Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  [ALSA] soc - fix S3C2410 i2s programming error
  [ALSA] soc - fix s3c2410 PCM breakage
  [ALSA] ac97 - Add a workaround for broken quirk for VT1617A codec
  [ALSA] Revert migration to alc_set_pin_output() in alc861_auto_set_output_and_unmute()
  [ALSA] fm801 - Fix kconfig dependency mess of fm801-tea575x
  [ALSA] hda - Support IDT 92HD206 codec
  [ALSA] pcsp: Fix build with CONFIG_PM=n

7 files changed, 22 insertions, 7 deletions

diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 59203511e77..54a1f9036c6 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 581debf37dc..7e474210957 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 39198e505b1..2da89810ca1 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 d9783a4263e..6d4df45e81e 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 b3a15d61687..393f7fd2b1b 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 4ebcd6a8bf2..1ed6afd4545 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 6c70a81c730..7806ae61461 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;
 		}