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] hda - Fix capture mute Widget for stac9250/9251
  [ALSA] snd-pcsp - fix pcsp_treble_info() to honour an item number
  [ALSA] hda - Added support for Foxconn P35AX-S mainboard
  [ALSA] hda - Fix COEF and EAPD in ALC889 auto-configuration mode
  [ALSA] hda - Fix noise on VT1708 codec
  [ALSA] hda - Add model for ASUS P5K-E/WIFI-AP

6 files changed, 31 insertions, 4 deletions

diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1ee1fe7..1d661f795e8c 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -24,7 +24,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 /* default timer freq for PC-Speaker: 18643 Hz */
 #define DIV_18KHZ 64
 #define MAX_DIV DIV_18KHZ
-#define CUR_DIV() (MAX_DIV >> chip->treble)
+#define CALC_DIV(d) (MAX_DIV >> (d))
+#define CUR_DIV() CALC_DIV(chip->treble)
 #define PCSP_MAX_TREBLE 1
 
 /* unfortunately, with hrtimers 37KHz does not work very well :( */
@@ -36,7 +37,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 #define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1)
 #define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV)
 #define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble))
-#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV())
+#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i))
+#define PCSP_RATE() PCSP_CALC_RATE(chip->treble)
 #define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE
 #define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE
 #define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1)
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 64a695fef74e..caeb0f57fcca 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.enumerated.items = chip->max_treble + 1;
 	if (uinfo->value.enumerated.item > chip->max_treble)
 		uinfo->value.enumerated.item = chip->max_treble;
-	sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE());
+	sprintf(uinfo->value.enumerated.name, "%d",
+			PCSP_CALC_RATE(uinfo->value.enumerated.item));
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde42..ff1b922c610b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = {
 static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f598c38..8f31247c52bd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -853,6 +853,7 @@ do_sku:
 		case 0x10ec0269:
 		case 0x10ec0862:
 		case 0x10ec0662:	
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x14, 0,
 					    AC_VERB_SET_EAPD_BTLENABLE, 2);
 			snd_hda_codec_write(codec, 0x15, 0,
@@ -877,6 +878,7 @@ do_sku:
 		case 0x10ec0883:
 		case 0x10ec0885:
 		case 0x10ec0888:
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_COEF_INDEX, 7);
 			tmp = snd_hda_codec_read(codec, 0x20, 0,
@@ -7743,6 +7745,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd2b1be..a4f44a00bae8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
 	{ } /* end */
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81a26f7..e7e43524f8c7 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
 	},
 };
 
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x10, /* NID to query formats and rates */
+	/* We got noisy outputs on the right channel on VT1708 when
+	 * 24bit samples are used.  Until any workaround is found,
+	 * disable the 24bit format, so far.
+	 */
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_pcm_prepare,
+		.cleanup = via_playback_pcm_cleanup
+	},
+};
+
 static struct hda_pcm_stream vt1708_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
@@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec)
 	
 	spec->stream_name_analog = "VT1708 Analog";
 	spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+	/* disable 32bit format on VT1708 */
+	if (codec->vendor_id == 0x11061708)
+		spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
 	spec->stream_analog_capture = &vt1708_pcm_analog_capture;
 
 	spec->stream_name_digital = "VT1708 Digital";