October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Aureal issues on resume
by Clemens Ladisch 27 Oct '11

27 Oct '11

(added recipients) -------- Original Message -------- Subject: Aureal issues on resume Date: Fri, 02 Sep 2011 13:41:42 +0200 From: Anders Eriksson <aeriksson(a)fastmail.fm> To: linux-kernel(a)vger.kernel.org, linux-sound(a)vger.kernel.org Hi, upon resuming from hibernation, I see this in my log: Sep 2 11:15:20 tippex kernel: [72436.287020] vortex_adb_addroutes: unending route! 0x10 Sep 2 11:15:20 tippex kernel: [72436.287127] vortex_adb_addroutes: unending route! 0x838 Sep 2 11:15:20 tippex kernel: [72436.287277] vortex_adb_addroutes: unending route! 0x11 Sep 2 11:15:20 tippex kernel: [72436.287418] vortex_adb_addroutes: unending route! 0x8b9 Sep 2 11:15:20 tippex kernel: [72436.292037] Vortex: vortex_fifo_setadbctrl fail Sep 2 11:15:55 tippex kernel: [72471.403490] vortex_adb_delroutes: route not found! 0x10 Sep 2 11:15:55 tippex kernel: [72471.403596] vortex_adb_delroutes: route not found! 0x838 Sep 2 11:15:55 tippex kernel: [72471.403710] vortex_adb_delroutes: route not found! 0x11 Sep 2 11:15:55 tippex kernel: [72471.403811] vortex_adb_delroutes: route not found! 0x8b9 And the sound is no more. I need to reboot to get it back. If memory serves, it's been there since 2.6.30 or so. Any ideas what can be triggering it? lspci says it's an 00:04.0 Multimedia audio controller: Aureal Semiconductor Vortex 1 (rev 02) Subsystem: Diamond Multimedia Systems Sonic Impact A3D Flags: bus master, medium devsel, latency 64, IRQ 11 Memory at 50000000 (32-bit, non-prefetchable) [size=128K] I/O ports at 10b0 [size=8] I/O ports at 10b8 [size=8] Capabilities: [dc] Power Management version 1 Kernel driver in use: snd_au8820 Kernel modules: snd-au8820 -Anders

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[alsa-devel] Help about LineIn capture on WM8904
by gnuser 27 Oct '11

27 Oct '11

Hi Mark and all, I'm working on an ASoC machine driver to include the support for WM8904. The ARM SoC is AT91SAM9G45 from Atmel. In Atmel's original reference board, Atmel uses WM8731 which works quite well. For my case, I replaced it with WM8904. Both Atmel's ref board and mine use I2S to talk with audio codec. So far, the audio playback is OK. But audio capture can not work well. The *hardware connection* in my case is, external 32768Hz crystal --> MCLK (Use FLL to get proper sysclk) Line In Jack --> IN2L Line In Jack --> IN2R MiC --> IN1L HeadPhone --> HPOUTL HeadPhone --> HPOUTR NC Pins: LINEOUTL, LINEOUTR, LINEOUTFB, IN1R Codec is in master mode, ARM Soc is in slave mode. My questions are, 1) Since WM8731 works fine (both playback and capture) on Atmel's ref board, and playback is also OK for my case, can we get the conclusion that the PCM driver and DAI driver are OK? Does the clock configuration for my case also look OK? 2) In my machine driver, I added the following DAPM components, static const struct snd_soc_dapm_widget mini_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_LINE("LineIn Jack", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* headphone jack connected to HPOUT */ {"Headphone Jack", NULL, "HPOUTL"}, {"Headphone Jack", NULL, "HPOUTR"}, {"MICBIAS", NULL, "Mic Jack"}, {"IN1L", NULL, "MICBIAS"}, {"IN2R", NULL, "LineIn Jack"}, {"IN2L", NULL, "LineIn Jack"}, }; - and - then, { snd_soc_dapm_nc_pin(dapm, "IN1R"); snd_soc_dapm_nc_pin(dapm, "IN3L"); snd_soc_dapm_nc_pin(dapm, "IN3R"); snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); snd_soc_dapm_new_controls(dapm, mini_dapm_widgets, ARRAY_SIZE(mini_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "LineIn Jack"); snd_soc_dapm_sync(dapm); } Does the above code look fine? 3) I used "arecord -d 8 -f S16_LE -c2 -r8000 8000.wav" to capture audio. But the captured data are something looks like 84 FF 73 FF 83 FF 70 FF 85 FF 72 FF 84 FF 71 FF 83 FF 72 FF 84 FF 72 FF 87 FF 71 FF 83 FF 71 FF (RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz) The captured file can not be played well (no sound actually) Both arecord, PCM driver, DAI driver and snd core driver are quiet without any warning or error. I tried to use alsamixer to configure the audo path. But I'm new to ALSA and not sure about the configuration. I just paste the amixer here, could you please point me where I've made mistakes? A piece of amixer scripts will greatly help me. :) Thanks in advance. Simple mixer control 'Headphone',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 63 Front Left: 46 [73%] [-11.00dB] Playback [on] Front Right: 45 [71%] [-12.00dB] Playback [on] Simple mixer control 'Headphone ZC',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Line Output',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 63 Front Left: 57 [90%] [0.00dB] Playback [on] Front Right: 57 [90%] [0.00dB] Playback [on] Simple mixer control 'Line Output ZC',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 5 [16%] [on] Front Right: Capture 5 [16%] [on] Simple mixer control 'ADC 128x OSR',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'AIFOUTL Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Left' Simple mixer control 'AIFOUTR Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Right' Simple mixer control 'DAC Deemphasis',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DAC OSRx2',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DACL Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Left' Simple mixer control 'DACR Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Right' Simple mixer control 'DRC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DRC Path',0 Capabilities: enum Items: 'ADC' 'DAC' Item0: 'ADC' Simple mixer control 'Digital',0 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 96 Capture 0 - 119 Front Left: Playback 96 [100%] [0.00dB] Capture 96 [81%] [0.00dB] Front Right: Playback 96 [100%] [0.00dB] Capture 96 [81%] [0.00dB] Simple mixer control 'Digital Playback Boost',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] [0.00dB] Simple mixer control 'Digital Sidetone',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 15 Front Left: 0 [0%] [-36.00dB] Front Right: 0 [0%] [-36.00dB] Simple mixer control 'EQ',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'EQ1',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ2',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ3',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ4',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ5',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'HPL Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'HPR Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'High Pass Filter',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'High Pass Filter Mode',0 Capabilities: enum Items: 'Hi-fi' 'Voice 1' 'Voice 2' 'Voice 3' Item0: 'Hi-fi' Simple mixer control 'LINEL Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'LINER Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'Left Capture Inverting Mux',0 Capabilities: enum Items: 'IN1L' 'IN2L' 'IN3L' Item0: 'IN1L' Simple mixer control 'Left Capture Mux',0 Capabilities: enum Items: 'IN1L' 'IN2L' 'IN3L' Item0: 'IN2L' Simple mixer control 'Left Caputure Mode',0 Capabilities: enum Items: 'Single-Ended' 'Differential Line' 'Differential Mic' Item0: 'Single-Ended' Simple mixer control 'Left Sidetone',0 Capabilities: enum Items: 'None' 'Left' 'Right' Item0: 'None' Simple mixer control 'Right Capture Inverting Mux',0 Capabilities: enum Items: 'IN1R' 'IN2R' 'IN3R' Item0: 'IN1R' Simple mixer control 'Right Capture Mode',0 Capabilities: enum Items: 'Single-Ended' 'Differential Line' 'Differential Mic' Item0: 'Single-Ended' Simple mixer control 'Right Capture Mux',0 Capabilities: enum Items: 'IN1R' 'IN2R' 'IN3R' Item0: 'IN2R' Simple mixer control 'Right Sidetone',0 Capabilities: enum Items: 'None' 'Left' 'Right' Item0: 'None'

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic23: Clear TLV320AIC23_MS_MASTER bit for slave mode
by Axel Lin 27 Oct '11

27 Oct '11

According to the datasheet: Digital Audio Interface Format (07h) register: BIT6: Master/slave mode 0: Slave 1: Master Current code sets TLV320AIC23_MS_MASTER bit for master mode, but does not clear it for slave mode. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index ab27dbc..336de8f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -430,6 +430,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg |= TLV320AIC23_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + iface_reg &= ~TLV320AIC23_MS_MASTER; break; default: return -EINVAL; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Remove needless unlikely()
by Mark Brown 27 Oct '11

27 Oct '11

There's no point in adding unlikely() annotations outside of hot paths and on systems using these features the annotation will always be wrong (as opposed to being something that only comes up once in a while) so the annotation may even be harmful. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 52a7259..49aa71e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,8 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time)) { + if (codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC:core:add flag ignore_pmdown_time in snd_soc_dai_link
by ramesh.babu＠linux.intel.com 27 Oct '11

27 Oct '11

Resending the patch by CCing Peter With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: Ramesh Babu K V <ramesh.babu(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 3 ++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..877fcc1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,9 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* pmdown_time is ignored at stop */ + unsigned int ignore_pmdown_time:1; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..52a7259 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,7 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time)) { + if (unlikely(codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.4.1

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[alsa-devel] [PATCH v2] ASoC: wm8940: Fix mask for setting BCLKDIV
by Axel Lin 27 Oct '11

27 Oct '11

According to the datasheet: BCLK is controlled by BIT[4:2] of WM8940_CLOCK(06h) register. Signed-off-by: Chris Paulson-Ellis <chris(a)edesix.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index dc5cb31..de9ec9b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -621,7 +621,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8940_BCLKDIV: - reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3; + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFE3; ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2)); break; case WM8940_MCLKDIV: -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Use snd_soc_update_bits for read-modify-write
by Axel Lin 27 Oct '11

27 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 41 +++++++++++++-------------------------- 1 files changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7a49390..a77f6ea 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -833,7 +833,6 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 d, pll_d = 1; - u8 reg; int clk; /* select data word length */ @@ -869,14 +868,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); /* disable PLL if it is bypassed */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0); } else { snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); /* enable PLL when it is used */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } /* Route Left DAC to left channel input and @@ -1155,7 +1153,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - u8 reg; switch (level) { case SND_SOC_BIAS_ON: @@ -1164,9 +1161,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } break; case SND_SOC_BIAS_STANDBY: @@ -1175,9 +1171,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, 0); } break; case SND_SOC_BIAS_OFF: @@ -1294,7 +1289,6 @@ static int aic3x_resume(struct snd_soc_codec *codec) static int aic3x_init(struct snd_soc_codec *codec) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int reg; snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); @@ -1316,20 +1310,13 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ - reg = snd_soc_read(codec, LLOPM_CTRL); - snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, RLOPM_CTRL); - snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLOUT_CTRL); - snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPROUT_CTRL); - snd_soc_write(codec, HPROUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLCOM_CTRL); - snd_soc_write(codec, HPLCOM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPRCOM_CTRL); - snd_soc_write(codec, HPRCOM_CTRL, reg | UNMUTE); + snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPRCOM_CTRL, UNMUTE, UNMUTE); /* ADC default volume and unmute */ snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC:core:add flag ignore_pmdown_time in snd_soc_dai_link
by ramesh.babu＠linux.intel.com 27 Oct '11

27 Oct '11

With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: Ramesh Babu K V <ramesh.babu(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 3 ++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..877fcc1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,9 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* pmdown_time is ignored at stop */ + unsigned int ignore_pmdown_time:1; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..52a7259 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,7 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time)) { + if (unlikely(codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.4.1

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 27 Oct '11

27 Oct '11

The following changes since commit 226d0f22d044f0151287bb7cf334b85182248f0e: ASoC: keep pointer to resource so it can be freed (2011-10-22 10:46:35 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (2): ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: wm8940: Properly set codec->dapm.bias_level sound/soc/codecs/max98095.c | 13 ++++++------- sound/soc/codecs/wm8940.c | 2 ++ 2 files changed, 8 insertions(+), 7 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: wm8940: Properly set codec->dapm.bias_level
by Axel Lin 27 Oct '11

27 Oct '11

Reported-by: Chris Paulson-Ellis <chris(a)edesix.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index fec3892..e8a4ca3 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -488,6 +488,8 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; } + codec->dapm.bias_level = level; + return ret; } -- 1.7.5.4

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