September 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] snd_pcm_delay issue - How to debug?
by Sander Jansen 18 Sep '13

18 Sep '13

Never mind. Issue not related to alsa. On Tue, Jul 16, 2013 at 6:27 PM, Sander Jansen <s.jansen(a)gmail.com> wrote: > Here the two alsa-info dumps for the hardware I'm having the problem with. > Also a log of my music player. It reports: > > The 'position' the number of frames it has written to the device (also > reported in seconds) and the delay at that time (after writing). > Also the 'timestamp' which is the reported playback progress in seconds > reported to the UI. When the second track is played (line 590), the > problems start to become noticeable eg. playback progress is a couple of > seconds out of sync. > > Any ideas? > > Thanks, > > Sander > > > > On Tue, Jul 16, 2013 at 9:28 AM, Sander Jansen <s.jansen(a)gmail.com> wrote: > >> Hi, >> >> I'm seeing an issue where snd_pcm_delay is returning bogus values >> (usually a lot larger than in actuality) in that it ultimately affects the >> progress timer in my audio player. Both are HDA based audio cards, though >> one connects to the HDMI output and while the other connects to analog out. >> >> I'm suspecting I'm seeing a driver bug since I only start noticing the >> issue since a few kernel releases ago (though not sure which one started >> it). >> >> How can i best debug this issue and find out the cause. >> >> Thanks, >> >> Sander >> > >

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Re: [alsa-devel] [PATCH v2 05/11] ASoC: davinci-mcasp: Remove interrupt property from DT bindings doc
by Mark Brown 18 Sep '13

18 Sep '13

On Wed, Sep 18, 2013 at 09:48:34AM -0500, Nishanth Menon wrote: > On 09/18/2013 09:46 AM, Mark Brown wrote: > > How about moving these to optional instead of removing them completely > > (or just leaving them alone)? That way if the driver does want to start > > using them people will have hooked them up already so no updates will be > > required. > the definitions as they stand now make no sense as the ordering of the > interrupts are undefined - example is the first index RX or TX? the > alternative will be PIO McASP driver, which might be nice, but having > a vague binding with 0 code behind is of no use for anyone right? So the sensible fix seems to be to specify the order rather than remove them then...

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[alsa-devel] [PATCH 1/2] ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component
by Sachin Kamat 18 Sep '13

18 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/spear/spdif_in.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 63acfeb..21a8c95 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); -} - -static int spdif_in_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); } static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, - .remove = spdif_in_remove, .driver = { .name = "spdif-in", .owner = THIS_MODULE, -- 1.7.9.5

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[alsa-devel] Incorrect sbits for snd-usb-audio
by Sander Jansen 17 Sep '13

17 Sep '13

Hi, I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

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[alsa-devel] [PATCH 0/4] add regmap_filelds and use it on Renesas Sound driver
by Kuninori Morimoto 17 Sep '13

17 Sep '13

Hi Mark These patches add new regmap_filelds API on kernel. ~ and use it on Renesas sound driver instead of original method. It can care about multi port register offset via regmap. 0xAAAA + 0x40 -- port 0 -- regX regY regZ 0xAAAA + 0x80 -- port 1 -- regX regY regZ This case, current API needs 2 (= port) x 3 (= regX/Y/Z) regmap_fileld, but this new API can care about all port via 3 regmap_filelds with port ID. I'm not sure that regmap_filelds is good naming or not. Please let me know if you have good naming idea. These are based on below branchs regmap/for-next + asoc/for-next Kuninori Morimoto (4): regmap: add regmap_field_update_bits() regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rsnd: use regmap instead of original register mapping method drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regmap.c | 117 +++++++++++++++++ include/linux/regmap.h | 14 +++ sound/soc/sh/rcar/core.c | 45 ------- sound/soc/sh/rcar/gen.c | 270 +++++++++++++++++++++++++--------------- 5 files changed, 307 insertions(+), 142 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/1] ASoC: fsl_ssi: Remove redundant dev_set_drvdata
by Sachin Kamat 17 Sep '13

17 Sep '13

Driver core sets the driver data to NULL on detach. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4973be7..6ac8730 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1135,7 +1135,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: fsl_spdif: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon at the end of brace is removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..c0fea02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: ak4642: Remove redundant break
by Sachin Kamat 17 Sep '13

17 Sep '13

'break' after return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/ak4642.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d03787..21c35ed 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: twl6040: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb..35059a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: blackfin: Add missing break statement to bf6xx
by Mark Brown 17 Sep '13

17 Sep '13

On Sat, Sep 14, 2013 at 02:20:37AM +0300, Valentin Ilie wrote: > SNDRV_PCM_FORMAT_S8 isn't supposed to fall through to SNDRV_PCM_FORMAT_S16_LE > > Signed-off-by: Valentin Ilie <valentin.ilie(a)gmail.com> Applied, thanks.

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