Barry Song | 18 May 2013 14:25
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[PATCH v4] ASoC: dfbmcs320: make the driver common for other BT modules

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips,
we don't want everyone to have a seperate codec driver. anyway, the
feature of Bluetooth SCO is same on all platforms, so this patch
makes the DFBM-CS320 driver become a common BT SCO link driver.

Cc: Graeme Gregory <graeme <at> openmoko.org>
Cc: Lars-Peter Clausen <lars <at> metafoo.de>
Signed-off-by: Barry Song <Baohua.Song <at> csr.com>
---
 v4:
 change "select SND_SOC_DFBMCS320" to "select SND_SOC_BT_SCO" for OpenMOKO user
 as v3 missed this

 sound/soc/codecs/Kconfig                   |  4 ++--
 sound/soc/codecs/Makefile                  |  4 ++--
 sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++-----------
 sound/soc/samsung/Kconfig                  |  2 +-
 sound/soc/samsung/neo1973_wm8753.c         |  2 +-
 5 files changed, 29 insertions(+), 20 deletions(-)
 rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d8c4f3d..386292d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
 <at>  <at>  -40,7 +40,7  <at>  <at>  config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7213 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
-	select SND_SOC_DFBMCS320
(Continue reading)

Barry Song | 18 May 2013 13:22
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[PATCH v3] ASoC: dfbmcs320: make the driver common for other BT modules

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips,
we don't want everyone to have a seperate codec driver. anyway, the
feature of Bluetooth SCO is same on all platforms, so this patch
makes the DFBM-CS320 driver become a common BT SCO link driver.

Cc: Graeme Gregory <graeme <at> openmoko.org>
Cc: Lars-Peter Clausen <lars <at> metafoo.de>
Signed-off-by: Barry Song <Baohua.Song <at> csr.com>
---
 -v3: rename to sco according to Mark's comment

 sound/soc/codecs/Kconfig                   |  4 ++--
 sound/soc/codecs/Makefile                  |  4 ++--
 sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++-----------
 sound/soc/samsung/neo1973_wm8753.c         |  2 +-
 4 files changed, 28 insertions(+), 19 deletions(-)
 rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d8c4f3d..386292d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
 <at>  <at>  -40,7 +40,7  <at>  <at>  config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7213 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
-	select SND_SOC_DFBMCS320
+	select SND_SOC_BT_SCO
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
(Continue reading)

Jordi Mallach | 17 May 2013 18:25
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[PATCH] pcm: Fix typo in documentation: PREPARE → PREPARED.

Signed-off-by: Jordi Mallach <jordi <at> debian.org>
---
 src/pcm/pcm.c |    2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c
index 438fb3f..7ec1f0e 100644
--- a/src/pcm/pcm.c
+++ b/src/pcm/pcm.c
 <at>  <at>  -437,7 +437,7  <at>  <at>  to #SND_PCM_STATE_SETUP
 if successfully finishes, otherwise the state #SND_PCM_STATE_OPEN
 is entered.
 When it is brought to SETUP state, this function automatically
-calls #snd_pcm_prepare() function to bring to the PREPARE state
+calls #snd_pcm_prepare() function to bring to the PREPARED state
 as below.

 \par snd_pcm_prepare
--

-- 
1.7.10.4

Charles Keepax | 17 May 2013 17:58
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[PATCH] ASoC: soc-compress: Send correct stream event for capture start

Signed-off-by: Charles Keepax <ckeepax <at> opensource.wolfsonmicro.com>
---

Sorry little bug I missed due to some integration bugs in our
setup here. Again this applies onto Takashi's sound for-next
branch.

Thanks,
Charles

 sound/soc/soc-compress.c |    9 +++++++--
 1 files changed, 7 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7e..a5107df 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
 <at>  <at>  -220,8 +220,13  <at>  <at>  static int soc_compr_set_params(struct snd_compr_stream *cstream,
 			goto err;
 	}

-	snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-				SND_SOC_DAPM_STREAM_START);
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+					SND_SOC_DAPM_STREAM_START);
+	} else {
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+					SND_SOC_DAPM_STREAM_START);
+	}
(Continue reading)

Mark Brown | 17 May 2013 17:33
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[PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A

On WM8958 and WM1811A separate control of the LRCLK inversion bit is
available for the DAC and ADC LRCLKs which for compatibility reasons is
done in a new register bit.

Since writes to each scheme have no effect on parts using the other just
always write to both for simplicity.

Signed-off-by: Mark Brown <broonie <at> opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c |   14 ++++++++++++++
 1 file changed, 14 insertions(+)

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index ab7ebd9..74238bd 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
 <at>  <at>  -2573,17 +2573,24  <at>  <at>  static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	struct wm8994 *control = wm8994->wm8994;
 	int ms_reg;
 	int aif1_reg;
+	int dac_reg;
+	int adc_reg;
 	int ms = 0;
 	int aif1 = 0;
+	int lrclk = 0;

 	switch (dai->id) {
 	case 1:
 		ms_reg = WM8994_AIF1_MASTER_SLAVE;
 		aif1_reg = WM8994_AIF1_CONTROL_1;
(Continue reading)

Mark Brown | 17 May 2013 17:12
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[PATCH] ASoC: wm5110: Add missing speaker initialisation

Add callback to initialise the speaker in the core following the recent
changes to handling of integration with the thermal interrupts.

Signed-off-by: Mark Brown <broonie <at> opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5110.c |    2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index e77410e..91a7015 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
 <at>  <at>  -1252,6 +1252,8  <at>  <at>  static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;

+	arizona_init_spk(codec);
+
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");

 	priv->core.arizona->dapm = &codec->dapm;
--

-- 
1.7.10.4

Mark Brown | 17 May 2013 14:29
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[PATCH 1/3] ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces

Signed-off-by: Mark Brown <broonie <at> opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c |    7 +++++++
 sound/soc/codecs/arizona.h |    1 +
 2 files changed, 8 insertions(+)

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 90c3db1..4a42041 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
 <at>  <at>  -1198,6 +1198,13  <at>  <at>  const struct snd_soc_dai_ops arizona_dai_ops = {
 };
 EXPORT_SYMBOL_GPL(arizona_dai_ops);

+const struct snd_soc_dai_ops arizona_simple_dai_ops = {
+	.startup = arizona_startup,
+	.hw_params = arizona_hw_params_rate,
+	.set_sysclk = arizona_dai_set_sysclk,
+};
+EXPORT_SYMBOL_GPL(arizona_simple_dai_ops);
+
 int arizona_init_dai(struct arizona_priv *priv, int id)
 {
 	struct arizona_dai_priv *dai_priv = &priv->dai[id];
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index af39f10..a1a5dad 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
 <at>  <at>  -213,6 +213,7  <at>  <at>  extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			      int source, unsigned int freq, int dir);
(Continue reading)

Jordi Mallach | 17 May 2013 14:30
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[PATCH alsa-utils] Require alsa-lib 1.0.27.

Both aplay and speaker-test use the new snd_pcm_abort() function, which was
introduced in alsa-lib 1.0.27.

Signed-off-by: Jordi Mallach <jordi <at> debian.org>
---
 configure.in |    2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/configure.in b/configure.in
index aae3e66..0023e4d 100644
--- a/configure.in
+++ b/configure.in
 <at>  <at>  -32,7 +32,7  <at>  <at>  AC_PROG_MKDIR_P
 AC_PROG_LN_S
 AC_PROG_SED
 PKG_PROG_PKG_CONFIG
-AM_PATH_ALSA(1.0.24)
+AM_PATH_ALSA(1.0.27)
 if test "x$enable_alsatest" = "xyes"; then
 AC_CHECK_FUNC([snd_ctl_elem_add_enumerated],
 	      , [AC_ERROR([No user enum control support in alsa-lib])])
--

-- 
1.7.10.4

Wang, Xingchao | 17 May 2013 11:55
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module dependency between i915 and snd-hda-intel

Hi Takashi,

Seems kernel building will determine the module loading sequence, this would fix the dependency for i915
and snd-hda-intel.
If we call the i915 module API directly in snd-hda-intel side, kernel building will change the module.dep
accordingly like:
"
kernel/sound/pci/hda/snd-hda-intel.ko: kernel/drivers/gpu/drm/i915/i915.ko
kernel/drivers/gpu/dr     m/drm_kms_helper.ko kernel/drivers/gpu/drm/drm.ko
kernel/drivers/i2c/algos/i2c-algo-bit.ko kerne     l/drivers/acpi/video.ko
kernel/sound/pci/hda/snd-hda-codec.ko kernel/sound/core/snd-hwdep.ko ker    
nel/sound/core/snd-pcm.ko kernel/sound/core/snd-timer.ko kernel/sound/core/snd.ko
kernel/sound/s     oundcore.ko kernel/sound/core/snd-page-alloc.ko
"
So snd-hda-intel will wait for i915 loading. Even I added i915 into blacklist.conf, it will force load i915.ko.

So I think the only bad case is no i915 module built-in, that would cause haswell hda initialize fail, and the
patch would output error message.

thanks
--xingchao

Uwe Kleine-König | 17 May 2013 11:26
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[PATCH] ASoC: simplify registration of snd-soc-dummy device

Signed-off-by: Uwe Kleine-König <u.kleine-koenig <at> pengutronix.de>
---
 sound/soc/soc-utils.c | 13 ++++---------
 1 file changed, 4 insertions(+), 9 deletions(-)

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 4b3be6c..29b211e 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
 <at>  <at>  -159,15 +159,10  <at>  <at>  int __init snd_soc_util_init(void)
 {
 	int ret;

-	soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1);
-	if (!soc_dummy_dev)
-		return -ENOMEM;
-
-	ret = platform_device_add(soc_dummy_dev);
-	if (ret != 0) {
-		platform_device_put(soc_dummy_dev);
-		return ret;
-	}
+	soc_dummy_dev =
+		platform_device_register_simple("snd-soc-dummy", -1, NULL, 0);
+	if (IS_ERR(soc_dummy_dev))
+		return PTR_ERR(soc_dummy_dev);

 	ret = platform_driver_register(&soc_dummy_driver);
 	if (ret != 0)
--

-- 
(Continue reading)

Takashi Iwai | 17 May 2013 08:44
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Re: [PATCH] ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format

At Thu, 16 May 2013 20:26:17 +0200,
Torstein Hegge wrote:
> 
> freqshift is only set for the data endpoint and syncmaxsize is only set
> for the sync endpoint. This results in a syncmaxsize of zero used in the
> proc output feedback format calculation, which gives a feedback format
> incorrectly shown as 8.16 for UAC2 devices.
> 
> As neither the data nor the sync endpoint gives all the relevant
> content, output the two combined.
> 
> Also remove the sync_endpoint "packet size" which is always zero
> and the sync_endpoint "momentary freq" which is constant.
> 
> Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async.
> 
> Reported-by: B. Zhang <bb.zhang <at> free.fr>
> Signed-off-by: Torstein Hegge <hegge <at> resisty.net>

Thanks, applied.

Takashi

> ---
> Cc: Takashi Iwai <tiwai <at> suse.de>
> Cc: Daniel Mack <zonque <at> gmail.com>
> Cc: Clemens Ladisch <clemens <at> ladisch.de>
> Cc: Davor Herga <davor.herga <at> guest.arnes.si>
> 
>  sound/usb/proc.c | 22 +++++++++++-----------
(Continue reading)


Gmane