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

[alsa-devel] [PATCH V2] ASOC:DAPM: update DPCM runtime when mixer/mux changes
by Nenghua Cao 17 Dec '13

17 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V2 -> V1: Removes the flag and update DPCM always when mixer/mux changes, according to Takashi's and Mark's suggestion. >> Usually backward compatibility is concerned when something new breaks >> the existing ones. In your case, always updating DPCM would work, >> too, even without an extra flag; it's just suboptimal. >Right, and if we have separate versions then at some point we'll end up >having to define DPCM versions of everything which is going to get >tedious and error prone. V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. > You don't need to extend the ALSA core API for such a purpose. It's > ASoC-specific, so rather extend struct soc_mixer_control instead, if > such a new flag is really needed. Nenghua Cao (1): ASOC:DAPM: update DPCM runtime when mixer/mux changes sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 1 files changed, 24 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap()
by Lars-Peter Clausen 16 Dec '13

16 Dec '13

From: Fengguang Wu <fengguang.wu(a)intel.com> devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/adi/axi-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 0822c77..7f91a86 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -192,9 +192,9 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_i2s_regmap_config); -- 1.8.0

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[alsa-devel] Volume issues: Logitech S-150 USB Digital Speaker / "usb_audio: Warning! Unlikely big volume range"
by Linus Lüssing 16 Dec '13

16 Dec '13

Hi! I'm currently trying to use a Logitech S-150 USB Digital Speaker. Unfortunately, the volume range is way off, i.e. only 1% and 2% settings in the alsamixer are actually usable for general playback. dmesg: ----- [ 9850.768093] usb 1-1: new full-speed USB device number 2 using uhci_hcd [ 9850.948103] usb 1-1: New USB device found, idVendor=1130, idProduct=1620 [ 9850.948123] usb 1-1: New USB device strings: Mfr=0, Product=2, SerialNumber=0 [ 9850.948138] usb 1-1: Product: USB AUDIO [ 9852.353857] hidraw: raw HID events driver (C) Jiri Kosina [ 9852.397038] usb_audio: Warning! Unlikely big volume range (=17152), cval->res is probably wrong. [ 9852.397054] usb_audio: [2] FU [PCM Playback Volume] ch = 2, val = 0/17152/1<6>[ 9852.399123] usbcore: registered new interface driver snd-usb-audio [ 9852.412386] usbcore: registered new interface driver usbhid [ 9852.412400] usbhid: USB HID core driver [ 9852.562358] input: USB AUDIO as /devices/pci0000:00/0000:00:1d.0/usb1/1-1/1-1:1.2/input/input12 [ 9852.563157] hid-generic 0003:1130:1620.0001: input,hidraw0: USB HID v1.10 Device [USB AUDIO ] on usb-0000:00:1d.0-1/input2 ----- lsusb -d 1130:1620 (for lsusb -vvv, see attachment) ----- Bus 001 Device 002: ID 1130:1620 Tenx Technology, Inc. ----- The detected vendor does not seem to match, so all in all looks like a kernel bug? If you need any additional information or someone testing a kernel patch, just let me know. Cheers, Linus PS: My current workaround is to use more fine grained settings via amixer.

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[alsa-devel] [PATCH -next] ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path
by Wei Yongjun 16 Dec '13

16 Dec '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing clk_disable_unprepare() before return from tegra20_ac97_platform_probe() in the error handling case. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/tegra/tegra20_ac97.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index d8b98d7..cf5e1cf 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -393,7 +393,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); if (ret) { dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, @@ -401,7 +401,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -417,6 +417,8 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_clk_disable_unprepare: + clk_disable_unprepare(ac97->clk_ac97); err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put:

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[alsa-devel] [PATCH -next] ASoC: tegra20-ac97: fix error return code in tegra20_ac97_platform_probe()
by Wei Yongjun 16 Dec '13

16 Dec '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return -ENODEV instead of 0 when no GPIO supplied in tegra20_ac97_platform_probe(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/tegra/tegra20_ac97.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index d8b98d7..3ea9738 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -358,6 +358,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } } else { dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + ret = -ENODEV; goto err_clk_put; } @@ -365,6 +366,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) "nvidia,codec-sync-gpio", 0); if (!gpio_is_valid(ac97->sync_gpio)) { dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + ret = -ENODEV; goto err_clk_put; }

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[alsa-devel] [PATCH 0/3]: tinycompress: header and rate fixes
by Vinod Koul 16 Dec '13

16 Dec '13

This patch series update the header file to verion 0.1.2. This brings the 64bit ABI compatiblity issue fix. Users have complianed about use of SNDRV_RATE_xxx for passing the sample rates as it doesnt scale to what decoder user. So time to junk this and use raw rates as is. Vinod Koul (3): tinycompress: update the header to kernel API v0.1.2 tinycompress: cplay: use the numeric sample rate tinycompress: remove usage of SNDRV_RATE_xxx cplay.c | 2 +- include/sound/compress_offload.h | 23 ++++++++++++++--------- include/sound/compress_params.h | 9 +++++++++ include/tinycompress/tinycompress.h | 20 -------------------- utils.c | 34 ---------------------------------- 5 files changed, 24 insertions(+), 64 deletions(-) -- ~Vinod

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[alsa-devel] [PATCH] ALSA: compress: change the way sample rates are sent to kernel
by Vinod Koul 16 Dec '13

16 Dec '13

The usage of SNDRV_RATES is not effective as we can have rates like 12000 or some other ones used by decoders. This change the usage of this to use the raw Hz values to be sent to kernel Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_params.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 602dc6c..1114e38 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -324,7 +324,7 @@ union snd_codec_options { /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels - * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this + * @sample_rates: Sampling rates in Hz, use values like 48000 for this * @bit_rate: Indexed array containing supported bit rates * @num_bitrates: Number of valid values in bit_rate array * @rate_control: value is specified by SND_RATECONTROLMODE defines. -- 1.7.0.4

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[alsa-devel] [PATCH] ALSA: Add SNDRV_PCM_STATE_PAUSED case in wait_for_avail function
by JongHo Kim 16 Dec '13

16 Dec '13

When the process is sleeping at the SNDRV_PCM_STATE_PAUSED state from the wait_for_avail function, the sleep process will be woken by timeout(10 seconds). Even if the sleep process wake up by timeout, by this patch, the process will continue with sleep and wait for the other state. Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e03b46..a210467 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", -- 1.7.9.5

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[alsa-devel] While writing to speaker sometimes segmentation fault comes.
by ranjith vs 16 Dec '13

16 Dec '13

Hello, I am streaming audio from vlc,in the following formats sampling frequency 44100 stereo 16 bit.On the receiver side receiveing pcm frames and writing to speaker using alsa interface. But sometimes i'm getting segmentation fault in the snd_pcm_writei function. And one more thing is that eventhough I try to set the buffer size to 4096. It is setting it's own value.But i'm writing the buffer of length which I received from pcm_avail_upadte function. Thanks & Regards, Ranjith

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Re: [alsa-devel] [PATCH]ASOC:DAPM: update DPCM runtime when mixer/mux changes
by anish singh 16 Dec '13

16 Dec '13

<This patch is dependent on the $subject so i have used $subject instead of different one> I can see only one usage of: snd_soc_dapm_mixer_update_power and zero user of: snd_soc_dapm_mux_update_power() And after this $subject patch I think both of these functions becomes redundant. It will help in reducing the code size if we can get rid of this dead code.We can also remove code from tlv320aic3x.c and just let it use the core implementation. grep -rniI "snd_soc_dapm_mixer_update_power" sound/ sound/soc/soc-dapm.c:2149: int snd_soc_dapm_mixer_update_power sound/soc/codecs/tlv320aic3x.c:174: snd_soc_dapm_mixer_update_power grep -rniI "snd_soc_dapm_mux_update_power" sound/ no users. I would send a patch if experts agree.

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