Wait for the LR signal to allow synchronisation to the L/R clock
* from the codec. May only be needed for slave mode.
@@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
{
u32 iiscon;
- unsigned long timeout = jiffies + msecs_to_jiffies(5);
+ unsigned long loops = msecs_to_loops(5);

pr_debug("Entered %s\n", __func__);

- while (1) {
+ while (--loops) {
iiscon = readl(i2s->regs + S3C2412_IISCON);
if (iiscon & S3C2412_IISCON_LRINDEX)
break;

- if (timeout < jiffies) {
- printk(KERN_ERR "%s: timeout\n", __func__);
- return -ETIMEDOUT;
- }
+ cpu_relax();
+ }
+
+ if (!loops) {
+ printk(KERN_ERR "%s: timeout\n", __func__);
+ return -ETIMEDOUT;
}

return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0d8b08e..f79711b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1131,9 +1131,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n",
w->name, w->power ? "On" : "Off", in, out);

- if (w->active && w->sname)
- ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
- w->sname);
+ if (w->sname)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+ w->sname,
+ w->active ? "active" : "inactive");

list_for_each_entry(p, &w->sources, list_sink) {
if (p->connect)

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