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-rw-r--r--sound/drivers/pcsp/pcsp.h6
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c3
-rw-r--r--sound/isa/sb/sb_mixer.c4
-rw-r--r--sound/pci/ac97/ac97_patch.c48
-rw-r--r--sound/pci/aw2/aw2-alsa.c4
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c15
-rw-r--r--sound/pci/hda/patch_analog.c51
-rw-r--r--sound/pci/hda/patch_cmedia.c1
-rw-r--r--sound/pci/hda/patch_realtek.c6
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/pci/hda/patch_via.c20
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c12
12 files changed, 115 insertions, 57 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;
}
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 91d14224f6b..73d4572d136 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -925,7 +925,7 @@ static unsigned char als4000_saved_regs[] = {
static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
{
unsigned char *val = chip->saved_regs;
- snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+ snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
for (; num_regs; num_regs--)
*val++ = snd_sbmixer_read(chip, *regs++);
}
@@ -933,7 +933,7 @@ static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
{
unsigned char *val = chip->saved_regs;
- snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+ snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
for (; num_regs; num_regs--)
snd_sbmixer_write(chip, *regs++, *val++);
}
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 2da89810ca1..1292dcee072 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd
val = ac97->regs[AC97_AD_MISC];
ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL);
+ if (ac97->spec.ad18xx.lo_as_master)
+ ucontrol->value.integer.value[0] =
+ !ucontrol->value.integer.value[0];
return 0;
}
@@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = !ucontrol->value.integer.value[0]
- ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
+ val = !ucontrol->value.integer.value[0];
+ if (ac97->spec.ad18xx.lo_as_master)
+ val = !val;
+ val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
return snd_ac97_update_bits(ac97, AC97_AD_MISC,
AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val);
}
@@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97)
{
unsigned short val = 0;
/* clear LODIS if shared jack is to be used for Surround out */
- if (is_shared_linein(ac97))
+ if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97))
val |= (1 << 12);
/* clear CLDIS if shared jack is to be used for C/LFE out */
if (is_shared_micin(ac97))
@@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
static int patch_ad1888_specific(struct snd_ac97 *ac97)
{
- /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
- snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback");
- snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+ if (!ac97->spec.ad18xx.lo_as_master) {
+ /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+ snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+ "Master Surround Playback");
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback",
+ "Master Playback");
+ }
return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
}
@@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97)
patch_ad1881(ac97);
ac97->build_ops = &patch_ad1888_build_ops;
- /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
- /* it seems that most vendors connect line-out connector to headphone out of AC'97 */
+
+ /*
+ * LO can be used as a real line-out on some devices,
+ * and we need to revert the front/surround mixer switches
+ */
+ if (ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */
+ ac97->spec.ad18xx.lo_as_master = 1;
+
+ misc = snd_ac97_read(ac97, AC97_AD_MISC);
/* AD-compatible mode */
/* Stereo mutes enabled */
- misc = snd_ac97_read(ac97, AC97_AD_MISC);
- snd_ac97_write_cache(ac97, AC97_AD_MISC, misc |
- AC97_AD198X_LOSEL |
- AC97_AD198X_HPSEL |
- AC97_AD198X_MSPLT |
- AC97_AD198X_AC97NC);
+ misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC;
+ if (!ac97->spec.ad18xx.lo_as_master)
+ /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
+ /* it seems that most vendors connect line-out connector to
+ * headphone out of AC'97
+ */
+ misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL;
+
+ snd_ac97_write_cache(ac97, AC97_AD_MISC, misc);
ac97->flags |= AC97_STEREO_MUTES;
return 0;
}
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 56f87cd33c1..3f00ddf450f 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card,
return -ENOMEM;
}
+ /* (2) initialization of the chip hardware */
+ snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
IRQF_SHARED, "Audiowerk2", chip)) {
@@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card,
}
chip->irq = pci->irq;
- /* (2) initialization of the chip hardware */
- snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
free_irq(chip->irq, (void *)chip);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index abde5b90188..548c9cc81af 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
}
emu->port = pci_resource_start(pci, 0);
- if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
- "EMU10K1", emu)) {
- err = -EBUSY;
- goto error;
- }
- emu->irq = pci->irq;
-
emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT;
if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
32 * 1024, &emu->ptb_pages) < 0) {
@@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
emu->fx8010.etram_pages.area = NULL;
emu->fx8010.etram_pages.bytes = 0;
+ /* irq handler must be registered after I/O ports are activated */
+ if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
+ "EMU10K1", emu)) {
+ err = -EBUSY;
+ goto error;
+ }
+ emu->irq = pci->irq;
+
/*
* Init to 0x02109204 :
* Clock accuracy = 0 (1000ppm)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde4..a99e86d7427 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),
{}
};
@@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
{ } /* end */
};
-static struct hda_input_mux ad1884a_mobile_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 }, /* port-C */
- { "Mix", 0x3 },
- },
-};
-
static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
{ } /* end */
};
@@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec)
present ? 0x00 : 0x02);
}
+/* switch to external mic if plugged */
+static void ad1884a_hp_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 0 : 1);
+}
+
#define AD1884A_HP_EVENT 0x37
+#define AD1884A_MIC_EVENT 0x36
/* unsolicited event for HP jack sensing */
static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
{
- if ((res >> 26) != AD1884A_HP_EVENT)
- return;
- ad1884a_hp_automute(codec);
+ switch (res >> 26) {
+ case AD1884A_HP_EVENT:
+ ad1884a_hp_automute(codec);
+ break;
+ case AD1884A_MIC_EVENT:
+ ad1884a_hp_automic(codec);
+ break;
+ }
}
/* initialize jack-sensing, too */
@@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec)
{
ad198x_init(codec);
ad1884a_hp_automute(codec);
+ ad1884a_hp_automic(codec);
return 0;
}
@@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
/* Port-F pin */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-C pin - internal mic-in */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
/* analog mix */
{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* unsolicited event for pin-sense */
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
{ } /* end */
};
@@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec)
spec->mixers[0] = ad1884a_mobile_mixers;
spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1884a_mobile_capture_source;
codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
codec->patch_ops.init = ad1884a_hp_init;
break;
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c73ce074a6e..6ef57fbfb6e 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = {
static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+ SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f598c3..b0a2a262ece 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,
@@ -940,7 +942,6 @@ do_sku:
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_HP_EVENT);
spec->unsol_event = alc_sku_unsol_event;
- spec->init_hook = alc_sku_automute;
}
/*
@@ -7743,6 +7744,7 @@ 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),
@@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
};
/* mute/unmute internal speaker according to the hp jack and mute state */
@@ -10510,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
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 */
};
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";
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index cc0cddadd58..6facac5aed9 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -936,11 +936,13 @@ static int add_controls(struct oxygen *chip,
for (i = 0; i < count; ++i) {
template = controls[i];
- err = chip->model->control_filter(&template);
- if (err < 0)
- return err;
- if (err == 1)
- continue;
+ if (chip->model->control_filter) {
+ err = chip->model->control_filter(&template);
+ if (err < 0)
+ return err;
+ if (err == 1)
+ continue;
+ }
if (!strcmp(template.name, "Master Playback Volume") &&
chip->model->dac_tlv) {
template.tlv.p = chip->model->dac_tlv;
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-23 17:29:39 +0000