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-rw-r--r--sound/pci/azt3328.c3
-rw-r--r--sound/pci/hda/hda_codec.c12
-rw-r--r--sound/pci/hda/hda_codec.h3
-rw-r--r--sound/pci/hda/patch_cirrus.c4
-rw-r--r--sound/pci/hda/patch_conexant.c24
-rw-r--r--sound/pci/hda/patch_realtek.c44
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/pci/rme9652/hdspm.c1
-rw-r--r--sound/soc/codecs/ak4642.c31
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/omap/ams-delta.c34
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c4
-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.c6
18 files changed, 116 insertions, 78 deletions
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6..496f14c1a73 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/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f63bf0..684307372d7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -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 e9f71dc0d46..f0f1943a4b2 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/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bc5a993d114..c83ccdba1e5 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;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733aa4d..d29d6d37790 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -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 1358987c49d..22c73b78ac6 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;
@@ -800,7 +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;
@@ -1854,7 +1856,7 @@ 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",
@@ -1871,7 +1873,7 @@ 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",
@@ -2066,12 +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);
@@ -3149,7 +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;
}
@@ -3792,7 +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 */
@@ -4362,6 +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,
@@ -4436,12 +4446,20 @@ 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);
}
}
@@ -4481,6 +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[]) {
@@ -4637,6 +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),
@@ -5454,8 +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);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a0..9dbb5735d77 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4629,7 +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
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cc9f6c83d66..bc030a2088d 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/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 16bd1e7d238..f8e10ced244 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/wm8962.c b/sound/soc/codecs/wm8962.c
index 5bcb350bacc..15d467ff91b 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1988,7 +1988,7 @@ 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);
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 9203cdd0a15..4f81ed45632 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/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 78563bbbbf0..41586b26ce9 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = {
};
-/* Board specific codec bias level control */
-static int ams_delta_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
- enum snd_soc_bias_level level)
-{
- switch (level) {
- case SND_SOC_BIAS_ON:
- case SND_SOC_BIAS_PREPARE:
- case SND_SOC_BIAS_STANDBY:
- if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
- AMS_DELTA_LATCH2_MODEM_NRESET);
- break;
- case SND_SOC_BIAS_OFF:
- if (card->dapm.bias_level != SND_SOC_BIAS_OFF)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
- 0);
- }
- card->dapm.bias_level = level;
-
- return 0;
-}
-
/* Digital mute implemented using modem/CPU multiplexer.
* Shares hardware with codec config pulse generation */
static bool ams_delta_muted = 1;
@@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
ams_delta_ops.shutdown = ams_delta_shutdown;
}
- /* Set codec bias level */
- ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
-
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
ret = snd_soc_jack_new(rtd->codec, "hook_switch",
@@ -598,7 +572,6 @@ static struct snd_soc_card ams_delta_audio_card = {
.owner = THIS_MODULE,
.dai_link = &ams_delta_dai_link,
.num_links = 1,
- .set_bias_level = ams_delta_set_bias_level,
};
/* Module init/exit */
@@ -635,7 +608,7 @@ err:
platform_device_put(ams_delta_audio_platform_device);
return ret;
}
-module_init(ams_delta_module_init);
+late_initcall(ams_delta_module_init);
static void __exit ams_delta_module_exit(void)
{
@@ -647,11 +620,6 @@ static void __exit ams_delta_module_exit(void)
ARRAY_SIZE(ams_delta_hook_switch_gpios),
ams_delta_hook_switch_gpios);
- /* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card,
- &ams_delta_audio_card.rtd[0].codec->dapm,
- SND_SOC_BIAS_STANDBY);
-
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
}
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 24bdb321269..321d51134e4 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -367,7 +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,
},
@@ -376,7 +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,
},
};
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dcd11609f93..6241490fff3 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3238,9 +3238,13 @@ 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);
}
}
@@ -3253,7 +3257,9 @@ 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 2cf87f5afed..fde9a7a29cb 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 a39edcc32a9..da5fa1ac4ed 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 e09aba19375..ddfef57c4c9 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.c b/sound/usb/quirks.c
index a3ddac0deff..27817266867 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);
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-17 18:05:29 +0000