UbiquitousOS
/
OpenCloudOS-Kernel
mirror of https://gitee.com/OpenCloudOS/OpenCloudOS-Kernel.git
11
0
Fork 0
Code Issues Projects Releases Wiki Activity

Merge branch 'fix/hda' into topic/hda

This commit is contained in:
Takashi Iwai 2011-12-01 13:51:18 +01:00
parent a4567cb389 88d686027b
commit 9eb6e9b16f
20 changed files with 168 additions and 76 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
MAINTAINERS
View File

@ -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

1
sound/pci/hda/hda_intel.c
View File

@ -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),

2
sound/pci/hda/patch_cirrus.c
View File

@ -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",
};

22
sound/pci/hda/patch_sigmatel.c
View File

@ -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 =

80
sound/pci/hda/patch_via.c
View File

@ -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)
{
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));
}
#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;
if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1
&& !is_aa_path_mute(spec->codec))
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_update_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);
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)

2
sound/soc/codecs/adau1373.c
View File

@ -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),

8
sound/soc/codecs/cs4271.c
View File

@ -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);

2
sound/soc/codecs/rt5631.c
View File

@ -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),

2
sound/soc/codecs/sgtl5000.c
View File

@ -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),
};

63
sound/soc/codecs/sta32x.c
View File

@ -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);

1
sound/soc/codecs/sta32x.h
View File

@ -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

1
sound/soc/codecs/wm8731.c
View File

@ -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;

3
sound/soc/codecs/wm8753.c
View File

@ -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;

4
sound/soc/codecs/wm8962.c
View File

@ -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),
};

2
sound/soc/codecs/wm8993.c
View File

@ -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),
};

10
sound/soc/codecs/wm9081.c
View File

@ -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);

6
sound/soc/codecs/wm9090.c
View File

@ -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),
};

2
sound/soc/codecs/wm_hubs.c
View File

@ -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),
};

1
sound/soc/fsl/fsl_ssi.c
View File

@ -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;

31
sound/usb/quirks-table.h
View File

@ -1632,6 +1632,37 @@ 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) {