August 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa-project.org is down
by Gabriel M. Beddingfield 08 Aug '12

08 Aug '12

Howdy, Several have noticed that alsa-project.org is down. I don't know if this is a planned outage or not. -gabriel

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Re: [alsa-devel] [PATCH] ASoC: imx-ssi: Fix mono playback
by Mark Brown 07 Aug '12

07 Aug '12

On Mon, Aug 06, 2012 at 04:36:25PM -0300, Fabio Estevam wrote: > Currently mono playback is played on a faster rate. > > Adjust channels_min to 2 channels in order to play it back correctly. This doesn't actually fix mono playback so much as remove support for it...

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix 6pin mux configuration
by Peter Ujfalusi 07 Aug '12

07 Aug '12

The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad <cfadad(a)rocketmail.com> Cc: stable(a)kernel.org (3.4+) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, The omap-mcbsp FSR/CLKR mux configuration code seams to be broken since 3.4. Would it be possible to apply this patch for the 3.6 release? CC-ing stable(a)kernel.org since this part is broken in 3.4 and 3.5 kernels as well. Regards, Peter sound/soc/omap/mcbsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 1b08d98..5bda7c9 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -766,7 +766,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { -- 1.7.8.6

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[alsa-devel] [PATCH] ALSA: Initialize power-down register on AC97 codec wm9711/12/15.
by m.kiepfer＠fs-net.de 07 Aug '12

07 Aug '12

From: Martin Kiepfer <m.kiepfer(a)fs-net.de> Signed-off-by: Martin Kiepfer <m.kiepfer(a)fs-net.de> --- sound/pci/ac97/ac97_patch.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a872d0a..b8e493ac 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -697,6 +697,8 @@ static int patch_wolfson11(struct snd_ac97 * ac97) ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_MIC | AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD; + + snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0); return 0; } -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: mxs-saif: Use devm_clk_get()
by Fabio Estevam 07 Aug '12

07 Aug '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_clk_get can make the code simpler and smaller. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 20 +++++++------------- 1 files changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b303071..aa037b2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -704,7 +704,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->clk = clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -717,8 +717,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_get_resource; + return -ENODEV; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -731,7 +730,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) &saif->dma_param.chan_num); if (ret) { dev_err(&pdev->dev, "failed to get dma channel\n"); - goto failed_get_resource; + return ret; } } else { saif->dma_param.chan_num = dmares->start; @@ -742,7 +741,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } saif->dev = &pdev->dev; @@ -750,7 +749,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_resource; + return ret; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -758,7 +757,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } platform_set_drvdata(pdev, saif); @@ -766,7 +765,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_get_resource; + return ret; } ret = mxs_pcm_platform_register(&pdev->dev); @@ -779,19 +778,14 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_get_resource: - clk_put(saif->clk); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct mxs_saif *saif = platform_get_drvdata(pdev); - mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_put(saif->clk); return 0; } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: core: Set dapm->idle_bias_off for DAIs not mapped with a codec
by Peter Ujfalusi 07 Aug '12

07 Aug '12

The idle_bias_off flag is not configured for DAIs not mapped with a codec. This causes the pm counter to be increased at probe time for the CPU dai which unbalances the pm counter handling. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Mark, I'm not sure if this is the correct way to fix the problem but this patch works for me. The issue in short: Since commit: '18d7564 ASoC: when removing a CPU DAI, clean up its DAPM context' After booting up the kernel on BeagleBoard for example I see that the clocks are enabled for the McBSP CPU dai. This alone causing the first stream start to fail due to unbalanced pm_runtime counter. Later it stabilizes, but I see failures with full duplex operation (arecord | aplay). The root cause of this is not commit 18d7564, but the fact that we did not configured the dapm->idle_bias_off for the CPU dai to be 1 (we do this for the platform driver). The pm_runtime_get_sync is called from dapm_pre_sequence_async() - since we add the cpu_dai dapm context to the list. With this patch we set the dapm->idle_bias_off for DAIs not linked with a codec which means cpu_dais. If this is not acceptable we need to set this flag in the cpu_dai driver itself. Regards, Peter sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..f10f00b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3715,6 +3715,9 @@ int snd_soc_register_dai(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); @@ -3803,6 +3806,9 @@ int snd_soc_register_dais(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); -- 1.7.8.6

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Re: [alsa-devel] [PATCH] ASoC: cs42l52: Remove duplicate inclusion of slab.h header file
by Mark Brown 07 Aug '12

07 Aug '12

On Tue, Aug 07, 2012 at 12:07:27PM +0530, Sachin Kamat wrote: > slab.h header file was included twice. Applied, thanks.

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Re: [alsa-devel] adding an external dsp under soc-dsp framework
by Mark Brown 07 Aug '12

07 Aug '12

On Mon, Aug 06, 2012 at 06:11:33PM +0300, Ziv Haziz wrote: > ok thanks. This means that I need to introduce two codecs and make a dai > link between them? > It doesn't really fits the dai structure - how do you suggest to do it? No, if they're part of the same physical chip then they should be handled internally to the device. > ok - finger crossed you will publish an agreed interface soon - > currently I saw in codeaurora kernel an implementation of exposing mixer > controls, so currently this is the only option yes? That's not upstream, standard Linux has no support for Slimbus at all.

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[alsa-devel] Plumbers: Please split audio topics into separate sessions
by David Henningsson 07 Aug '12

07 Aug '12

Looking at: http://summit.linuxplumbersconf.org/lpc-2012/track/lpc2012-audio/ 45 minutes is not enough to handle two topics: Audio uConf: HD-audio cooking && QA/Testing Audio uConf: Expose Routing && DSPs in ASoC Audio uConf: Time Alignment && PA on Android Please split these three sessions into six. It is better to have a little more time than necessary (even if I believe we easily fill 45 minutes per topic), than having to skip a topic just because the other topic took more time than expected. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/2] ALSA: hda - remove quirk for Dell Vostro 1015
by David Henningsson 07 Aug '12

07 Aug '12

This computer is confirmed working with model=auto on kernel 3.2. Also, parsing fails with hda-emu with the current model. Cc: stable(a)kernel.org (3.2+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) Alsa info: see hda-emu/codecs/canonical/cx20585-dell-vostro-1015-ccert-201010-6649 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1436118..6f538eb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2972,7 +2972,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), -- 1.7.9.5

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