October 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH -next] ASoC: tegra: tegra_rt5640: use devm_snd_soc_register_card()
by Qinglang Miao 02 Oct '20

02 Oct '20

Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Qinglang Miao <miaoqinglang(a)huawei.com> --- sound/soc/tegra/tegra_rt5640.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index d66d86593..c73bd23b3 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -192,7 +192,7 @@ static int tegra_rt5640_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -202,15 +202,6 @@ static int tegra_rt5640_probe(struct platform_device *pdev) return 0; } -static int tegra_rt5640_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id tegra_rt5640_of_match[] = { { .compatible = "nvidia,tegra-audio-rt5640", }, {}, @@ -223,7 +214,6 @@ static struct platform_driver tegra_rt5640_driver = { .of_match_table = tegra_rt5640_of_match, }, .probe = tegra_rt5640_probe, - .remove = tegra_rt5640_remove, }; module_platform_driver(tegra_rt5640_driver); -- 2.23.0

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[PATCH -next] ASoC: fsl: mx27vis-aic32x4: use devm_snd_soc_register_card()
by Qinglang Miao 02 Oct '20

02 Oct '20

Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Qinglang Miao <miaoqinglang(a)huawei.com> --- sound/soc/fsl/mx27vis-aic32x4.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 4ead537e0..8d3b18973 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -176,7 +176,7 @@ static int mx27vis_aic32x4_probe(struct platform_device *pdev) mx27vis_amp_muter_gpio = pdata->amp_muter_gpio; mx27vis_aic32x4.dev = &pdev->dev; - ret = snd_soc_register_card(&mx27vis_aic32x4); + ret = devm_snd_soc_register_card(&pdev->dev, &mx27vis_aic32x4); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -199,19 +199,11 @@ static int mx27vis_aic32x4_probe(struct platform_device *pdev) return ret; } -static int mx27vis_aic32x4_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&mx27vis_aic32x4); - - return 0; -} - static struct platform_driver mx27vis_aic32x4_audio_driver = { .driver = { .name = "mx27vis", }, .probe = mx27vis_aic32x4_probe, - .remove = mx27vis_aic32x4_remove, }; module_platform_driver(mx27vis_aic32x4_audio_driver); -- 2.23.0

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[PATCH] ASoC: fsl_spdif: Fix unnecessary check in fsl_spdif_probe()
by Tang Bin 02 Oct '20

02 Oct '20

The function fsl_spdif_probe() is only called with an openfirmware platform device. Therefore there is no need to check that the passed in device is NULL. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_spdif.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7858a5499..395c50167 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1221,9 +1221,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) void __iomem *regs; int irq, ret, i; - if (!np) - return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; -- 2.20.1.windows.1

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[PATCH -next] ASoC: fsl: imx-mc13783: use devm_snd_soc_register_card()
by Qinglang Miao 02 Oct '20

02 Oct '20

Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Qinglang Miao <miaoqinglang(a)huawei.com> --- sound/soc/fsl/imx-mc13783.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index dd9c1ac81..d9dca7bbc 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -96,7 +96,7 @@ static int imx_mc13783_probe(struct platform_device *pdev) imx_mc13783.dev = &pdev->dev; - ret = snd_soc_register_card(&imx_mc13783); + ret = devm_snd_soc_register_card(&pdev->dev, &imx_mc13783); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -140,19 +140,11 @@ static int imx_mc13783_probe(struct platform_device *pdev) return ret; } -static int imx_mc13783_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&imx_mc13783); - - return 0; -} - static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", }, .probe = imx_mc13783_probe, - .remove = imx_mc13783_remove }; module_platform_driver(imx_mc13783_audio_driver); -- 2.23.0

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Webcam Microphone not working
by Adam Baker 02 Oct '20

02 Oct '20

Hi, I've recently acquired a couple of different cheap webcams and on both of them the microphone is not working under Linux but works under Windows. Both cameras appear to use the same chip ID 1b3f:2002 Generalplus Technology Inc. 808 Camera which on one endpoint provides a UAC V1 audio signal. Comparing wireshark USB captures between Windows and Linux I saw that windows is using much larger URBs than Linux so as a quick hack added to endpoint.c +++ sound/usb/endpoint.c 2020-10-01 22:46:30.191648731 +0100 @@ -687,6 +687,10 @@ static int data_ep_set_params(struct snd ep->freqmax = (data_maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); } + if (maxsize < 128) + maxsize = 128; + dev_info(&ep->chip->dev->dev, "max: %d, epmax : %d, frq: %d\n", + maxsize, ep->maxpacksize, ep->freqmax); if (ep->fill_max) ep->curpacksize = ep->maxpacksize; This was enough to make the microphone work suggesting that I am at least looking in the right area. The code in endpoint.c appears to be attempting to optimise the size of the URB based upon the sample frequency, sample size and the maximum rate at which packets can be sent in order to minimise latency and allow accurate start / stop timing. Looking at the packet capture however I suspect that that calculation is failing to account for some fixed header sizes. Before I delve into understanding this code and isochronous USB better I'd like to know if anyone has quick answers to the following questions: 1) Has this already been addressed in a newer kernel, I'm currently using Debian 4.19 because that is where I first found the problem? 2) What is the preferred kernel against which patches to this subsystem should be provided, is it the latest released kernel on Linus' git tree, the latest RC or a subsystem tree? 3) The sellers of these cameras both claimed they work under Linux. Do I need to try to find if there was a time in the past when they did work and if so provide a fixes tag against the commit that broke support? Would doing so affect the chances of a fixes making it to stable kernels? 4) Is there any documentation other than code comments indicating how the URB size calculation is supposed to work that I should read? 5) Are there any specific gotchas in this driver that I should be aware of? Thanks Adam Baker P.S. I do realise that once I have a properly tested patch you will want to see it sent with git-send-email and a signed off by line.

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[PATCH] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 01 Oct '20

01 Oct '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first. If pause and resume is allowed during draining, user experience can be enhanced. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/core/compress_offload.c | 34 ++++++++++++++++++++++++++-------- 1 file changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..90b437f302a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -708,11 +708,20 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -720,11 +729,20 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } base-commit: a1b8638ba1320e6684aa98233c15255eb803fac7 -- 2.21.0

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unable to initialize sum ring buffer error
by GitHub issues - edited 01 Oct '20

01 Oct '20

alsa-project/alsa-lib issue #83 was edited from Komal1809: i am using alsa-lib-1.1.4.1 library version. Whenever trying to play audio from file using aplay getting error as alsa-lib-1.1.4.1/src/pcm/pcm_dmix.c:1177: **(snd_pcm_dmix_open) unable to initialize sum ring buffer** Note : audio is getting played at the start once system reboot Please let me know how to debug this issue Issue URL : https://github.com/alsa-project/alsa-lib/issues/83 Repository URL: https://github.com/alsa-project/alsa-lib

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unable to initialize sum ring buffer error
by GitHub issues - opened 01 Oct '20

01 Oct '20

alsa-project/alsa-lib issue #83 was opened from Komal1809: i am using alsa-lib-1.1.4.1 library version. Whenever trying to play audio from file using aplay getting error as alsa-lib-1.1.4.1/src/pcm/pcm_dmix.c:1177: **(snd_pcm_dmix_open) unable to initialize sum ring buffer** Note : audio is getting played at the start once system reboot Issue URL : https://github.com/alsa-project/alsa-lib/issues/83 Repository URL: https://github.com/alsa-project/alsa-lib

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