February 2014 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] 答复: [PATCH] ASoC: pass snd_soc_jack_gpio to jack_status_check callback
by Mark Brown 23 Feb '14

23 Feb '14

On Wed, Feb 19, 2014 at 02:20:07PM +0000, 肖翔 wrote: > The new parameter isn't used to get the internal field of snd_soc_jack_gpio. > the usage is very like work_struct, here is the demo code: > struct xxx_priv { > truct snd_soc_jack_gpio gpio; > }; > > static int xxx_ jack_status_check(struct snd_soc_jack_gpio *gpio) > { > struct xxx_priv *xxx = container_of(gpio, struct xxx_priv, gpio); > ...... > } > Without this parameter, codec driver has to save xxx_priv as a global variable. OK, this makes sense as a reason for passing the argument in however how about instead of passing in the structure adding a void * to it which gets passed as the argument instead? The user can then assign it to point to the data they want directly without it looking like they should be using the private data for the core code.

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[alsa-devel] [PATCH v3 0/3] Add TDM slot info OF parsing supports
by Xiubo Li 23 Feb '14

23 Feb '14

Change in v3: - Add .of_xlate_tdm_slot_mask(). - Add snd_soc_of_xlate_tdm_slot_mask(). - Remove TDM slot masks OF parsing... - Split the TDM infomation into two properties. Xiubo Li (3): ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports .../devicetree/bindings/sound/simple-card.txt | 5 ++ .../devicetree/bindings/sound/tdm-slot.txt | 20 ++++++++ include/sound/simple_card.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc.h | 3 ++ sound/soc/generic/simple-card.c | 18 +++++++ sound/soc/soc-core.c | 59 ++++++++++++++++++++++ 7 files changed, 109 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt -- 1.8.4

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[alsa-devel] [PATCH v6 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 23 Feb '14

23 Feb '14

Hi, This series adds master clock and regulators to the tlv320aic32x4 driver. In v6 I added error handling for the regulator setup. master clock is required for the driver now but still optional in the DT bindings. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Asoc: codecs: Remove tlv320aic32x4 from compatibles of tlv320aic3x .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 177 +++++++++++++++++++-- 3 files changed, 173 insertions(+), 17 deletions(-) -- 1.8.5.3

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[alsa-devel] [PATCH 029/102] ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL()
by Takashi Iwai 23 Feb '14

23 Feb '14

Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/adau1373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 46b1595d7c50..5765c224cd9a 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static const struct soc_enum adau1373_decimator_enum = - SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); +static SOC_ENUM_SINGLE_DECL(adau1373_decimator_enum, + 0, 0, adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); -- 1.8.5.2

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[alsa-devel] [PATCH] ASoC: pxa: Remove superfluous locking
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

The locking here was added in commit 71a295602e ("ASoC: Lock the CODEC in PXA external jack controls") to protect the DAPM changes that are made inside of ${board}_ext_control() against concurrent updates. The ASoC core was updated in commit a73fb2df01 ("ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex") to use a card wide lock rather the CODEC mutex to protect DAPM operations. We now have proper locking inside ${board}_ext_control() itself, so taking the CODEC lock can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/magician.c | 4 ---- sound/soc/pxa/poodle.c | 4 ---- sound/soc/pxa/spitz.c | 4 ---- sound/soc/pxa/tosa.c | 4 ---- 5 files changed, 20 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 9d9c8ad..b4db3b9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -103,13 +103,9 @@ static int corgi_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ corgi_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 31242be..41ab6678 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -77,13 +77,9 @@ static int magician_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ magician_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index c93e138..27c6c03 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -76,13 +76,9 @@ static int poodle_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ poodle_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 04dbb5a..a3a13c9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -113,13 +113,9 @@ static int spitz_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ spitz_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2a4b438..cead165 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -84,13 +84,9 @@ static int tosa_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ tosa_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: sta32x: Fix cache sync
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

codec->control_data contains a pointer to the regmap struct of the device, not to the device private data. Use snd_soc_codec_get_drvdata() instead. The issue was introduced in commit 29fdf4fbbe ("ASoC: sta32x: Convert to regmap"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 42c5b45..ea78c17 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; -- 1.8.0

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[alsa-devel] No sound from Headphone Jack over Lenovo Docking Station with T440p Laptop
by Nicolas Kalkhof 22 Feb '14

22 Feb '14

Hello, I have a Lenovo ThinkPad T440p laptop connected to an Ultra Docking Station. Sound works fine over Laptop Speakers and the 3,5mm Headphone Jack. However that is not the case when I connect the T440p to the Docking station Type 40A10065EU and use the Headphone Jack at the Dock. I've tried loading the alsa module with the "model=thinkpad" or "model=lenovo-dock" without success. My Distro is Gentoo and the Kernel is 3.13-rc4. alsa.conf: # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion ## alias snd-card-0 snd-interwave ## alias snd-card-1 snd-ens1371 ## OSS/Free portion ## alias sound-slot-0 snd-card-0 ## alias sound-slot-1 snd-card-1 ## # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 ## alias sound-service-1-0 snd-mixer-oss ## alias sound-service-1-3 snd-pcm-oss ## alias sound-service-1-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 #options snd-hda-intel model=thinkpad options snd-hda-intel model=thinkpad enable=1 index=0 options snd-hda-intel model=hdmi enable=1 index=-2 lspci -vv and dmesg goes here: [1]http://pastebin.ca/2510939 Any Help? References 1. http://pastebin.ca/2510939

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[alsa-devel] [PATCH 0/3] alsa-lib - improve surround 2.1 support
by David Henningsson 21 Feb '14

21 Feb '14

So, in order to support 2.1 surround on both 4-channels (e g FL FR LFE LFE, on internal subwoofer) and 6-channels (e g FL FR RL RR C LFE, on desktops with 5.1 out), I needed to make the route plugin more chmap aware. This also included allowing a new syntax for the ttable, which I think is quite nice. The new syntax also falls back to ALSAs standard chmap in case the hardware does not support it. David Henningsson (3): route: Allow chmap syntax for slave channels in ttable route: Select slave chmap based on ttable information conf: Allow 2.1 surround to use different number of channels src/conf/pcm/surround21.conf | 7 +- src/pcm/pcm_route.c | 304 ++++++++++++++++++++++++++++++++++++++----- 2 files changed, 271 insertions(+), 40 deletions(-) -- 1.9.0

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[alsa-devel] [PATCH] rawmidi: remove undefined functions.
by Takashi Sakamoto 21 Feb '14

21 Feb '14

'snd_rawmidi_transmit_reset()' and 'snd_rawmidi_receive_reset()' are declared but not defined. This state has been continue over 10 years. So let us remove them. --- include/sound/rawmidi.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index adf0885..311dafe 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -157,10 +157,8 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, /* callbacks */ -void snd_rawmidi_receive_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count); -void snd_rawmidi_transmit_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count); -- 1.8.3.2

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[alsa-devel] Enable more HP machine to enable mute led
by Kailang 21 Feb '14

21 Feb '14

Hi Takashi, HP want to enable mute led functions for more machine. I created patch as attach patch. Thanks. BR, Kailang

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