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

[alsa-devel] [patch] ALSA: ASoC: tas5086: signedness bug in tas5086_hw_params()
by Dan Carpenter 13 Mar '13

13 Mar '13

"val" has to be signed for the error handling to work. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 008bea4..41d03ae 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -251,7 +251,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - unsigned int val; + int val; int ret; priv->rate = params_rate(params);

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[alsa-devel] [PATCH] ASoC: fsi: remove unused irq
by Kuninori Morimoto 13 Mar '13

13 Mar '13

FSI is using devm_request_irq() from 1ddd82868cc888e008ed520465c172a6cdddd689 (ASoC: fsi: use devm_request_irq()) master->irq is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a606d0f..7a874a3 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -300,7 +300,6 @@ struct fsi_core { struct fsi_master { void __iomem *base; - int irq; struct fsi_priv fsia; struct fsi_priv fsib; struct fsi_core *core; @@ -2086,7 +2085,6 @@ static int fsi_probe(struct platform_device *pdev) } /* master setting */ - master->irq = irq; master->core = (struct fsi_core *)id_entry->driver_data; spin_lock_init(&master->lock); -- 1.7.9.5

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[alsa-devel] [PATCH 2/2] ASoC: fsl: imx-audmux: Use devm_clk_get()
by Fabio Estevam 13 Mar '13

13 Mar '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> By using devm_clk_get() we can save a call to clk_put(). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3f333e5..47f046a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -262,7 +262,7 @@ static int imx_audmux_probe(struct platform_device *pdev) return PTR_ERR(pinctrl); } - audmux_clk = clk_get(&pdev->dev, "audmux"); + audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", PTR_ERR(audmux_clk)); @@ -282,7 +282,6 @@ static int imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); - clk_put(audmux_clk); return 0; } -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: samsung: remove last traces of neo1973-gta01
by Mark Brown 13 Mar '13

13 Mar '13

On Tue, Mar 12, 2013 at 11:09:35PM +0100, Paul Bolle wrote: > The (rudimentary) support for the Openmoko Neo1973 GTA01 got removed in > commit 1ae5cbc52e7c6619a3f44b87809fd25370df31bb ("ASoC: neo1973_wm8753: > remove references to the neo1973-gta01 machine"). Remove its last traces > in the Kconfig file too. The support wasn't rudimentary, it was a pretty much complete audio driver. Anyway I've applied this with a fixed up commit message.

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Re: [alsa-devel] [PATCH] ASoC: samsung: remove last traces of neo1973-gta01
by Kukjin Kim 13 Mar '13

13 Mar '13

Paul Bolle wrote: > > The (rudimentary) support for the Openmoko Neo1973 GTA01 got removed in > commit 1ae5cbc52e7c6619a3f44b87809fd25370df31bb ("ASoC: neo1973_wm8753: > remove references to the neo1973-gta01 machine"). Remove its last traces > in the Kconfig file too. > > Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> Acked-by: Kukjin Kim <kgene.kim(a)samsung.com> - Kukjin > --- > Untested. > > sound/soc/samsung/Kconfig | 5 ++--- > 1 file changed, 2 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig > index 90e7e66..475fb0d 100644 > --- a/sound/soc/samsung/Kconfig > +++ b/sound/soc/samsung/Kconfig > @@ -35,11 +35,10 @@ config SND_SAMSUNG_I2S > tristate > > config SND_SOC_SAMSUNG_NEO1973_WM8753 > - tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)" > - depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02) > + tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" > + depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 > select SND_S3C24XX_I2S > select SND_SOC_WM8753 > - select SND_SOC_LM4857 if MACH_NEO1973_GTA01 > select SND_SOC_DFBMCS320 > help > Say Y here to enable audio support for the Openmoko Neo1973 > -- > 1.7.11.7

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Re: [alsa-devel] HDA regression on Fujitsu S7020 laptop (ALC260 codec)
by Jonathan Woithe 12 Mar '13

12 Mar '13

Hi Takashi > > On Wed, Feb 20, 2013 at 02:14:20PM +0100, Takashi Iwai wrote: > > > > Sounds good - I will test in the next day or so and report back. What's the > > > > most efficient way to get hold of this tree / branch given that I have the > > > > for-next branch of sound.git? Can I somehow leverage this, or do I need to > > > > do a completely separate checkout? Apologies if this is a basic question - > > > > I'm still coming to terms with git. > > > > > > sound-unstable git tree always contains the sound git tree master > > > branch, so everything is there basically. ... > > > > What's the next step in the process? I've reported the outcome of the tests > > of sound-unstable as per that report progress is being made. Is it a case > > of waiting for things to stabilise, or do you need me to do something to > > move things along? If so, what? > > I merged the patches to the main sound git tree for-next branch now. > > Also a few updates are merged there, too: now the hp mic jack mode > should be able to change the pin control no matter whether which > capture source is. Also, the capture source won't change the hp/mic > pinctl when the jack mode enum is created. This reduces the > automation but also reduces the confusion, too. After all, when jack > mode is present, user wants to control the pin by himself. > > Let me know if anything else is missing or broken. I've given this new revision a quick test; sorry for the delay. With this revision, when using "model=fujitsu-jwse" things seem to be working pretty well. In particular, the updates you mention greatly improve usability; the lack of automatic interaction between mixer controls makes it much easier to use and understand. I think everything is present and accounted for in this latest revision. Thanks for your work on this. The only slightly odd thing I noticed was that whenever the driver is loaded the "headphone mic jack mode" defaults to "mic bias 80pc", rather than "line out" or "headphone" which I would have expected. Obviously this can be manually or automatically overridden after module load, but in many ways it would be good if it defaulted to one of the output modes (given that the jack's primary purpose is for output). Where does this leave us now? Is the "fujitsu-jwse" a permanent thing, or will the respective changes be rolled into something more generic? Regards jonathan

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Re: [alsa-devel] [PATCH linux-next v2] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Mark Brown 12 Mar '13

12 Mar '13

On Tue, Mar 12, 2013 at 01:17:26PM -0600, Tim Gardner wrote: > > threads (I nearly deleted this unread) and avoid including > > enormous CC lists like this in the body of the mail. > I'm just going by the default output of scripts/get_maintainer.pl. Are > there better options to filter the roles ? In this case I received a > review from a supporter (Lars), and the patch is being committed by > yet another supporter (yourself). Would you have noticed this patch > had I restricted the Cc list to just maintainers and lists ? I specifically mean don't include the lists in the body of the mail - it means they either litter the git log or need to be manually deleted. If you're just using unfiltered get_maintainer output then I'd suggest that you generate the CCs in your scripts when sending rather than putting them in the changelog.

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[alsa-devel] [PATCH] ALSA: ASoC: davinci-mcasp: clean ip davinci_hw_common_param()
by Daniel Mack 12 Mar '13

12 Mar '13

As pointed of by Vaibhav, commit 2952b27e2 ("ASoC: davinci-mcasp: Add support for multichannel playback") duplicated the logic of counting the active serializers. That can be avoided by shifting the code around a bit. Also, drop two unused defines introduced by the same commit. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b386cad..8a9c0ee 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -237,8 +237,6 @@ #define RXSTATE BIT(5) #define SRMOD_MASK 3 #define SRMOD_INACTIVE 0 -#define SRMOD_TX 1 -#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -691,27 +689,6 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } for (i = 0; i < dev->num_serializer; i++) { - if (dev->serial_dir[i] == TX_MODE) - tx_ser++; - if (dev->serial_dir[i] == RX_MODE) - rx_ser++; - } - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - ser = tx_ser; - else - ser = rx_ser; - - if (ser < max_active_serializers) { - dev_warn(dev->dev, "stream has more channels (%d) than are " - "enabled in mcasp (%d)\n", channels, ser * slots); - return -EINVAL; - } - - tx_ser = 0; - rx_ser = 0; - - for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); if (dev->serial_dir[i] == TX_MODE && @@ -730,6 +707,17 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; -- 1.8.1.4

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Re: [alsa-devel] [PATCH 07/10] ASoC: phycore-ac97: Add DT support
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:08PM +0100, Markus Pargmann wrote: > Add devicetree support for this audio soc fabric driver. > > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> > --- > sound/soc/fsl/phycore-ac97.c | 148 ++++++++++++++++++++++++++++++++++++------- > 1 file changed, 125 insertions(+), 23 deletions(-) A binding document is mandatory for new bindings. > +#ifdef CONFIG_MACH_IMX27_DT > + .cpu_dai_name = "10010000.ssi", > + .platform_name = "imx-fiq-pcm-audio", > +#else This looks wrong, at least the CPU DAI name should be being looked up via the DT. > +static const struct of_device_id imx_phycore_ac97_of_dev_id[] = { > + { > + .compatible = "phytec,imx27-pca100-ac97", > + .data = (void *)MX27_PCA100, > + }, { > + .compatible = "phytec,imx27-pcm043-ac97", > + .data = (void *)MX27_PCM043 > + }, { > + /* sentinel */ > + } > +}; > +MODULE_DEVICE_TABLE(of, imx_phycore_ac97_of_dev_id); It seems like we ought to be able to describe the properties of the boards in this class rather than just enumerating the boards. What are the differences? > MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); > -MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); > +MODULE_DESCRIPTION(DRV_NAME ": PhyCORE ALSA SoC fabric driver"); Please don't make up terms like "fabric driver".

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Re: [alsa-devel] [PATCH 05/10] ASoC: pcm030 audio fabric: remove __init from probe
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:06PM +0100, Markus Pargmann wrote: > Remove probe function from the init section. > > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> Applied, thanks.

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