From 8ac60a6866e58b861f2a15689c6513faf1602a3d Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 29 May 2013 20:01:19 +0200 Subject: [PATCH 1/9] ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 030f53c96ec0..756c204b62d8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); 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), @@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { 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), From 7d8acf2cba81d7c64842b5dac0d7b3dae16f0378 Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 29 May 2013 20:01:20 +0200 Subject: [PATCH 2/9] ASoC: cs42l52: fix hp_gain_enum shift value. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 756c204b62d8..987f728718c5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -262,7 +262,7 @@ static const char * const hp_gain_num_text[] = { }; 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[] = { From 9e43088bb015397930d6c9ea5edba92abc0dc655 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 May 2013 18:38:46 +0100 Subject: [PATCH 3/9] ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index dfd997aaadfc..19e0b2048af6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3836,7 +3836,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) 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, From 7afce3f5e56e9cb97cf1f35832bf8e8dde08cc45 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 May 2013 13:42:27 +0100 Subject: [PATCH 4/9] ASoC: wm8994: Ensure microphone detection state is reset on removal Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 19e0b2048af6..29e95f93d482 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3842,6 +3842,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, 0, SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + wm8994->mic_detecting = true; goto out; } From d9f1f489c034829c1082c0e7efa09450e716d23e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 1 Jun 2013 19:31:15 +0100 Subject: [PATCH 5/9] MAINTAINERS: Remove myself from Wolfson maintainers I no longer work for Wolfson. Signed-off-by: Mark Brown --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index fd3a495a0005..859cf064bbab 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8985,7 +8985,7 @@ S: Maintained F: drivers/net/wireless/wl3501* WM97XX TOUCHSCREEN DRIVERS -M: Mark Brown +M: Mark Brown M: Liam Girdwood L: linux-input@vger.kernel.org T: git git://opensource.wolfsonmicro.com/linux-2.6-touch @@ -8995,7 +8995,6 @@ F: drivers/input/touchscreen/*wm97* F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS -M: Mark Brown 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 From 056790923e1c4eed5d8cc502e1092944a2b23025 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 1 Jun 2013 23:13:53 +0100 Subject: [PATCH 6/9] ASoC: pcm: Require both CODEC and CPU support when declaring stream caps When declaring playback and capture capabilities check for both CODEC side and CPU side support rather than only checking for CODEC side support. While it is unusual some CPUs do have unidirectional DAIs. Reported-by: Fabio Estevam Tested-by: Fabio Estevam Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8eefa491..a9fddf0fea19 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2011,9 +2011,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) 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; } From ee4b7c7fe0c50b97d074f9185dba9558d9440c21 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Jun 2013 14:51:30 +0100 Subject: [PATCH 7/9] ASoC: arizona: Correct AEC loopback enable Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d3939eef..100fdadda56a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, 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 ba38f0679662..88ad7db52dde 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, 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), From 4616274d3382fa7698536d61b351e63cf0ce27f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 5 Jun 2013 19:36:11 +0100 Subject: [PATCH 8/9] ASoC: dapm: Treat DAI widgets like AIF widgets for power Even though they are virtual widgets DAI widgets still get counted for the DAPM context power management so we can't just use the active state to check if they should be powered as they may not be part of a complete path. Instead split them into input and output widgets and do the same power checks as we perform on AIFs. Reported-by: Stephen Warren Tested-by: Stephen Warren Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 49 +++++++++++++++++++++------------------- sound/soc/soc-pcm.c | 7 +++++- 3 files changed, 34 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d4609029f014..385c6329a967 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -450,7 +450,8 @@ enum snd_soc_dapm_type { 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/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8be..c7051c457b75 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [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 @@ static int dapm_down_seq[] = { [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 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, 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 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, 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 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, 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 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) 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 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, 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 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, 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 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, 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 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, 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 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } 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 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* 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 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) 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 a9fddf0fea19..ccb6be4d658d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* 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); From 2894770ec17ff732f911c8495ae0504f06a5dad5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= Date: Wed, 5 Jun 2013 08:49:47 +0200 Subject: [PATCH 9/9] ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is prevented. This problem was introduced in Linux 3.5 Signed-off-by: Andreas Irestål Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d60b7c6..1514bf845e4b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, 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; }