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Date: Mon, 05 May 2008 14:27:56 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: linux-kernel@...r.kernel.org
Subject: [GIT PULL] ALSA fixes for 2.6.26-rc1
Linus,
please pull ALSA fixes for 2.6.26-rc1 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Davide Rizzo (2):
[ALSA] soc - fix s3c2410 PCM breakage
[ALSA] soc - fix S3C2410 i2s programming error
Jacek Luczak (1):
[ALSA] Revert migration to alc_set_pin_output() in alc861_auto_set_output_and_unmute()
Johann Felix Soden (1):
[ALSA] pcsp: Fix build with CONFIG_PM=n
Takashi Iwai (3):
[ALSA] hda - Support IDT 92HD206 codec
[ALSA] fm801 - Fix kconfig dependency mess of fm801-tea575x
[ALSA] ac97 - Add a workaround for broken quirk for VT1617A codec
sound/drivers/pcsp/pcsp.c | 4 ++++
sound/pci/Kconfig | 5 +----
sound/pci/ac97/ac97_patch.c | 9 ++++++++-
sound/pci/hda/patch_realtek.c | 5 ++++-
sound/pci/hda/patch_sigmatel.c | 2 ++
sound/soc/s3c24xx/s3c24xx-i2s.c | 2 ++
sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +-
7 files changed, 22 insertions(+), 7 deletions(-)
---
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 5920351..54a1f90 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -194,6 +194,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
spin_unlock_irq(&chip->substream_lock);
}
+#ifdef CONFIG_PM
static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
{
struct snd_pcsp *chip = platform_get_drvdata(dev);
@@ -201,6 +202,9 @@ static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
snd_pcm_suspend_all(chip->pcm);
return 0;
}
+#else
+#define pcsp_suspend NULL
+#endif /* CONFIG_PM */
static void pcsp_shutdown(struct platform_device *dev)
{
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 581debf..7e47421 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -515,19 +515,16 @@ config SND_FM801
config SND_FM801_TEA575X_BOOL
bool "ForteMedia FM801 + TEA5757 tuner"
depends on SND_FM801
+ depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801
help
Say Y here to include support for soundcards based on the ForteMedia
FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media
Forte SF256-PCS-02) into the snd-fm801 driver.
- This will enable support for the old V4L1 API.
-
config SND_FM801_TEA575X
tristate
depends on SND_FM801_TEA575X_BOOL
default SND_FM801
- select VIDEO_V4L1
- select VIDEO_DEV
config SND_HDA_INTEL
tristate "Intel HD Audio"
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 39198e5..2da8981 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3446,6 +3446,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = {
int patch_vt1617a(struct snd_ac97 * ac97)
{
int err = 0;
+ int val;
/* we choose to not fail out at this point, but we tell the
caller when we return */
@@ -3456,7 +3457,13 @@ int patch_vt1617a(struct snd_ac97 * ac97)
/* bring analog power consumption to normal by turning off the
* headphone amplifier, like WinXP driver for EPIA SP
*/
- snd_ac97_write_cache(ac97, 0x5c, 0x20);
+ /* We need to check the bit before writing it.
+ * On some (many?) hardwares, setting bit actually clears it!
+ */
+ val = snd_ac97_read(ac97, 0x5c);
+ if (!(val & 0x20))
+ snd_ac97_write_cache(ac97, 0x5c, 0x20);
+
ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
ac97->build_ops = &patch_vt1616_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d9783a4..6d4df45 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11902,7 +11902,10 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid,
int pin_type, int dac_idx)
{
- alc_set_pin_output(codec, nid, pin_type);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_type);
+ snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
}
static void alc861_auto_init_multi_out(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b3a15d6..393f7fd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4289,6 +4289,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
{ .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
{ .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
+ { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
+ { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
/* The following does not take into account .id=0x83847661 when subsys =
* 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
* currently not fully supported.
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 4ebcd6a..1ed6afd 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -224,6 +224,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
iismod |= S3C2410_IISMOD_SLAVE;
break;
case SND_SOC_DAIFMT_CBS_CFS:
+ iismod &= ~S3C2410_IISMOD_SLAVE;
break;
default:
return -EINVAL;
@@ -234,6 +235,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
iismod |= S3C2410_IISMOD_MSB;
break;
case SND_SOC_DAIFMT_I2S:
+ iismod &= ~S3C2410_IISMOD_MSB;
break;
default:
return -EINVAL;
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 6c70a81..7806ae6 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
ret = s3c2410_dma_request(prtd->params->channel,
prtd->params->client, NULL);
- if (ret) {
+ if (ret < 0) {
DBG(KERN_ERR "failed to get dma channel\n");
return ret;
}
--
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