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Message-ID: <s5h3ao6l46a.wl%tiwai@suse.de>
Date: Sun, 25 May 2008 18:49:33 +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-rc3
Hi Linus,
please pull another ALSA updates for 2.6.26-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This includes a few trivial fixes for HD-audio and one fix for
snd-pcsp.
Thanks!
Takashi
===
Mauro Carvalho Chehab (1):
[ALSA] hda - Fix capture mute Widget for stac9250/9251
Stas Sergeev (1):
[ALSA] snd-pcsp - fix pcsp_treble_info() to honour an item number
Takashi Iwai (2):
[ALSA] hda - Fix noise on VT1708 codec
[ALSA] hda - Fix COEF and EAPD in ALC889 auto-configuration mode
Travis Place (2):
[ALSA] hda - Add model for ASUS P5K-E/WIFI-AP
[ALSA] hda - Added support for Foxconn P35AX-S mainboard
sound/drivers/pcsp/pcsp.h | 6 ++++--
sound/drivers/pcsp/pcsp_mixer.c | 3 ++-
sound/pci/hda/patch_analog.c | 1 +
sound/pci/hda/patch_realtek.c | 3 +++
sound/pci/hda/patch_sigmatel.c | 2 +-
sound/pci/hda/patch_via.c | 20 ++++++++++++++++++++
6 files changed, 31 insertions(+), 4 deletions(-)
---
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1e..1d661f7 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 64a695f..caeb0f5 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 e0a605a..ff1b922 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 864b2f5..8f31247 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 393f7fd..a4f44a0 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 52b1d81..e7e4352 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";
--
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