Merge tag 'asoc-v3.10-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound
Pull ASoC sound updates from Mark Brown:
"Takashi is travelling at the minute and it'd be good to get the
MAINTAINERS update in here merged so sending directly.
As well as the usual driver specifics we've got a couple of core fixes
here, one fixing capabilities for unidirectional streams and the other
fixing suspend while audio streams are active.
The suspend fix is a little involved but mostly as a result of
removing some special casing that was doing the wrong thing."
* tag 'asoc-v3.10-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound:
ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x()
ASoC: dapm: Treat DAI widgets like AIF widgets for power
ASoC: arizona: Correct AEC loopback enable
ASoC: pcm: Require both CODEC and CPU support when declaring stream caps
MAINTAINERS: Remove myself from Wolfson maintainers
ASoC: wm8994: Ensure microphone detection state is reset on removal
ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race
ASoC: cs42l52: fix hp_gain_enum shift value.
ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet.
diff --git a/MAINTAINERS b/MAINTAINERS
index 0c9dc71..5be702c 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -9004,7 +9004,7 @@
F: drivers/net/wireless/wl3501*
WM97XX TOUCHSCREEN DRIVERS
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
+M: Mark Brown <broonie@kernel.org>
M: Liam Girdwood <lrg@slimlogic.co.uk>
L: linux-input@vger.kernel.org
T: git git://opensource.wolfsonmicro.com/linux-2.6-touch
@@ -9014,7 +9014,6 @@
F: include/linux/wm97xx.h
WOLFSON MICROELECTRONICS DRIVERS
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
L: patches@opensource.wolfsonmicro.com
T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index d460902..385c632 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -450,7 +450,8 @@
snd_soc_dapm_aif_in, /* audio interface input */
snd_soc_dapm_aif_out, /* audio interface output */
snd_soc_dapm_siggen, /* signal generator */
- snd_soc_dapm_dai, /* link to DAI structure */
+ snd_soc_dapm_dai_in, /* link to DAI structure */
+ snd_soc_dapm_dai_out,
snd_soc_dapm_dai_link, /* link between two DAI structures */
};
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 030f53c..987f728 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -193,6 +193,8 @@
static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
+static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+
static const unsigned int limiter_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -260,7 +262,7 @@
};
static const struct soc_enum hp_gain_enum =
- SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+ SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
static const char * const beep_pitch_text[] = {
@@ -441,7 +443,7 @@
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
- 0, 0x7f, 0x19, hl_tlv),
+ 0, 0x7f, 0x19, mix_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 65d09d6..1514bf8 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -187,14 +187,14 @@
break;
}
-
- if (found)
- snd_soc_dapm_sync(widget->dapm);
}
- ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
-
mutex_unlock(&widget->codec->mutex);
+
+ if (found)
+ snd_soc_dapm_sync(widget->dapm);
+
+ ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
return ret;
}
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e895d39..100fdad 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1120,7 +1120,8 @@
ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm5102_aec_loopback_mux),
SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index ba38f06..88ad7db 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -503,7 +503,8 @@
NULL, 0),
SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux),
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm5110_aec_loopback_mux),
SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index dfd997a..29e95f9 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3836,12 +3836,13 @@
ret);
} else if (!(ret & WM1811_JACKDET_LVL)) {
dev_dbg(codec->dev, "Ignoring removed jack\n");
- return IRQ_HANDLED;
+ goto out;
}
} else if (!(reg & WM8958_MICD_STS)) {
snd_soc_jack_report(wm8994->micdet[0].jack, 0,
SND_JACK_MECHANICAL | SND_JACK_HEADSET |
wm8994->btn_mask);
+ wm8994->mic_detecting = true;
goto out;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a80c883..c7051c4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -55,7 +55,8 @@
[snd_soc_dapm_clock_supply] = 1,
[snd_soc_dapm_micbias] = 2,
[snd_soc_dapm_dai_link] = 2,
- [snd_soc_dapm_dai] = 3,
+ [snd_soc_dapm_dai_in] = 3,
+ [snd_soc_dapm_dai_out] = 3,
[snd_soc_dapm_aif_in] = 3,
[snd_soc_dapm_aif_out] = 3,
[snd_soc_dapm_mic] = 4,
@@ -92,7 +93,8 @@
[snd_soc_dapm_value_mux] = 9,
[snd_soc_dapm_aif_in] = 10,
[snd_soc_dapm_aif_out] = 10,
- [snd_soc_dapm_dai] = 10,
+ [snd_soc_dapm_dai_in] = 10,
+ [snd_soc_dapm_dai_out] = 10,
[snd_soc_dapm_dai_link] = 11,
[snd_soc_dapm_clock_supply] = 12,
[snd_soc_dapm_regulator_supply] = 12,
@@ -419,7 +421,8 @@
case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
@@ -820,7 +823,7 @@
switch (widget->id) {
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_out:
if (widget->active) {
widget->outputs = snd_soc_dapm_suspend_check(widget);
return widget->outputs;
@@ -916,7 +919,7 @@
switch (widget->id) {
case snd_soc_dapm_dac:
case snd_soc_dapm_aif_in:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
if (widget->active) {
widget->inputs = snd_soc_dapm_suspend_check(widget);
return widget->inputs;
@@ -1135,16 +1138,6 @@
return out != 0 && in != 0;
}
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
- DAPM_UPDATE_STAT(w, power_checks);
-
- if (w->active)
- return w->active;
-
- return dapm_generic_check_power(w);
-}
-
/* Check to see if an ADC has power */
static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
{
@@ -2318,7 +2311,8 @@
case snd_soc_dapm_clock_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
case snd_soc_dapm_dai_link:
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &wsink->sources);
@@ -3129,10 +3123,12 @@
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
+ case snd_soc_dapm_dai_out:
w->power_check = dapm_adc_check_power;
break;
case snd_soc_dapm_dac:
case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_dai_in:
w->power_check = dapm_dac_check_power;
break;
case snd_soc_dapm_pga:
@@ -3152,9 +3148,6 @@
case snd_soc_dapm_clock_supply:
w->power_check = dapm_supply_check_power;
break;
- case snd_soc_dapm_dai:
- w->power_check = dapm_dai_check_power;
- break;
default:
w->power_check = dapm_always_on_check_power;
break;
@@ -3375,7 +3368,7 @@
template.reg = SND_SOC_NOPM;
if (dai->driver->playback.stream_name) {
- template.id = snd_soc_dapm_dai;
+ template.id = snd_soc_dapm_dai_in;
template.name = dai->driver->playback.stream_name;
template.sname = dai->driver->playback.stream_name;
@@ -3393,7 +3386,7 @@
}
if (dai->driver->capture.stream_name) {
- template.id = snd_soc_dapm_dai;
+ template.id = snd_soc_dapm_dai_out;
template.name = dai->driver->capture.stream_name;
template.sname = dai->driver->capture.stream_name;
@@ -3423,8 +3416,13 @@
/* For each DAI widget... */
list_for_each_entry(dai_w, &card->widgets, list) {
- if (dai_w->id != snd_soc_dapm_dai)
+ switch (dai_w->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
+ break;
+ default:
continue;
+ }
dai = dai_w->priv;
@@ -3433,8 +3431,13 @@
if (w->dapm != dai_w->dapm)
continue;
- if (w->id == snd_soc_dapm_dai)
+ switch (w->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
continue;
+ default:
+ break;
+ }
if (!w->sname)
continue;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 73bb8ee..ccb6be4 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -928,8 +928,13 @@
/* Create any new FE <--> BE connections */
for (i = 0; i < list->num_widgets; i++) {
- if (list->widgets[i]->id != snd_soc_dapm_dai)
+ switch (list->widgets[i]->id) {
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_dai_out:
+ break;
+ default:
continue;
+ }
/* is there a valid BE rtd for this widget */
be = dpcm_get_be(card, list->widgets[i], stream);
@@ -2011,9 +2016,11 @@
if (cpu_dai->driver->capture.channels_min)
capture = 1;
} else {
- if (codec_dai->driver->playback.channels_min)
+ if (codec_dai->driver->playback.channels_min &&
+ cpu_dai->driver->playback.channels_min)
playback = 1;
- if (codec_dai->driver->capture.channels_min)
+ if (codec_dai->driver->capture.channels_min &&
+ cpu_dai->driver->capture.channels_min)
capture = 1;
}
