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authorMauro Carvalho Chehab <mchehab@redhat.com>2012-03-19 13:41:24 -0300
committerMauro Carvalho Chehab <mchehab@redhat.com>2012-03-19 13:41:24 -0300
commit9ce28d827f74d0acdd058bded8bab5309b0f5c8f (patch)
tree634f22e8df9c7fd3966b3639e3e997436751ca50 /sound
parentf074ff92b5b26f3a559fab1203c36e140ea8d067 (diff)
parentc16fa4f2ad19908a47c63d8fa436a1178438c7e7 (diff)
Merge tag 'v3.3' into staging/for_v3.4
* tag 'v3.3': (1646 commits) Linux 3.3 Don't limit non-nested epoll paths netfilter: ctnetlink: fix race between delete and timeout expiration ipv6: Don't dev_hold(dev) in ip6_mc_find_dev_rcu. nilfs2: fix NULL pointer dereference in nilfs_load_super_block() nilfs2: clamp ns_r_segments_percentage to [1, 99] afs: Remote abort can cause BUG in rxrpc code afs: Read of file returns EBADMSG C6X: remove dead code from entry.S wimax/i2400m: fix erroneous NETDEV_TX_BUSY use net/hyperv: fix erroneous NETDEV_TX_BUSY use net/usbnet: reserve headroom on rx skbs bnx2x: fix memory leak in bnx2x_init_firmware() bnx2x: fix a crash on corrupt firmware file sch_sfq: revert dont put new flow at the end of flows ipv6: fix icmp6_dst_alloc() MAINTAINERS: Add Serge as maintainer of capabilities drivers/video/backlight/s6e63m0.c: fix corruption storing gamma mode MAINTAINERS: add entry for exynos mipi display drivers MAINTAINERS: fix link to Gustavo Padovans tree ...
Diffstat (limited to 'sound')
-rw-r--r--sound/core/compress_offload.c13
-rw-r--r--sound/isa/sb/emu8000_patch.c1
-rw-r--r--sound/pci/azt3328.c3
-rw-r--r--sound/pci/hda/alc880_quirks.c17
-rw-r--r--sound/pci/hda/alc882_quirks.c15
-rw-r--r--sound/pci/hda/hda_codec.c14
-rw-r--r--sound/pci/hda/hda_codec.h3
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/hda_jack.c24
-rw-r--r--sound/pci/hda/patch_ca0132.c33
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c26
-rw-r--r--sound/pci/hda/patch_realtek.c175
-rw-r--r--sound/pci/hda/patch_sigmatel.c25
-rw-r--r--sound/pci/hda/patch_via.c287
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c25
-rw-r--r--sound/pci/rme9652/hdspm.c1
-rw-r--r--sound/pci/ymfpci/ymfpci.c21
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c21
-rw-r--r--sound/soc/codecs/ak4642.c31
-rw-r--r--sound/soc/codecs/cs42l73.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c17
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c110
-rw-r--r--sound/soc/codecs/wm2000.c31
-rw-r--r--sound/soc/codecs/wm5100.c15
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8962.c10
-rw-r--r--sound/soc/codecs/wm8994.c16
-rw-r--r--sound/soc/codecs/wm8996.c9
-rw-r--r--sound/soc/codecs/wm8996.h4
-rw-r--r--sound/soc/codecs/wm_hubs.c18
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/mxs/mxs-saif.c5
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c69
-rw-r--r--sound/soc/sh/fsi.c6
-rw-r--r--sound/soc/soc-core.c11
-rw-r--r--sound/soc/soc-dapm.c12
-rw-r--r--sound/usb/caiaq/audio.c5
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/format.c4
-rw-r--r--sound/usb/quirks-table.h8
-rw-r--r--sound/usb/quirks.c6
43 files changed, 638 insertions, 482 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index dac3633507c9..a68aed7fce02 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
- if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
- return -EFAULT;
+ if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
+ retval = -EFAULT;
+ goto out;
+ }
retval = snd_compr_allocate_buffer(stream, params);
if (retval) {
- kfree(params);
- return -ENOMEM;
+ retval = -ENOMEM;
+ goto out;
}
retval = stream->ops->set_params(stream, params);
if (retval)
goto out;
stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- } else
+ } else {
return -EPERM;
+ }
out:
kfree(params);
return retval;
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index e09f144177f5..c99c6078be33 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -22,7 +22,6 @@
#include "emu8000_local.h"
#include <asm/uaccess.h>
#include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
static int emu8000_reset_addr;
module_param(emu8000_reset_addr, int, 0444);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e..496f14c1a731 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
if (err < 0)
goto out_err;
+ opl3->private_data = chip;
}
- opl3->private_data = chip;
-
sprintf(card->longname, "%s at 0x%lx, irq %i",
card->shortname, chip->ctrl_io, chip->irq);
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index 5b68435d195b..501501ef36a9 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
- switch (res >> 28) {
+ res >>= 28;
+ switch (res) {
case ALC_MIC_EVENT:
alc88x_simple_mic_automute(codec);
break;
default:
- alc_sku_unsol_event(codec, res);
+ alc_exec_unsol_event(codec, res);
break;
}
}
+static void alc880_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ alc_exec_unsol_event(codec, res >> 28);
+}
+
static void alc880_uniwill_p53_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
- if ((res >> 28) == ALC_DCVOL_EVENT)
+ res >>= 28;
+ if (res == ALC_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
else
- alc_sku_unsol_event(codec, res);
+ alc_exec_unsol_event(codec, res);
}
/*
@@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_lg_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc880_unsol_event,
.setup = alc880_lg_setup,
.init_hook = alc_hp_automute,
#ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index bdf0ed4ab3e2..bb364a53f546 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
alc889A_mb31_automute(codec);
}
+static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ alc_exec_unsol_event(codec, res >> 26);
+}
+
/*
* configuration and preset
*/
@@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = {
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mba21_setup,
.init_hook = alc_hp_automute,
},
@@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mbp3_setup,
.init_hook = alc_hp_automute,
},
@@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_mb5_setup,
.init_hook = alc_hp_automute,
},
@@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &macmini3_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_macmini3_setup,
.init_hook = alc_hp_automute,
},
@@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = {
.input_mux = &alc889A_imac91_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
+ .unsol_event = alc882_unsol_event,
.setup = alc885_imac91_setup,
.init_hook = alc_hp_automute,
},
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4df72c0e8c37..684307372d73 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
for (i = 0; i < c->cvt_setups.used; i++) {
p = snd_array_elem(&c->cvt_setups, i);
if (!p->active && p->stream_tag == stream_tag &&
- get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+ get_wcaps_type(get_wcaps(c, p->nid)) == type)
p->dirty = 1;
}
}
@@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
parm |= index << AC_AMP_SET_INDEX_SHIFT;
- parm |= val;
+ if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+ (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+ ; /* set the zero value as a fake mute */
+ else
+ parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val;
}
@@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
val1 += ofs;
val1 = ((int)val1) * ((int)val2);
- if (min_mute)
+ if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
val2 |= TLV_DB_SCALE_MUTE;
if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
return -EFAULT;
@@ -5114,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
const char *pfx = "", *sfx = "";
/* handle as a speaker if it's a fixed line-out */
- if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT)
+ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
name = "Speaker";
/* check the location */
switch (attr) {
@@ -5173,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
switch (get_defcfg_device(def_conf)) {
case AC_JACK_LINE_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Line-Out",
+ return fill_audio_out_name(codec, nid, cfg, "Line Out",
label, maxlen, indexp);
case AC_JACK_SPEAKER:
return fill_audio_out_name(codec, nid, cfg, "Speaker",
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e9f71dc0d464..f0f1943a4b2c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -298,6 +298,9 @@ enum {
#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
#define AC_AMPCAP_MUTE_SHIFT 31
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */
+
/* Connection list */
#define AC_CLIST_LENGTH (0x7f<<0)
#define AC_CLIST_LONG (1<<7)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fb35474c1203..95dfb6874941 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -469,6 +469,7 @@ struct azx {
unsigned int irq_pending_warned :1;
unsigned int probing :1; /* codec probing phase */
unsigned int snoop:1;
+ unsigned int align_buffer_size:1;
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
@@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rates = hinfo->rates;
snd_pcm_limit_hw_rates(runtime);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- if (align_buffer_size)
+ if (chip->align_buffer_size)
/* constrain buffer sizes to be multiple of 128
bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and
@@ -2773,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
+ chip->align_buffer_size = align_buffer_size;
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
- align_buffer_size = 0;
+ chip->align_buffer_size = 0;
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d8a35da0803f..9d819c4b4923 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg)
+ const struct auto_pin_cfg *cfg,
+ char *lastname, int *lastidx)
{
unsigned int def_conf, conn;
char name[44];
@@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
return 0;
snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+ if (!strcmp(name, lastname) && idx == *lastidx)
+ idx++;
+ strncpy(lastname, name, 44);
+ *lastidx = idx;
err = snd_hda_jack_add_kctl(codec, nid, name, idx);
if (err < 0)
return err;
@@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
const hda_nid_t *p;
- int i, err;
+ int i, err, lastidx = 0;
+ char lastname[44] = "";
for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0; i < cfg->num_inputs; i++) {
- err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
+ err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
- err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
+ err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
- err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
+ err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
return 0;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 35abe3c62908..21d91d580da8 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_MUTE, &data);
if (err < 0)
- return err;
+ goto exit;
/* *valp 0 is mute, 1 is unmute */
data = (data & 0x7f) | (*valp ? 0 : 0x80);
- chipio_write(codec, REG_CODEC_MUTE, data);
+ err = chipio_write(codec, REG_CODEC_MUTE, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_hp_switch = *valp;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
@@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_MUTE, &data);
if (err < 0)
- return err;
+ goto exit;
/* *valp 0 is mute, 1 is unmute */
data = (data & 0xef) | (*valp ? 0 : 0x10);
- chipio_write(codec, REG_CODEC_MUTE, data);
+ err = chipio_write(codec, REG_CODEC_MUTE, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_speaker_switch = *valp;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
@@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
if (err < 0)
- return err;
+ goto exit;
val = 31 - left_vol;
data = (data & 0xe0) | val;
- chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+ err = chipio_write(codec, REG_CODEC_HP_VOL_L, data);
if (err < 0)
- return err;
+ goto exit;
val = 31 - right_vol;
data = (data & 0xe0) | val;
- chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+ err = chipio_write(codec, REG_CODEC_HP_VOL_R, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_hp_volume[0] = left_vol;
spec->curr_hp_volume[1] = right_vol;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
@@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
err = add_in_volume(codec, spec->dig_in, "IEC958");
+ if (err < 0)
+ return err;
}
return 0;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 0e99357e822c..c83ccdba1e5a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
"Front Speaker", "Surround Speaker", "Bass Speaker"
};
static const char * const line_outs[] = {
- "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+ "Front Line Out", "Surround Line Out", "Bass Line Out"
};
fix_volume_caps(codec, dac);
@@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
if (num_ctls > 1)
name = line_outs[idx];
else
- name = "Line-Out";
+ name = "Line Out";
break;
}
@@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec)
change_cur_input(codec, !spec->automic_idx, 0);
} else {
if (present) {
- spec->last_input = spec->cur_input;
- spec->cur_input = spec->automic_idx;
+ if (spec->cur_input != spec->automic_idx) {
+ spec->last_input = spec->cur_input;
+ spec->cur_input = spec->automic_idx;
+ }
} else {
spec->cur_input = spec->last_input;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8a32a69c83c3..d29d6d377904 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
@@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
@@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
- if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+ if (!(query_amp_caps(codec, nid, hda_dir) &
+ (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
break;
}
return 0;
@@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = {
{}
};
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+ static hda_nid_t out_nids[] = {
+ 0x10, 0x11, 0
+ };
+ hda_nid_t *p;
+
+ for (p = out_nids; *p; p++)
+ snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+ AC_AMPCAP_MIN_MUTE |
+ query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
static int patch_conexant_auto(struct hda_codec *codec)
{
struct conexant_spec *spec;
@@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
case 0x14f15045:
spec->single_adc_amp = 1;
break;
+ case 0x14f15051:
+ add_cx5051_fake_mutes(codec);
+ break;
}
apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e82acf77c5a..22c73b78ac6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */
};
+#define MAX_VOL_NIDS 0x40
+
struct alc_spec {
/* codec parameterization */
const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
hda_nid_t mixer_nid; /* analog-mixer NID */
- DECLARE_BITMAP(vol_ctls, 0x20 << 1);
- DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+ DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+ DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
/* capture setup for dynamic dual-adc switch */
hda_nid_t cur_adc;
@@ -177,6 +179,7 @@ struct alc_spec {
unsigned int detect_lo:1; /* Line-out detection enabled */
unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
unsigned int automute_lo_possible:1; /* there are line outs and HP */
+ unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
@@ -185,7 +188,6 @@ struct alc_spec {
unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
- unsigned int use_jack_tbl:1; /* 1 for model=auto */
/* auto-mute control */
int automute_mode;
@@ -496,13 +498,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
+ unsigned int val;
if (!nid)
break;
switch (spec->automute_mode) {
case ALC_AUTOMUTE_PIN:
+ /* don't reset VREF value in case it's controlling
+ * the amp (see alc861_fixup_asus_amp_vref_0f())
+ */
+ if (spec->keep_vref_in_automute) {
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ val &= ~PIN_HP;
+ } else
+ val = 0;
+ val |= pin_bits;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_bits);
+ val);
break;
case ALC_AUTOMUTE_AMP:
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -621,17 +634,10 @@ static void alc_mic_automute(struct hda_codec *codec)
alc_mux_select(codec, 0, spec->int_mic_idx, false);
}
-/* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+/* handle the specified unsol action (ALC_XXX_EVENT) */
+static void alc_exec_unsol_event(struct hda_codec *codec, int action)
{
- struct alc_spec *spec = codec->spec;
- if (codec->vendor_id == 0x10ec0880)
- res >>= 28;
- else
- res >>= 26;
- if (spec->use_jack_tbl)
- res = snd_hda_jack_get_action(codec, res);
- switch (res) {
+ switch (action) {
case ALC_HP_EVENT:
alc_hp_automute(codec);
break;
@@ -645,6 +651,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
snd_hda_jack_report_sync(codec);
}
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ res = snd_hda_jack_get_action(codec, res);
+ alc_exec_unsol_event(codec, res);
+}
+
/* call init functions of standard auto-mute helpers */
static void alc_inithook(struct hda_codec *codec)
{
@@ -785,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
@@ -1839,7 +1856,9 @@ static const char * const alc_slave_vols[] = {
"Headphone Playback Volume",
"Speaker Playback Volume",
"Mono Playback Volume",
- "Line-Out Playback Volume",
+ "Line Out Playback Volume",
+ "CLFE Playback Volume",
+ "Bass Speaker Playback Volume",
"PCM Playback Volume",
NULL,
};
@@ -1854,7 +1873,9 @@ static const char * const alc_slave_sws[] = {
"Speaker Playback Switch",
"Mono Playback Switch",
"IEC958 Playback Switch",
- "Line-Out Playback Switch",
+ "Line Out Playback Switch",
+ "CLFE Playback Switch",
+ "Bass Speaker Playback Switch",
"PCM Playback Switch",
NULL,
};
@@ -1883,7 +1904,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = {
};
#endif
-static int alc_build_controls(struct hda_codec *codec)
+static int __alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct snd_kcontrol *kctl = NULL;
@@ -2029,11 +2050,16 @@ static int alc_build_controls(struct hda_codec *codec)
alc_free_kctls(codec); /* no longer needed */
- err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
+ return 0;
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err = __alc_build_controls(codec);
if (err < 0)
return err;
-
- return 0;
+ return snd_hda_jack_add_kctls(codec, &spec->autocfg);
}
@@ -2042,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec)
*/
static void alc_init_special_input_src(struct hda_codec *codec);
+static int alc269_fill_coef(struct hda_codec *codec);
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
+ if (codec->vendor_id == 0x10ec0269)
+ alc269_fill_coef(codec);
+
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
@@ -2298,7 +2328,7 @@ static int alc_build_pcms(struct hda_codec *codec)
"%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
- if (spec->multiout.dac_nids > 0) {
+ if (spec->multiout.num_dacs > 0) {
p = spec->stream_analog_playback;
if (!p)
p = &alc_pcm_analog_playback;
@@ -3125,7 +3155,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
static inline unsigned int get_ctl_pos(unsigned int data)
{
hda_nid_t nid = get_amp_nid_(data);
- unsigned int dir = get_amp_direction_(data);
+ unsigned int dir;
+ if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+ return 0;
+ dir = get_amp_direction_(data);
return (nid << 1) | dir;
}
@@ -3233,7 +3266,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
int i, err, noutputs;
noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
+ if (spec->multi_ios > 0 && cfg->line_outs < 3)
noutputs += spec->multi_ios;
for (i = 0; i < noutputs; i++) {
@@ -3768,7 +3801,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
else
nums = spec->num_adc_nids;
for (c = 0; c < nums; c++)
- alc_mux_select(codec, 0, spec->cur_mux[c], true);
+ alc_mux_select(codec, c, spec->cur_mux[c], true);
}
/* add mic boosts if needed */
@@ -3904,7 +3937,6 @@ static void set_capture_mixer(struct hda_codec *codec)
static void alc_auto_init_std(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- spec->use_jack_tbl = 1;
alc_auto_init_multi_out(codec);
alc_auto_init_extra_out(codec);
alc_auto_init_analog_input(codec);
@@ -4168,6 +4200,8 @@ static int patch_alc880(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc880_loopbacks;
@@ -4297,6 +4331,8 @@ static int patch_alc260(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
spec->shutup = alc_eapd_shutup;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -4335,6 +4371,7 @@ enum {
ALC882_FIXUP_PB_M5210,
ALC882_FIXUP_ACER_ASPIRE_7736,
ALC882_FIXUP_ASUS_W90V,
+ ALC889_FIXUP_CD,
ALC889_FIXUP_VAIO_TT,
ALC888_FIXUP_EEE1601,
ALC882_FIXUP_EAPD,
@@ -4347,6 +4384,7 @@ enum {
ALC882_FIXUP_ACER_ASPIRE_8930G,
ALC882_FIXUP_ASPIRE_8930G_VERBS,
ALC885_FIXUP_MACPRO_GPIO,
+ ALC889_FIXUP_DAC_ROUTE,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -4400,6 +4438,31 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec,
alc882_gpio_mute(codec, 1, 0);
}
+/* Fix the connection of some pins for ALC889:
+ * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't
+ * work correctly (bko#42740)
+ */
+static void alc889_fixup_dac_route(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+ /* fake the connections during parsing the tree */
+ hda_nid_t conn1[2] = { 0x0c, 0x0d };
+ hda_nid_t conn2[2] = { 0x0e, 0x0f };
+ snd_hda_override_conn_list(codec, 0x14, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x15, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x18, 2, conn2);
+ snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+ } else if (action == ALC_FIXUP_ACT_PROBE) {
+ /* restore the connections */
+ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+ snd_hda_override_conn_list(codec, 0x14, 5, conn);
+ snd_hda_override_conn_list(codec, 0x15, 5, conn);
+ snd_hda_override_conn_list(codec, 0x18, 5, conn);
+ snd_hda_override_conn_list(codec, 0x1a, 5, conn);
+ }
+}
+
static const struct alc_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = ALC_FIXUP_PINS,
@@ -4436,6 +4499,13 @@ static const struct alc_fixup alc882_fixups[] = {
{ }
}
},
+ [ALC889_FIXUP_CD] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1c, 0x993301f0 }, /* CD */
+ { }
+ }
+ },
[ALC889_FIXUP_VAIO_TT] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
@@ -4547,6 +4617,10 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc885_fixup_macpro_gpio,
},
+ [ALC889_FIXUP_DAC_ROUTE] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc889_fixup_dac_route,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -4571,6 +4645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+ SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
@@ -4587,6 +4662,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -4691,6 +4767,8 @@ static int patch_alc882(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC_MODEL_AUTO)
spec->init_hook = alc_auto_init_std;
+ else
+ codec->patch_ops.build_controls = __alc_build_controls;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -4722,7 +4800,6 @@ enum {
ALC262_FIXUP_FSC_H270,
ALC262_FIXUP_HP_Z200,
ALC262_FIXUP_TYAN,
- ALC262_FIXUP_TOSHIBA_RX1,
ALC262_FIXUP_LENOVO_3000,
ALC262_FIXUP_BENQ,
ALC262_FIXUP_BENQ_T31,
@@ -4752,16 +4829,6 @@ static const struct alc_fixup alc262_fixups[] = {
{ }
}
},
- [ALC262_FIXUP_TOSHIBA_RX1] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x90170110 }, /* speaker */
- { 0x15, 0x0421101f }, /* HP */
- { 0x1a, 0x40f000f0 }, /* N/A */
- { 0x1b, 0x40f000f0 }, /* N/A */
- { 0x1e, 0x40f000f0 }, /* N/A */
- }
- },
[ALC262_FIXUP_LENOVO_3000] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -4794,8 +4861,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN),
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
- ALC262_FIXUP_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270),
SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ),
@@ -5364,7 +5429,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC),
@@ -5416,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = {
static int alc269_fill_coef(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
int val;
+ if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+ return 0;
+
if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8817);
@@ -5573,8 +5641,28 @@ static const struct hda_amp_list alc861_loopbacks[] = {
/* Pin config fixes */
enum {
PINFIX_FSC_AMILO_PI1505,
+ PINFIX_ASUS_A6RP,
};
+/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
+static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ val = snd_hda_codec_read(codec, 0x0f, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
+ val |= AC_PINCTL_IN_EN;
+ val |= AC_PINCTL_VREF_50;
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ spec->keep_vref_in_automute = 1;
+}
+
static const struct alc_fixup alc861_fixups[] = {
[PINFIX_FSC_AMILO_PI1505] = {
.type = ALC_FIXUP_PINS,
@@ -5584,9 +5672,16 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
+ [PINFIX_ASUS_A6RP] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc861_fixup_asus_amp_vref_0f,
+ },
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP),
+ SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3556408d6ece..9dbb5735d778 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
"Alienware M17x", STAC_ALIENWARE_M17X),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
- "Alienware M17x", STAC_ALIENWARE_M17X),
+ "Alienware M17x R3", STAC_DELL_EQ),
{} /* terminator */
};
@@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
return 1;
}
-static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
for (i = 0; i < cfg->hp_outs; i++)
if (cfg->hp_pins[i] == nid)
return 1; /* nid is a HP-Out */
-
+ for (i = 0; i < cfg->line_outs; i++)
+ if (cfg->line_out_pins[i] == nid)
+ return 1; /* nid is a line-Out */
return 0; /* nid is not a HP-Out */
};
@@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec)
continue;
}
- if (is_nid_hp_pin(cfg, nid))
+ if (is_nid_out_jack_pin(cfg, nid))
continue; /* already has an unsol event */
pinctl = snd_hda_codec_read(codec, nid, 0,
@@ -4627,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
if (no_hp_sensing(spec, i))
continue;
- if (presence)
+ if (1 /*presence*/)
stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
#if 0 /* FIXME */
/* Resetting the pinctl like below may lead to (a sort of) regressions
@@ -4868,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
/* BIOS bug: unfilled OEM string */
if (strstr(dev->name, "HP_Mute_LED_P_G")) {
set_hp_led_gpio(codec);
- spec->gpio_led_polarity = 1;
+ switch (codec->subsystem_id) {
+ case 0x103c148a:
+ spec->gpio_led_polarity = 0;
+ break;
+ default:
+ spec->gpio_led_polarity = 1;
+ break;
+ }
return 1;
}
}
@@ -5069,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
spec->gpio_dir, spec->gpio_data);
} else {
notmtd_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD;
+ AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
muted_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
+ AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 03e63fed9caf..dff9a00ee8fb 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -199,6 +199,9 @@ struct via_spec {
unsigned int no_pin_power_ctl;
enum VIA_HDA_CODEC codec_type;
+ /* analog low-power control */
+ bool alc_mode;
+
/* smart51 setup */
unsigned int smart51_nums;
hda_nid_t smart51_pins[2];
@@ -663,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init input-src */
for (i = 0; i < spec->num_adc_nids; i++) {
int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+ /* secondary ADCs must have the unique MUX */
+ if (i > 0 && !spec->mux_nids[i])
+ break;
if (spec->mux_nids[adc_idx]) {
int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
@@ -687,6 +693,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
}
}
+static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int parm)
+{
+ if (snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0) == parm)
+ return;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+}
+
static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int *affected_parm)
{
@@ -709,7 +724,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
} else
parm = AC_PWRST_D3;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, nid, parm);
}
static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol,
@@ -749,6 +764,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol,
return 0;
spec->no_pin_power_ctl = val;
set_widgets_power_state(codec);
+ analog_low_current_mode(codec);
return 1;
}
@@ -1036,13 +1052,19 @@ static bool is_aa_path_mute(struct hda_codec *codec)
}
/* enter/exit analog low-current mode */
-static void analog_low_current_mode(struct hda_codec *codec)
+static void __analog_low_current_mode(struct hda_codec *codec, bool force)
{
struct via_spec *spec = codec->spec;
bool enable;
unsigned int verb, parm;
- enable = is_aa_path_mute(codec) && (spec->opened_streams != 0);
+ if (spec->no_pin_power_ctl)
+ enable = false;
+ else
+ enable = is_aa_path_mute(codec) && !spec->opened_streams;
+ if (enable == spec->alc_mode && !force)
+ return;
+ spec->alc_mode = enable;
/* decide low current mode's verb & parameter */
switch (spec->codec_type) {
@@ -1074,6 +1096,11 @@ static void analog_low_current_mode(struct hda_codec *codec)
snd_hda_codec_write(codec, codec->afg, 0, verb, parm);
}
+static void analog_low_current_mode(struct hda_codec *codec)
+{
+ return __analog_low_current_mode(codec, false);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -1446,6 +1473,7 @@ static int via_build_controls(struct hda_codec *codec)
struct snd_kcontrol *kctl;
int err, i;
+ spec->no_pin_power_ctl = 1;
if (spec->set_widgets_power_state)
if (!via_clone_control(spec, &via_pin_power_ctl_enum))
return -ENOMEM;
@@ -1499,10 +1527,6 @@ static int via_build_controls(struct hda_codec *codec)
return err;
}
- /* init power states */
- set_widgets_power_state(codec);
- analog_low_current_mode(codec);
-
via_free_kctls(codec); /* no longer needed */
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
@@ -2295,10 +2319,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
if (mux) {
/* switch to D0 beofre change index */
- if (snd_hda_codec_read(codec, mux, 0,
- AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
- snd_hda_codec_write(codec, mux, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, mux, AC_PWRST_D0);
snd_hda_codec_write(codec, mux, 0,
AC_VERB_SET_CONNECT_SEL,
spec->inputs[cur].mux_idx);
@@ -2776,6 +2797,10 @@ static int via_init(struct hda_codec *codec)
for (i = 0; i < spec->num_iverbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ /* init power states */
+ set_widgets_power_state(codec);
+ __analog_low_current_mode(codec, true);
+
via_auto_init_multi_out(codec);
via_auto_init_hp_out(codec);
via_auto_init_speaker_out(codec);
@@ -2922,9 +2947,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW 0/1 (13h/14h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -2932,8 +2957,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x19, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW6 (22h), SW2 (26h), AOW2 (24h) */
if (is_8ch) {
@@ -2941,20 +2966,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x22, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x26, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x24, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x24, parm);
} else if (codec->vendor_id == 0x11064397) {
/* PW7(23h), SW2(27h), AOW2(25h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x23, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
+ update_power_state(codec, 0x25, parm);
}
/* PW 3/4/7 (1ch/1dh/23h) */
@@ -2966,17 +2987,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x23, &parm);
/* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
if (is_8ch) {
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
+ update_power_state(codec, 0x27, parm);
} else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
}
static int patch_vt1708S(struct hda_codec *codec);
@@ -3149,10 +3166,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */
/* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x12, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x20, parm);
/* outputs */
/* PW 3/4 (16h/17h) */
@@ -3160,10 +3177,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
set_pin_power_state(codec, 0x17, &parm);
set_pin_power_state(codec, 0x16, &parm);
/* MW0 (1ah), AOW 0/1 (10h/1dh) */
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x1d, parm);
}
static int patch_vt1702(struct hda_codec *codec)
@@ -3228,52 +3244,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* PW3 (27h), MW2 (1ah), AOW3 (bh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x27, &parm);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, parm);
+ update_power_state(codec, 0xb, parm);
/* PW2 (26h), AOW2 (ah) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2b, &parm);
- snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0xa, parm);
/* PW0 (24h), AOW0 (8h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (!spec->hp_independent_mode) /* check for redirected HP */
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
- snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm);
/* PW1 (25h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2a, &parm);
- snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x9, parm);
if (spec->hp_independent_mode) {
/* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xc, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1b, parm);
+ update_power_state(codec, 0x34, parm);
+ update_power_state(codec, 0xc, parm);
}
}
@@ -3433,8 +3445,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW0(13h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x1e, &parm);
@@ -3442,12 +3454,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (spec->dmic_enabled)
set_pin_power_state(codec, 0x22, &parm);
else
- snd_hda_codec_write(codec, 0x22, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x22, AC_PWRST_D3);
/* SW2(26h), AIW1(14h) */
- snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -3456,8 +3467,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW2(1bh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW7 (23h), SW3 (27h), AOW3 (25h) */
parm = AC_PWRST_D3;
@@ -3465,12 +3476,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW1(1ah) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
/* Smart 5.1 PW5(1eh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1e, &parm);
- snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* Mono out */
/* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
@@ -3486,9 +3497,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
mono_out = 1;
}
parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
- snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x28, parm);
+ update_power_state(codec, 0x29, parm);
+ update_power_state(codec, 0x2a, parm);
/* PW 3/4 (1ch/1dh) */
parm = AC_PWRST_D3;
@@ -3496,15 +3507,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
set_pin_power_state(codec, 0x1d, &parm);
/* HP Independent Mode, power on AOW3 */
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* force to D0 for internal Speaker */
/* MW0 (16h), AOW0 (10h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
- mono_out ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm);
}
static int patch_vt1716S(struct hda_codec *codec)
@@ -3580,54 +3588,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
if (spec->codec_type == VT1802) {
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
} else {
/* PW4 (26h), MW4 (1ch), MUX4(37h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x37, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x37, parm);
}
if (spec->codec_type == VT1802) {
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
} else {
/* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x19, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x19, parm);
+ update_power_state(codec, 0x35, parm);
}
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Class-D */
/* PW0 (24h), MW0(18h/14h), MUX0(34h) */
@@ -3637,12 +3636,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x24, &parm);
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802)
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x14, parm);
else
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x34, parm);
/* Mono Out */
present = snd_hda_jack_detect(codec, 0x26);
@@ -3650,28 +3647,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802) {
/* PW15 (33h), MW8(1ch), MUX8(3ch) */
- snd_hda_codec_write(codec, 0x33, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x33, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x3c, parm);
} else {
/* PW15 (31h), MW8(17h), MUX8(3bh) */
- snd_hda_codec_write(codec, 0x31, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x17, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3b, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x31, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x3b, parm);
}
/* MW9 (21h) */
if (imux_is_smixer || !is_aa_path_mute(codec))
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x21, AC_PWRST_D0);
else
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x21, AC_PWRST_D3);
}
/* patch for vt2002P */
@@ -3731,30 +3720,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x8, AC_PWRST_D0);
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Internal Speaker */
/* PW0 (24h), MW0(14h), MUX0(34h) */
@@ -3763,15 +3750,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x14, AC_PWRST_D3);
+ update_power_state(codec, 0x34, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x14, AC_PWRST_D0);
+ update_power_state(codec, 0x34, AC_PWRST_D0);
}
@@ -3782,26 +3765,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x31, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x1c, AC_PWRST_D3);
+ update_power_state(codec, 0x3c, AC_PWRST_D3);
+ update_power_state(codec, 0x3e, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x1c, AC_PWRST_D0);
+ update_power_state(codec, 0x3c, AC_PWRST_D0);
+ update_power_state(codec, 0x3e, AC_PWRST_D0);
}
/* PW15 (33h), MW15 (1dh), MUX15(3dh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x33, &parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1d, parm);
+ update_power_state(codec, 0x3d, parm);
}
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 9f3b01bb72c8..e0a4263baa20 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x161f,
+ .subdevice = 0x202f,
+ .name = "Gateway M520",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x161f,
.subdevice = 0x203a,
.name = "Gateway 4525GZ", /* AD1981B */
.type = AC97_TUNE_INV_EAPD
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 26c7e8bcb229..c0dbb52d45be 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl,
mutex_lock(&chip->mutex);
reg = oxygen_read_ac97(chip, codec, index);
mutex_unlock(&chip->mutex);
- value->value.integer.value[0] = 31 - (reg & 0x1f);
- if (stereo)
- value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f);
+ if (!stereo) {
+ value->value.integer.value[0] = 31 - (reg & 0x1f);
+ } else {
+ value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f);
+ value->value.integer.value[1] = 31 - (reg & 0x1f);
+ }
return 0;
}
@@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl,
mutex_lock(&chip->mutex);
oldreg = oxygen_read_ac97(chip, codec, index);
- newreg = oldreg;
- newreg = (newreg & ~0x1f) |
- (31 - (value->value.integer.value[0] & 0x1f));
- if (stereo)
- newreg = (newreg & ~0x1f00) |
- ((31 - (value->value.integer.value[1] & 0x1f)) << 8);
- else
- newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8);
+ if (!stereo) {
+ newreg = oldreg & ~0x1f;
+ newreg |= 31 - (value->value.integer.value[0] & 0x1f);
+ } else {
+ newreg = oldreg & ~0x1f1f;
+ newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8;
+ newreg |= 31 - (value->value.integer.value[1] & 0x1f);
+ }
change = newreg != oldreg;
if (change)
oxygen_write_ac97(chip, codec, index, newreg);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cc9f6c83d661..bc030a2088da 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
hw->ops.open = snd_hdspm_hwdep_dummy_op;
hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+ hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl;
hw->ops.release = snd_hdspm_hwdep_dummy_op;
return 0;
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index e57b89e8aa89..94ab728f5ca8 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) {
+ err = snd_ymfpci_mixer(chip, rear_switch[dev]);
+ if (err < 0) {
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) {
- snd_card_free(card);
- return err;
- }
- if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) {
- snd_card_free(card);
- return err;
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ err = snd_ymfpci_pcm_4ch(chip, 2, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_ymfpci_pcm2(chip, 3, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
}
if ((err = snd_ymfpci_timer(chip, 0)) < 0) {
snd_card_free(card);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 03ee4e365311..12a9a2b03387 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "4ch Duplication",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_ymfpci_info_dup4ch,
+ .get = snd_ymfpci_get_dup4ch,
+ .put = snd_ymfpci_put_dup4ch,
+};
static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = {
{
@@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "4ch Duplication",
- .info = snd_ymfpci_info_dup4ch,
- .get = snd_ymfpci_get_dup4ch,
- .put = snd_ymfpci_put_dup4ch,
-},
};
@@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch)
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0)
return err;
}
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip);
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ }
/* add S/PDIF control */
if (snd_BUG_ON(!chip->pcm_spdif))
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..278c0a0575f5 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 9d38db8f1919..78979b3e0e95 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
priv->config[id].mmcc &= 0xC0;
priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
priv->config[id].spc &= 0xFC;
- priv->config[id].spc &= MCK_SCLK_64FS;
+ priv->config[id].spc |= MCK_SCLK_MCLK;
} else {
/* CS42L73 Slave */
priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f8863ebb4304..7f4ba819a9f6 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
/* restore regular registers */
for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
- /* this regs depends on the others */
+ /* These regs should restore in particular order */
if (reg == SGTL5000_CHIP_ANA_POWER ||
reg == SGTL5000_CHIP_CLK_CTRL ||
reg == SGTL5000_CHIP_LINREG_CTRL ||
reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
- reg == SGTL5000_CHIP_CLK_CTRL)
+ reg == SGTL5000_CHIP_REF_CTRL)
continue;
snd_soc_write(codec, reg, cache[reg]);
@@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
snd_soc_write(codec, reg, cache[reg]);
/*
- * restore power and other regs according
- * to set_power() and set_clock()
+ * restore these regs according to the power setting sequence in
+ * sgtl5000_set_power_regs() and clock setting sequence in
+ * sgtl5000_set_clock().
+ *
+ * The order of restore is:
+ * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after
+ * SGTL5000_CHIP_ANA_POWER PLL bits set
+ * 2. SGTL5000_CHIP_LINREG_CTRL should be set before
+ * SGTL5000_CHIP_ANA_POWER LINREG_D restored
+ * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage,
+ * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored
*/
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL]);
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index eb401ef021fb..372b0b83bd9f 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -60,7 +60,6 @@ struct aic32x4_rate_divs {
struct aic32x4_priv {
u32 sysclk;
- s32 master;
u8 page_no;
void *control_data;
u32 power_cfg;
@@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u8 iface_reg_1;
u8 iface_reg_2;
u8 iface_reg_3;
@@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- aic32x4->master = 1;
iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- aic32x4->master = 0;
break;
default:
printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
@@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-
switch (level) {
case SND_SOC_BIAS_ON:
- if (aic32x4->master) {
- /* Switch on PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, AIC32X4_PLLEN);
-
- /* Switch on NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, AIC32X4_NDACEN);
-
- /* Switch on MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, AIC32X4_MDACEN);
-
- /* Switch on NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, AIC32X4_NADCEN);
-
- /* Switch on MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, AIC32X4_MADCEN);
-
- /* Switch on BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, AIC32X4_BCLKEN);
- }
+ /* Switch on PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, AIC32X4_PLLEN);
+
+ /* Switch on NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, AIC32X4_NDACEN);
+
+ /* Switch on MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, AIC32X4_MDACEN);
+
+ /* Switch on NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, AIC32X4_NADCEN);
+
+ /* Switch on MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, AIC32X4_MADCEN);
+
+ /* Switch on BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, AIC32X4_BCLKEN);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (aic32x4->master) {
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
-
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
-
- /* Switch off MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, 0);
-
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
-
- /* Switch off MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, 0);
-
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
- }
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
+
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
+
+ /* Switch off MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, 0);
+
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
+
+ /* Switch off MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, 0);
+
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
break;
case SND_SOC_BIAS_OFF:
break;
@@ -651,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
}
- if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
- snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
- }
+
+ tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
+ AIC32X4_LDOCTLEN : 0;
+ snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
+
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
tmp_reg |= AIC32X4_LDOIN_18_36;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c2880907fced..a75c3766aede 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
struct wm2000_priv *wm2000;
struct wm2000_platform_data *pdata;
const char *filename;
- const struct firmware *fw;
- int reg, ret;
+ const struct firmware *fw = NULL;
+ int ret;
+ int reg;
u16 id;
wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv),
@@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = PTR_ERR(wm2000->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ goto out;
}
/* Verify that this is a WM2000 */
@@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (id != 0x2000) {
dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id);
ret = -ENODEV;
- goto err_regmap;
+ goto out_regmap_exit;
}
reg = wm2000_read(i2c, WM2000_REG_REVISON);
@@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = request_firmware(&fw, filename, &i2c->dev);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret);
- goto err_regmap;
+ goto out_regmap_exit;
}
/* Pre-cook the concatenation of the register address onto the image */
@@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (wm2000->anc_download == NULL) {
dev_err(&i2c->dev, "Out of memory\n");
ret = -ENOMEM;
- goto err_fw;
+ goto out_regmap_exit;
}
wm2000->anc_download[0] = 0x80;
wm2000->anc_download[1] = 0x00;
memcpy(wm2000->anc_download + 2, fw->data, fw->size);
- release_firmware(fw);
-
wm2000->anc_eng_ena = 1;
wm2000->anc_active = 1;
wm2000->spk_ena = 1;
@@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
wm2000_reset(wm2000);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
- NULL, 0);
- if (ret != 0)
- goto err_fw;
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0);
+ if (!ret)
+ goto out;
- return 0;
-
-err_fw:
- release_firmware(fw);
-err_regmap:
+out_regmap_exit:
regmap_exit(wm2000->regmap);
-err:
+out:
+ release_firmware(fw);
return ret;
}
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 8b24323d6b2c..89f2af77b1c3 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1377,6 +1377,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
switch (wm5100->rev) {
case 0:
+ regcache_cache_bypass(wm5100->regmap, true);
snd_soc_write(codec, 0x11, 0x3);
snd_soc_write(codec, 0x203, 0xc);
snd_soc_write(codec, 0x206, 0);
@@ -1392,6 +1393,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec,
wm5100_reva_patches[i].reg,
wm5100_reva_patches[i].val);
+ regcache_cache_bypass(wm5100->regmap, false);
break;
default:
break;
@@ -1402,6 +1404,8 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
+ regcache_cache_only(wm5100->regmap, true);
+ regcache_mark_dirty(wm5100->regmap);
if (wm5100->pdata.ldo_ena)
gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
@@ -2180,6 +2184,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Microphone detected\n");
wm5100->jack_mic = true;
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack,
SND_JACK_HEADSET,
SND_JACK_HEADSET | SND_JACK_BTN_0);
@@ -2218,6 +2223,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
SND_JACK_BTN_0);
} else if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Headphone detected\n");
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE,
SND_JACK_HEADPHONE);
@@ -2607,6 +2613,13 @@ static const struct regmap_config wm5100_regmap = {
.cache_type = REGCACHE_RBTREE,
};
+static const unsigned int wm5100_mic_ctrl_reg[] = {
+ WM5100_IN1L_CONTROL,
+ WM5100_IN2L_CONTROL,
+ WM5100_IN3L_CONTROL,
+ WM5100_IN4L_CONTROL,
+};
+
static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2739,7 +2752,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
}
for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) {
- regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL,
+ regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i],
WM5100_IN1_MODE_MASK |
WM5100_IN1_DMIC_SUP_MASK,
(wm5100->pdata.in_mode[i] <<
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 8d4ea43d40a3..40ac888faf3d 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
return 0;
if (fw->size < 32) {
- dev_err(codec->dev, "%s: firmware too short (%d bytes)\n",
+ dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n",
name, fw->size);
goto err;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 296de4e30d26..0ac228b7dc04 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -96,7 +96,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \
struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8962->regmap, true); \
+ regcache_mark_dirty(wm8962->regmap); \
} \
return 0; \
}
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
@@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- aif0 |= 0x40;
+ aif0 |= 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- aif0 |= 0x80;
+ aif0 |= 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- aif0 |= 0xc0;
+ aif0 |= 0xc;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 93d27b660257..ec69a6c152fe 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ pm_runtime_get_sync(codec->dev);
+
wm8994->vmid_refcount++;
dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
@@ -783,7 +785,12 @@ static void vmid_reference(struct snd_soc_codec *codec)
WM8994_VMID_RAMP_MASK,
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
- (0x11 << WM8994_VMID_RAMP_SHIFT));
+ (0x3 << WM8994_VMID_RAMP_SHIFT));
+
+ /* Remove discharge for line out */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH, 0);
/* Main bias enable, VMID=2x40k */
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
@@ -837,6 +844,8 @@ static void vmid_dereference(struct snd_soc_codec *codec)
WM8994_VMID_BUF_ENA |
WM8994_VMID_RAMP_MASK, 0);
}
+
+ pm_runtime_put(codec->dev);
}
static int vmid_event(struct snd_soc_dapm_widget *w,
@@ -2753,11 +2762,6 @@ static int wm8994_resume(struct snd_soc_codec *codec)
codec->cache_only = 0;
}
- /* Restore the registers */
- ret = snd_soc_cache_sync(codec);
- if (ret != 0)
- dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
-
wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d8da10fe5b52..61f7daa4d0e6 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -108,7 +108,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \
struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8996->regmap, true); \
+ regcache_mark_dirty(wm8996->regmap); \
} \
return 0; \
}
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0),
@@ -2007,6 +2008,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
int lfclk = 0;
int ratediv = 0;
+ int sync = WM8996_REG_SYNC;
int src;
int old;
@@ -2051,6 +2053,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
case 32000:
case 32768:
lfclk = WM8996_LFCLK_ENA;
+ sync = 0;
break;
default:
dev_warn(codec->dev, "Unsupported clock rate %dHz\n",
@@ -2064,6 +2067,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK,
src << WM8996_SYSCLK_SRC_SHIFT | ratediv);
snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk);
+ snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1,
+ WM8996_REG_SYNC, sync);
snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1,
WM8996_SYSCLK_ENA, old);
diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h
index 0fde643194ce..de9ac3e44aec 100644
--- a/sound/soc/codecs/wm8996.h
+++ b/sound/soc/codecs/wm8996.h
@@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
/*
* R257 (0x101) - Control Interface (1)
*/
+#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */
+#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */
#define WM8996_AUTO_INC 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2a61094075f8..8a68cea4a3ee 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -586,14 +586,14 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new line2_mix[] = {
-SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new line2n_mix[] = {
-SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
};
static const struct snd_kcontrol_new line2p_mix[] = {
@@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -844,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
- { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
- { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
+ { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" },
+ { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
@@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf02..b6adbed6e506 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
break;
}
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index dccfb37a9626..f204dbac11d4 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -124,6 +124,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
*
* If MCLK is not used, we just set saif clk to 512*fs.
*/
+ clk_prepare_enable(master_saif->clk);
+
if (master_saif->mclk_in_use) {
if (mclk % 32 == 0) {
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
@@ -133,6 +135,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
ret = clk_set_rate(master_saif->clk, 384 * rate);
} else {
/* SAIF MCLK should be either 32x or 48x */
+ clk_disable_unprepare(master_saif->clk);
return -EINVAL;
}
} else {
@@ -140,6 +143,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
}
+ clk_disable_unprepare(master_saif->clk);
+
if (ret)
return ret;
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 7ac0ba2025c3..d23b19a59d83 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
/* GTA02 specific routes and controls */
-#ifdef CONFIG_MACH_NEO1973_GTA02
-
static int gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
return 0;
}
-#else
-static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
-#endif
-
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
int ret;
/* set up NC codec pins */
- if (machine_is_neo1973_gta01()) {
- snd_soc_dapm_nc_pin(dapm, "LOUT2");
- snd_soc_dapm_nc_pin(dapm, "ROUT2");
- }
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "OUT4");
snd_soc_dapm_nc_pin(dapm, "LINE1");
@@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-/* GTA01 specific controls */
-
-#ifdef CONFIG_MACH_NEO1973_GTA01
-
-static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
- {"Amp IN", NULL, "ROUT1"},
- {"Amp IN", NULL, "LOUT1"},
-
- {"Handset Spk", NULL, "Amp EP"},
- {"Stereo Out", NULL, "Amp LS"},
- {"Headphone", NULL, "Amp HP"},
-};
-
-static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
- SND_SOC_DAPM_SPK("Stereo Out", NULL),
- SND_SOC_DAPM_HP("Headphone", NULL),
-};
-
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
-{
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
- ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
- ARRAY_SIZE(neo1973_lm4857_routes));
- if (ret)
- return ret;
-
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
- snd_soc_dapm_ignore_suspend(dapm, "Headphone");
-
- return 0;
-}
-
-#else
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
-#endif
-
static struct snd_soc_dai_link neo1973_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
@@ -421,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -430,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.stream_name = "Voice",
.cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.ops = &neo1973_voice_ops,
},
};
@@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = {
.name = "dfbmcs320",
.codec_name = "dfbmcs320.0",
},
- {
- .name = "lm4857",
- .codec_name = "lm4857.0-007c",
- .init = neo1973_lm4857_init,
- },
};
static struct snd_soc_codec_conf neo1973_codec_conf[] = {
@@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = {
},
};
-#ifdef CONFIG_MACH_NEO1973_GTA02
static const struct gpio neo1973_gta02_gpios[] = {
{ GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
{ GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
};
-#else
-static const struct gpio neo1973_gta02_gpios[] = {};
-#endif
static struct snd_soc_card neo1973 = {
.name = "neo1973",
@@ -480,7 +417,7 @@ static int __init neo1973_init(void)
{
int ret;
- if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
+ if (!machine_is_neo1973_gta02())
return -ENODEV;
if (machine_is_neo1973_gta02()) {
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db6c89a28bda..ea4a82d01160 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
- int samples_pos = io->buff_sample_pos - 1;
- if (samples_pos < 0)
- samples_pos = 0;
-
- return fsi_sample2frame(fsi, samples_pos);
+ return fsi_sample2frame(fsi, io->buff_sample_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b5ecf6d23214..92cee24ed2dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev)
if (!codec->suspended && codec->driver->suspend) {
switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
+ /*
+ * If the CODEC is capable of idle
+ * bias off then being in STANDBY
+ * means it's doing something,
+ * otherwise fall through.
+ */
+ if (codec->dapm.idle_bias_off) {
+ dev_dbg(codec->dev,
+ "idle_bias_off CODEC on over suspend\n");
+ break;
+ }
case SND_SOC_BIAS_OFF:
codec->driver->suspend(codec);
codec->suspended = 1;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1f55ded4047f..1315663c1c09 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+ if (dapm->bias_level == SND_SOC_BIAS_ON)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_STANDBY);
}
}
@@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&codec->dapm,
+ SND_SOC_BIAS_OFF);
}
}
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed4..fde9a7a29cb6 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
spin_lock(&dev->spinlock);
- if (dev->input_panic || dev->output_panic)
+ if (dev->input_panic || dev->output_panic) {
ptr = SNDRV_PCM_POS_XRUN;
+ goto unlock;
+ }
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
ptr = bytes_to_frames(sub->runtime,
dev->audio_in_buf_pos[index]);
+unlock:
spin_unlock(&dev->spinlock);
return ptr;
}
diff --git a/sound/usb/card.h b/sound/usb/card.h
index a39edcc32a93..da5fa1ac4eda 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -1,6 +1,7 @@
#ifndef __USBAUDIO_CARD_H
#define __USBAUDIO_CARD_H
+#define MAX_NR_RATES 1024
#define MAX_PACKS 20
#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */
#define MAX_URBS 8
diff --git a/sound/usb/format.c b/sound/usb/format.c
index e09aba19375c..ddfef57c4c9f 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
return 0;
}
-#define MAX_UAC2_NR_RATES 1024
-
/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
@@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
nr_rates++;
- if (nr_rates >= MAX_UAC2_NR_RATES) {
+ if (nr_rates >= MAX_NR_RATES) {
snd_printk(KERN_ERR "invalid uac2 rates\n");
break;
}
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 8edc5035fc8f..d89ab4c7d44b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1618,6 +1618,14 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
{
+ /* Edirol UM-3G */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .ifnum = 0,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
/* Boss JS-8 Jam Station */
USB_DEVICE(0x0582, 0x0109),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a3ddac0deffd..27817266867a 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
unsigned *rate_table = NULL;
fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
- if (! fp) {
+ if (!fp) {
snd_printk(KERN_ERR "cannot memdup\n");
return -ENOMEM;
}
+ if (fp->nr_rates > MAX_NR_RATES) {
+ kfree(fp);
+ return -EINVAL;
+ }
if (fp->nr_rates > 0) {
rate_table = kmemdup(fp->rate_table,
sizeof(int) * fp->nr_rates, GFP_KERNEL);

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