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author | Linus Torvalds <torvalds@linux-foundation.org> | 2008-05-25 21:59:27 (GMT) |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2008-05-25 21:59:27 (GMT) |
commit | 32522bfdaed094e447f71cce68c349847ae9c7d5 (patch) | |
tree | 36b13887f66ab8daf7a2121b58d7a6ce53b6cb9c | |
parent | eb90d81d03c0917b0fd629f6342554a3b58ea52c (diff) | |
parent | 587755f1f6a983a9f0f3322d284034f4e146891a (diff) | |
download | linux-fsl-qoriq-32522bfdaed094e447f71cce68c349847ae9c7d5.tar.xz |
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
-rw-r--r-- | sound/drivers/pcsp/pcsp.h | 6 | ||||
-rw-r--r-- | sound/drivers/pcsp/pcsp_mixer.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/patch_analog.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 2 | ||||
-rw-r--r-- | 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"; |