From 50b6bce59d154b5db137907a5c0ed45a4e7a3829 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
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 <jy0922.shim@samsung.com>
Tested-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
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 d89f6dc00908..66d4c165f99b 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 <anuj.aggarwal@ti.com>
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 <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
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 6b24d8bb02bb..90a0264f7538 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 <rmk+kernel@arm.linux.org.uk>
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 <rmk+kernel@arm.linux.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/arm/aaci.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1f0f8213e2d5..1cb7c282a1fb 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 <rmk+kernel@arm.linux.org.uk>
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 <rmk+kernel@arm.linux.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 1cb7c282a1fb..6c160a038b23 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;