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authorLinus Torvalds <torvalds@linux-foundation.org>2012-03-02 15:20:41 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2012-03-02 15:20:41 -0800
commit5e8063d758725a5a6f38ac34a37d13509ccea0da (patch)
tree61a56d860787d7f82348006addb359f6e38f9355
parent200e9ef7ab51f3dce4f35f90ea458cf43ea83bb8 (diff)
parente49a3434f1bc64dc49ff3a56e416bb5894868dde (diff)
downloadlinux-stericsson-5e8063d758725a5a6f38ac34a37d13509ccea0da.tar.gz
Merge tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
sound fixes for 3.3-rc6 from Takashi Iwai This contains again regression fixes for various HD-audio and ASoC regarding SSI and dapm shutdown path. In addition, a minor azt3328 fix and the correction of the new jack-notification strings in HD-audio. * tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - Kill hyphenated names ALSA: hda - Add a fake mute feature ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3 ALSA: hda/realtek - Fix resume of multiple input sources ASoC: i.MX SSI: Fix DSP_A format. ASoC: dapm: Check for bias level when powering down
-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.c8
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/soc-dapm.c12
9 files changed, 51 insertions, 19 deletions
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/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f63bf06..684307372d73 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 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/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bc5a993d1146..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;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733aa4d2..d29d6d377904 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 3647baa9bfed..f286bb8fda13 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -802,7 +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;
@@ -1856,7 +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",
@@ -1873,7 +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",
@@ -3797,7 +3797,7 @@ 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 */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a00..9dbb5735d778 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/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/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);
}
}