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-rw-r--r--sound/aoa/codecs/tas.c2
-rw-r--r--sound/pci/au88x0/au88x0_pcm.c7
-rw-r--r--sound/pci/hda/hda_codec.c4
-rw-r--r--sound/pci/hda/patch_realtek.c55
-rw-r--r--sound/pci/hda/patch_via.c10
-rw-r--r--sound/soc/codecs/jz4740.c2
-rw-r--r--sound/soc/codecs/sn95031.c2
-rw-r--r--sound/soc/codecs/ssm2602.c10
-rw-r--r--sound/soc/codecs/uda134x.c2
-rw-r--r--sound/soc/codecs/wm8903.c40
-rw-r--r--sound/soc/codecs/wm8994.c16
-rw-r--r--sound/soc/codecs/wm_hubs.c8
-rw-r--r--sound/soc/davinci/davinci-mcasp.c19
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c2
-rw-r--r--sound/soc/mid-x86/sst_platform.c16
-rw-r--r--sound/soc/samsung/goni_wm8994.c8
-rw-r--r--sound/soc/samsung/pcm.c4
-rw-r--r--sound/soc/sh/fsi.c22
-rw-r--r--sound/soc/soc-core.c7
-rw-r--r--sound/soc/tegra/harmony.c1
-rw-r--r--sound/usb/format.c4
-rw-r--r--sound/usb/quirks.c1
22 files changed, 160 insertions, 82 deletions
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index 58804c7acfcf..fd2188c3df2b 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@ static void tas_set_volume(struct tas *tas)
/* analysing the volume and mixer tables shows
* that they are similar enough when we shift
* the mixer table down by 4 bits. The error
- * is minuscule, in just one item the error
+ * is miniscule, in just one item the error
* is 1, at a value of 0x07f17b (mixer table
* value is 0x07f17a) */
tmp = tas_gaintable[left];
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 33f0ba5559a7..62e959120c44 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 0x10000,
- .period_bytes_min = 0x1,
+ .period_bytes_min = 0x20,
.period_bytes_max = 0x1000,
.periods_min = 2,
- .periods_max = 32,
+ .periods_max = 1024,
};
#ifndef CHIP_AU8820
@@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
return err;
+ snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
+
if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
#ifndef CHIP_AU8820
if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 430f41db6044..759ade12e758 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
}
EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
+#ifdef SND_HDA_NEEDS_RESUME
/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
static void restore_shutup_pins(struct hda_codec *codec)
{
@@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec)
}
codec->pins_shutup = 0;
}
+#endif
static void init_hda_cache(struct hda_cache_rec *cache,
unsigned int record_size);
@@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
}
}
+#ifdef SND_HDA_NEEDS_RESUME
/* clean up all streams; called from suspend */
static void hda_cleanup_all_streams(struct hda_codec *codec)
{
@@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
really_cleanup_stream(codec, p);
}
}
+#endif
/*
* amp access functions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52928d9a72da..c82979a8cd09 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1704,11 +1704,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
codec->chip_name, fix->type);
break;
}
- if (!fix[id].chained)
+ if (!fix->chained)
break;
if (++depth > 10)
break;
- id = fix[id].chain_id;
+ id = fix->chain_id;
}
}
@@ -5645,6 +5645,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
static struct snd_pci_quirk beep_white_list[] = {
SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+ SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
{}
};
@@ -9863,6 +9864,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -10699,7 +10701,6 @@ enum {
PINFIX_LENOVO_Y530,
PINFIX_PB_M5210,
PINFIX_ACER_ASPIRE_7736,
- PINFIX_GIGABYTE_880GM,
};
static const struct alc_fixup alc882_fixups[] = {
@@ -10731,13 +10732,6 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_SKU,
.v.sku = ALC_FIXUP_SKU_IGNORE,
},
- [PINFIX_GIGABYTE_880GM] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x1114410 }, /* set as speaker */
- { }
- }
- },
};
static struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -10745,7 +10739,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
{}
};
@@ -14868,6 +14861,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
alc_write_coef_idx(codec, 0x1e, coef | 0x80);
}
+static void alc271_fixup_dmic(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ static struct hda_verb verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+ {}
+ };
+ unsigned int cfg;
+
+ if (strcmp(codec->chip_name, "ALC271X"))
+ return;
+ cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+ if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+ snd_hda_sequence_write(codec, verbs);
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -14876,6 +14886,7 @@ enum {
ALC269_FIXUP_ASUS_G73JW,
ALC269_FIXUP_LENOVO_EAPD,
ALC275_FIXUP_SONY_HWEQ,
+ ALC271_FIXUP_DMIC,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -14929,7 +14940,11 @@ static const struct alc_fixup alc269_fixups[] = {
.v.func = alc269_fixup_hweq,
.chained = true,
.chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
- }
+ },
+ [ALC271_FIXUP_DMIC] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc271_fixup_dmic,
+ },
};
static struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -14938,6 +14953,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -18782,6 +18798,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+ ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -19455,7 +19473,7 @@ enum {
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
ALC662_FIXUP_CZC_P10T,
- ALC662_FIXUP_GIGABYTE,
+ ALC662_FIXUP_SKU_IGNORE,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19484,20 +19502,17 @@ static const struct alc_fixup alc662_fixups[] = {
{}
}
},
- [ALC662_FIXUP_GIGABYTE] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x1114410 }, /* set as speaker */
- { }
- }
+ [ALC662_FIXUP_SKU_IGNORE] = {
+ .type = ALC_FIXUP_SKU,
+ .v.sku = ALC_FIXUP_SKU_IGNORE,
},
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 1371b57c11e8..0997031c48d2 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1292,14 +1292,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec)
{
int i;
struct snd_ctl_elem_id id;
- const char *labels[] = {"Mic", "Front Mic", "Line"};
+ const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
+ struct snd_kcontrol *ctl;
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
for (i = 0; i < ARRAY_SIZE(labels); i++) {
sprintf(id.name, "%s Playback Volume", labels[i]);
- snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &id);
+ ctl = snd_hda_find_mixer_ctl(codec, id.name);
+ if (ctl)
+ snd_ctl_notify(codec->bus->card,
+ SNDRV_CTL_EVENT_MASK_VALUE,
+ &ctl->id);
}
}
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346fd727..f5ccdbf7ebc6 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
ARRAY_SIZE(jz4740_codec_dapm_routes));
- snd_soc_dapm_new_widgets(codec);
-
jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index a54d2a5b28f6..4d9fb279e146 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -927,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
.owner = THIS_MODULE,
},
.probe = sn95031_device_probe,
- .remove = sn95031_device_remove,
+ .remove = __devexit_p(sn95031_device_remove),
};
static int __init sn95031_init(void)
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 2727befd158e..b04d28039c16 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -139,7 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
-SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
@@ -602,7 +602,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.read = ssm2602_read_reg_cache,
.write = ssm2602_write,
.set_bias_level = ssm2602_set_bias_level,
- .reg_cache_size = sizeof(ssm2602_reg),
+ .reg_cache_size = ARRAY_SIZE(ssm2602_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ssm2602_reg,
};
@@ -614,7 +614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
* low = 0x1a
* high = 0x1b
*/
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
+static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ssm2602_priv *ssm2602;
@@ -635,7 +635,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
return ret;
}
-static int ssm2602_i2c_remove(struct i2c_client *client)
+static int __devexit ssm2602_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
@@ -655,7 +655,7 @@ static struct i2c_driver ssm2602_i2c_driver = {
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
- .remove = ssm2602_i2c_remove,
+ .remove = __devexit_p(ssm2602_i2c_remove),
.id_table = ssm2602_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 48ffd406a71d..a7b8f301bad3 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -601,9 +601,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
.write = uda134x_write,
-#ifdef POWER_OFF_ON_STANDBY
.set_bias_level = uda134x_set_bias_level,
-#endif
};
static int __devinit uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadfae84c..824d1c8c8a35 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
case WM8903_REVISION_NUMBER:
case WM8903_INTERRUPT_STATUS_1:
case WM8903_WRITE_SEQUENCER_4:
- case WM8903_POWER_MANAGEMENT_3:
- case WM8903_POWER_MANAGEMENT_2:
case WM8903_DC_SERVO_READBACK_1:
case WM8903_DC_SERVO_READBACK_2:
case WM8903_DC_SERVO_READBACK_3:
@@ -694,7 +692,7 @@ SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
- WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
+ WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
- 4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+ 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
0, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
NULL, 0),
SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
- { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
- { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
- { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
- { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+ { "HPL_ENA", NULL, "Left Headphone Output PGA" },
+ { "HPR_ENA", NULL, "Right Headphone Output PGA" },
+ { "HPL_ENA_DLY", NULL, "HPL_ENA" },
+ { "HPR_ENA_DLY", NULL, "HPR_ENA" },
+ { "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+ { "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+ { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+ { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
{ "HPL_DCS", NULL, "DCS Master" },
{ "HPR_DCS", NULL, "DCS Master" },
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3290333b2bb9..84e1bd1d2822 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch volume updates (right only; we always do left then right). */
+ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+ WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+ WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+ snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+ WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+ WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+ snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+ WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+ snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+ WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+ snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+ WM8994_DAC1_VU, WM8994_DAC1_VU);
snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
WM8994_DAC1_VU, WM8994_DAC1_VU);
+ snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+ WM8994_DAC2_VU, WM8994_DAC2_VU);
snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
WM8994_DAC2_VU, WM8994_DAC2_VU);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c18e299..4005e9af5d61 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
{ "SPKL", "Input Switch", "MIXINL" },
{ "SPKL", "IN1LP Switch", "IN1LP" },
- { "SPKL", "Output Switch", "Left Output Mixer" },
+ { "SPKL", "Output Switch", "Left Output PGA" },
{ "SPKL", NULL, "TOCLK" },
{ "SPKR", "Input Switch", "MIXINR" },
{ "SPKR", "IN1RP Switch", "IN1RP" },
- { "SPKR", "Output Switch", "Right Output Mixer" },
+ { "SPKR", "Output Switch", "Right Output PGA" },
{ "SPKR", NULL, "TOCLK" },
{ "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
{ "SPKOUTRP", NULL, "SPKR Driver" },
{ "SPKOUTRN", NULL, "SPKR Driver" },
- { "Left Headphone Mux", "Mixer", "Left Output Mixer" },
- { "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+ { "Left Headphone Mux", "Mixer", "Left Output PGA" },
+ { "Right Headphone Mux", "Mixer", "Right Output PGA" },
{ "Headphone PGA", NULL, "Left Headphone Mux" },
{ "Headphone PGA", NULL, "Right Headphone Mux" },
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index a5af834c8ef5..4ddc6d3b6678 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -434,17 +434,21 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26));
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | AHCLKX | AFSX);
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
- mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
- mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26));
+ mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | ACLKR);
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+ AFSX | AFSR);
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is clock and frame master */
@@ -454,7 +458,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26));
+ mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
break;
default:
@@ -644,7 +649,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
- if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
else
@@ -660,7 +665,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
AHCLKRE);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
- if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
else
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 419bf4f5534a..cd22a54b2f14 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -133,7 +133,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream,
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
uint32_t conf;
- if (!dai->active)
+ if (dai->active)
return;
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index b2e9198a983a..6b1f9d3bf34e 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
int state)
{
- spin_lock(&stream->status_lock);
+ unsigned long flags;
+ spin_lock_irqsave(&stream->status_lock, flags);
stream->stream_status = state;
- spin_unlock(&stream->status_lock);
+ spin_unlock_irqrestore(&stream->status_lock, flags);
}
static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
{
int state;
+ unsigned long flags;
- spin_lock(&stream->status_lock);
+ spin_lock_irqsave(&stream->status_lock, flags);
state = stream->stream_status;
- spin_unlock(&stream->status_lock);
+ spin_unlock_irqrestore(&stream->status_lock, flags);
return state;
}
@@ -374,6 +376,11 @@ static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
static struct snd_pcm_ops sst_platform_ops = {
.open = sst_platform_open,
.close = sst_platform_close,
@@ -382,6 +389,7 @@ static struct snd_pcm_ops sst_platform_ops = {
.trigger = sst_platform_pcm_trigger,
.pointer = sst_platform_pcm_pointer,
.hw_params = sst_platform_pcm_hw_params,
+ .hw_free = sst_platform_pcm_hw_free,
};
static void sst_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index f6b3a3ce5919..0e80daee8b6f 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -236,18 +236,18 @@ static struct snd_soc_dai_link goni_dai[] = {
.name = "WM8994",
.stream_name = "WM8994 HiFi",
.cpu_dai_name = "samsung-i2s.0",
- .codec_dai_name = "wm8994-hifi",
+ .codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-audio",
- .codec_name = "wm8994-codec.0-0x1a",
+ .codec_name = "wm8994-codec.0-001a",
.init = goni_wm8994_init,
.ops = &goni_hifi_ops,
}, {
.name = "WM8994 Voice",
.stream_name = "Voice",
.cpu_dai_name = "goni-voice-dai",
- .codec_dai_name = "wm8994-voice",
+ .codec_dai_name = "wm8994-aif2",
.platform_name = "samsung-audio",
- .codec_name = "wm8994-codec.0-0x1a",
+ .codec_name = "wm8994-codec.0-001a",
.ops = &goni_voice_ops,
},
};
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 38aac7d57a59..9c7e8b48aed6 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
ctl = readl(regs + S3C_PCM_CTL);
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_NB_NF:
- /* Nothing to do, NB_NF by default */
+ case SND_SOC_DAIFMT_IB_NF:
+ /* Nothing to do, IB_NF by default */
break;
default:
dev_err(pcm->dev, "Unsupported clock inversion!\n");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0c9997e2d8c0..23c0e83d4c19 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev)
master->fsib.master = master;
pm_runtime_enable(&pdev->dev);
- pm_runtime_resume(&pdev->dev);
dev_set_drvdata(&pdev->dev, master);
+ pm_runtime_get_sync(&pdev->dev);
fsi_soft_all_reset(master);
+ pm_runtime_put_sync(&pdev->dev);
ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
id_entry->name, master);
@@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev)
goto exit_free_irq;
}
- return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+ ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
+ ARRAY_SIZE(fsi_soc_dai));
+ if (ret < 0) {
+ dev_err(&pdev->dev, "cannot snd dai register\n");
+ goto exit_snd_soc;
+ }
+
+ return ret;
+exit_snd_soc:
+ snd_soc_unregister_platform(&pdev->dev);
exit_free_irq:
free_irq(irq, master);
exit_iounmap:
@@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev)
master = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
- snd_soc_unregister_platform(&pdev->dev);
-
+ free_irq(master->irq, master);
pm_runtime_disable(&pdev->dev);
- free_irq(master->irq, master);
+ snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+ snd_soc_unregister_platform(&pdev->dev);
iounmap(master->base);
kfree(master);
@@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b76b74db0968..dd55d1069468 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rates |= codec_dai_drv->capture.rates;
}
+ ret = -EINVAL;
snd_pcm_limit_hw_rates(runtime);
if (!runtime->hw.rates) {
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
@@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
codec_dai->name, cpu_dai->name);
goto config_err;
}
- if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+ if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+ runtime->hw.channels_min > runtime->hw.channels_max) {
printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
codec_dai->name, cpu_dai->name);
goto config_err;
@@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = {
.resume = snd_soc_resume,
.poweroff = snd_soc_poweroff,
};
+EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
/* ASoC platform driver */
static struct platform_driver soc_driver = {
@@ -3288,6 +3291,8 @@ int snd_soc_register_card(struct snd_soc_card *card)
if (!card->name || !card->dev)
return -EINVAL;
+ dev_set_drvdata(card->dev, card);
+
snd_soc_initialize_card_lists(card);
soc_init_card_debugfs(card);
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 8585957477eb..556a57133925 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = {
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = tegra_snd_harmony_probe,
.remove = __devexit_p(tegra_snd_harmony_remove),
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5b792d2c8061..f079b5e2ab28 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -176,9 +176,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
if (!rate)
continue;
/* C-Media CM6501 mislabels its 96 kHz altsetting */
+ /* Terratec Aureon 7.1 USB C-Media 6206, too */
if (rate == 48000 && nr_rates == 1 &&
(chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
- chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+ chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+ chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
fp->altsetting == 5 && fp->maxpacksize == 392)
rate = 96000;
/* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ec07e62e53f3..1b94ec3a3368 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
case USB_ID(0x0d8c, 0x0102):
/* C-Media CM6206 / CM106-Like Sound Device */
+ case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
return snd_usb_cm6206_boot_quirk(dev);
case USB_ID(0x133e, 0x0815):
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-20 10:15:54 +0100