From b9e16bc548600124da9d24186364ee8d06040569 Mon Sep 17 00:00:00 2001
From: Travis Place <wishie@wishie.net>
Date: Wed, 21 May 2008 16:57:20 +0200
Subject: [ALSA] hda - Add model for ASUS P5K-E/WIFI-AP
Added a config table entry for the ASUS P5K-E/WIFI-AP mainboard (ID
1043:8227) to use AD1988_6STACK_DIG
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_analog.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde4..ff1b922c610 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),
{}
};
--
cgit v1.2.3-18-g5258
From bc9b56238eedda865070dcaed6694d65b517c8d6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 May 2008 17:50:27 +0200
Subject: [ALSA] hda - Fix noise on VT1708 codec
We get quite noisy output on the right channel on VT1708 codec
when 24bit samples are used. Suppress the 24bit support until any
real fix is found.
https://bugzilla.novell.com/show_bug.cgi?id=390473
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 20 ++++++++++++++++++++
1 file changed, 20 insertions(+)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81a26f..e7e43524f8c 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";
--
cgit v1.2.3-18-g5258
From 20a3a05dd66ad0f678a587688cc85f0b36869876 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 May 2008 17:52:53 +0200
Subject: [ALSA] hda - Fix COEF and EAPD in ALC889 auto-configuration mode
Fix the missing COEF and EAPD initialization in ALC889 auto-configuration
mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f598c3..d42864a1989 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,
--
cgit v1.2.3-18-g5258
From 97ec710cab76f90a6bece76a04e76aa50096a470 Mon Sep 17 00:00:00 2001
From: Travis Place <wishie@wishie.net>
Date: Fri, 23 May 2008 18:31:46 +0200
Subject: [ALSA] hda - Added support for Foxconn P35AX-S mainboard
Added IDs for the Foxconn P35AX-S mainboard to patch_realtek.c, so
that ALC883_6ST_DIG is used by default.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d42864a1989..8f31247c52b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7745,6 +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),
--
cgit v1.2.3-18-g5258
From 97e08f5d732bbfd5180f73aa7875d328421bee8a Mon Sep 17 00:00:00 2001
From: Stas Sergeev <stsp@aknet.ru>
Date: Sat, 24 May 2008 18:05:47 +0200
Subject: [ALSA] snd-pcsp - fix pcsp_treble_info() to honour an item number
This solves the problem with mixers wrongly displaying the PWM freq.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/drivers/pcsp/pcsp.h | 6 ++++--
sound/drivers/pcsp/pcsp_mixer.c | 3 ++-
2 files changed, 6 insertions(+), 3 deletions(-)
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1ee1fe..1d661f795e8 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 64a695fef74..caeb0f57fcc 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;
}
--
cgit v1.2.3-18-g5258
From 587755f1f6a983a9f0f3322d284034f4e146891a Mon Sep 17 00:00:00 2001
From: Mauro Carvalho Chehab <mchehab@infradead.org>
Date: Sun, 25 May 2008 18:20:06 +0200
Subject: [ALSA] hda - Fix capture mute Widget for stac9250/9251
Fix capture mute widget for STAC9250/9251 codecs. The widget 0x09
has no mute but 0x14 does actually.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
---
sound/pci/hda/patch_sigmatel.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd2b1b..a4f44a00bae 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 */
};
--
cgit v1.2.3-18-g5258