summaryrefslogtreecommitdiff
path: root/sound/pci
diff options
context:
space:
mode:
Diffstat (limited to 'sound/pci')
-rw-r--r--sound/pci/Kconfig10
-rw-r--r--sound/pci/ac97/ac97_patch.c1
-rw-r--r--sound/pci/asihpi/hpicmn.c5
-rw-r--r--sound/pci/azt3328.c11
-rw-r--r--sound/pci/fm801.c15
-rw-r--r--sound/pci/hda/alc268_quirks.c36
-rw-r--r--sound/pci/hda/alc269_quirks.c7
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c31
-rw-r--r--sound/pci/hda/hda_intel.c9
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c57
-rw-r--r--sound/pci/hda/patch_realtek.c71
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/rme9652/hdspm.c19
16 files changed, 188 insertions, 106 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5b..8816804 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
# ALSA PCI drivers
+config SND_TEA575X
+ tristate
+ depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+ default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
menuconfig SND_PCI
bool "PCI sound devices"
depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
SF64-PCR) into the snd-fm801 driver.
-config SND_TEA575X
- tristate
- depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
- default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
source "sound/pci/hda/Kconfig"
config SND_HDSP
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
0x103c0944, /* HP nc6220 */
0x103c0934, /* HP nc8220 */
0x103c006d, /* HP nx9105 */
+ 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
0x17340088, /* FSC Scenic-W */
0 /* end */
};
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca1..bd47521 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
if (!p_cache)
return NULL;
- p_cache->p_info =
- kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+ p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+ GFP_KERNEL);
if (!p_cache->p_info) {
kfree(p_cache);
return NULL;
}
- memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
p_cache->cache_size_in_bytes = size_in_bytes;
p_cache->control_count = control_count;
p_cache->p_cache = p_dsp_control_buffer;
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a2..579fc0d 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
int err;
snd_azf3328_dbgcallenter();
- if (dev >= SNDRV_CARDS)
- return -ENODEV;
+ if (dev >= SNDRV_CARDS) {
+ err = -ENODEV;
+ goto out;
+ }
if (!enable[dev]) {
dev++;
- return -ENOENT;
+ err = -ENOENT;
+ goto out;
}
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
- return err;
+ goto out;
strcpy(card->driver, "AZF3328");
strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f0..32b02d9 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
module_param_array(tea575x_tuner, int, NULL, 0444);
MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
+#define TUNER_DISABLED (1<<3)
#define TUNER_ONLY (1<<4)
#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF)
@@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip)
__end_hw:
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
- snd_tea575x_exit(&chip->tea);
+ if (!(chip->tea575x_tuner & TUNER_DISABLED))
+ snd_tea575x_exit(&chip->tea);
#endif
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -1236,7 +1238,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
if (snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- snd_fm801_free(chip);
return -ENODEV;
}
} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1251,11 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card,
}
if (tea575x_tuner == 4) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- snd_fm801_free(chip);
- return -ENODEV;
+ chip->tea575x_tuner = TUNER_DISABLED;
}
}
- strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+ if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+ strlcpy(chip->tea.card,
+ snd_fm801_tea575x_gpios[(tea575x_tuner &
+ TUNER_TYPE_MASK) - 1].name,
+ sizeof(chip->tea.card));
+ }
#endif
*rchip = chip;
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2..2e5876c 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
static const struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
- .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
- .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_DMIC] = {
- .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_ASPIRE_ONE] = {
- .mixers = { alc268_acer_aspire_one_mixer,
- alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_DELL] = {
- .mixers = { alc268_dell_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
- .mixers = { alc268_test_mixer, alc268_capture_mixer },
+ .mixers = { alc268_test_mixer },
+ .cap_mixer = alc268_capture_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs,
alc268_beep_init_verbs },
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf2..5ac0e21 100644
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..f3aefef 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
return -1;
}
recursive++;
- for (i = 0; i < nums; i++)
+ for (i = 0; i < nums; i++) {
+ unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+ if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+ continue;
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
+ }
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..c34f730 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
-
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
-
- if ((val & AC_ELDD_ELD_VALID) == 0) {
- snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
- byte_index);
- val = 0;
- }
-
- return val & AC_ELDD_ELD_DATA;
+ return val;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
if (!buf)
return -ENOMEM;
- for (i = 0; i < size; i++)
- buf[i] = hdmi_get_eld_byte(codec, nid, i);
+ for (i = 0; i < size; i++) {
+ unsigned int val = hdmi_get_eld_data(codec, nid, i);
+ if (!(val & AC_ELDD_ELD_VALID)) {
+ if (!i) {
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data byte %d\n", i);
+ val = 0;
+ } else
+ val &= AC_ELDD_ELD_DATA;
+ buf[i] = val;
+ }
ret = hdmi_update_eld(eld, buf, size);
+error:
kfree(buf);
return ret;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index be69822..e9a2a87 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1924,7 +1924,8 @@ static unsigned int azx_via_get_position(struct azx *chip,
}
static unsigned int azx_get_position(struct azx *chip,
- struct azx_dev *azx_dev)
+ struct azx_dev *azx_dev,
+ bool with_check)
{
unsigned int pos;
int stream = azx_dev->substream->stream;
@@ -1940,7 +1941,7 @@ static unsigned int azx_get_position(struct azx *chip,
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
- if (chip->position_fix[stream] == POS_FIX_AUTO) {
+ if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos || pos == (u32)-1) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
@@ -1964,7 +1965,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
struct azx *chip = apcm->chip;
struct azx_dev *azx_dev = get_azx_dev(substream);
return bytes_to_frames(substream->runtime,
- azx_get_position(chip, azx_dev));
+ azx_get_position(chip, azx_dev, false));
}
/*
@@ -1987,7 +1988,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
return -1; /* bogus (too early) interrupt */
stream = azx_dev->substream->stream;
- pos = azx_get_position(chip, azx_dev);
+ pos = azx_get_position(chip, azx_dev, true);
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i;
+ int i, idx;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
- if (*idxp >= 0)
+ idx = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (idx >= 0) {
+ *idxp = idx;
return nid;
+ }
}
return 0;
}
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[32];
+ char tmp[44];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..7696d05 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
+#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
+
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ if (filled[nums].dac)
+ continue;
+ if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+ filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+ filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
}
return nums;
}
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info,
- AUTO_PIN_LINE_OUT);
- nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_HP_OUT);
- nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info, 0,
+ AUTO_PIN_LINE_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_HP_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
+ else if (nid & DAC_SLAVE_FLAG)
+ nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ if (dac && !(dac & DAC_SLAVE_FLAG)) {
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
+ }
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60..7a73621 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */
unsigned int automute:1; /* HP automute enabled */
unsigned int detect_line:1; /* Line-out detection enabled */
- unsigned int automute_lines:1; /* automute line-out as well */
+ unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */
unsigned int automute_hp_lo:1; /* both HP and LO available */
/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
return;
- if (!spec->automute_lines || !spec->automute)
+ if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
on = 0;
else
on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute)
- return;
spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
+ if (!spec->automute)
+ return;
update_speakers(codec);
}
@@ -578,11 +578,15 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
+ /* check LO jack only when it's different from HP */
+ if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
return;
+
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
+ if (!spec->automute || !spec->detect_line)
+ return;
update_speakers(codec);
}
@@ -803,7 +807,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
unsigned int val;
if (!spec->automute)
val = 0;
- else if (!spec->automute_lines)
+ else if (!spec->automute_hp_lo || !spec->automute_lines)
val = 1;
else
val = 2;
@@ -824,7 +828,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
spec->automute = 0;
break;
case 1:
- if (spec->automute && !spec->automute_lines)
+ if (spec->automute &&
+ (!spec->automute_hp_lo || !spec->automute_lines))
return 0;
spec->automute = 1;
spec->automute_lines = 0;
@@ -1320,7 +1325,9 @@ do_sku:
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!spec->autocfg.hp_pins[0]) {
+ if (!spec->autocfg.hp_pins[0] &&
+ !(spec->autocfg.line_out_pins[0] &&
+ spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
@@ -1784,6 +1791,7 @@ static const char * const alc_slave_vols[] = {
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
+ "PCM Playback Volume",
NULL,
};
@@ -1798,6 +1806,7 @@ static const char * const alc_slave_sws[] = {
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
+ "PCM Playback Switch",
NULL,
};
@@ -3081,16 +3090,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.hp_nid);
+ if (pin) {
+ dac = spec->multiout.hp_nid;
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+ }
pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
- spec->multiout.extra_out_nid[0]);
+ if (pin) {
+ dac = spec->multiout.extra_out_nid[0];
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+ }
}
/*
@@ -4484,6 +4499,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
}
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ int coef;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ /* The digital-mic unit sends PDM (differential signal) instead of
+ * the standard PCM, thus you can't record a valid mono stream as is.
+ * Below is a workaround specific to ALC269 to control the dmic
+ * signal source as mono.
+ */
+ coef = alc_read_coef_idx(codec, 0x07);
+ alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4525,7 @@ enum {
ALC275_FIXUP_SONY_HWEQ,
ALC271_FIXUP_DMIC,
ALC269_FIXUP_PCM_44K,
+ ALC269_FIXUP_STEREO_DMIC,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4588,19 @@ static const struct alc_fixup alc269_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_pcm_44k,
},
+ [ALC269_FIXUP_STEREO_DMIC] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_stereo_dmic,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b5..987e3cf 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
+#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -5628,6 +5630,7 @@ again:
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76df:
case 0x111d76e5:
case 0x111d7666:
case 0x111d7667:
@@ -6571,6 +6574,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798..4ebfbd8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
struct nid_path *path;
bool check_dac;
- hda_nid_t pin, dac;
+ hda_nid_t pin, dac = 0;
int err;
pin = spec->autocfg.speaker_pins[0];
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67c..493e394 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
break;
case MADIface:
freq_const = 131072000000000ULL;
+ break;
+ default:
+ snd_BUG();
+ return 0;
}
return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
switch (hdspm->io_type) {
case MADIface:
- n = 131072000000000ULL; /* 125 MHz */
- break;
+ n = 131072000000000ULL; /* 125 MHz */
+ break;
case MADI:
case AES32:
- n = 110069313433624ULL; /* 105 MHz */
- break;
+ n = 110069313433624ULL; /* 105 MHz */
+ break;
case RayDAT:
case AIO:
- n = 104857600000000ULL; /* 100 MHz */
- break;
+ n = 104857600000000ULL; /* 100 MHz */
+ break;
+ default:
+ snd_BUG();
+ return;
}
n = div_u64(n, rate);
generated by cgit v0.10.2 at 2024-10-31 17:46:01 (GMT)