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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.c14
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-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/soc-dapm.c12
-rw-r--r--sound/usb/caiaq/audio.c5
12 files changed, 79 insertions, 35 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 65c01798d84..76bac4fc047 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;
@@ -5123,7 +5127,7 @@ 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) {
@@ -5182,7 +5186,7 @@ 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 654d2e41e25..9a9f372e1be 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 f3b79031fcc..5a56fda8362 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3470,7 +3470,7 @@ 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;
@@ -4112,7 +4112,8 @@ 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;
@@ -4413,6 +4414,22 @@ 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;
@@ -4431,6 +4448,9 @@ 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 e142f6f5c49..01179d53edc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -841,7 +841,7 @@ 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;
@@ -1856,7 +1856,7 @@ DEFINE_CAPMIX_NOSRC(3);
*/
static const char * const alc_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Speaker", "Mono", "Line-Out",
+ "Headphone", "Speaker", "Mono", "Line Out",
"CLFE", "Bass Speaker", "PCM",
NULL,
};
@@ -4147,7 +4147,7 @@ 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 */
@@ -5082,12 +5082,20 @@ 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);
}
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8c346ac59d4..5988dbdedc4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4638,7 +4638,7 @@ 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/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e..278c0a0575f 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 29c4b02c479..0ac228b7dc0 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -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);
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf0..b6adbed6e50 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/soc-dapm.c b/sound/soc/soc-dapm.c
index 1f55ded4047..1315663c1c0 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 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;
}
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-24 07:34:44 +0000