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authorLinus Torvalds <torvalds@linux-foundation.org>2008-07-27 09:45:59 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2008-07-27 09:45:59 -0700
commit375614422509c98a1f3dbef410206bf81775169b (patch)
tree02e65184a80446d56b6c05b76417791a3b68b234 /sound
parenteeb61f719c00c626115852bbc91189dc3011a844 (diff)
parent536319afd1f25383009c0c88f6fb00104f49c178 (diff)
downloadop-kernel-dev-375614422509c98a1f3dbef410206bf81775169b.zip
op-kernel-dev-375614422509c98a1f3dbef410206bf81775169b.tar.gz
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: Allow to force model to intel-mac-v3 in snd_hda_intel (sigmatel). ALSA: cs4232: fix crash during chip PNP detection ALSA: hda - Add automatic model setting for the Acer Aspire 5920G laptop ALSA: make snd_ac97_add_vmaster() static ALSA: sound/pci/azt3328.h: no variables for enums ALSA: soc - wm9712 mono mixer ALSA: hda - Add support of ASUS Eeepc P90* ALSA: opti9xx: no isapnp param for !CONFIG_PNP ALSA: opti93x - Fix NULL dereference ALSA: hda - Added support for Asus V1Sn ALSA: ASoC: Factor PGA DAPM handling into main ALSA: ASoC: Refactor DAPM event handler ALSA: ALSA: ens1370: communicate PCI device to AC97 ALSA: ens1370: SRC stands for Sample Rate Converter ALSA: hda - Align BDL position adjustment parameter ALSA: Au1xpsc: psc not disabled when TX is idle ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name.
Diffstat (limited to 'sound')
-rw-r--r--sound/isa/cs423x/cs4236.c1
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c6
-rw-r--r--sound/pci/ac97/ac97_codec.c3
-rw-r--r--sound/pci/ac97/ac97_patch.c4
-rw-r--r--sound/pci/azt3328.h4
-rw-r--r--sound/pci/ens1370.c3
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/patch_realtek.c181
-rw-r--r--sound/pci/hda/patch_sigmatel.c14
-rw-r--r--sound/soc/au1x/psc-i2s.c2
-rw-r--r--sound/soc/codecs/wm9712.c10
-rw-r--r--sound/soc/soc-dapm.c105
12 files changed, 259 insertions, 80 deletions
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index dbe63db4..4d4b8dd 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev)
static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard,
struct pnp_dev *pdev)
{
+ acard->wss = pdev;
if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0)
return -EBUSY;
cport[dev] = -1;
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 41c047e..0797ca4 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -68,7 +68,9 @@ MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)},"
static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
//static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */
+#ifdef CONFIG_PNP
static int isapnp = 1; /* Enable ISA PnP detection */
+#endif
static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */
static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */
static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */
@@ -85,8 +87,10 @@ module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard.");
//module_param(enable, bool, 0444);
//MODULE_PARM_DESC(enable, "Enable opti9xx soundcard.");
+#ifdef CONFIG_PNP
module_param(isapnp, bool, 0444);
MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard.");
+#endif
module_param(port, long, 0444);
MODULE_PARM_DESC(port, "WSS port # for opti9xx driver.");
module_param(mpu_port, long, 0444);
@@ -688,7 +692,7 @@ static void snd_card_opti9xx_free(struct snd_card *card)
if (chip) {
#ifdef OPTi93X
struct snd_cs4231 *codec = chip->codec;
- if (codec->irq > 0) {
+ if (codec && codec->irq > 0) {
disable_irq(codec->irq);
free_irq(codec->irq, codec);
}
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 07364c0..8c49a00a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -161,6 +161,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL },
{ 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH },
{ 0x54524102, 0xffffffff, "TR28022", NULL, NULL },
+{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL },
{ 0x54524106, 0xffffffff, "TR28026", NULL, NULL },
{ 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99]
{ 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)]
@@ -169,7 +170,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF
{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL },
{ 0x57454301, 0xffffffff, "W83971D", NULL, NULL },
-{ 0x574d4c00, 0xffffffff, "WM9701A", NULL, NULL },
+{ 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL },
{ 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL},
{ 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL},
{ 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL},
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 0746e9c..f4fbc79 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3381,8 +3381,8 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
}
/* create a virtual master control and add slaves */
-int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
- const unsigned int *tlv, const char **slaves)
+static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
+ const unsigned int *tlv, const char **slaves)
{
struct snd_kcontrol *kctl;
const char **s;
diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h
index 7e3e894..974e051 100644
--- a/sound/pci/azt3328.h
+++ b/sound/pci/azt3328.h
@@ -94,7 +94,7 @@ enum azf_freq_t {
AZF_FREQ(48000),
AZF_FREQ(66200),
#undef AZF_FREQ
-} AZF_FREQUENCIES;
+};
/** recording area (see also: playback bit flag definitions) **/
#define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */
@@ -210,7 +210,7 @@ enum azf_freq_t {
enum {
AZF_GAME_LEGACY_IO_PORT = 0x200
-} AZF_GAME_CONFIGS;
+};
#define IDX_GAME_LEGACY_COMPATIBLE 0x00
/* in some operation mode, writing anything to this port
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index fbf1124..9bf9536 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -522,7 +522,7 @@ static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq)
return r;
cond_resched();
}
- snd_printk(KERN_ERR "wait source ready timeout 0x%lx [0x%x]\n",
+ snd_printk(KERN_ERR "wait src ready timeout 0x%lx [0x%x]\n",
ES_REG(ensoniq, 1371_SMPRATE), r);
return 0;
}
@@ -1629,6 +1629,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq,
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = ensoniq;
ac97.private_free = snd_ensoniq_mixer_free_ac97;
+ ac97.pci = ensoniq->pci;
ac97.scaps = AC97_SCAP_AUDIO;
if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0)
return err;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 16715a6..ef9f072 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1047,9 +1047,13 @@ static int azx_setup_periods(struct azx *chip,
pos_adj = bdl_pos_adj[chip->dev_index];
if (pos_adj > 0) {
struct snd_pcm_runtime *runtime = substream->runtime;
+ int pos_align = pos_adj;
pos_adj = (pos_adj * runtime->rate + 47999) / 48000;
if (!pos_adj)
- pos_adj = 1;
+ pos_adj = pos_align;
+ else
+ pos_adj = ((pos_adj + pos_align - 1) / pos_align) *
+ pos_align;
pos_adj = frames_to_bytes(runtime, pos_adj);
if (pos_adj >= period_bytes) {
snd_printk(KERN_WARNING "Too big adjustment %d\n",
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2807bc8..add4e87 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -122,6 +122,8 @@ enum {
/* ALC269 models */
enum {
ALC269_BASIC,
+ ALC269_ASUS_EEEPC_P703,
+ ALC269_ASUS_EEEPC_P901,
ALC269_AUTO,
ALC269_MODEL_LAST /* last tag */
};
@@ -7905,6 +7907,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -10946,7 +10949,23 @@ static int patch_alc268(struct hda_codec *codec)
static hda_nid_t alc269_adc_nids[1] = {
/* ADC1 */
- 0x07,
+ 0x08,
+};
+
+static struct hda_input_mux alc269_eeepc_dmic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x5 },
+ { "e-Mic", 0x0 },
+ },
+};
+
+static struct hda_input_mux alc269_eeepc_amic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x1 },
+ { "e-Mic", 0x0 },
+ },
};
#define alc269_modes alc260_modes
@@ -10968,10 +10987,27 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
{ } /* end */
};
+/* bind volumes of both NID 0x0c and 0x0d */
+static struct hda_bind_ctls alc269_epc_bind_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
+ HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
+ HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
/* capture mixer elements */
static struct snd_kcontrol_new alc269_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
@@ -10987,6 +11023,13 @@ static struct snd_kcontrol_new alc269_capture_mixer[] = {
{ } /* end */
};
+/* capture mixer elements */
+static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -10994,7 +11037,7 @@ static struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
* analog-loopback mixer widget
@@ -11057,6 +11100,98 @@ static struct hda_verb alc269_init_verbs[] = {
{ }
};
+static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned int bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? AMP_IN_MUTE(0) : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 0 : 5);
+}
+
+static void alc269_eeepc_amic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0));
+ snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1));
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc269_speaker_automute(codec);
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc269_eeepc_dmic_automute(codec);
+}
+
+static void alc269_eeepc_dmic_inithook(struct hda_codec *codec)
+{
+ alc269_speaker_automute(codec);
+ alc269_eeepc_dmic_automute(codec);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc269_speaker_automute(codec);
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc269_eeepc_amic_automute(codec);
+}
+
+static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
+{
+ alc269_speaker_automute(codec);
+ alc269_eeepc_amic_automute(codec);
+}
+
/* add playback controls from the parsed DAC table */
static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -11188,6 +11323,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ spec->mixers[spec->num_mixers] = alc269_capture_mixer;
+ spec->num_mixers++;
+
return 1;
}
@@ -11215,12 +11353,16 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
};
static struct snd_pci_quirk alc269_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
+ ALC269_ASUS_EEEPC_P703),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
+ ALC269_ASUS_EEEPC_P901),
{}
};
static struct alc_config_preset alc269_presets[] = {
[ALC269_BASIC] = {
- .mixers = { alc269_base_mixer },
+ .mixers = { alc269_base_mixer, alc269_capture_mixer },
.init_verbs = { alc269_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
@@ -11229,6 +11371,32 @@ static struct alc_config_preset alc269_presets[] = {
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
},
+ [ALC269_ASUS_EEEPC_P703] = {
+ .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
+ .init_verbs = { alc269_init_verbs,
+ alc269_eeepc_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_eeepc_amic_capture_source,
+ .unsol_event = alc269_eeepc_amic_unsol_event,
+ .init_hook = alc269_eeepc_amic_inithook,
+ },
+ [ALC269_ASUS_EEEPC_P901] = {
+ .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer},
+ .init_verbs = { alc269_init_verbs,
+ alc269_eeepc_dmic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_eeepc_dmic_capture_source,
+ .unsol_event = alc269_eeepc_dmic_unsol_event,
+ .init_hook = alc269_eeepc_dmic_inithook,
+ },
};
static int patch_alc269(struct hda_codec *codec)
@@ -11282,8 +11450,6 @@ static int patch_alc269(struct hda_codec *codec)
spec->adc_nids = alc269_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
- spec->mixers[spec->num_mixers] = alc269_capture_mixer;
- spec->num_mixers++;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC269_AUTO)
@@ -12994,6 +13160,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 08cb77f..7fdafcb 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -94,6 +94,9 @@ enum {
STAC_INTEL_MAC_V3,
STAC_INTEL_MAC_V4,
STAC_INTEL_MAC_V5,
+ STAC_INTEL_MAC_AUTO, /* This model is selected if no module parameter
+ * is given, one of the above models will be
+ * chosen according to the subsystem id. */
/* for backward compatibility */
STAC_MACMINI,
STAC_MACBOOK,
@@ -1483,6 +1486,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
[STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs,
[STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs,
[STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs,
+ [STAC_INTEL_MAC_AUTO] = intel_mac_v3_pin_configs,
/* for backward compatibility */
[STAC_MACMINI] = intel_mac_v3_pin_configs,
[STAC_MACBOOK] = intel_mac_v5_pin_configs,
@@ -1505,6 +1509,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = {
[STAC_INTEL_MAC_V3] = "intel-mac-v3",
[STAC_INTEL_MAC_V4] = "intel-mac-v4",
[STAC_INTEL_MAC_V5] = "intel-mac-v5",
+ [STAC_INTEL_MAC_AUTO] = "intel-mac-auto",
/* for backward compatibility */
[STAC_MACMINI] = "macmini",
[STAC_MACBOOK] = "macbook",
@@ -1576,9 +1581,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707,
"Intel D945P", STAC_D945GTP5),
/* other systems */
- /* Apple Mac Mini (early 2006) */
+ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */
SND_PCI_QUIRK(0x8384, 0x7680,
- "Mac Mini", STAC_INTEL_MAC_V3),
+ "Mac", STAC_INTEL_MAC_AUTO),
/* Dell systems */
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7,
"unknown Dell", STAC_922X_DELL_D81),
@@ -3725,7 +3730,7 @@ static int patch_stac922x(struct hda_codec *codec)
spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS,
stac922x_models,
stac922x_cfg_tbl);
- if (spec->board_config == STAC_INTEL_MAC_V3) {
+ if (spec->board_config == STAC_INTEL_MAC_AUTO) {
spec->gpio_mask = spec->gpio_dir = 0x03;
spec->gpio_data = 0x03;
/* Intel Macs have all same PCI SSID, so we need to check
@@ -3757,6 +3762,9 @@ static int patch_stac922x(struct hda_codec *codec)
case 0x106b2200:
spec->board_config = STAC_INTEL_MAC_V5;
break;
+ default:
+ spec->board_config = STAC_INTEL_MAC_V3;
+ break;
}
}
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index ba4b5c1..9384702 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -231,7 +231,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
/* if both TX and RX are idle, disable PSC */
stat = au_readl(I2S_STAT(pscdata));
- if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) {
+ if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
au_writel(0, I2S_CFG(pscdata));
au_sync();
au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 9fc8edd..1fb7f9a 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -427,20 +427,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"HPOUTR", NULL, "Headphone PGA"},
{"Headphone PGA", NULL, "Right HP Mixer"},
- /* mono hp mixer */
- {"Mono HP Mixer", NULL, "Left HP Mixer"},
- {"Mono HP Mixer", NULL, "Right HP Mixer"},
+ /* mono mixer */
+ {"Mono Mixer", NULL, "Left HP Mixer"},
+ {"Mono Mixer", NULL, "Right HP Mixer"},
/* Out3 Mux */
{"Out3 Mux", "Left", "Left HP Mixer"},
{"Out3 Mux", "Mono", "Phone Mixer"},
- {"Out3 Mux", "Left + Right", "Mono HP Mixer"},
+ {"Out3 Mux", "Left + Right", "Mono Mixer"},
{"Out 3 PGA", NULL, "Out3 Mux"},
{"OUT3", NULL, "Out 3 PGA"},
/* speaker Mux */
{"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
- {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"},
+ {"Speaker Mux", "Headphone Mix", "Mono Mixer"},
{"Speaker PGA", NULL, "Speaker Mux"},
{"LOUT2", NULL, "Speaker PGA"},
{"ROUT2", NULL, "Speaker PGA"},
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2c87061..820347c 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -523,24 +523,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
continue;
}
- /* programmable gain/attenuation */
- if (w->id == snd_soc_dapm_pga) {
- int on;
- in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
- out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
- w->power = on = (out != 0 && in != 0) ? 1 : 0;
-
- if (!on)
- dapm_set_pga(w, on); /* lower volume to reduce pops */
- dapm_update_bits(w);
- if (on)
- dapm_set_pga(w, on); /* restore volume from zero */
-
- continue;
- }
-
/* pre and post event widgets */
if (w->id == snd_soc_dapm_pre) {
if (!w->event)
@@ -586,45 +568,56 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
power_change = (w->power == power) ? 0: 1;
w->power = power;
+ if (!power_change)
+ continue;
+
/* call any power change event handlers */
- if (power_change) {
- if (w->event) {
- pr_debug("power %s event for %s flags %x\n",
- w->power ? "on" : "off", w->name, w->event_flags);
- if (power) {
- /* power up event */
- if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMU);
- if (ret < 0)
- return ret;
- }
- dapm_update_bits(w);
- if (w->event_flags & SND_SOC_DAPM_POST_PMU){
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMU);
- if (ret < 0)
- return ret;
- }
- } else {
- /* power down event */
- if (w->event_flags & SND_SOC_DAPM_PRE_PMD) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMD);
- if (ret < 0)
- return ret;
- }
- dapm_update_bits(w);
- if (w->event_flags & SND_SOC_DAPM_POST_PMD) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMD);
- if (ret < 0)
- return ret;
- }
- }
- } else
- /* no event handler */
- dapm_update_bits(w);
+ if (w->event)
+ pr_debug("power %s event for %s flags %x\n",
+ w->power ? "on" : "off",
+ w->name, w->event_flags);
+
+ /* power up pre event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down pre event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* Lower PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && !power)
+ dapm_set_pga(w, power);
+
+ dapm_update_bits(w);
+
+ /* Raise PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && power)
+ dapm_set_pga(w, power);
+
+ /* power up post event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down post event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
}
}
}
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