From 50b6bce59d154b5db137907a5c0ed45a4e7a3829 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Nov 2009 13:11:53 +0000 Subject: [PATCH 1/4] ASoC: Fix suspend with active audio streams When we get a stream suspend event force the power down since otherwise the stream would remain marked as active. In future we'll probably want to make this stream-specific and add an interface to make the power down of other widgets optional in order to support leaving bypass paths active while suspending the processor. Cc: stable@kernel.org Reported-by: Joonyoung Shim Tested-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d89f6dc0090..66d4c165f99 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (!w->power_check) continue; - power = w->power_check(w); - if (power) - sys_power = 1; + /* If we're suspending then pull down all the + * power. */ + switch (event) { + case SND_SOC_DAPM_STREAM_SUSPEND: + power = 0; + break; + + default: + power = w->power_check(w); + if (power) + sys_power = 1; + break; + } if (w->power == power) continue; @@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_SUSPEND: + sys_power = 0; + break; case SND_SOC_DAPM_STREAM_NOP: sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + break; default: break; } From e9ff5eb2ae018fe2298c68746c873bf828c6b10e Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 27 Nov 2009 17:40:58 +0530 Subject: [PATCH 2/4] ASoC: AIC23: Fixing infinite loop in resume path This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Cc: stable@kernel.org Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8bb02b..90a0264f753 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } From 4acd57c3de62374fe5bb52e5cd24538190f4eab2 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:52 +0000 Subject: [PATCH 3/4] ALSA: AACI: fix AC97 multiple-open bug Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d..1cb7c282a1f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -504,6 +504,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } err = devdma_hw_alloc(NULL, substream, params_buffer_bytes(params)); From 8ee763b9c82c6ca0a59a7271ce4fa29d7baf5c09 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:59 +0000 Subject: [PATCH 4/4] ALSA: AACI: fix recording bug pcm->r[1].slots is the double rate slot information, not the capture information. For capture, 'pcm' will already be the capture ac97 pcm structure. Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1cb7c282a1f..6c160a038b2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -521,7 +521,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, else err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), params_channels(params), - aacirun->pcm->r[1].slots); + aacirun->pcm->r[0].slots); if (err) goto out;