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authorTakashi Iwai2011-12-01 13:51:18 +0100
committerTakashi Iwai2011-12-01 13:51:18 +0100
commit9eb6e9b16f86ef94f05427bd7fc84d521aa57169 (patch)
treeb115e55cf7f86c7bdb5453f83af35361a9c6643b
parentALSA: hda - Increase the max number of coverters/pins in patch_hdmi.c (diff)
parentALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED (diff)
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Merge branch 'fix/hda' into topic/hda
-rw-r--r--MAINTAINERS1
-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c22
-rw-r--r--sound/pci/hda/patch_via.c76
-rw-r--r--sound/soc/codecs/adau1373.c2
-rw-r--r--sound/soc/codecs/cs4271.c8
-rw-r--r--sound/soc/codecs/rt5631.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sta32x.c63
-rw-r--r--sound/soc/codecs/sta32x.h1
-rw-r--r--sound/soc/codecs/wm8731.c1
-rw-r--r--sound/soc/codecs/wm8753.c3
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm9081.c10
-rw-r--r--sound/soc/codecs/wm9090.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/usb/quirks-table.h31
20 files changed, 166 insertions, 74 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index c802e5fa2d11..fd7e441b5ea7 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5648,7 +5648,6 @@ F: drivers/media/video/*7146*
F: include/media/*7146*
SAMSUNG AUDIO (ASoC) DRIVERS
-M: Jassi Brar <jassisinghbrar@gmail.com>
M: Sangbeom Kim <sbkim73@samsung.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ddd7f3b21cdc..d1582ddb426c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2507,7 +2507,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7bd2a52f2bac..70a7abda7e22 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1278,7 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
[CS420X_MBP53] = "mbp53",
[CS420X_MBP55] = "mbp55",
[CS420X_IMAC27] = "imac27",
- [CS420X_IMAC27] = "apple",
+ [CS420X_APPLE] = "apple",
[CS420X_AUTO] = "auto",
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f3658658548e..d8d2f9dccd9b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4441,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec)
int pinctl, def_conf;
/* power on when no jack detection is available */
- if (!spec->hp_detect) {
+ /* or when the VREF is used for controlling LED */
+ if (!spec->hp_detect ||
+ (spec->gpio_led > 8 && spec->gpio_led == nid)) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -5055,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
return 0;
}
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
- if (spec->gpio_led > 8) {
- /* with vref-out pin used for mute led control
- * codec AFG is prevented from D3 state, but on
- * system suspend it can (and should) be used
- */
- snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- }
- return 0;
-}
-
static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
@@ -5668,8 +5656,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
@@ -5983,8 +5969,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 431c0d417eeb..b5137629f8e9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -208,6 +208,7 @@ struct via_spec {
/* work to check hp jack state */
struct hda_codec *codec;
struct delayed_work vt1708_hp_work;
+ int hp_work_active;
int vt1708_jack_detect;
int vt1708_hp_present;
@@ -305,27 +306,35 @@ enum {
static void analog_low_current_mode(struct hda_codec *codec);
static bool is_aa_path_mute(struct hda_codec *codec);
-static void vt1708_start_hp_work(struct via_spec *spec)
+#define hp_detect_with_aa(codec) \
+ (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1 && \
+ !is_aa_path_mute(codec))
+
+static void vt1708_stop_hp_work(struct via_spec *spec)
{
if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
return;
- snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detect);
- if (!delayed_work_pending(&spec->vt1708_hp_work))
- schedule_delayed_work(&spec->vt1708_hp_work,
- msecs_to_jiffies(100));
+ if (spec->hp_work_active) {
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 1);
+ cancel_delayed_work_sync(&spec->vt1708_hp_work);
+ spec->hp_work_active = 0;
+ }
}
-static void vt1708_stop_hp_work(struct via_spec *spec)
+static void vt1708_update_hp_work(struct via_spec *spec)
{
if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
return;
- if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1
- && !is_aa_path_mute(spec->codec))
- return;
- snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detect);
- cancel_delayed_work_sync(&spec->vt1708_hp_work);
+ if (spec->vt1708_jack_detect &&
+ (spec->active_streams || hp_detect_with_aa(spec->codec))) {
+ if (!spec->hp_work_active) {
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 0);
+ schedule_delayed_work(&spec->vt1708_hp_work,
+ msecs_to_jiffies(100));
+ spec->hp_work_active = 1;
+ }
+ } else if (!hp_detect_with_aa(spec->codec))
+ vt1708_stop_hp_work(spec);
}
static void set_widgets_power_state(struct hda_codec *codec)
@@ -343,12 +352,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
set_widgets_power_state(codec);
analog_low_current_mode(snd_kcontrol_chip(kcontrol));
- if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
- if (is_aa_path_mute(codec))
- vt1708_start_hp_work(codec->spec);
- else
- vt1708_stop_hp_work(codec->spec);
- }
+ vt1708_update_hp_work(codec->spec);
return change;
}
@@ -1154,7 +1158,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
spec->cur_dac_stream_tag = stream_tag;
spec->cur_dac_format = format;
mutex_unlock(&spec->config_mutex);
- vt1708_start_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1174,7 +1178,7 @@ static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo,
spec->cur_hp_stream_tag = stream_tag;
spec->cur_hp_format = format;
mutex_unlock(&spec->config_mutex);
- vt1708_start_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1188,7 +1192,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
spec->active_streams &= ~STREAM_MULTI_OUT;
mutex_unlock(&spec->config_mutex);
- vt1708_stop_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1203,7 +1207,7 @@ static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0);
spec->active_streams &= ~STREAM_INDEP_HP;
mutex_unlock(&spec->config_mutex);
- vt1708_stop_hp_work(spec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -1645,7 +1649,8 @@ static void via_hp_automute(struct hda_codec *codec)
int nums;
struct via_spec *spec = codec->spec;
- if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0])
+ if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] &&
+ (spec->codec_type != VT1708 || spec->vt1708_jack_detect))
present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
if (spec->smart51_enabled)
@@ -2612,8 +2617,6 @@ static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol,
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detect =
- !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1);
ucontrol->value.integer.value[0] = spec->vt1708_jack_detect;
return 0;
}
@@ -2623,18 +2626,22 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- int change;
+ int val;
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detect = ucontrol->value.integer.value[0];
- change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8))
- == !spec->vt1708_jack_detect;
- if (spec->vt1708_jack_detect) {
+ val = !!ucontrol->value.integer.value[0];
+ if (spec->vt1708_jack_detect == val)
+ return 0;
+ spec->vt1708_jack_detect = val;
+ if (spec->vt1708_jack_detect &&
+ snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") != 1) {
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
}
- return change;
+ via_hp_automute(codec);
+ vt1708_update_hp_work(spec);
+ return 1;
}
static const struct snd_kcontrol_new vt1708_jack_detect_ctl = {
@@ -2771,6 +2778,7 @@ static int via_init(struct hda_codec *codec)
via_auto_init_unsol_event(codec);
via_hp_automute(codec);
+ vt1708_update_hp_work(spec);
return 0;
}
@@ -2787,7 +2795,9 @@ static void vt1708_update_hp_jack_state(struct work_struct *work)
spec->vt1708_hp_present ^= 1;
via_hp_automute(spec->codec);
}
- vt1708_start_hp_work(spec);
+ if (spec->vt1708_jack_detect)
+ schedule_delayed_work(&spec->vt1708_hp_work,
+ msecs_to_jiffies(100));
}
static int get_mux_nids(struct hda_codec *codec)
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ccf8dd47576..45c63028b40d 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = {
};
static const unsigned int adau1373_bass_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(3),
0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 23d1bd5dadda..69fde1506fe1 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
{
int ret;
/* Set power-down bit */
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
+ CS4271_MODE2_PDN);
if (ret < 0)
return ret;
return 0;
@@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec)
return ret;
}
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
- CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
if (ret < 0)
return ret;
ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 27a078cbb6eb..4646e808b90a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */
static unsigned int mic_bst_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(7),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d15695d1c273..bbcf921166f7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
/* tlv for mic gain, 0db 20db 30db 40db */
static const unsigned int mic_gain_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
};
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index bb82408ab8e1..d2f37152f940 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -76,6 +76,8 @@ struct sta32x_priv {
unsigned int mclk;
unsigned int format;
+
+ u32 coef_shadow[STA32X_COEF_COUNT];
};
static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
@@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
int numcoef = kcontrol->private_value >> 16;
int index = kcontrol->private_value & 0xffff;
unsigned int cfud;
@@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
snd_soc_write(codec, STA32X_CFUD, cfud);
snd_soc_write(codec, STA32X_CFADDR2, index);
+ for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++)
+ sta32x->coef_shadow[index + i] =
+ (ucontrol->value.bytes.data[3 * i] << 16)
+ | (ucontrol->value.bytes.data[3 * i + 1] << 8)
+ | (ucontrol->value.bytes.data[3 * i + 2]);
for (i = 0; i < 3 * numcoef; i++)
snd_soc_write(codec, STA32X_B1CF1 + i,
ucontrol->value.bytes.data[i]);
@@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
return 0;
}
+int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ unsigned int cfud;
+ int i;
+
+ /* preserve reserved bits in STA32X_CFUD */
+ cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+
+ for (i = 0; i < STA32X_COEF_COUNT; i++) {
+ snd_soc_write(codec, STA32X_CFADDR2, i);
+ snd_soc_write(codec, STA32X_B1CF1,
+ (sta32x->coef_shadow[i] >> 16) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF2,
+ (sta32x->coef_shadow[i] >> 8) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF3,
+ (sta32x->coef_shadow[i]) & 0xff);
+ /* chip documentation does not say if the bits are
+ * self-clearing, so do it explicitly */
+ snd_soc_write(codec, STA32X_CFUD, cfud);
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x01);
+ }
+ return 0;
+}
+
+int sta32x_cache_sync(struct snd_soc_codec *codec)
+{
+ unsigned int mute;
+ int rc;
+
+ if (!codec->cache_sync)
+ return 0;
+
+ /* mute during register sync */
+ mute = snd_soc_read(codec, STA32X_MMUTE);
+ snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE);
+ sta32x_sync_coef_shadow(codec);
+ rc = snd_soc_cache_sync(codec);
+ snd_soc_write(codec, STA32X_MMUTE, mute);
+ return rc;
+}
+
#define SINGLE_COEF(xname, index) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = sta32x_coefficient_info, \
@@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
- snd_soc_cache_sync(codec);
+ sta32x_cache_sync(codec);
}
/* Power up to mute */
@@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec)
STA32X_CxCFG_OM_MASK,
2 << STA32X_CxCFG_OM_SHIFT);
+ /* initialize coefficient shadow RAM with reset values */
+ for (i = 4; i <= 49; i += 5)
+ sta32x->coef_shadow[i] = 0x400000;
+ for (i = 50; i <= 54; i++)
+ sta32x->coef_shadow[i] = 0x7fffff;
+ sta32x->coef_shadow[55] = 0x5a9df7;
+ sta32x->coef_shadow[56] = 0x7fffff;
+ sta32x->coef_shadow[59] = 0x7fffff;
+ sta32x->coef_shadow[60] = 0x400000;
+ sta32x->coef_shadow[61] = 0x400000;
+
sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Bias level configuration will have done an extra enable */
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
index b97ee5a75667..d8e32a6262ee 100644
--- a/sound/soc/codecs/sta32x.h
+++ b/sound/soc/codecs/sta32x.h
@@ -19,6 +19,7 @@
/* STA326 register addresses */
#define STA32X_REGISTER_COUNT 0x2d
+#define STA32X_COEF_COUNT 62
#define STA32X_CONFA 0x00
#define STA32X_CONFB 0x01
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7e5ec03f6f8d..a7c9ae17fc7e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
+ codec->cache_sync = 1;
break;
}
codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a9504710bb69..3a629d0d690e 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
u16 ioctl;
+ if (wm8753->dai_func == ucontrol->value.integer.value[0])
+ return 0;
+
if (codec->active)
return -EBUSY;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 91d3c6dbeba3..53edd9a8c758 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec)
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0);
static const unsigned int mixinpga_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(5),
0, 1, TLV_DB_SCALE_ITEM(0, 600, 0),
2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0),
3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0),
@@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0);
static const unsigned int classd_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index eec8e1435116..d1a142f48b09 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
static const unsigned int drc_max_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
};
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 3cd35a02c28c..4a398c3bfe84 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
mdelay(100);
/* Normal bias enable & soft start off */
- reg |= WM9081_BIAS_ENA;
reg &= ~WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
}
/* VMID 2*240k */
- reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg &= ~WM9081_VMID_SEL_MASK;
reg |= 0x04;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
- /* Startup bias source */
+ /* Startup bias source and disable bias */
reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_BIAS_SRC;
+ reg &= ~WM9081_BIAS_ENA;
snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
- /* Disable VMID and biases with soft ramping */
+ /* Disable VMID with soft ramping */
reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
- reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+ reg &= ~WM9081_VMID_SEL_MASK;
reg |= WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 2b5252c9e377..f94c06057c64 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
}
static const unsigned int in_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(3),
0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
4, 6, TLV_DB_SCALE_ITEM(600, 600, 0),
};
static const unsigned int mix_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 84f33d4ea2cd..48e61e912400 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0268cf989736..83c4bd5b2dd7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
+ sysfs_attr_init(&dev_attr->attr);
dev_attr->attr.name = "statistics";
dev_attr->attr.mode = S_IRUGO;
dev_attr->show = fsl_sysfs_ssi_show;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f3af9e..32d2a21f2e3b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
{
+ /* Roland GAIA SH-01 */
+ USB_DEVICE(0x0582, 0x0111),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "GAIA",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &(const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
USB_DEVICE(0x0582, 0x0113),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
/* .vendor_name = "BOSS", */