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[alsa-devel] alsa-lib 1.0.27 incorrect definition of __kernel_off_t?
by Kevin Groeneveld 23 Jul '13

23 Jul '13

I tried doing a cross compile of alsa-lib 1.0.27.2 today and ended up with the following error: error: conflicting types for 'off_t' Commit 9b5beef1 adds the following line to local.h: #define __kernel_off_t off_t Not only is __kernel_off_t already defined elsewhere in my header files, the above line tries to define __kernel_off_t (which I think should be 32 bit on my platform) to off_t which is 64 bit. I just deleted the line and everything seems to be working so far. Kevin

2 2

[alsa-devel] [PATCH] ARM: kirkwood: enable S/PDIF
by Jean-Francois Moine 23 Jul '13

23 Jul '13

This patch enables S/PDIF. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..ad94c50 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -159,6 +159,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | KIRKWOOD_RECCTL_I2S_EN; @@ -169,6 +170,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_20; ctl_play = KIRKWOOD_PLAYCTL_SIZE_20 | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_20 | KIRKWOOD_RECCTL_I2S_EN; @@ -177,6 +179,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | KIRKWOOD_RECCTL_I2S_EN; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

4 7

[alsa-devel] [PATCH] ARM: kirkwood: extend the kirkwood pcm/dma driver for DT usage
by Jean-Francois Moine 23 Jul '13

23 Jul '13

The kirkwood pcm/dma driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Documentation/devicetree/bindings/sound/kirkwood-dma.txt | 11 +++++++++++ sound/soc/kirkwood/kirkwood-dma.c | 11 +++++++++++ 2 files changed, 22 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/kirkwood-dma.txt b/Documentation/devicetree/bindings/sound/kirkwood-dma.txt new file mode 100644 index 0000000..9dd2989 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-dma.txt @@ -0,0 +1,11 @@ +* mvebu (Kirkwood, Dove, Armada 370) PCM/DMA controller + +Required properties: + +- compatible: "marvell,mvebu-pcm-audio" + +Example: + +pcm: pcm { + compatible = "marvell,mvebu-pcm-audio"; +}; diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index a9f1453..b91045a 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -19,6 +19,8 @@ #include <linux/dma-mapping.h> #include <linux/mbus.h> #include <sound/soc.h> +#include <linux/of.h> + #include "kirkwood.h" #define KIRKWOOD_RATES \ @@ -383,10 +385,19 @@ static int kirkwood_soc_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_pcm_of_match[] = { + { .compatible = "marvell,mvebu-pcm-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_pcm_of_match); +#endif + static struct platform_driver kirkwood_pcm_driver = { .driver = { .name = "kirkwood-pcm-audio", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_pcm_of_match), }, .probe = kirkwood_soc_platform_probe, -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 3

[alsa-devel] [PATCH] alsa-utils: fix wrong alias for signal.h
by John Spencer 23 Jul '13

23 Jul '13

in POSIX, there is no <sys/signal.h>. it's merely a legacy alias used by glibc, and the header does nothing else than including <signal.h>. so let's do the right thing and use the right name, which works everywhere. Signed-Off-By: John Spencer <maillist-alsa(a)barfooze.de>

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: hiface: return correct XRUN indication
by Eldad Zack 23 Jul '13

23 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2..c21a3df 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- 1.8.1.5

3 3

[alsa-devel] [PATCH] ASoC: samsung: Enable COMPILE_TEST builds
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> Platform using OF and some others don't have any dependencies on arch/arm so can be built on any platform. Enable that for better compile coverage when COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/Kconfig | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9855dfc..df28dc8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG + depends on PLAT_SAMSUNG || COMPILE_TEST select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help @@ -150,7 +150,7 @@ config SND_SOC_SMARTQ config SND_SOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) + depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA || COMPILE_TEST) depends on I2C=y && GENERIC_HARDIRQS select SND_SAMSUNG_I2S select MFD_WM8994 @@ -186,7 +186,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -195,13 +195,13 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM8962 config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -211,14 +211,14 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: mxs: Depends on COMMON_CLK
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> The SAIF driver is a clock provider so specifically needs the common clock implementedation. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 7daf860..219235c 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

2 1

Re: [alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 22 Jul '13

22 Jul '13

On Sat, 2013-07-06 at 16:22 +0600, Alexey Galakhov wrote: > Hi, > > > The DMA engine emits a lot of messages complaining of (I think) a read > > fault: > > > dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 > > dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 > > I'm debugging the same error right now. FTC-40000 means "Read error", > i.e. invalid address of the DMA source block. > > Regards, > Alex In my case it turned out the TZPC hadn't been configured properly by the bootloader. I had to modify to the bootloader to make sure the I2S was non secured (it's secure by default). Hope this helps, Mike

2 1

[alsa-devel] [patch] ASoC: core: double free in snd_soc_add_platform()
by Dan Carpenter 22 Jul '13

22 Jul '13

There are three callers for this function, and none of them want it to free platform for them. It leads to a double free. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070c..d82ee38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3908,10 +3908,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) { - kfree(platform); + if (platform->name == NULL) return -ENOMEM; - } platform->dev = dev; platform->driver = platform_drv;

2 1

[alsa-devel] [PATCH] ASoC: mxs: Remove unneeded mach-types.h inclusions
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54511c5..b56b8a0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -31,7 +31,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> #include "mxs-saif.h" diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d7..b2e372d 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -25,7 +25,6 @@ #include <sound/soc.h> #include <sound/jack.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include "../codecs/sgtl5000.h" #include "mxs-saif.h" -- 1.8.3.2

2 1

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Daniel Mack 22 Jul '13

22 Jul '13

Hi, On 20.07.2013 19:25, Jean-Francois Moine wrote: > This generic simple card driver uses DT values to instanciate an audio > system in which the real work is done by the subdrivers (audio > controller, audio codec and pcm/dma controller). > > Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> > --- > diff --git a/sound/soc/generic/simple-dt-card.c b/sound/soc/generic/simple-dt-card.c > new file mode 100644 There is a simple-card driver in the tree already, and there were several attempts to add DT bindings for it in the past - have you seen that? Search for "ASoC: add simple-card DT support" in the archives ... Daniel

3 3

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Check the return value from clk_prepare_enable()
by Fabio Estevam 22 Jul '13

22 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e260f1f..1a5da1e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -73,8 +73,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); @@ -224,14 +227,19 @@ EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr) { + int ret; + if (audmux_type != IMX31_AUDMUX) return -EINVAL; if (!audmux_base) return -ENOSYS; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Support SND_SOC_DAIFMT_CBM_CFS for omap3/4
by Michael Trimarchi 22 Jul '13

22 Jul '13

Add SND_SOC_DAIFMT_CBM_CFS support for omap3/omap4. The patch was tested on a pandaboard-es board connected to the pcm1792a codec. mcbspx_fsx must configured as output and mcbspx_clkx must be configured as input. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..6c19bba 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -433,6 +433,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Sample rate generator drives the FS */ regs->srgr2 |= FSGM; break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP slave. FS clock as output */ + regs->srgr2 |= FSGM; + regs->pcr0 |= FSXM; + break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ break; -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Mark Brown 22 Jul '13

22 Jul '13

On Sun, Jul 21, 2013 at 10:34:09AM +0800, Richard Zhao wrote: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct Applied, thanks.

1 0

[alsa-devel] [PATCH 1/3] ASoC: fsl: fsl_ssi: Add MODULE_ALIAS
by Fabio Estevam 22 Jul '13

22 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add MODULE_ALIAS, so that auto module loading can work. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9974a4..429fbfd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,6 +900,7 @@ static struct platform_driver fsl_ssi_driver = { module_platform_driver(fsl_ssi_driver); +MODULE_ALIAS("platform:fsl-ssi-dai"); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL v2"); -- 1.8.1.2

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Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Mark Brown 22 Jul '13

22 Jul '13

On Sat, Jul 20, 2013 at 07:25:54PM +0200, Jean-Francois Moine wrote: > Documentation/devicetree/bindings/sound/simple-dt-card.txt | 18 ++++ > sound/soc/generic/Kconfig | 7 ++ > sound/soc/generic/Makefile | 2 + > sound/soc/generic/simple-dt-card.c | 137 ++++++++++++++++++++++++ > 4 files changed, 163 insertions(+) The obvious question here is why is this a separate driver to the existing simple card driver rather than DT bindings for that? I don't see that mentioned here.

1 0

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Lucas Stach 21 Jul '13

21 Jul '13

Am Sonntag, den 21.07.2013, 10:34 +0800 schrieb Richard Zhao: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. > > Signed-off-by: Richard Zhao <rizhao(a)nvidia.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: <stable(a)vger.kernel.org> # 3.10 Acked-by: Lucas Stach <dev(a)lynxeye.de> > --- > sound/soc/tegra/tegra20_ac97.c | 6 +++--- > sound/soc/tegra/tegra20_spdif.c | 4 ++-- > 2 files changed, 5 insertions(+), 5 deletions(-) > > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > index e58233f..6c48662 100644 > --- a/sound/soc/tegra/tegra20_ac97.c > +++ b/sound/soc/tegra/tegra20_ac97.c > @@ -389,9 +389,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) > ac97->capture_dma_data.slave_id = of_dma[1]; > > ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; > - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - ac97->capture_dma_data.maxburst = 4; > - ac97->capture_dma_data.slave_id = of_dma[0]; > + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + ac97->playback_dma_data.maxburst = 4; > + ac97->playback_dma_data.slave_id = of_dma[1]; > > ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); > if (ret) > diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c > index 5eaa12c..551b3c9 100644 > --- a/sound/soc/tegra/tegra20_spdif.c > +++ b/sound/soc/tegra/tegra20_spdif.c > @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) > } > > spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; > - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - spdif->capture_dma_data.maxburst = 4; > + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + spdif->playback_dma_data.maxburst = 4; > spdif->playback_dma_data.slave_id = dmareq->start; > > pm_runtime_enable(&pdev->dev);

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Re: [alsa-devel] [PATCH] sound: replace strict_strto*() with kstrto*()
by Takashi Iwai 21 Jul '13

21 Jul '13

At Fri, 19 Jul 2013 16:24:59 +0900, Jingoo Han wrote: > > The usage of strict_strto*() is not preferred, because > strict_strto*() is obsolete. Thus, kstrto*() should be > used. > > Signed-off-by: Jingoo Han <jg1.han(a)samsung.com> Thanks, applied. Takashi > --- > sound/drivers/dummy.c | 2 +- > sound/pci/hda/hda_hwdep.c | 6 +++--- > sound/soc/codecs/wm8962.c | 2 +- > sound/soc/omap/mcbsp.c | 2 +- > sound/soc/soc-core.c | 8 +++++--- > 5 files changed, 11 insertions(+), 9 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index 11048cc..915b4d7 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry, > if (i >= ARRAY_SIZE(fields)) > continue; > snd_info_get_str(item, ptr, sizeof(item)); > - if (strict_strtoull(item, 0, &val)) > + if (kstrtoull(item, 0, &val)) > continue; > if (fields[i].size == sizeof(int)) > *get_dummy_int_ptr(dummy, fields[i].offset) = val; > diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c > index ce67608..fe0bda1 100644 > --- a/sound/pci/hda/hda_hwdep.c > +++ b/sound/pci/hda/hda_hwdep.c > @@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \ > struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ > struct hda_codec *codec = hwdep->private_data; \ > unsigned long val; \ > - int err = strict_strtoul(buf, 0, &val); \ > + int err = kstrtoul(buf, 0, &val); \ > if (err < 0) \ > return err; \ > codec->type = val; \ > @@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) > p = snd_hda_get_hint(codec, key); > if (!p) > ret = -ENOENT; > - else if (strict_strtoul(p, 0, &val)) > + else if (kstrtoul(p, 0, &val)) > ret = -EINVAL; > else { > *valp = val; > @@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ > struct hda_codec **codecp) \ > { \ > unsigned long val; \ > - if (!strict_strtoul(buf, 0, &val)) \ > + if (!kstrtoul(buf, 0, &val)) \ > (*codecp)->name = val; \ > } > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > index e2de9ec..e37c06f 100644 > --- a/sound/soc/codecs/wm8962.c > +++ b/sound/soc/codecs/wm8962.c > @@ -3175,7 +3175,7 @@ static ssize_t wm8962_beep_set(struct device *dev, > long int time; > int ret; > > - ret = strict_strtol(buf, 10, &time); > + ret = kstrtol(buf, 10, &time); > if (ret != 0) > return ret; > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 361e4c0..83433fd 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \ > unsigned long val; \ > int status; \ > \ > - status = strict_strtoul(buf, 0, &val); \ > + status = kstrtoul(buf, 0, &val); \ > if (status) \ > return status; \ > \ > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index 0ec070c..88daa64 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -192,7 +192,7 @@ static ssize_t pmdown_time_set(struct device *dev, > struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); > int ret; > > - ret = strict_strtol(buf, 10, &rtd->pmdown_time); > + ret = kstrtol(buf, 10, &rtd->pmdown_time); > if (ret) > return ret; > > @@ -237,6 +237,7 @@ static ssize_t codec_reg_write_file(struct file *file, > char *start = buf; > unsigned long reg, value; > struct snd_soc_codec *codec = file->private_data; > + int ret; > > buf_size = min(count, (sizeof(buf)-1)); > if (copy_from_user(buf, user_buf, buf_size)) > @@ -248,8 +249,9 @@ static ssize_t codec_reg_write_file(struct file *file, > reg = simple_strtoul(start, &start, 16); > while (*start == ' ') > start++; > - if (strict_strtoul(start, 16, &value)) > - return -EINVAL; > + ret = kstrtoul(start, 16, &value); > + if (ret) > + return ret; > > /* Userspace has been fiddling around behind the kernel's back */ > add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE); > -- > 1.7.10.4 > >

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: 6fire: return correct XRUN indication
by Eldad Zack 21 Jul '13

21 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index c5b9cac..60ac8fb 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add snd_hda_jack_detect_state() helper function
by Takashi Iwai 21 Jul '13

21 Jul '13

snd_hda_jack_detect() function returns a boolean value for a jack plugged in or not, but it also returns always true when the corresponding pin is phantom (i.e. fixed). This is OK in most cases, but it makes the generic parser misbehaving about the auto-mute or auto-mic switching, e.g. when one of headphone pins is a fixed. Namely, the driver decides whether to mute the speaker or not, just depending on the headphone plug state: if one of the headphone jacks is seen as active, then the speaker is muted. Thus this will result always in the muted speaker output. So, the problem is the function returns a boolean, after all, although we need to think of "phantom" jack. Now a new function, snd_hda_jack_detect_state() is introduced to return these tristates. The generic parser uses this function for checking the headphone or mic jack states. Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for keeping compatibility in other driver codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 +++++--- sound/pci/hda/hda_jack.c | 18 ++++++++++++------ sound/pci/hda/hda_jack.h | 13 ++++++++++++- 3 files changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..f5c2d1f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3724,7 +3724,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, /* check each pin in the given array; returns true if any of them is plugged */ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - int i, present = 0; + int i; + bool present = false; for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; @@ -3733,7 +3734,8 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* don't detect pins retasked as inputs */ if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN) continue; - present |= snd_hda_jack_detect(codec, nid); + if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT) + present = true; } return present; } @@ -3887,7 +3889,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; - if (snd_hda_jack_detect(codec, pin)) { + if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..dc93761 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_pin_sense); /** - * snd_hda_jack_detect - query pin Presence Detect status + * snd_hda_jack_detect_state - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense * - * Query and return the pin's Presence Detect status. + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return get_jack_plug_state(sense); + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack && jack->phantom_jack) + return HDA_JACK_PHANTOM; + else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + return HDA_JACK_PRESENT; + else + return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd..379420c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +/* the jack state returned from snd_hda_jack_detect_state() */ +enum { + HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, +}; + +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); + +static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; +} bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); -- 1.8.3.1

2 2

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 08:34 PM, Richard Zhao wrote: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. Acked-by: Stephen Warren <swarren(a)nvidia.com> Hmm, that's what I get for only testing I2S and not SPDIF or AC'97 (not that I can test the latter).

1 0

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 11:25 AM, Jean-Francois Moine wrote: > This generic simple card driver uses DT values to instanciate an audio > system in which the real work is done by the subdrivers (audio > controller, audio codec and pcm/dma controller). New DT bindings should be sent to devicetree(a)vger.kernel.org for review. Note that's a branch new list (it moved from a different server), so it might be best to wait a few days for people to subscribe before sending mail to it.

1 0

[alsa-devel] [PATCH RFC 00/10] Refactor set_format()
by Eldad Zack 20 Jul '13

20 Jul '13

Hi, This is a low-priority series (applies against current for-next). I had the idea of refactoring set_format to make it easier to read (I actually indended to look at the implicit feedback setup race, but stumbled on this again). This series is not intended to be applied as-is (it has some whitespace issues and I wrote the commit messages quickly), but to get some feedback - if you think it's a good idea in general, and also specific feedback to individual patches. Assuming it makes sense, I'll fix the whitespace issues, commit messages and anything else before resubmitting. Cheers, Eldad Eldad Zack (10): ALSA: usb-audio: remove disabled debug code in set_format ALSA: usb-audio: remove assignment from if condition ALSA: usb-audio: separate sync endpoint setting from set_format ALSA: usb-audio: move implicit fb quirks to separate function ALSA: usb-audio: reverse condition logic in set_sync_endpoint ALSA: usb-audio: do not initialize and check implicit_fb ALSA: usb-audio: remove is_playback from implicit feedback quirks ALSA: usb-audio: remove implicit_fb from quirk ALSA: usb-audio: clarify implicit fb quirk for Roland ALSA: usb-audio: check alts is not NULL before adding ep sound/usb/pcm.c | 249 ++++++++++++++++++++++++++++++++------------------------ 1 file changed, 144 insertions(+), 105 deletions(-) -- 1.8.1.5

2 12

Re: [alsa-devel] [ecasound] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 20 Jul '13

20 Jul '13

On 20.07.2013 11:57, Julien Claassen wrote: > Hello Knut! > No idea yet, who's to blame. but have you tried repeating the experiment > with another tool. You can use audacity, so you can use the GUI. Audacity is even more broken. That´s true for the version from opensuse (2.03), and it´s true for the current development code, If I mark that 1s of audio described in the original post, a lot of audio is missing at the beginning and a bit of audio is missing at end. In total only about 460 msec of audio is captured, 540 msec is lost. Audacity currently does not playback correctly with non-memory-mapped alsa devices. But that´s another story ;-( > There are other recording applications. Or you might even try with the alsa utilities > aplay and arecord. this will require two console or terminals. If I put arecord -f cd -D hw:0,0 -d 2 testarec.wav & aplay testin.wav -D hw:0,0 into a script I get a perfect recording: 47 msec silence, exactly 1s of audio data, silence up to the end of the file. Well, that result seems to indicate that I found an ecasound bug. Unless it´s some alsa code unused by aplay ;-) > You might also try the same experiment with JACK, if you're not averse to it. In the best case that could only mask the bug. > I did digitise a few tapes and prelistened to the volume and audio quality. > So it's not a problem in general. > Half a second silence at the end of the audio masks the bug. Because of that I agree that it´s not a general problem, most people will never notice it. But it´s still a bug. cu, Knut

1 0

[alsa-devel] [PATCH] ASoC: twl6040: Use power efficient workqueue
by Mark Brown 20 Jul '13

20 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621dd..caf8784 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.8.3.2

3 2

[alsa-devel] [PATCH v2] compress: fix the typo in compress_get_alsa_rate()
by Vinod Koul 20 Jul '13

20 Jul '13

Reported-by: Ravi <ralama(a)codeaurora.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- utils.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/utils.c b/utils.c index fa37f23..a835bef 100644 --- a/utils.c +++ b/utils.c @@ -72,7 +72,7 @@ unsigned int compress_get_alsa_rate(unsigned int rate) return SNDRV_PCM_RATE_11025; case 16000: return SNDRV_PCM_RATE_16000; - case 220500: + case 22050: return SNDRV_PCM_RATE_22050; case 32000: return SNDRV_PCM_RATE_32000; -- 1.7.0.4

1 0

[alsa-devel] [PATCH] compress: fix the typo is get rates
by Vinod Koul 20 Jul '13

20 Jul '13

Reported-by: Ravi <ralama(a)codeaurora.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- utils.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/utils.c b/utils.c index fa37f23..a835bef 100644 --- a/utils.c +++ b/utils.c @@ -72,7 +72,7 @@ unsigned int compress_get_alsa_rate(unsigned int rate) return SNDRV_PCM_RATE_11025; case 16000: return SNDRV_PCM_RATE_16000; - case 220500: + case 22050: return SNDRV_PCM_RATE_22050; case 32000: return SNDRV_PCM_RATE_32000; -- 1.7.0.4

1 0

Re: [alsa-devel] [PATCH] ASoC: ep93xx: fix build of ep93xx-ac97.c
by Mark Brown 19 Jul '13

19 Jul '13

On Fri, Jul 19, 2013 at 09:53:25AM -0700, H Hartley Sweeten wrote: > Fix the build of this driver. It was broken by: > > Commit 453807f3006757a5661c4000262d7d9284b5214c > ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: atmel: add wm8904 based audio machine driver
by Bo Shen 19 Jul '13

19 Jul '13

Add wm8904 based audio machine driver for Atmel EK board The following ek board based on it - at91sam9n12ek - sama5d3xek (d31, d33, d34, d35) Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- This patch is only for Atmel SoC with wm8904 codec machine driver only, temp no change for dts file, as there are many changes to dts file in mailing list for atmel SoC now, so hold on for dts change, after that will send the dts file patches Change in v2: - Update the binding doc, add widgets describe of codec and board - set sysclk after set pll - using bias level trasition to decide enable/disable clock - return real error code instead of return -1 Todo: - After the at91 clock move to common clock framework, then rework the clock in this driver --- .../devicetree/bindings/sound/atmel-wm8904.txt | 55 +++++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_wm8904.c | 254 ++++++++++++++++++++ 4 files changed, 321 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 sound/soc/atmel/atmel_wm8904.c diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt new file mode 100644 index 0000000..8bbe50c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,55 @@ +Atmel ASoC driver with wm8904 audio codec complex + +Required properties: + - compatible: "atmel,asoc-wm8904" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the WM8904's pins, and the jacks on the board: + + WM8904 pins: + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + + Board connectors: + + * Headphone Jack + * Line In Jack + * Mic + + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8904 audio codec + +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; +}; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1c0b185..986323b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -33,6 +33,16 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967cc..922d4da 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,8 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 0000000..7222380 --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,254 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen <voice.shen(a)atmel.com> + * + * GPLv2 or later + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/pinctrl/consumer.h> + +#include <sound/soc.h> + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + /* + * As here wm8904 use FLL output as its system clock + * so calling set_sysclk won't care freq parameter + * then we pass 0 + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_hw_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + switch (level) { + case SND_SOC_BIAS_PREPARE: + clk_prepare_enable(mclk); + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(mclk); + break; + default: + break; + } + } + + return 0; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) { + dev_err(&pdev->dev, "only device tree supported\n"); + return -EINVAL; + } + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + return ret; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(id); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen <voice.shen(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk()
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. Quoting Mark Brown [1]: "The issue here is triggered by the machine switching from the FLL to direct MCLK usage where the MCLK isn't generating a useful ratio. I suspect we should just kill the configure_bclk() in set_sysclk(), that one isn't safe as we can't reconfigure a live SYSCLK and it's probably the one that generates your warnings." Confirmed that the "Unsupported sysclk ratio" error message comes from wm8962_set_dai_sysclk(), so get rid of wm8962_configure_bclk() inside this function. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064241.html Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Remove wm8962_configure_bclk() from wm8962_set_dai_sysclk sound/soc/codecs/wm8962.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ec..8b8905d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2621,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/13] Remove static quirks for HD-audio AD codecs
by Takashi Iwai 19 Jul '13

19 Jul '13

OK, it's time to clean up the craps in the old room. Let's start from AD codecs, which I could test a few of them, and worked well with the generic parser. [PATCH 01/13] ALSA: hda - Add GPIO control to AD1884 HP fixup [PATCH 02/13] ALSA: hda - Add fixup for Lenovo Thinkpad with AD1984 [PATCH 03/13] ALSA: hda - Add fixup for HP TouchSmart with AD1984A [PATCH 04/13] ALSA: hda - Remove static quirks for AD1882 [PATCH 05/13] ALSA: hda - Remove static quirks for AD1884/1984 & [PATCH 06/13] ALSA: hda - Remove static quirks for AD1981 and AD1983 [PATCH 07/13] ALSA: hda - Remove static quirks for AD1988 codecs [PATCH 08/13] ALSA: hda - Convert the static quirk for Samsung Q1 [PATCH 09/13] ALSA: hda - Convert static quirks for AD1986A Samsung [PATCH 10/13] ALSA: hda - Drop static quirks for other AD1986A [PATCH 11/13] ALSA: hda - Drop static quirk for Toshiba Satellite [PATCH 12/13] ALSA: hda - Drop a few other static quirks for AD1986A [PATCH 13/13] ALSA: hda - Remove static quirks for AD1986A codec Takashi

1 13

[alsa-devel] [RFC][PATCH 0/4] firewire-lib: add payload callback function for each driver
by Takashi Sakamoto 19 Jul '13

19 Jul '13

Hi Clemens, (CCed Rober von Chamier and Damien Zammit, they work for Dice and 003Rack driver) Would you please review my patches and comment? Currently snd-firewire-lib handles the contents of data block for AMDTP packet in itself. But to implement drivers for Fireworks(Echo Audio)/BeBoB(BridgeCo), there is an inconvinience. Fireworks don't use AM824 label for PCM data. BeBoB needs to remap PCM and MIDI channels depending on devices. And Dice(TC Applied Technologies) - I think you understand this than me - needs the mode called as "dual wire". In this mode, the sampling rate is a half of requested one and each data block includes the data as twice as usual sampling rate. The each data block handle two PCM frames, usually it's one. Your snd-firewire-lib in alsa-kprivate.git repository manages to solve these issue. But it has restriction about the PCM format (S32 only) and for me it seems to be hard to maintain. Additionally to implement full duplex with synchronization, I need to restart streams for MIDI. Then current amdtp_out_stream_set_pcm_format() is not convinient. I want to add condition for PCM format into each drivers. Furthermore, 003Rack(Digidesign) - I don't work for this - has more issue. The differences related to payload are: - the value of syt field in AMDTP packet is always zero - The data in each channel is encoded by its own way The driver needs to refer to and modify the contents of payload. To solve these device-dependent issues, I want to move the processing of contents in payload from firewire-lib to each devices. This series of patches is for this purpose. sakamocchi (4): firewire-lib: add helper function to set data block size firewire-lib: add callback function for payload firewire-lib: remove unused members and functions firewire-speakers: add functions to handle payload amdtp.c | 144 ++++++------------------------------------------------------ amdtp.h | 45 +++++++------------ speakers.c | 94 ++++++++++++++++++++++++++++++++++++--- 3 files changed, 117 insertions(+), 166 deletions(-) -- 1.7.10.4

2 6

[alsa-devel] [PATCH] ASoC: wm8962: Provide a .hw_free function
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. In order to avoid such error message, let's add a .hw_free function which will zero out wm8962->lrclk and wm8962->bclk, which according to the code below: if (!wm8962->bclk || !wm8962->lrclk) { dev_dbg(codec->dev, "No audio clocks configured\n"); return; } ,will cause the wm8962_configure_bclk() function not to search a valid sysclk ratio after playback has finished. Suggested-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6a5066c..01033c2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,6 +2914,18 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) WM8962_DAC_MUTE, val); } +static int wm8962_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962->lrclk = 0; + wm8962->bclk = 0; + + return 0; +} + #define WM8962_RATES SNDRV_PCM_RATE_8000_96000 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2921,6 +2933,7 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) static const struct snd_soc_dai_ops wm8962_dai_ops = { .hw_params = wm8962_hw_params, + .hw_free = wm8962_hw_free, .set_sysclk = wm8962_set_dai_sysclk, .set_fmt = wm8962_set_dai_fmt, .digital_mute = wm8962_mute, -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Provide register I/O functions by default
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> Use the ARM version by default as that's the more generally portable one, it doesn't matter if they work well on random platforms when the goal is only build coverage. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9974a4..e12a997 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -36,7 +36,7 @@ #define read_ssi(addr) in_be32(addr) #define write_ssi(val, addr) out_be32(addr, val) #define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) -#elif defined ARM +#else #define read_ssi(addr) readl(addr) #define write_ssi(val, addr) writel(val, addr) /* -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: atmel: add wm8904 based audio machine driver
by Bo Shen 19 Jul '13

19 Jul '13

add wm8904 based audio machine driver the following ek board based on it - at91sam9n12ek - sama5d3xek (d31, d33, d34, d35) Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- This patch is only for Atmel SoC with wm8904 codec machine driver only, temp no change for dts file, as there are many changes to dts file in mailing list for atmel SoC now, so hold on for dts change, after that will send the dts file patches --- .../devicetree/bindings/sound/atmel-wm8904.txt | 30 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_wm8904.c | 263 ++++++++++++++++++++ 4 files changed, 305 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 sound/soc/atmel/atmel_wm8904.c diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt new file mode 100644 index 0000000..5990271 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,30 @@ +Atmel ASoC driver with wm8904 audio codec complex + +Required properties: + - compatible: "atmel,asoc-wm8904" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8904 audio codec +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; +}; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1c0b185..986323b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -33,6 +33,16 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967cc..922d4da 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,8 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 0000000..e309607 --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,263 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen <voice.shen(a)atmel.com> + * + * GPLv2 or later + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/pinctrl/consumer.h> + +#include <sound/soc.h> + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static int atmel_asoc_wm8904_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) { + ret = clk_prepare_enable(mclk); + if (ret == 0) + mclk_on = 1; + } + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + if (mclk_on) + clk_disable_unprepare(mclk); + mclk_on = 0; + break; + } + + return ret; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = atmel_asoc_wm8904_init, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) + return -1; + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + goto err_set_audio; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC 0 for audio\n"); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(0); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen <voice.shen(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: wm8962: No need to call wm8962_configure_bclk() multiple times
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. There is no need to re-calculate the bit clock after the playback has finished, so call wm8962_configure_bclk only once inside wm8962_hw_params(). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8962.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6a5066c..1bf79dc 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2497,8 +2497,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*50k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x80); - - wm8962_configure_bclk(codec); break; case SND_SOC_BIAS_STANDBY: @@ -2621,8 +2619,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- 1.8.1.2

3 2

[alsa-devel] [PATCH] hda-emu: Allow Realtek codecs to access node 0x51
by David Henningsson 19 Jul '13

19 Jul '13

There is sometimes a secret node 0x51 on Realtek codecs, controlling an equally secret DSP. At least make sure the automated test suite doesn't fail when trying to access this secret node. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 6 ++++++ hda-spec.c | 13 +++++++++++++ include/hda-types.h | 1 + 3 files changed, 20 insertions(+) diff --git a/hda-int.c b/hda-int.c index fbd3cee..ef90fe8 100644 --- a/hda-int.c +++ b/hda-int.c @@ -1125,6 +1125,12 @@ int hda_cmd(struct xhda_codec *codec, unsigned int cmd) unsigned int nid = (cmd >> 20) & 0x7f; unsigned int verb = (cmd >> 8) & 0xfff; + if (codec->extended_cmd) { + int r = codec->extended_cmd(codec, cmd); + if (r != -ENXIO) + return r; + } + tbl = find_matching_verb(codec, verb); if (!tbl) return -ENXIO; diff --git a/hda-spec.c b/hda-spec.c index 8f95914..e4bb547 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -354,6 +354,16 @@ static struct verb_ext_list extensions[] = { { } }; +static int realtek_dsp_cmd(struct xhda_codec *codec, unsigned int cmd) +{ + unsigned int nid = (cmd >> 20) & 0x7f; + if (nid != 0x51) + return -ENXIO; + /* There might be a secret DSP connected to node 0x51 */ + codec->rc = 0; + return 0; +} + /* * fixups */ @@ -466,4 +476,7 @@ void add_codec_extensions(struct xhda_codec *codec) else if (!strcmp(codec->parsed_name, "Realtek ALC661")) hda_set_proc_coef(codec, 0x20, 0x00, 0x8020); + if (!strncmp(codec->parsed_name, "Realtek", strlen("Realtek"))) + codec->extended_cmd = realtek_dsp_cmd; + } diff --git a/include/hda-types.h b/include/hda-types.h index fac5084..fe6b31f 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -114,6 +114,7 @@ struct xhda_codec { unsigned int rc; struct xhda_verb_table *extended_verbs; struct xhda_verb_table *extended_parameters; + int (*extended_cmd) (struct xhda_codec *codec, unsigned int cmd); struct xhda_value_cache *value_cache; unsigned short pci_vendor, pci_device; unsigned short pci_subvendor, pci_subdevice; -- 1.7.9.5

2 1

[alsa-devel] [RFC] ASoC: wm892: Set an initial non-mute volume for the headphones
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> By default the headphone left/right registers have a volume of zero. Instead of having to change the volume via alsamixer in order to be able to hear something, let's set an initial value of -20dB, so that audio playback can be heard by default. Tested on a mx6qsabresd board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Mark, I posted it as RFC as I don't know if this is a valid approach. Suggestions are welcome. Thanks sound/soc/codecs/wm8962.c | 6 ++++++ sound/soc/codecs/wm8962.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ec..6a5066c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3448,6 +3448,12 @@ static int wm8962_probe(struct snd_soc_codec *codec) WM8962_MICBIAS_LVL, pdata->mic_cfg); + /* set an initial volume for playback of -20 dB */ + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUTL_VOL_MASK, WM8962_HPOUTx_MINUS_20_DB); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUTR_VOL_MASK, WM8962_HPOUTx_MINUS_20_DB); + /* Latch volume update bits */ snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, WM8962_IN_VU, WM8962_IN_VU); diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d52..5dbe972 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -27,6 +27,8 @@ #define WM8962_FLL_OSC 3 #define WM8962_FLL_INT 4 +#define WM8962_HPOUTx_MINUS_20_DB 0x65 + /* * Register values. */ -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: jack: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..7aa26b5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8350: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8350.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0e8b3aa..af1318d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.8.3.2

2 4

[alsa-devel] [PATCH] ASoC: sta32x: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sta32x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe..06edb39 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work) } if (!sta32x->shutdown) - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } static void sta32x_watchdog_start(struct sta32x_priv *sta32x) { if (sta32x->pdata->needs_esd_watchdog) { sta32x->shutdown = 0; - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: max98090: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98090.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ad5313f..0569a4c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2084,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data) pm_wakeup_event(codec->dev, 100); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); } if (active & M98090_DRCACT_MASK) @@ -2132,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec, snd_soc_jack_report(max98090->jack, 0, SND_JACK_HEADSET | SND_JACK_BTN_0); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); return 0; } -- 1.8.3.2

2 1

[alsa-devel] cs42l52.c:undefined reference to `devm_input_allocate_device'
by Fengguang Wu 19 Jul '13

19 Jul '13

Greetings, Here is an old link error, unveiled by this build fix: tree: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master head: 7a62711aacda8887d94c40daa199b37abb1d54e1 commit: 446d64e3e1154806092ac27de198dff1225797d9 block: fix part_pack_uuid() build error date: 5 months ago config: i386-randconfig-x002-0719 (attached as .config) All error/warnings: sound/built-in.o: In function `cs42l52_beep_set': cs42l52.c:(.text+0x425b3): undefined reference to `input_event' sound/built-in.o: In function `cs42l52_probe': >> cs42l52.c:(.text+0x42776): undefined reference to `devm_input_allocate_device' cs42l52.c:(.text+0x42869): undefined reference to `input_register_device' sound/built-in.o: In function `wm8962_beep_set': wm8962.c:(.text+0x7733b): undefined reference to `input_event' sound/built-in.o: In function `wm8962_probe': >> wm8962.c:(.text+0x786c5): undefined reference to `devm_input_allocate_device' wm8962.c:(.text+0x787ea): undefined reference to `input_register_device' --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation # # Automatically generated file; DO NOT EDIT. # Linux/i386 3.8.0 Kernel Configuration # # CONFIG_64BIT is not set CONFIG_X86_32=y CONFIG_X86=y CONFIG_INSTRUCTION_DECODER=y CONFIG_OUTPUT_FORMAT="elf32-i386" CONFIG_ARCH_DEFCONFIG="arch/x86/configs/i386_defconfig" CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_MMU=y CONFIG_NEED_SG_DMA_LENGTH=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y CONFIG_GENERIC_GPIO=y CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_RWSEM_XCHGADD_ALGORITHM=y CONFIG_GENERIC_CALIBRATE_DELAY=y CONFIG_ARCH_HAS_CPU_RELAX=y CONFIG_ARCH_HAS_DEFAULT_IDLE=y CONFIG_ARCH_HAS_CACHE_LINE_SIZE=y CONFIG_ARCH_HAS_CPU_AUTOPROBE=y CONFIG_HAVE_SETUP_PER_CPU_AREA=y CONFIG_NEED_PER_CPU_EMBED_FIRST_CHUNK=y CONFIG_NEED_PER_CPU_PAGE_FIRST_CHUNK=y CONFIG_ARCH_HIBERNATION_POSSIBLE=y CONFIG_ARCH_SUSPEND_POSSIBLE=y # CONFIG_ZONE_DMA32 is not set # CONFIG_AUDIT_ARCH is not set CONFIG_ARCH_SUPPORTS_OPTIMIZED_INLINING=y CONFIG_ARCH_SUPPORTS_DEBUG_PAGEALLOC=y CONFIG_X86_32_LAZY_GS=y CONFIG_ARCH_HWEIGHT_CFLAGS="-fcall-saved-ecx -fcall-saved-edx" CONFIG_ARCH_SUPPORTS_UPROBES=y CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" CONFIG_CONSTRUCTORS=y CONFIG_IRQ_WORK=y CONFIG_BUILDTIME_EXTABLE_SORT=y # # General setup # CONFIG_EXPERIMENTAL=y CONFIG_BROKEN_ON_SMP=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_CROSS_COMPILE="" CONFIG_LOCALVERSION="" CONFIG_LOCALVERSION_AUTO=y CONFIG_HAVE_KERNEL_GZIP=y CONFIG_HAVE_KERNEL_BZIP2=y CONFIG_HAVE_KERNEL_LZMA=y CONFIG_HAVE_KERNEL_XZ=y CONFIG_HAVE_KERNEL_LZO=y # CONFIG_KERNEL_GZIP is not set CONFIG_KERNEL_BZIP2=y # CONFIG_KERNEL_LZMA is not set # CONFIG_KERNEL_XZ is not set # CONFIG_KERNEL_LZO is not set CONFIG_DEFAULT_HOSTNAME="(none)" # CONFIG_SYSVIPC is not set # CONFIG_FHANDLE is not set CONFIG_HAVE_GENERIC_HARDIRQS=y # # IRQ subsystem # CONFIG_GENERIC_HARDIRQS=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_IRQ_SHOW=y CONFIG_GENERIC_IRQ_CHIP=y CONFIG_IRQ_DOMAIN=y # CONFIG_IRQ_DOMAIN_DEBUG is not set CONFIG_IRQ_FORCED_THREADING=y CONFIG_SPARSE_IRQ=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_ALWAYS_USE_PERSISTENT_CLOCK=y CONFIG_KTIME_SCALAR=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_GENERIC_CLOCKEVENTS_MIN_ADJUST=y CONFIG_GENERIC_CMOS_UPDATE=y # # Timers subsystem # CONFIG_TICK_ONESHOT=y # CONFIG_NO_HZ is not set CONFIG_HIGH_RES_TIMERS=y # # CPU/Task time and stats accounting # CONFIG_TICK_CPU_ACCOUNTING=y # CONFIG_IRQ_TIME_ACCOUNTING is not set # CONFIG_BSD_PROCESS_ACCT is not set # # RCU Subsystem # CONFIG_TREE_PREEMPT_RCU=y # CONFIG_TINY_PREEMPT_RCU is not set CONFIG_PREEMPT_RCU=y CONFIG_RCU_STALL_COMMON=y CONFIG_RCU_FANOUT=32 CONFIG_RCU_FANOUT_LEAF=16 # CONFIG_RCU_FANOUT_EXACT is not set # CONFIG_TREE_RCU_TRACE is not set # CONFIG_RCU_BOOST is not set # CONFIG_RCU_NOCB_CPU is not set # CONFIG_IKCONFIG is not set CONFIG_LOG_BUF_SHIFT=17 CONFIG_HAVE_UNSTABLE_SCHED_CLOCK=y CONFIG_ARCH_SUPPORTS_NUMA_BALANCING=y CONFIG_ARCH_WANTS_PROT_NUMA_PROT_NONE=y # CONFIG_CHECKPOINT_RESTORE is not set CONFIG_NAMESPACES=y # CONFIG_UTS_NS is not set CONFIG_USER_NS=y CONFIG_PID_NS=y CONFIG_UIDGID_CONVERTED=y CONFIG_UIDGID_STRICT_TYPE_CHECKS=y # CONFIG_SCHED_AUTOGROUP is not set CONFIG_SYSFS_DEPRECATED=y CONFIG_SYSFS_DEPRECATED_V2=y # CONFIG_RELAY is not set # CONFIG_BLK_DEV_INITRD is not set CONFIG_CC_OPTIMIZE_FOR_SIZE=y CONFIG_ANON_INODES=y CONFIG_EXPERT=y CONFIG_HAVE_UID16=y # CONFIG_UID16 is not set CONFIG_SYSCTL_EXCEPTION_TRACE=y CONFIG_KALLSYMS=y CONFIG_KALLSYMS_ALL=y CONFIG_HOTPLUG=y CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y # CONFIG_PCSPKR_PLATFORM is not set CONFIG_HAVE_PCSPKR_PLATFORM=y # CONFIG_BASE_FULL is not set # CONFIG_FUTEX is not set CONFIG_EPOLL=y # CONFIG_SIGNALFD is not set # CONFIG_TIMERFD is not set # CONFIG_EVENTFD is not set CONFIG_SHMEM=y CONFIG_AIO=y CONFIG_EMBEDDED=y CONFIG_HAVE_PERF_EVENTS=y # # Kernel Performance Events And Counters # CONFIG_PERF_EVENTS=y # CONFIG_DEBUG_PERF_USE_VMALLOC is not set CONFIG_VM_EVENT_COUNTERS=y # CONFIG_PCI_QUIRKS is not set CONFIG_COMPAT_BRK=y CONFIG_SLAB=y # CONFIG_SLUB is not set # CONFIG_SLOB is not set CONFIG_PROFILING=y CONFIG_TRACEPOINTS=y # CONFIG_OPROFILE is not set CONFIG_HAVE_OPROFILE=y CONFIG_OPROFILE_NMI_TIMER=y # CONFIG_KPROBES is not set # CONFIG_JUMP_LABEL is not set CONFIG_UPROBES=y CONFIG_HAVE_EFFICIENT_UNALIGNED_ACCESS=y CONFIG_ARCH_USE_BUILTIN_BSWAP=y CONFIG_HAVE_IOREMAP_PROT=y CONFIG_HAVE_KPROBES=y CONFIG_HAVE_KRETPROBES=y CONFIG_HAVE_OPTPROBES=y CONFIG_HAVE_KPROBES_ON_FTRACE=y CONFIG_HAVE_ARCH_TRACEHOOK=y CONFIG_HAVE_DMA_ATTRS=y CONFIG_HAVE_DMA_CONTIGUOUS=y CONFIG_GENERIC_SMP_IDLE_THREAD=y CONFIG_HAVE_REGS_AND_STACK_ACCESS_API=y CONFIG_HAVE_DMA_API_DEBUG=y CONFIG_HAVE_HW_BREAKPOINT=y CONFIG_HAVE_MIXED_BREAKPOINTS_REGS=y CONFIG_HAVE_USER_RETURN_NOTIFIER=y CONFIG_HAVE_PERF_EVENTS_NMI=y CONFIG_HAVE_PERF_REGS=y CONFIG_HAVE_PERF_USER_STACK_DUMP=y CONFIG_HAVE_ARCH_JUMP_LABEL=y CONFIG_ARCH_HAVE_NMI_SAFE_CMPXCHG=y CONFIG_HAVE_CMPXCHG_LOCAL=y CONFIG_HAVE_CMPXCHG_DOUBLE=y CONFIG_ARCH_WANT_IPC_PARSE_VERSION=y CONFIG_HAVE_ARCH_SECCOMP_FILTER=y CONFIG_HAVE_IRQ_TIME_ACCOUNTING=y CONFIG_HAVE_ARCH_TRANSPARENT_HUGEPAGE=y CONFIG_MODULES_USE_ELF_REL=y CONFIG_CLONE_BACKWARDS=y CONFIG_OLD_SIGSUSPEND3=y CONFIG_OLD_SIGACTION=y # # GCOV-based kernel profiling # CONFIG_GCOV_KERNEL=y # CONFIG_GCOV_PROFILE_ALL is not set CONFIG_HAVE_GENERIC_DMA_COHERENT=y CONFIG_RT_MUTEXES=y CONFIG_BASE_SMALL=1 CONFIG_MODULES=y CONFIG_MODULE_FORCE_LOAD=y CONFIG_MODULE_UNLOAD=y # CONFIG_MODULE_FORCE_UNLOAD is not set CONFIG_MODVERSIONS=y CONFIG_MODULE_SRCVERSION_ALL=y CONFIG_MODULE_SIG=y # CONFIG_MODULE_SIG_FORCE is not set # CONFIG_MODULE_SIG_SHA1 is not set # CONFIG_MODULE_SIG_SHA224 is not set # CONFIG_MODULE_SIG_SHA256 is not set CONFIG_MODULE_SIG_SHA384=y # CONFIG_MODULE_SIG_SHA512 is not set # CONFIG_BLOCK is not set CONFIG_ASN1=y CONFIG_UNINLINE_SPIN_UNLOCK=y # CONFIG_FREEZER is not set # # Processor type and features # CONFIG_ZONE_DMA=y # CONFIG_SMP is not set CONFIG_X86_MPPARSE=y # CONFIG_X86_EXTENDED_PLATFORM is not set # CONFIG_X86_GOLDFISH is not set CONFIG_X86_SUPPORTS_MEMORY_FAILURE=y CONFIG_X86_32_IRIS=y CONFIG_SCHED_OMIT_FRAME_POINTER=y # CONFIG_PARAVIRT_GUEST is not set CONFIG_NO_BOOTMEM=y # CONFIG_MEMTEST is not set # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set # CONFIG_M586MMX is not set # CONFIG_M686 is not set # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set # CONFIG_MK8 is not set # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MELAN is not set # CONFIG_MGEODEGX1 is not set CONFIG_MGEODE_LX=y # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MCORE2 is not set # CONFIG_MATOM is not set # CONFIG_X86_GENERIC is not set CONFIG_X86_INTERNODE_CACHE_SHIFT=5 CONFIG_X86_L1_CACHE_SHIFT=5 CONFIG_X86_USE_PPRO_CHECKSUM=y CONFIG_X86_USE_3DNOW=y CONFIG_X86_TSC=y CONFIG_X86_CMPXCHG64=y CONFIG_X86_CMOV=y CONFIG_X86_MINIMUM_CPU_FAMILY=5 CONFIG_X86_DEBUGCTLMSR=y CONFIG_PROCESSOR_SELECT=y CONFIG_CPU_SUP_INTEL=y # CONFIG_CPU_SUP_CYRIX_32 is not set # CONFIG_CPU_SUP_AMD is not set CONFIG_CPU_SUP_CENTAUR=y # CONFIG_CPU_SUP_TRANSMETA_32 is not set CONFIG_CPU_SUP_UMC_32=y # CONFIG_HPET_TIMER is not set CONFIG_DMI=y CONFIG_NR_CPUS=1 # CONFIG_PREEMPT_NONE is not set # CONFIG_PREEMPT_VOLUNTARY is not set CONFIG_PREEMPT=y CONFIG_PREEMPT_COUNT=y CONFIG_X86_UP_APIC=y CONFIG_X86_UP_IOAPIC=y CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y # CONFIG_X86_REROUTE_FOR_BROKEN_BOOT_IRQS is not set CONFIG_X86_MCE=y # CONFIG_X86_MCE_INTEL is not set # CONFIG_X86_MCE_AMD is not set # CONFIG_X86_ANCIENT_MCE is not set # CONFIG_X86_MCE_INJECT is not set CONFIG_VM86=y CONFIG_TOSHIBA=m CONFIG_I8K=m # CONFIG_X86_REBOOTFIXUPS is not set CONFIG_MICROCODE=m CONFIG_MICROCODE_INTEL=y CONFIG_MICROCODE_AMD=y CONFIG_MICROCODE_OLD_INTERFACE=y CONFIG_MICROCODE_INTEL_LIB=y # CONFIG_X86_MSR is not set CONFIG_X86_CPUID=y CONFIG_NOHIGHMEM=y # CONFIG_HIGHMEM4G is not set # CONFIG_HIGHMEM64G is not set # CONFIG_VMSPLIT_3G is not set CONFIG_VMSPLIT_2G=y # CONFIG_VMSPLIT_1G is not set CONFIG_PAGE_OFFSET=0x80000000 CONFIG_X86_PAE=y CONFIG_ARCH_PHYS_ADDR_T_64BIT=y CONFIG_ARCH_FLATMEM_ENABLE=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_ILLEGAL_POINTER_VALUE=0 CONFIG_SELECT_MEMORY_MODEL=y CONFIG_FLATMEM_MANUAL=y # CONFIG_SPARSEMEM_MANUAL is not set CONFIG_FLATMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_SPARSEMEM_STATIC=y CONFIG_HAVE_MEMBLOCK=y CONFIG_HAVE_MEMBLOCK_NODE_MAP=y CONFIG_ARCH_DISCARD_MEMBLOCK=y # CONFIG_HAVE_BOOTMEM_INFO_NODE is not set CONFIG_PAGEFLAGS_EXTENDED=y CONFIG_SPLIT_PTLOCK_CPUS=999999 CONFIG_COMPACTION=y CONFIG_MIGRATION=y CONFIG_PHYS_ADDR_T_64BIT=y CONFIG_ZONE_DMA_FLAG=1 CONFIG_VIRT_TO_BUS=y # CONFIG_KSM is not set CONFIG_DEFAULT_MMAP_MIN_ADDR=4096 CONFIG_ARCH_SUPPORTS_MEMORY_FAILURE=y # CONFIG_MEMORY_FAILURE is not set CONFIG_TRANSPARENT_HUGEPAGE=y # CONFIG_TRANSPARENT_HUGEPAGE_ALWAYS is not set CONFIG_TRANSPARENT_HUGEPAGE_MADVISE=y # CONFIG_CROSS_MEMORY_ATTACH is not set CONFIG_NEED_PER_CPU_KM=y CONFIG_CLEANCACHE=y CONFIG_X86_CHECK_BIOS_CORRUPTION=y # CONFIG_X86_BOOTPARAM_MEMORY_CORRUPTION_CHECK is not set CONFIG_X86_RESERVE_LOW=64 CONFIG_MATH_EMULATION=y # CONFIG_MTRR is not set CONFIG_ARCH_RANDOM=y CONFIG_X86_SMAP=y CONFIG_SECCOMP=y # CONFIG_CC_STACKPROTECTOR is not set # CONFIG_HZ_100 is not set # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set CONFIG_HZ_1000=y CONFIG_HZ=1000 CONFIG_SCHED_HRTICK=y CONFIG_KEXEC=y CONFIG_PHYSICAL_START=0x1000000 # CONFIG_RELOCATABLE is not set CONFIG_PHYSICAL_ALIGN=0x1000000 CONFIG_COMPAT_VDSO=y CONFIG_CMDLINE_BOOL=y CONFIG_CMDLINE="" # CONFIG_CMDLINE_OVERRIDE is not set # # Power management and ACPI options # # CONFIG_SUSPEND is not set CONFIG_PM_RUNTIME=y CONFIG_PM=y # CONFIG_PM_DEBUG is not set # CONFIG_ACPI is not set CONFIG_SFI=y # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=y CONFIG_CPU_FREQ_GOV_COMMON=y # CONFIG_CPU_FREQ_STAT is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_POWERSAVE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y # CONFIG_CPU_FREQ_GOV_POWERSAVE is not set CONFIG_CPU_FREQ_GOV_USERSPACE=y CONFIG_CPU_FREQ_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_GOV_CONSERVATIVE is not set # # x86 CPU frequency scaling drivers # CONFIG_X86_INTEL_PSTATE=y CONFIG_X86_POWERNOW_K6=y # CONFIG_X86_POWERNOW_K7 is not set CONFIG_X86_GX_SUSPMOD=y # CONFIG_X86_SPEEDSTEP_CENTRINO is not set CONFIG_X86_SPEEDSTEP_ICH=y # CONFIG_X86_SPEEDSTEP_SMI is not set CONFIG_X86_P4_CLOCKMOD=m CONFIG_X86_CPUFREQ_NFORCE2=y CONFIG_X86_LONGRUN=y # CONFIG_X86_E_POWERSAVER is not set # # shared options # CONFIG_X86_SPEEDSTEP_LIB=y CONFIG_X86_SPEEDSTEP_RELAXED_CAP_CHECK=y CONFIG_CPU_IDLE=y CONFIG_CPU_IDLE_MULTIPLE_DRIVERS=y CONFIG_CPU_IDLE_GOV_LADDER=y # CONFIG_ARCH_NEEDS_CPU_IDLE_COUPLED is not set # CONFIG_INTEL_IDLE is not set # # Bus options (PCI etc.) # CONFIG_PCI=y CONFIG_PCI_GOBIOS=y # CONFIG_PCI_GOMMCONFIG is not set # CONFIG_PCI_GODIRECT is not set # CONFIG_PCI_GOANY is not set CONFIG_PCI_BIOS=y CONFIG_PCI_DOMAINS=y # CONFIG_PCI_CNB20LE_QUIRK is not set CONFIG_PCIEPORTBUS=y CONFIG_HOTPLUG_PCI_PCIE=m # CONFIG_PCIEAER is not set CONFIG_PCIEASPM=y # CONFIG_PCIEASPM_DEBUG is not set CONFIG_PCIEASPM_DEFAULT=y # CONFIG_PCIEASPM_POWERSAVE is not set # CONFIG_PCIEASPM_PERFORMANCE is not set CONFIG_ARCH_SUPPORTS_MSI=y # CONFIG_PCI_MSI is not set # CONFIG_PCI_DEBUG is not set CONFIG_PCI_REALLOC_ENABLE_AUTO=y # CONFIG_PCI_STUB is not set # CONFIG_HT_IRQ is not set CONFIG_PCI_ATS=y # CONFIG_PCI_IOV is not set # CONFIG_PCI_PRI is not set CONFIG_PCI_PASID=y CONFIG_PCI_LABEL=y CONFIG_ISA_DMA_API=y # CONFIG_ISA is not set # CONFIG_SCx200 is not set CONFIG_ALIX=y # CONFIG_NET5501 is not set # CONFIG_GEOS is not set CONFIG_PCCARD=m CONFIG_PCMCIA=m CONFIG_PCMCIA_LOAD_CIS=y # CONFIG_CARDBUS is not set # # PC-card bridges # # CONFIG_YENTA is not set CONFIG_PD6729=m CONFIG_I82092=m CONFIG_PCCARD_NONSTATIC=y CONFIG_HOTPLUG_PCI=y CONFIG_HOTPLUG_PCI_COMPAQ=y CONFIG_HOTPLUG_PCI_COMPAQ_NVRAM=y # CONFIG_HOTPLUG_PCI_IBM is not set # CONFIG_HOTPLUG_PCI_CPCI is not set CONFIG_HOTPLUG_PCI_SHPC=y # CONFIG_RAPIDIO is not set # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_ARCH_BINFMT_ELF_RANDOMIZE_PIE=y CONFIG_CORE_DUMP_DEFAULT_ELF_HEADERS=y CONFIG_HAVE_AOUT=y CONFIG_BINFMT_AOUT=y CONFIG_BINFMT_MISC=y CONFIG_COREDUMP=y CONFIG_HAVE_ATOMIC_IOMAP=y CONFIG_HAVE_TEXT_POKE_SMP=y # CONFIG_NET is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="" CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y CONFIG_FIRMWARE_IN_KERNEL=y CONFIG_EXTRA_FIRMWARE="" CONFIG_FW_LOADER_USER_HELPER=y # CONFIG_DEBUG_DRIVER is not set CONFIG_DEBUG_DEVRES=y # CONFIG_SYS_HYPERVISOR is not set # CONFIG_GENERIC_CPU_DEVICES is not set CONFIG_REGMAP=y CONFIG_REGMAP_I2C=y CONFIG_REGMAP_SPI=y CONFIG_REGMAP_MMIO=y CONFIG_REGMAP_IRQ=y CONFIG_DMA_SHARED_BUFFER=y # CONFIG_CMA is not set # # Bus devices # # CONFIG_MTD is not set CONFIG_PARPORT=y CONFIG_PARPORT_PC=m # CONFIG_PARPORT_SERIAL is not set CONFIG_PARPORT_PC_FIFO=y CONFIG_PARPORT_PC_SUPERIO=y # CONFIG_PARPORT_PC_PCMCIA is not set # CONFIG_PARPORT_GSC is not set CONFIG_PARPORT_AX88796=m CONFIG_PARPORT_1284=y CONFIG_PARPORT_NOT_PC=y # # Misc devices # CONFIG_SENSORS_LIS3LV02D=m # CONFIG_AD525X_DPOT is not set CONFIG_IBM_ASM=m CONFIG_PHANTOM=y # CONFIG_INTEL_MID_PTI is not set CONFIG_SGI_IOC4=y CONFIG_TIFM_CORE=y CONFIG_TIFM_7XX1=y # CONFIG_ICS932S401 is not set CONFIG_ATMEL_SSC=y CONFIG_ENCLOSURE_SERVICES=m # CONFIG_HP_ILO is not set # CONFIG_APDS9802ALS is not set CONFIG_ISL29003=m CONFIG_ISL29020=m CONFIG_SENSORS_TSL2550=m CONFIG_SENSORS_BH1780=m CONFIG_SENSORS_BH1770=m # CONFIG_SENSORS_APDS990X is not set CONFIG_HMC6352=m CONFIG_DS1682=y # CONFIG_TI_DAC7512 is not set # CONFIG_VMWARE_BALLOON is not set CONFIG_BMP085=y CONFIG_BMP085_I2C=y CONFIG_BMP085_SPI=m CONFIG_PCH_PHUB=m CONFIG_USB_SWITCH_FSA9480=y CONFIG_LATTICE_ECP3_CONFIG=m CONFIG_C2PORT=m CONFIG_C2PORT_DURAMAR_2150=m # # EEPROM support # CONFIG_EEPROM_AT24=m CONFIG_EEPROM_AT25=m CONFIG_EEPROM_LEGACY=m CONFIG_EEPROM_MAX6875=m # CONFIG_EEPROM_93CX6 is not set # CONFIG_EEPROM_93XX46 is not set CONFIG_CB710_CORE=y # CONFIG_CB710_DEBUG is not set CONFIG_CB710_DEBUG_ASSUMPTIONS=y # # Texas Instruments shared transport line discipline # CONFIG_SENSORS_LIS3_SPI=m CONFIG_SENSORS_LIS3_I2C=m # # Altera FPGA firmware download module # CONFIG_ALTERA_STAPL=m # CONFIG_VMWARE_VMCI is not set CONFIG_HAVE_IDE=y # # SCSI device support # CONFIG_SCSI_MOD=y # CONFIG_SCSI_DMA is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_FUSION is not set # # IEEE 1394 (FireWire) support # # CONFIG_FIREWIRE is not set CONFIG_FIREWIRE_NOSY=y CONFIG_I2O=m CONFIG_I2O_LCT_NOTIFY_ON_CHANGES=y # CONFIG_I2O_EXT_ADAPTEC is not set # CONFIG_I2O_CONFIG is not set # CONFIG_I2O_BUS is not set CONFIG_I2O_PROC=m # CONFIG_MACINTOSH_DRIVERS is not set # # Input device support # CONFIG_INPUT=m CONFIG_INPUT_FF_MEMLESS=m CONFIG_INPUT_POLLDEV=m CONFIG_INPUT_SPARSEKMAP=m CONFIG_INPUT_MATRIXKMAP=m # # Userland interfaces # # CONFIG_INPUT_MOUSEDEV is not set CONFIG_INPUT_JOYDEV=m CONFIG_INPUT_EVDEV=m # CONFIG_INPUT_EVBUG is not set # # Input Device Drivers # # CONFIG_INPUT_KEYBOARD is not set # CONFIG_INPUT_MOUSE is not set CONFIG_INPUT_JOYSTICK=y CONFIG_JOYSTICK_ANALOG=m # CONFIG_JOYSTICK_A3D is not set CONFIG_JOYSTICK_ADI=m CONFIG_JOYSTICK_COBRA=m CONFIG_JOYSTICK_GF2K=m CONFIG_JOYSTICK_GRIP=m CONFIG_JOYSTICK_GRIP_MP=m # CONFIG_JOYSTICK_GUILLEMOT is not set # CONFIG_JOYSTICK_INTERACT is not set CONFIG_JOYSTICK_SIDEWINDER=m CONFIG_JOYSTICK_TMDC=m # CONFIG_JOYSTICK_IFORCE is not set CONFIG_JOYSTICK_WARRIOR=m # CONFIG_JOYSTICK_MAGELLAN is not set CONFIG_JOYSTICK_SPACEORB=m # CONFIG_JOYSTICK_SPACEBALL is not set # CONFIG_JOYSTICK_STINGER is not set # CONFIG_JOYSTICK_TWIDJOY is not set CONFIG_JOYSTICK_ZHENHUA=m # CONFIG_JOYSTICK_DB9 is not set CONFIG_JOYSTICK_GAMECON=m CONFIG_JOYSTICK_TURBOGRAFX=m CONFIG_JOYSTICK_AS5011=m CONFIG_JOYSTICK_JOYDUMP=m CONFIG_JOYSTICK_XPAD=m CONFIG_JOYSTICK_XPAD_FF=y CONFIG_JOYSTICK_XPAD_LEDS=y CONFIG_JOYSTICK_WALKERA0701=m CONFIG_INPUT_TABLET=y # CONFIG_TABLET_USB_ACECAD is not set CONFIG_TABLET_USB_AIPTEK=m CONFIG_TABLET_USB_GTCO=m CONFIG_TABLET_USB_HANWANG=m # CONFIG_TABLET_USB_KBTAB is not set CONFIG_TABLET_USB_WACOM=m CONFIG_INPUT_TOUCHSCREEN=y # CONFIG_TOUCHSCREEN_ADS7846 is not set CONFIG_TOUCHSCREEN_AD7877=m # CONFIG_TOUCHSCREEN_AD7879 is not set CONFIG_TOUCHSCREEN_ATMEL_MXT=m # CONFIG_TOUCHSCREEN_AUO_PIXCIR is not set # CONFIG_TOUCHSCREEN_BU21013 is not set CONFIG_TOUCHSCREEN_CY8CTMG110=m CONFIG_TOUCHSCREEN_CYTTSP_CORE=m CONFIG_TOUCHSCREEN_CYTTSP_I2C=m CONFIG_TOUCHSCREEN_CYTTSP_SPI=m # CONFIG_TOUCHSCREEN_DYNAPRO is not set # CONFIG_TOUCHSCREEN_HAMPSHIRE is not set # CONFIG_TOUCHSCREEN_EETI is not set CONFIG_TOUCHSCREEN_FUJITSU=m # CONFIG_TOUCHSCREEN_ILI210X is not set CONFIG_TOUCHSCREEN_GUNZE=m # CONFIG_TOUCHSCREEN_ELO is not set CONFIG_TOUCHSCREEN_WACOM_W8001=m # CONFIG_TOUCHSCREEN_WACOM_I2C is not set # CONFIG_TOUCHSCREEN_MAX11801 is not set CONFIG_TOUCHSCREEN_MCS5000=m # CONFIG_TOUCHSCREEN_MMS114 is not set # CONFIG_TOUCHSCREEN_MTOUCH is not set CONFIG_TOUCHSCREEN_INEXIO=m CONFIG_TOUCHSCREEN_MK712=m CONFIG_TOUCHSCREEN_PENMOUNT=m # CONFIG_TOUCHSCREEN_EDT_FT5X06 is not set # CONFIG_TOUCHSCREEN_TOUCHRIGHT is not set CONFIG_TOUCHSCREEN_TOUCHWIN=m # CONFIG_TOUCHSCREEN_TI_AM335X_TSC is not set CONFIG_TOUCHSCREEN_PIXCIR=m CONFIG_TOUCHSCREEN_WM831X=m CONFIG_TOUCHSCREEN_USB_COMPOSITE=m CONFIG_TOUCHSCREEN_MC13783=m CONFIG_TOUCHSCREEN_USB_EGALAX=y CONFIG_TOUCHSCREEN_USB_PANJIT=y CONFIG_TOUCHSCREEN_USB_3M=y CONFIG_TOUCHSCREEN_USB_ITM=y CONFIG_TOUCHSCREEN_USB_ETURBO=y CONFIG_TOUCHSCREEN_USB_GUNZE=y CONFIG_TOUCHSCREEN_USB_DMC_TSC10=y CONFIG_TOUCHSCREEN_USB_IRTOUCH=y # CONFIG_TOUCHSCREEN_USB_IDEALTEK is not set CONFIG_TOUCHSCREEN_USB_GENERAL_TOUCH=y # CONFIG_TOUCHSCREEN_USB_GOTOP is not set # CONFIG_TOUCHSCREEN_USB_JASTEC is not set CONFIG_TOUCHSCREEN_USB_ELO=y CONFIG_TOUCHSCREEN_USB_E2I=y # CONFIG_TOUCHSCREEN_USB_ZYTRONIC is not set CONFIG_TOUCHSCREEN_USB_ETT_TC45USB=y # CONFIG_TOUCHSCREEN_USB_NEXIO is not set CONFIG_TOUCHSCREEN_USB_EASYTOUCH=y # CONFIG_TOUCHSCREEN_TOUCHIT213 is not set CONFIG_TOUCHSCREEN_TSC_SERIO=m CONFIG_TOUCHSCREEN_TSC2005=m CONFIG_TOUCHSCREEN_TSC2007=m CONFIG_TOUCHSCREEN_PCAP=m CONFIG_TOUCHSCREEN_ST1232=m # CONFIG_TOUCHSCREEN_TPS6507X is not set CONFIG_INPUT_MISC=y # CONFIG_INPUT_88PM80X_ONKEY is not set CONFIG_INPUT_AD714X=m CONFIG_INPUT_AD714X_I2C=m # CONFIG_INPUT_AD714X_SPI is not set CONFIG_INPUT_ARIZONA_HAPTICS=m CONFIG_INPUT_BMA150=m CONFIG_INPUT_MC13783_PWRBUTTON=m CONFIG_INPUT_MMA8450=m # CONFIG_INPUT_MPU3050 is not set CONFIG_INPUT_APANEL=m CONFIG_INPUT_GP2A=m # CONFIG_INPUT_GPIO_TILT_POLLED is not set CONFIG_INPUT_WISTRON_BTNS=m CONFIG_INPUT_ATI_REMOTE2=m CONFIG_INPUT_KEYSPAN_REMOTE=m # CONFIG_INPUT_KXTJ9 is not set CONFIG_INPUT_POWERMATE=m CONFIG_INPUT_YEALINK=m # CONFIG_INPUT_CM109 is not set # CONFIG_INPUT_RETU_PWRBUTTON is not set # CONFIG_INPUT_TWL4030_PWRBUTTON is not set CONFIG_INPUT_TWL4030_VIBRA=m CONFIG_INPUT_UINPUT=m # CONFIG_INPUT_PCF50633_PMU is not set CONFIG_INPUT_PCF8574=m CONFIG_INPUT_PWM_BEEPER=m CONFIG_INPUT_GPIO_ROTARY_ENCODER=m # CONFIG_INPUT_WM831X_ON is not set CONFIG_INPUT_PCAP=m CONFIG_INPUT_ADXL34X=m CONFIG_INPUT_ADXL34X_I2C=m # CONFIG_INPUT_ADXL34X_SPI is not set CONFIG_INPUT_CMA3000=m CONFIG_INPUT_CMA3000_I2C=m # # Hardware I/O ports # CONFIG_SERIO=m CONFIG_SERIO_I8042=m CONFIG_SERIO_SERPORT=m # CONFIG_SERIO_CT82C710 is not set CONFIG_SERIO_PARKBD=m CONFIG_SERIO_PCIPS2=m CONFIG_SERIO_LIBPS2=m CONFIG_SERIO_RAW=m # CONFIG_SERIO_ALTERA_PS2 is not set CONFIG_SERIO_PS2MULT=m # CONFIG_SERIO_ARC_PS2 is not set CONFIG_GAMEPORT=y CONFIG_GAMEPORT_NS558=y CONFIG_GAMEPORT_L4=m # CONFIG_GAMEPORT_EMU10K1 is not set CONFIG_GAMEPORT_FM801=y # # Character devices # CONFIG_TTY=y # CONFIG_VT is not set CONFIG_UNIX98_PTYS=y CONFIG_DEVPTS_MULTIPLE_INSTANCES=y # CONFIG_LEGACY_PTYS is not set CONFIG_SERIAL_NONSTANDARD=y CONFIG_ROCKETPORT=m # CONFIG_CYCLADES is not set CONFIG_MOXA_INTELLIO=m # CONFIG_MOXA_SMARTIO is not set # CONFIG_SYNCLINK is not set # CONFIG_SYNCLINKMP is not set CONFIG_SYNCLINK_GT=y CONFIG_NOZOMI=m CONFIG_ISI=m CONFIG_N_HDLC=y CONFIG_TRACE_ROUTER=m CONFIG_TRACE_SINK=m CONFIG_DEVKMEM=y # CONFIG_STALDRV is not set # # Serial drivers # CONFIG_SERIAL_8250=m CONFIG_FIX_EARLYCON_MEM=y CONFIG_SERIAL_8250_PCI=m CONFIG_SERIAL_8250_CS=m CONFIG_SERIAL_8250_NR_UARTS=4 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 # CONFIG_SERIAL_8250_EXTENDED is not set CONFIG_SERIAL_8250_DW=m # # Non-8250 serial port support # CONFIG_SERIAL_KGDB_NMI=y CONFIG_SERIAL_MAX3100=m CONFIG_SERIAL_MAX310X=y # CONFIG_SERIAL_MFD_HSU is not set # CONFIG_SERIAL_UARTLITE is not set CONFIG_SERIAL_CORE=y CONFIG_CONSOLE_POLL=y # CONFIG_SERIAL_JSM is not set # CONFIG_SERIAL_SCCNXP is not set CONFIG_SERIAL_TIMBERDALE=m CONFIG_SERIAL_ALTERA_JTAGUART=y # CONFIG_SERIAL_ALTERA_JTAGUART_CONSOLE is not set CONFIG_SERIAL_ALTERA_UART=m CONFIG_SERIAL_ALTERA_UART_MAXPORTS=4 CONFIG_SERIAL_ALTERA_UART_BAUDRATE=115200 # CONFIG_SERIAL_IFX6X60 is not set CONFIG_SERIAL_PCH_UART=y # CONFIG_SERIAL_PCH_UART_CONSOLE is not set CONFIG_SERIAL_ARC=y # CONFIG_SERIAL_ARC_CONSOLE is not set CONFIG_SERIAL_ARC_NR_PORTS=1 CONFIG_SERIAL_RP2=y CONFIG_SERIAL_RP2_NR_UARTS=32 # CONFIG_TTY_PRINTK is not set CONFIG_PRINTER=m CONFIG_LP_CONSOLE=y # CONFIG_PPDEV is not set CONFIG_HVC_DRIVER=y CONFIG_VIRTIO_CONSOLE=m CONFIG_IPMI_HANDLER=y CONFIG_IPMI_PANIC_EVENT=y # CONFIG_IPMI_PANIC_STRING is not set CONFIG_IPMI_DEVICE_INTERFACE=y CONFIG_IPMI_SI=m CONFIG_IPMI_WATCHDOG=y CONFIG_IPMI_POWEROFF=m CONFIG_HW_RANDOM=m CONFIG_HW_RANDOM_TIMERIOMEM=m CONFIG_HW_RANDOM_INTEL=m CONFIG_HW_RANDOM_AMD=m # CONFIG_HW_RANDOM_GEODE is not set CONFIG_HW_RANDOM_VIA=m CONFIG_HW_RANDOM_VIRTIO=m CONFIG_HW_RANDOM_TPM=m # CONFIG_NVRAM is not set CONFIG_R3964=y CONFIG_APPLICOM=m CONFIG_SONYPI=m # # PCMCIA character devices # # CONFIG_SYNCLINK_CS is not set CONFIG_CARDMAN_4000=m # CONFIG_CARDMAN_4040 is not set CONFIG_MWAVE=m CONFIG_PC8736x_GPIO=y CONFIG_NSC_GPIO=y # CONFIG_HANGCHECK_TIMER is not set CONFIG_TCG_TPM=y CONFIG_TCG_TIS=y # CONFIG_TCG_TIS_I2C_INFINEON is not set CONFIG_TCG_NSC=m CONFIG_TCG_ATMEL=y # CONFIG_TCG_ST33_I2C is not set # CONFIG_TELCLOCK is not set CONFIG_DEVPORT=y CONFIG_I2C=y CONFIG_I2C_BOARDINFO=y CONFIG_I2C_COMPAT=y # CONFIG_I2C_CHARDEV is not set CONFIG_I2C_MUX=m # # Multiplexer I2C Chip support # # CONFIG_I2C_MUX_GPIO is not set CONFIG_I2C_MUX_PCA9541=m CONFIG_I2C_MUX_PCA954x=m # CONFIG_I2C_HELPER_AUTO is not set CONFIG_I2C_SMBUS=m # # I2C Algorithms # CONFIG_I2C_ALGOBIT=y # CONFIG_I2C_ALGOPCF is not set CONFIG_I2C_ALGOPCA=m # # I2C Hardware Bus support # # # PC SMBus host controller drivers # CONFIG_I2C_ALI1535=m CONFIG_I2C_ALI1563=y CONFIG_I2C_ALI15X3=m CONFIG_I2C_AMD756=y CONFIG_I2C_AMD756_S4882=y # CONFIG_I2C_AMD8111 is not set # CONFIG_I2C_I801 is not set # CONFIG_I2C_ISCH is not set # CONFIG_I2C_PIIX4 is not set CONFIG_I2C_NFORCE2=y # CONFIG_I2C_NFORCE2_S4985 is not set # CONFIG_I2C_SIS5595 is not set CONFIG_I2C_SIS630=y CONFIG_I2C_SIS96X=y CONFIG_I2C_VIA=m # CONFIG_I2C_VIAPRO is not set # # I2C system bus drivers (mostly embedded / system-on-chip) # # CONFIG_I2C_CBUS_GPIO is not set # CONFIG_I2C_DESIGNWARE_PCI is not set # CONFIG_I2C_EG20T is not set # CONFIG_I2C_GPIO is not set # CONFIG_I2C_INTEL_MID is not set CONFIG_I2C_OCORES=m CONFIG_I2C_PCA_PLATFORM=m # CONFIG_I2C_PXA_PCI is not set CONFIG_I2C_SIMTEC=y CONFIG_I2C_XILINX=m # # External I2C/SMBus adapter drivers # # CONFIG_I2C_DIOLAN_U2C is not set # CONFIG_I2C_PARPORT is not set CONFIG_I2C_PARPORT_LIGHT=m CONFIG_I2C_TAOS_EVM=m CONFIG_I2C_TINY_USB=m # # Other I2C/SMBus bus drivers # # CONFIG_SCx200_ACB is not set CONFIG_I2C_STUB=m # CONFIG_I2C_DEBUG_CORE is not set # CONFIG_I2C_DEBUG_ALGO is not set # CONFIG_I2C_DEBUG_BUS is not set CONFIG_SPI=y # CONFIG_SPI_DEBUG is not set CONFIG_SPI_MASTER=y # # SPI Master Controller Drivers # CONFIG_SPI_ALTERA=m CONFIG_SPI_BITBANG=y # CONFIG_SPI_BUTTERFLY is not set CONFIG_SPI_GPIO=y # CONFIG_SPI_LM70_LLP is not set CONFIG_SPI_OC_TINY=m CONFIG_SPI_PXA2XX_DMA=y CONFIG_SPI_PXA2XX=m CONFIG_SPI_PXA2XX_PCI=m CONFIG_SPI_SC18IS602=y CONFIG_SPI_TOPCLIFF_PCH=y CONFIG_SPI_XCOMM=m CONFIG_SPI_XILINX=m CONFIG_SPI_DESIGNWARE=m # CONFIG_SPI_DW_PCI is not set # # SPI Protocol Masters # CONFIG_SPI_SPIDEV=m CONFIG_SPI_TLE62X0=y CONFIG_HSI=m CONFIG_HSI_BOARDINFO=y # # HSI clients # CONFIG_HSI_CHAR=m # # PPS support # CONFIG_PPS=m # CONFIG_PPS_DEBUG is not set CONFIG_NTP_PPS=y # # PPS clients support # CONFIG_PPS_CLIENT_KTIMER=m CONFIG_PPS_CLIENT_LDISC=m CONFIG_PPS_CLIENT_PARPORT=m CONFIG_PPS_CLIENT_GPIO=m # # PPS generators support # # # PTP clock support # CONFIG_PTP_1588_CLOCK=m # # Enable PHYLIB and NETWORK_PHY_TIMESTAMPING to see the additional clocks. # CONFIG_PTP_1588_CLOCK_PCH=m CONFIG_ARCH_WANT_OPTIONAL_GPIOLIB=y CONFIG_GPIOLIB=y CONFIG_DEBUG_GPIO=y # CONFIG_GPIO_SYSFS is not set CONFIG_GPIO_GENERIC=y CONFIG_GPIO_MAX730X=y # # Memory mapped GPIO drivers: # CONFIG_GPIO_GENERIC_PLATFORM=y # CONFIG_GPIO_IT8761E is not set # CONFIG_GPIO_TS5500 is not set # CONFIG_GPIO_SCH is not set # CONFIG_GPIO_ICH is not set # CONFIG_GPIO_VX855 is not set # # I2C GPIO expanders: # CONFIG_GPIO_ARIZONA=y CONFIG_GPIO_MAX7300=y # CONFIG_GPIO_MAX732X is not set CONFIG_GPIO_PCA953X=m CONFIG_GPIO_PCF857X=y # CONFIG_GPIO_RC5T583 is not set CONFIG_GPIO_SX150X=y # CONFIG_GPIO_TC3589X is not set # CONFIG_GPIO_TWL4030 is not set CONFIG_GPIO_WM831X=y CONFIG_GPIO_WM8350=m # CONFIG_GPIO_WM8994 is not set CONFIG_GPIO_ADP5588=m # # PCI GPIO expanders: # CONFIG_GPIO_BT8XX=m CONFIG_GPIO_AMD8111=y # CONFIG_GPIO_LANGWELL is not set CONFIG_GPIO_PCH=y CONFIG_GPIO_ML_IOH=m # CONFIG_GPIO_TIMBERDALE is not set CONFIG_GPIO_RDC321X=m # # SPI GPIO expanders: # # CONFIG_GPIO_MAX7301 is not set CONFIG_GPIO_MCP23S08=y # CONFIG_GPIO_MC33880 is not set CONFIG_GPIO_74X164=y # # AC97 GPIO expanders: # # # MODULbus GPIO expanders: # # CONFIG_GPIO_JANZ_TTL is not set # CONFIG_GPIO_PALMAS is not set CONFIG_GPIO_TPS65910=y # # USB GPIO expanders: # CONFIG_W1=y # # 1-wire Bus Masters # # CONFIG_W1_MASTER_MATROX is not set CONFIG_W1_MASTER_DS2490=m # CONFIG_W1_MASTER_DS2482 is not set # CONFIG_W1_MASTER_DS1WM is not set # CONFIG_W1_MASTER_GPIO is not set # # 1-wire Slaves # CONFIG_W1_SLAVE_THERM=y CONFIG_W1_SLAVE_SMEM=m # CONFIG_W1_SLAVE_DS2408 is not set # CONFIG_W1_SLAVE_DS2423 is not set CONFIG_W1_SLAVE_DS2431=m CONFIG_W1_SLAVE_DS2433=m CONFIG_W1_SLAVE_DS2433_CRC=y CONFIG_W1_SLAVE_DS2760=y CONFIG_W1_SLAVE_DS2780=m CONFIG_W1_SLAVE_DS2781=y CONFIG_W1_SLAVE_DS28E04=m # CONFIG_W1_SLAVE_BQ27000 is not set CONFIG_POWER_SUPPLY=y CONFIG_POWER_SUPPLY_DEBUG=y CONFIG_PDA_POWER=m CONFIG_GENERIC_ADC_BATTERY=y CONFIG_WM831X_BACKUP=m # CONFIG_WM831X_POWER is not set CONFIG_WM8350_POWER=y CONFIG_TEST_POWER=m CONFIG_BATTERY_DS2760=y CONFIG_BATTERY_DS2780=m CONFIG_BATTERY_DS2781=y CONFIG_BATTERY_DS2782=y # CONFIG_BATTERY_SBS is not set # CONFIG_BATTERY_BQ27x00 is not set # CONFIG_BATTERY_MAX17040 is not set CONFIG_BATTERY_MAX17042=y CONFIG_CHARGER_PCF50633=m # CONFIG_BATTERY_RX51 is not set CONFIG_CHARGER_ISP1704=y CONFIG_CHARGER_MAX8903=y CONFIG_CHARGER_TWL4030=m CONFIG_CHARGER_LP8727=y CONFIG_CHARGER_GPIO=y # CONFIG_CHARGER_MANAGER is not set # CONFIG_CHARGER_MAX8997 is not set CONFIG_CHARGER_MAX8998=m CONFIG_CHARGER_BQ2415X=m CONFIG_CHARGER_SMB347=y CONFIG_BATTERY_GOLDFISH=y # CONFIG_POWER_RESET is not set CONFIG_POWER_AVS=y CONFIG_HWMON=m CONFIG_HWMON_VID=m # CONFIG_HWMON_DEBUG_CHIP is not set # # Native drivers # CONFIG_SENSORS_ABITUGURU=m CONFIG_SENSORS_ABITUGURU3=m CONFIG_SENSORS_AD7314=m CONFIG_SENSORS_AD7414=m # CONFIG_SENSORS_AD7418 is not set CONFIG_SENSORS_ADCXX=m CONFIG_SENSORS_ADM1021=m CONFIG_SENSORS_ADM1025=m # CONFIG_SENSORS_ADM1026 is not set CONFIG_SENSORS_ADM1029=m # CONFIG_SENSORS_ADM1031 is not set # CONFIG_SENSORS_ADM9240 is not set CONFIG_SENSORS_ADT7410=m CONFIG_SENSORS_ADT7411=m CONFIG_SENSORS_ADT7462=m # CONFIG_SENSORS_ADT7470 is not set CONFIG_SENSORS_ADT7475=m CONFIG_SENSORS_ASC7621=m CONFIG_SENSORS_K8TEMP=m CONFIG_SENSORS_K10TEMP=m CONFIG_SENSORS_FAM15H_POWER=m CONFIG_SENSORS_ASB100=m # CONFIG_SENSORS_ATXP1 is not set # CONFIG_SENSORS_DS620 is not set # CONFIG_SENSORS_DS1621 is not set # CONFIG_SENSORS_I5K_AMB is not set CONFIG_SENSORS_F71805F=m CONFIG_SENSORS_F71882FG=m # CONFIG_SENSORS_F75375S is not set CONFIG_SENSORS_FSCHMD=m # CONFIG_SENSORS_G760A is not set # CONFIG_SENSORS_GL518SM is not set CONFIG_SENSORS_GL520SM=m # CONFIG_SENSORS_GPIO_FAN is not set CONFIG_SENSORS_HIH6130=m # CONFIG_SENSORS_CORETEMP is not set CONFIG_SENSORS_IBMAEM=m # CONFIG_SENSORS_IBMPEX is not set CONFIG_SENSORS_IT87=m CONFIG_SENSORS_JC42=m # CONFIG_SENSORS_LINEAGE is not set # CONFIG_SENSORS_LM63 is not set # CONFIG_SENSORS_LM70 is not set # CONFIG_SENSORS_LM73 is not set CONFIG_SENSORS_LM75=m # CONFIG_SENSORS_LM77 is not set # CONFIG_SENSORS_LM78 is not set CONFIG_SENSORS_LM80=m # CONFIG_SENSORS_LM83 is not set CONFIG_SENSORS_LM85=m CONFIG_SENSORS_LM87=m CONFIG_SENSORS_LM90=m CONFIG_SENSORS_LM92=m CONFIG_SENSORS_LM93=m # CONFIG_SENSORS_LTC4151 is not set # CONFIG_SENSORS_LTC4215 is not set CONFIG_SENSORS_LTC4245=m CONFIG_SENSORS_LTC4261=m # CONFIG_SENSORS_LM95241 is not set CONFIG_SENSORS_LM95245=m CONFIG_SENSORS_MAX1111=m CONFIG_SENSORS_MAX16065=m CONFIG_SENSORS_MAX1619=m # CONFIG_SENSORS_MAX1668 is not set CONFIG_SENSORS_MAX197=m # CONFIG_SENSORS_MAX6639 is not set CONFIG_SENSORS_MAX6642=m # CONFIG_SENSORS_MAX6650 is not set CONFIG_SENSORS_MAX6697=m CONFIG_SENSORS_MCP3021=m # CONFIG_SENSORS_NTC_THERMISTOR is not set CONFIG_SENSORS_PC87360=m CONFIG_SENSORS_PC87427=m # CONFIG_SENSORS_PCF8591 is not set # CONFIG_PMBUS is not set CONFIG_SENSORS_SHT15=m CONFIG_SENSORS_SHT21=m # CONFIG_SENSORS_SIS5595 is not set CONFIG_SENSORS_SMM665=m CONFIG_SENSORS_DME1737=m # CONFIG_SENSORS_EMC1403 is not set CONFIG_SENSORS_EMC2103=m # CONFIG_SENSORS_EMC6W201 is not set CONFIG_SENSORS_SMSC47M1=m # CONFIG_SENSORS_SMSC47M192 is not set CONFIG_SENSORS_SMSC47B397=m # CONFIG_SENSORS_SCH56XX_COMMON is not set CONFIG_SENSORS_ADS1015=m CONFIG_SENSORS_ADS7828=m # CONFIG_SENSORS_ADS7871 is not set # CONFIG_SENSORS_AMC6821 is not set # CONFIG_SENSORS_INA209 is not set CONFIG_SENSORS_INA2XX=m CONFIG_SENSORS_THMC50=m CONFIG_SENSORS_TMP102=m CONFIG_SENSORS_TMP401=m CONFIG_SENSORS_TMP421=m CONFIG_SENSORS_TWL4030_MADC=m CONFIG_SENSORS_VIA_CPUTEMP=m # CONFIG_SENSORS_VIA686A is not set CONFIG_SENSORS_VT1211=m CONFIG_SENSORS_VT8231=m CONFIG_SENSORS_W83781D=m # CONFIG_SENSORS_W83791D is not set # CONFIG_SENSORS_W83792D is not set CONFIG_SENSORS_W83793=m CONFIG_SENSORS_W83795=m CONFIG_SENSORS_W83795_FANCTRL=y # CONFIG_SENSORS_W83L785TS is not set CONFIG_SENSORS_W83L786NG=m # CONFIG_SENSORS_W83627HF is not set CONFIG_SENSORS_W83627EHF=m # CONFIG_SENSORS_WM831X is not set # CONFIG_SENSORS_WM8350 is not set # CONFIG_SENSORS_APPLESMC is not set CONFIG_SENSORS_MC13783_ADC=m CONFIG_THERMAL=y # CONFIG_THERMAL_DEFAULT_GOV_STEP_WISE is not set CONFIG_THERMAL_DEFAULT_GOV_FAIR_SHARE=y # CONFIG_THERMAL_DEFAULT_GOV_USER_SPACE is not set CONFIG_FAIR_SHARE=y # CONFIG_STEP_WISE is not set # CONFIG_USER_SPACE is not set # CONFIG_CPU_THERMAL is not set # CONFIG_WATCHDOG is not set CONFIG_SSB_POSSIBLE=y # # Sonics Silicon Backplane # CONFIG_SSB=y CONFIG_SSB_PCIHOST_POSSIBLE=y # CONFIG_SSB_PCIHOST is not set CONFIG_SSB_SILENT=y CONFIG_SSB_DRIVER_GPIO=y CONFIG_BCMA_POSSIBLE=y # # Broadcom specific AMBA # CONFIG_BCMA=m CONFIG_BCMA_HOST_PCI_POSSIBLE=y CONFIG_BCMA_HOST_PCI=y CONFIG_BCMA_DRIVER_GMAC_CMN=y CONFIG_BCMA_DRIVER_GPIO=y CONFIG_BCMA_DEBUG=y # # Multifunction device drivers # CONFIG_MFD_CORE=y # CONFIG_MFD_88PM860X is not set CONFIG_MFD_88PM800=y CONFIG_MFD_88PM805=m # CONFIG_MFD_SM501 is not set CONFIG_MFD_RTSX_PCI=m CONFIG_MFD_TI_AM335X_TSCADC=m # CONFIG_HTC_PASIC3 is not set CONFIG_HTC_I2CPLD=y # CONFIG_MFD_LM3533 is not set CONFIG_TPS6105X=m # CONFIG_TPS65010 is not set CONFIG_TPS6507X=m CONFIG_MFD_TPS65217=m # CONFIG_MFD_TPS6586X is not set CONFIG_MFD_TPS65910=y # CONFIG_MFD_TPS65912_I2C is not set # CONFIG_MFD_TPS65912_SPI is not set # CONFIG_MFD_TPS80031 is not set CONFIG_TWL4030_CORE=y CONFIG_TWL4030_MADC=m CONFIG_MFD_TWL4030_AUDIO=y # CONFIG_TWL6040_CORE is not set # CONFIG_MFD_STMPE is not set CONFIG_MFD_TC3589X=y # CONFIG_MFD_TMIO is not set CONFIG_MFD_SMSC=y # CONFIG_PMIC_DA903X is not set # CONFIG_MFD_DA9052_SPI is not set # CONFIG_MFD_DA9052_I2C is not set # CONFIG_MFD_DA9055 is not set # CONFIG_PMIC_ADP5520 is not set # CONFIG_MFD_LP8788 is not set CONFIG_MFD_MAX77686=y # CONFIG_MFD_MAX77693 is not set # CONFIG_MFD_MAX8907 is not set # CONFIG_MFD_MAX8925 is not set CONFIG_MFD_MAX8997=y CONFIG_MFD_MAX8998=y CONFIG_MFD_SEC_CORE=y CONFIG_MFD_ARIZONA=y # CONFIG_MFD_ARIZONA_I2C is not set CONFIG_MFD_ARIZONA_SPI=y CONFIG_MFD_WM5102=y # CONFIG_MFD_WM5110 is not set # CONFIG_MFD_WM8400 is not set CONFIG_MFD_WM831X=y CONFIG_MFD_WM831X_I2C=y # CONFIG_MFD_WM831X_SPI is not set CONFIG_MFD_WM8350=y CONFIG_MFD_WM8350_I2C=y CONFIG_MFD_WM8994=y CONFIG_MFD_PCF50633=m # CONFIG_PCF50633_ADC is not set CONFIG_PCF50633_GPIO=m CONFIG_MFD_MC13783=y CONFIG_MFD_MC13XXX=y CONFIG_MFD_MC13XXX_SPI=y CONFIG_MFD_MC13XXX_I2C=m # CONFIG_ABX500_CORE is not set CONFIG_EZX_PCAP=y # CONFIG_MFD_CS5535 is not set CONFIG_MFD_TIMBERDALE=y CONFIG_LPC_SCH=y CONFIG_LPC_ICH=y CONFIG_MFD_RDC321X=m CONFIG_MFD_JANZ_CMODIO=m CONFIG_MFD_VX855=y # CONFIG_MFD_WL1273_CORE is not set # CONFIG_MFD_TPS65090 is not set CONFIG_MFD_AAT2870_CORE=y CONFIG_MFD_RC5T583=y CONFIG_MFD_PALMAS=y # CONFIG_MFD_VIPERBOARD is not set CONFIG_MFD_RETU=m # CONFIG_MFD_AS3711 is not set CONFIG_REGULATOR=y # CONFIG_REGULATOR_DEBUG is not set # CONFIG_REGULATOR_DUMMY is not set CONFIG_REGULATOR_FIXED_VOLTAGE=m CONFIG_REGULATOR_VIRTUAL_CONSUMER=y # CONFIG_REGULATOR_USERSPACE_CONSUMER is not set # CONFIG_REGULATOR_GPIO is not set # CONFIG_REGULATOR_AD5398 is not set # CONFIG_REGULATOR_AAT2870 is not set CONFIG_REGULATOR_ARIZONA=y CONFIG_REGULATOR_FAN53555=m CONFIG_REGULATOR_MC13XXX_CORE=m CONFIG_REGULATOR_MC13783=m # CONFIG_REGULATOR_MC13892 is not set CONFIG_REGULATOR_ISL6271A=y CONFIG_REGULATOR_MAX1586=y # CONFIG_REGULATOR_MAX8649 is not set CONFIG_REGULATOR_MAX8660=m CONFIG_REGULATOR_MAX8952=m # CONFIG_REGULATOR_MAX8973 is not set CONFIG_REGULATOR_MAX8997=m CONFIG_REGULATOR_MAX8998=m CONFIG_REGULATOR_MAX77686=m # CONFIG_REGULATOR_PCAP is not set # CONFIG_REGULATOR_LP3971 is not set # CONFIG_REGULATOR_LP3972 is not set # CONFIG_REGULATOR_LP872X is not set CONFIG_REGULATOR_LP8755=m CONFIG_REGULATOR_PCF50633=m CONFIG_REGULATOR_RC5T583=m CONFIG_REGULATOR_S2MPS11=m CONFIG_REGULATOR_S5M8767=m CONFIG_REGULATOR_PALMAS=m # CONFIG_REGULATOR_TPS51632 is not set CONFIG_REGULATOR_TPS6105X=m # CONFIG_REGULATOR_TPS62360 is not set CONFIG_REGULATOR_TPS65023=y CONFIG_REGULATOR_TPS6507X=m # CONFIG_REGULATOR_TPS65217 is not set CONFIG_REGULATOR_TPS6524X=y CONFIG_REGULATOR_TPS65910=m CONFIG_REGULATOR_TWL4030=y CONFIG_REGULATOR_WM831X=y # CONFIG_REGULATOR_WM8350 is not set CONFIG_REGULATOR_WM8994=y CONFIG_MEDIA_SUPPORT=m # # Multimedia core support # CONFIG_MEDIA_CAMERA_SUPPORT=y # CONFIG_MEDIA_ANALOG_TV_SUPPORT is not set # CONFIG_MEDIA_DIGITAL_TV_SUPPORT is not set # CONFIG_MEDIA_RADIO_SUPPORT is not set # CONFIG_MEDIA_RC_SUPPORT is not set # CONFIG_MEDIA_CONTROLLER is not set CONFIG_VIDEO_DEV=m CONFIG_VIDEO_V4L2=m CONFIG_VIDEO_ADV_DEBUG=y # CONFIG_VIDEO_FIXED_MINOR_RANGES is not set CONFIG_VIDEO_TUNER=m CONFIG_V4L2_MEM2MEM_DEV=m CONFIG_VIDEOBUF_GEN=m CONFIG_VIDEOBUF_DMA_SG=m CONFIG_VIDEOBUF_VMALLOC=m CONFIG_VIDEOBUF2_CORE=m CONFIG_VIDEOBUF2_MEMOPS=m CONFIG_VIDEOBUF2_DMA_CONTIG=m CONFIG_VIDEOBUF2_VMALLOC=m # CONFIG_VIDEO_V4L2_INT_DEVICE is not set # CONFIG_TTPCI_EEPROM is not set # # Media drivers # CONFIG_MEDIA_USB_SUPPORT=y # # Webcam devices # # CONFIG_USB_VIDEO_CLASS is not set CONFIG_USB_GSPCA=m CONFIG_USB_M5602=m # CONFIG_USB_STV06XX is not set CONFIG_USB_GL860=m CONFIG_USB_GSPCA_BENQ=m CONFIG_USB_GSPCA_CONEX=m # CONFIG_USB_GSPCA_CPIA1 is not set # CONFIG_USB_GSPCA_ETOMS is not set CONFIG_USB_GSPCA_FINEPIX=m CONFIG_USB_GSPCA_JEILINJ=m CONFIG_USB_GSPCA_JL2005BCD=m CONFIG_USB_GSPCA_KINECT=m CONFIG_USB_GSPCA_KONICA=m CONFIG_USB_GSPCA_MARS=m # CONFIG_USB_GSPCA_MR97310A is not set CONFIG_USB_GSPCA_NW80X=m # CONFIG_USB_GSPCA_OV519 is not set CONFIG_USB_GSPCA_OV534=m CONFIG_USB_GSPCA_OV534_9=m CONFIG_USB_GSPCA_PAC207=m CONFIG_USB_GSPCA_PAC7302=m # CONFIG_USB_GSPCA_PAC7311 is not set # CONFIG_USB_GSPCA_SE401 is not set CONFIG_USB_GSPCA_SN9C2028=m CONFIG_USB_GSPCA_SN9C20X=m CONFIG_USB_GSPCA_SONIXB=m CONFIG_USB_GSPCA_SONIXJ=m CONFIG_USB_GSPCA_SPCA500=m CONFIG_USB_GSPCA_SPCA501=m # CONFIG_USB_GSPCA_SPCA505 is not set # CONFIG_USB_GSPCA_SPCA506 is not set CONFIG_USB_GSPCA_SPCA508=m CONFIG_USB_GSPCA_SPCA561=m # CONFIG_USB_GSPCA_SPCA1528 is not set # CONFIG_USB_GSPCA_SQ905 is not set CONFIG_USB_GSPCA_SQ905C=m # CONFIG_USB_GSPCA_SQ930X is not set # CONFIG_USB_GSPCA_STK014 is not set CONFIG_USB_GSPCA_STV0680=m # CONFIG_USB_GSPCA_SUNPLUS is not set CONFIG_USB_GSPCA_T613=m CONFIG_USB_GSPCA_TOPRO=m # CONFIG_USB_GSPCA_TV8532 is not set CONFIG_USB_GSPCA_VC032X=m # CONFIG_USB_GSPCA_VICAM is not set CONFIG_USB_GSPCA_XIRLINK_CIT=m CONFIG_USB_GSPCA_ZC3XX=m # CONFIG_USB_PWC is not set CONFIG_VIDEO_CPIA2=m CONFIG_USB_ZR364XX=m CONFIG_USB_STKWEBCAM=m CONFIG_USB_S2255=m # CONFIG_USB_SN9C102 is not set # # Webcam, TV (analog/digital) USB devices # CONFIG_VIDEO_EM28XX=m # CONFIG_VIDEO_EM28XX_ALSA is not set CONFIG_MEDIA_PCI_SUPPORT=y # # Media capture support # CONFIG_V4L_PLATFORM_DRIVERS=y # CONFIG_VIDEO_CAFE_CCIC is not set # CONFIG_SOC_CAMERA is not set # CONFIG_V4L_MEM2MEM_DRIVERS is not set CONFIG_V4L_TEST_DRIVERS=y CONFIG_VIDEO_MEM2MEM_TESTDEV=m # # Supported MMC/SDIO adapters # CONFIG_MEDIA_PARPORT_SUPPORT=y # CONFIG_VIDEO_BWQCAM is not set CONFIG_VIDEO_CQCAM=m CONFIG_VIDEO_W9966=m CONFIG_VIDEO_TVEEPROM=m # # Media ancillary drivers (tuners, sensors, i2c, frontends) # # CONFIG_MEDIA_SUBDRV_AUTOSELECT is not set # # Encoders, decoders, sensors and other helper chips # # # Audio decoders, processors and mixers # # CONFIG_VIDEO_TVAUDIO is not set CONFIG_VIDEO_TDA7432=m CONFIG_VIDEO_TDA9840=m # CONFIG_VIDEO_TEA6415C is not set CONFIG_VIDEO_TEA6420=m # CONFIG_VIDEO_MSP3400 is not set # CONFIG_VIDEO_CS5345 is not set CONFIG_VIDEO_CS53L32A=m CONFIG_VIDEO_TLV320AIC23B=m # CONFIG_VIDEO_WM8775 is not set CONFIG_VIDEO_WM8739=m CONFIG_VIDEO_VP27SMPX=m # # RDS decoders # CONFIG_VIDEO_SAA6588=m # # Video decoders # CONFIG_VIDEO_ADV7180=m CONFIG_VIDEO_ADV7183=m CONFIG_VIDEO_BT819=m CONFIG_VIDEO_BT856=m # CONFIG_VIDEO_BT866 is not set # CONFIG_VIDEO_KS0127 is not set # CONFIG_VIDEO_SAA7110 is not set # CONFIG_VIDEO_SAA711X is not set # CONFIG_VIDEO_SAA7191 is not set CONFIG_VIDEO_TVP514X=m CONFIG_VIDEO_TVP5150=m CONFIG_VIDEO_TVP7002=m CONFIG_VIDEO_VPX3220=m # # Video and audio decoders # # CONFIG_VIDEO_SAA717X is not set CONFIG_VIDEO_CX25840=m # # Video encoders # # CONFIG_VIDEO_SAA7127 is not set CONFIG_VIDEO_SAA7185=m CONFIG_VIDEO_ADV7170=m CONFIG_VIDEO_ADV7175=m CONFIG_VIDEO_ADV7343=m CONFIG_VIDEO_ADV7393=m # CONFIG_VIDEO_AK881X is not set # # Camera sensor devices # CONFIG_VIDEO_OV7670=m CONFIG_VIDEO_VS6624=m CONFIG_VIDEO_MT9V011=m CONFIG_VIDEO_SR030PC30=m # # Flash devices # # # Video improvement chips # # CONFIG_VIDEO_UPD64031A is not set CONFIG_VIDEO_UPD64083=m # # Miscelaneous helper chips # CONFIG_VIDEO_THS7303=m # CONFIG_VIDEO_M52790 is not set # # Sensors used on soc_camera driver # # # Customise DVB Frontends # # CONFIG_DVB_AU8522_V4L is not set CONFIG_DVB_TUNER_DIB0070=m CONFIG_DVB_TUNER_DIB0090=m # # Tools to develop new frontends # CONFIG_DVB_DUMMY_FE=m # # Graphics support # CONFIG_AGP=m CONFIG_AGP_ALI=m CONFIG_AGP_ATI=m # CONFIG_AGP_AMD is not set CONFIG_AGP_INTEL=m CONFIG_AGP_NVIDIA=m CONFIG_AGP_SIS=m CONFIG_AGP_SWORKS=m # CONFIG_AGP_VIA is not set # CONFIG_AGP_EFFICEON is not set # CONFIG_VGA_ARB is not set # CONFIG_DRM is not set # CONFIG_VGASTATE is not set CONFIG_VIDEO_OUTPUT_CONTROL=y # CONFIG_FB is not set CONFIG_EXYNOS_VIDEO=y CONFIG_BACKLIGHT_LCD_SUPPORT=y CONFIG_LCD_CLASS_DEVICE=y # CONFIG_LCD_L4F00242T03 is not set CONFIG_LCD_LMS283GF05=m CONFIG_LCD_LTV350QV=m CONFIG_LCD_ILI9320=y CONFIG_LCD_TDO24M=y CONFIG_LCD_VGG2432A4=y CONFIG_LCD_PLATFORM=m CONFIG_LCD_S6E63M0=m CONFIG_LCD_LD9040=m CONFIG_LCD_AMS369FG06=m # CONFIG_LCD_LMS501KF03 is not set # CONFIG_LCD_HX8357 is not set CONFIG_BACKLIGHT_CLASS_DEVICE=m CONFIG_BACKLIGHT_GENERIC=m CONFIG_BACKLIGHT_PWM=m CONFIG_BACKLIGHT_SAHARA=m # CONFIG_BACKLIGHT_WM831X is not set CONFIG_BACKLIGHT_ADP8860=m CONFIG_BACKLIGHT_ADP8870=m CONFIG_BACKLIGHT_PCF50633=m CONFIG_BACKLIGHT_AAT2870=m # CONFIG_BACKLIGHT_LM3630 is not set CONFIG_BACKLIGHT_LM3639=m # CONFIG_BACKLIGHT_LP855X is not set CONFIG_BACKLIGHT_PANDORA=m # CONFIG_BACKLIGHT_TPS65217 is not set CONFIG_SOUND=y CONFIG_SOUND_OSS_CORE=y # CONFIG_SOUND_OSS_CORE_PRECLAIM is not set CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_HWDEP=y CONFIG_SND_RAWMIDI=y CONFIG_SND_COMPRESS_OFFLOAD=y # CONFIG_SND_SEQUENCER is not set CONFIG_SND_OSSEMUL=y CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_PCM_OSS_PLUGINS=y CONFIG_SND_HRTIMER=m CONFIG_SND_DYNAMIC_MINORS=y CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_DEBUG is not set CONFIG_SND_VMASTER=y CONFIG_SND_DMA_SGBUF=y # CONFIG_SND_RAWMIDI_SEQ is not set # CONFIG_SND_OPL3_LIB_SEQ is not set # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set # CONFIG_SND_EMU10K1_SEQ is not set CONFIG_SND_VX_LIB=m CONFIG_SND_DRIVERS=y # CONFIG_SND_DUMMY is not set CONFIG_SND_ALOOP=y # CONFIG_SND_MTPAV is not set CONFIG_SND_MTS64=y CONFIG_SND_SERIAL_U16550=m # CONFIG_SND_MPU401 is not set # CONFIG_SND_PORTMAN2X4 is not set # CONFIG_SND_PCI is not set # CONFIG_SND_SPI is not set CONFIG_SND_USB=y # CONFIG_SND_USB_AUDIO is not set CONFIG_SND_USB_UA101=m CONFIG_SND_USB_USX2Y=m CONFIG_SND_USB_CAIAQ=y # CONFIG_SND_USB_US122L is not set CONFIG_SND_USB_6FIRE=y CONFIG_SND_PCMCIA=y CONFIG_SND_VXPOCKET=m # CONFIG_SND_PDAUDIOCF is not set CONFIG_SND_SOC=y CONFIG_SND_ATMEL_SOC=y CONFIG_SND_SOC_I2C_AND_SPI=y CONFIG_SND_SOC_ALL_CODECS=y CONFIG_SND_SOC_ARIZONA=y CONFIG_SND_SOC_WM_HUBS=y CONFIG_SND_SOC_WM_ADSP=y CONFIG_SND_SOC_AD1836=y CONFIG_SND_SOC_AD193X=y CONFIG_SND_SOC_AD73311=y CONFIG_SND_SOC_ADAU1373=y CONFIG_SND_SOC_ADAV80X=y CONFIG_SND_SOC_ADS117X=y CONFIG_SND_SOC_AK4104=y CONFIG_SND_SOC_AK4535=y CONFIG_SND_SOC_AK4641=y CONFIG_SND_SOC_AK4642=y CONFIG_SND_SOC_AK4671=y CONFIG_SND_SOC_ALC5623=y CONFIG_SND_SOC_ALC5632=y CONFIG_SND_SOC_CS42L51=y CONFIG_SND_SOC_CS42L52=y CONFIG_SND_SOC_CS42L73=y CONFIG_SND_SOC_CS4270=y CONFIG_SND_SOC_CS4271=y CONFIG_SND_SOC_CX20442=y CONFIG_SND_SOC_JZ4740_CODEC=y CONFIG_SND_SOC_L3=y CONFIG_SND_SOC_DA7210=y CONFIG_SND_SOC_DA7213=y CONFIG_SND_SOC_DA732X=y CONFIG_SND_SOC_DA9055=y CONFIG_SND_SOC_DFBMCS320=y CONFIG_SND_SOC_ISABELLE=y CONFIG_SND_SOC_LM49453=y CONFIG_SND_SOC_MAX98088=y CONFIG_SND_SOC_MAX98090=y CONFIG_SND_SOC_MAX98095=y CONFIG_SND_SOC_MAX9850=y CONFIG_SND_SOC_PCM3008=y CONFIG_SND_SOC_RT5631=y CONFIG_SND_SOC_SGTL5000=y CONFIG_SND_SOC_SPDIF=y CONFIG_SND_SOC_SSM2602=y CONFIG_SND_SOC_STA32X=y CONFIG_SND_SOC_STA529=y CONFIG_SND_SOC_TLV320AIC23=y CONFIG_SND_SOC_TLV320AIC26=y CONFIG_SND_SOC_TLV320AIC32X4=y CONFIG_SND_SOC_TLV320AIC3X=y CONFIG_SND_SOC_TLV320DAC33=y CONFIG_SND_SOC_TWL4030=y CONFIG_SND_SOC_UDA134X=y CONFIG_SND_SOC_UDA1380=y CONFIG_SND_SOC_WM0010=y CONFIG_SND_SOC_WM1250_EV1=y CONFIG_SND_SOC_WM2000=y CONFIG_SND_SOC_WM2200=y CONFIG_SND_SOC_WM5100=y CONFIG_SND_SOC_WM5102=y CONFIG_SND_SOC_WM8350=y CONFIG_SND_SOC_WM8510=y CONFIG_SND_SOC_WM8523=y CONFIG_SND_SOC_WM8580=y CONFIG_SND_SOC_WM8711=y CONFIG_SND_SOC_WM8727=y CONFIG_SND_SOC_WM8728=y CONFIG_SND_SOC_WM8731=y CONFIG_SND_SOC_WM8737=y CONFIG_SND_SOC_WM8741=y CONFIG_SND_SOC_WM8750=y CONFIG_SND_SOC_WM8753=y CONFIG_SND_SOC_WM8770=y CONFIG_SND_SOC_WM8776=y CONFIG_SND_SOC_WM8782=y CONFIG_SND_SOC_WM8804=y CONFIG_SND_SOC_WM8900=y CONFIG_SND_SOC_WM8903=y CONFIG_SND_SOC_WM8904=y CONFIG_SND_SOC_WM8940=y CONFIG_SND_SOC_WM8955=y CONFIG_SND_SOC_WM8960=y CONFIG_SND_SOC_WM8961=y CONFIG_SND_SOC_WM8962=y CONFIG_SND_SOC_WM8971=y CONFIG_SND_SOC_WM8974=y CONFIG_SND_SOC_WM8978=y CONFIG_SND_SOC_WM8983=y CONFIG_SND_SOC_WM8985=y CONFIG_SND_SOC_WM8988=y CONFIG_SND_SOC_WM8990=y CONFIG_SND_SOC_WM8991=y CONFIG_SND_SOC_WM8993=y CONFIG_SND_SOC_WM8994=y CONFIG_SND_SOC_WM8995=y CONFIG_SND_SOC_WM8996=y CONFIG_SND_SOC_WM9081=y CONFIG_SND_SOC_WM9090=y CONFIG_SND_SOC_LM4857=y CONFIG_SND_SOC_MAX9768=y CONFIG_SND_SOC_MAX9877=y CONFIG_SND_SOC_MC13783=y CONFIG_SND_SOC_ML26124=y CONFIG_SND_SOC_TPA6130A2=y # CONFIG_SND_SIMPLE_CARD is not set CONFIG_SOUND_PRIME=m # CONFIG_SOUND_OSS is not set # # HID support # CONFIG_HID=m # CONFIG_HIDRAW is not set CONFIG_UHID=m CONFIG_HID_GENERIC=m # # Special HID drivers # CONFIG_HID_A4TECH=m CONFIG_HID_ACRUX=m CONFIG_HID_ACRUX_FF=y CONFIG_HID_APPLE=m # CONFIG_HID_AUREAL is not set CONFIG_HID_BELKIN=m # CONFIG_HID_CHERRY is not set CONFIG_HID_CHICONY=m # CONFIG_HID_PRODIKEYS is not set # CONFIG_HID_CYPRESS is not set # CONFIG_HID_DRAGONRISE is not set CONFIG_HID_EMS_FF=m CONFIG_HID_EZKEY=m # CONFIG_HID_HOLTEK is not set # CONFIG_HID_KEYTOUCH is not set # CONFIG_HID_KYE is not set # CONFIG_HID_UCLOGIC is not set CONFIG_HID_WALTOP=m # CONFIG_HID_GYRATION is not set # CONFIG_HID_TWINHAN is not set CONFIG_HID_KENSINGTON=m # CONFIG_HID_LCPOWER is not set CONFIG_HID_LENOVO_TPKBD=m # CONFIG_HID_LOGITECH is not set CONFIG_HID_MICROSOFT=m # CONFIG_HID_MONTEREY is not set CONFIG_HID_MULTITOUCH=m CONFIG_HID_NTRIG=m CONFIG_HID_ORTEK=m # CONFIG_HID_PANTHERLORD is not set CONFIG_HID_PETALYNX=m # CONFIG_HID_PICOLCD is not set CONFIG_HID_PRIMAX=m CONFIG_HID_ROCCAT=m CONFIG_HID_SAITEK=m CONFIG_HID_SAMSUNG=m # CONFIG_HID_SONY is not set CONFIG_HID_SPEEDLINK=m CONFIG_HID_STEELSERIES=m # CONFIG_HID_SUNPLUS is not set CONFIG_HID_GREENASIA=m CONFIG_GREENASIA_FF=y # CONFIG_HID_SMARTJOYPLUS is not set # CONFIG_HID_TIVO is not set CONFIG_HID_TOPSEED=m # CONFIG_HID_THINGM is not set # CONFIG_HID_THRUSTMASTER is not set # CONFIG_HID_ZEROPLUS is not set CONFIG_HID_ZYDACRON=m CONFIG_HID_SENSOR_HUB=m # # USB HID support # CONFIG_USB_HID=m # CONFIG_HID_PID is not set CONFIG_USB_HIDDEV=y # # USB HID Boot Protocol drivers # CONFIG_USB_KBD=m CONFIG_USB_MOUSE=m # # I2C HID support # CONFIG_I2C_HID=m CONFIG_USB_ARCH_HAS_OHCI=y CONFIG_USB_ARCH_HAS_EHCI=y CONFIG_USB_ARCH_HAS_XHCI=y CONFIG_USB_SUPPORT=y CONFIG_USB_COMMON=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB=y # CONFIG_USB_DEBUG is not set # CONFIG_USB_ANNOUNCE_NEW_DEVICES is not set # # Miscellaneous USB options # CONFIG_USB_DYNAMIC_MINORS=y # CONFIG_USB_SUSPEND is not set # CONFIG_USB_OTG_WHITELIST is not set CONFIG_USB_OTG_BLACKLIST_HUB=y CONFIG_USB_DWC3=m # CONFIG_USB_DWC3_HOST is not set CONFIG_USB_DWC3_GADGET=y # CONFIG_USB_DWC3_DUAL_ROLE is not set # CONFIG_USB_DWC3_DEBUG is not set CONFIG_USB_MON=y CONFIG_USB_WUSB_CBAF=m CONFIG_USB_WUSB_CBAF_DEBUG=y # # USB Host Controller Drivers # CONFIG_USB_C67X00_HCD=y CONFIG_USB_XHCI_HCD=y CONFIG_USB_XHCI_PLATFORM=m # CONFIG_USB_XHCI_HCD_DEBUGGING is not set CONFIG_USB_EHCI_HCD=y CONFIG_USB_EHCI_ROOT_HUB_TT=y CONFIG_USB_EHCI_TT_NEWSCHED=y CONFIG_USB_EHCI_PCI=y CONFIG_USB_OXU210HP_HCD=y CONFIG_USB_ISP116X_HCD=m CONFIG_USB_ISP1760_HCD=m # CONFIG_USB_ISP1362_HCD is not set # CONFIG_USB_OHCI_HCD is not set CONFIG_USB_EHCI_HCD_PLATFORM=m CONFIG_USB_UHCI_HCD=y CONFIG_USB_U132_HCD=m CONFIG_USB_SL811_HCD=y # CONFIG_USB_SL811_HCD_ISO is not set # CONFIG_USB_SL811_CS is not set CONFIG_USB_R8A66597_HCD=m # CONFIG_USB_HCD_BCMA is not set CONFIG_USB_HCD_SSB=m CONFIG_USB_MUSB_HDRC=m # CONFIG_USB_MUSB_TUSB6010 is not set CONFIG_MUSB_PIO_ONLY=y CONFIG_USB_CHIPIDEA=y CONFIG_USB_CHIPIDEA_HOST=y # CONFIG_USB_CHIPIDEA_DEBUG is not set # CONFIG_USB_RENESAS_USBHS is not set # # USB Device Class drivers # CONFIG_USB_ACM=y CONFIG_USB_PRINTER=y CONFIG_USB_WDM=y CONFIG_USB_TMC=y # # NOTE: USB_STORAGE depends on SCSI but BLK_DEV_SD may # # # also be needed; see USB_STORAGE Help for more info # # # USB Imaging devices # CONFIG_USB_MDC800=m # # USB port drivers # CONFIG_USB_USS720=y CONFIG_USB_SERIAL=y CONFIG_USB_SERIAL_CONSOLE=y # CONFIG_USB_SERIAL_GENERIC is not set # CONFIG_USB_SERIAL_AIRCABLE is not set CONFIG_USB_SERIAL_ARK3116=m CONFIG_USB_SERIAL_BELKIN=y # CONFIG_USB_SERIAL_CH341 is not set CONFIG_USB_SERIAL_WHITEHEAT=y CONFIG_USB_SERIAL_DIGI_ACCELEPORT=y # CONFIG_USB_SERIAL_CP210X is not set CONFIG_USB_SERIAL_CYPRESS_M8=y # CONFIG_USB_SERIAL_EMPEG is not set # CONFIG_USB_SERIAL_FTDI_SIO is not set # CONFIG_USB_SERIAL_FUNSOFT is not set # CONFIG_USB_SERIAL_VISOR is not set CONFIG_USB_SERIAL_IPAQ=y # CONFIG_USB_SERIAL_IR is not set CONFIG_USB_SERIAL_EDGEPORT=m CONFIG_USB_SERIAL_EDGEPORT_TI=m CONFIG_USB_SERIAL_F81232=y CONFIG_USB_SERIAL_GARMIN=m CONFIG_USB_SERIAL_IPW=y CONFIG_USB_SERIAL_IUU=m CONFIG_USB_SERIAL_KEYSPAN_PDA=m # CONFIG_USB_SERIAL_KEYSPAN is not set CONFIG_USB_SERIAL_KLSI=m CONFIG_USB_SERIAL_KOBIL_SCT=m CONFIG_USB_SERIAL_MCT_U232=m # CONFIG_USB_SERIAL_METRO is not set CONFIG_USB_SERIAL_MOS7720=y CONFIG_USB_SERIAL_MOS7715_PARPORT=y CONFIG_USB_SERIAL_MOS7840=y CONFIG_USB_SERIAL_MOTOROLA=m CONFIG_USB_SERIAL_NAVMAN=y # CONFIG_USB_SERIAL_PL2303 is not set CONFIG_USB_SERIAL_OTI6858=m # CONFIG_USB_SERIAL_QCAUX is not set CONFIG_USB_SERIAL_QUALCOMM=m # CONFIG_USB_SERIAL_SPCP8X5 is not set CONFIG_USB_SERIAL_HP4X=m CONFIG_USB_SERIAL_SAFE=m CONFIG_USB_SERIAL_SAFE_PADDED=y CONFIG_USB_SERIAL_SIEMENS_MPI=m CONFIG_USB_SERIAL_SIERRAWIRELESS=y CONFIG_USB_SERIAL_SYMBOL=m CONFIG_USB_SERIAL_TI=m # CONFIG_USB_SERIAL_CYBERJACK is not set CONFIG_USB_SERIAL_XIRCOM=m CONFIG_USB_SERIAL_WWAN=y # CONFIG_USB_SERIAL_OPTION is not set CONFIG_USB_SERIAL_OMNINET=y CONFIG_USB_SERIAL_OPTICON=m CONFIG_USB_SERIAL_VIVOPAY_SERIAL=y CONFIG_USB_SERIAL_XSENS_MT=m CONFIG_USB_SERIAL_ZIO=m CONFIG_USB_SERIAL_ZTE=y CONFIG_USB_SERIAL_SSU100=y CONFIG_USB_SERIAL_QT2=y CONFIG_USB_SERIAL_DEBUG=y # # USB Miscellaneous drivers # # CONFIG_USB_EMI62 is not set # CONFIG_USB_EMI26 is not set CONFIG_USB_ADUTUX=y CONFIG_USB_SEVSEG=m CONFIG_USB_RIO500=y CONFIG_USB_LEGOTOWER=m # CONFIG_USB_LCD is not set # CONFIG_USB_LED is not set # CONFIG_USB_CYPRESS_CY7C63 is not set CONFIG_USB_CYTHERM=m # CONFIG_USB_IDMOUSE is not set CONFIG_USB_FTDI_ELAN=y CONFIG_USB_APPLEDISPLAY=m # CONFIG_USB_SISUSBVGA is not set # CONFIG_USB_LD is not set # CONFIG_USB_TRANCEVIBRATOR is not set CONFIG_USB_IOWARRIOR=m # CONFIG_USB_TEST is not set CONFIG_USB_ISIGHTFW=m # CONFIG_USB_YUREX is not set CONFIG_USB_EZUSB_FX2=y CONFIG_USB_HSIC_USB3503=m # # USB Physical Layer drivers # CONFIG_OMAP_USB3=y CONFIG_OMAP_CONTROL_USB=y CONFIG_USB_ISP1301=y CONFIG_USB_RCAR_PHY=y CONFIG_USB_GADGET=m CONFIG_USB_GADGET_DEBUG=y CONFIG_USB_GADGET_DEBUG_FS=y CONFIG_USB_GADGET_VBUS_DRAW=2 CONFIG_USB_GADGET_STORAGE_NUM_BUFFERS=2 # # USB Peripheral Controller # CONFIG_USB_R8A66597=m CONFIG_USB_MV_UDC=m # CONFIG_USB_GADGET_MUSB_HDRC is not set CONFIG_USB_M66592=m # CONFIG_USB_AMD5536UDC is not set CONFIG_USB_NET2272=m # CONFIG_USB_NET2272_DMA is not set CONFIG_USB_NET2280=m CONFIG_USB_GOKU=m # CONFIG_USB_EG20T is not set CONFIG_USB_DUMMY_HCD=m CONFIG_USB_LIBCOMPOSITE=m CONFIG_USB_F_ACM=m CONFIG_USB_F_SS_LB=m CONFIG_USB_U_SERIAL=m CONFIG_USB_ZERO=m # CONFIG_USB_AUDIO is not set CONFIG_USB_GADGETFS=m # CONFIG_USB_FUNCTIONFS is not set CONFIG_USB_G_SERIAL=m CONFIG_USB_MIDI_GADGET=m # CONFIG_USB_G_PRINTER is not set # CONFIG_USB_G_HID is not set CONFIG_USB_G_DBGP=m CONFIG_USB_G_DBGP_PRINTK=y # CONFIG_USB_G_DBGP_SERIAL is not set CONFIG_USB_G_WEBCAM=m # # OTG and related infrastructure # CONFIG_USB_OTG_UTILS=y CONFIG_USB_GPIO_VBUS=m # CONFIG_NOP_USB_XCEIV is not set # CONFIG_UWB is not set CONFIG_MMC=m CONFIG_MMC_DEBUG=y CONFIG_MMC_UNSAFE_RESUME=y # CONFIG_MMC_CLKGATE is not set # # MMC/SD/SDIO Card Drivers # # CONFIG_SDIO_UART is not set # CONFIG_MMC_TEST is not set # # MMC/SD/SDIO Host Controller Drivers # CONFIG_MMC_SDHCI=m CONFIG_MMC_SDHCI_PCI=m # CONFIG_MMC_RICOH_MMC is not set # CONFIG_MMC_SDHCI_PLTFM is not set CONFIG_MMC_WBSD=m CONFIG_MMC_TIFM_SD=m CONFIG_MMC_SPI=m # CONFIG_MMC_SDRICOH_CS is not set CONFIG_MMC_CB710=m CONFIG_MMC_VIA_SDMMC=m # CONFIG_MMC_VUB300 is not set CONFIG_MMC_USHC=m CONFIG_MMC_REALTEK_PCI=m CONFIG_MEMSTICK=y # CONFIG_MEMSTICK_DEBUG is not set # # MemoryStick drivers # CONFIG_MEMSTICK_UNSAFE_RESUME=y # # MemoryStick Host Controller Drivers # CONFIG_MEMSTICK_TIFM_MS=y # CONFIG_MEMSTICK_JMICRON_38X is not set # CONFIG_MEMSTICK_R592 is not set CONFIG_MEMSTICK_REALTEK_PCI=m CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=y # # LED drivers # # CONFIG_LEDS_LM3530 is not set # CONFIG_LEDS_LM3642 is not set CONFIG_LEDS_PCA9532=m CONFIG_LEDS_PCA9532_GPIO=y CONFIG_LEDS_GPIO=y CONFIG_LEDS_LP3944=y # CONFIG_LEDS_LP5521 is not set CONFIG_LEDS_LP5523=m CONFIG_LEDS_CLEVO_MAIL=m CONFIG_LEDS_PCA955X=y # CONFIG_LEDS_PCA9633 is not set # CONFIG_LEDS_WM831X_STATUS is not set CONFIG_LEDS_WM8350=m # CONFIG_LEDS_DAC124S085 is not set # CONFIG_LEDS_REGULATOR is not set CONFIG_LEDS_BD2802=m CONFIG_LEDS_INTEL_SS4200=m CONFIG_LEDS_LT3593=m CONFIG_LEDS_MC13783=m CONFIG_LEDS_TCA6507=y # CONFIG_LEDS_MAX8997 is not set # CONFIG_LEDS_LM355x is not set CONFIG_LEDS_OT200=y CONFIG_LEDS_BLINKM=y CONFIG_LEDS_TRIGGERS=y # # LED Triggers # CONFIG_LEDS_TRIGGER_TIMER=y CONFIG_LEDS_TRIGGER_ONESHOT=m CONFIG_LEDS_TRIGGER_HEARTBEAT=y # CONFIG_LEDS_TRIGGER_BACKLIGHT is not set CONFIG_LEDS_TRIGGER_CPU=y CONFIG_LEDS_TRIGGER_GPIO=y CONFIG_LEDS_TRIGGER_DEFAULT_ON=m # # iptables trigger is under Netfilter config (LED target) # CONFIG_LEDS_TRIGGER_TRANSIENT=y # CONFIG_ACCESSIBILITY is not set CONFIG_EDAC=y CONFIG_EDAC_LEGACY_SYSFS=y CONFIG_EDAC_DEBUG=y CONFIG_EDAC_MM_EDAC=m CONFIG_EDAC_AMD76X=m CONFIG_EDAC_E7XXX=m CONFIG_EDAC_E752X=m # CONFIG_EDAC_I82875P is not set CONFIG_EDAC_I82975X=m CONFIG_EDAC_I3000=m CONFIG_EDAC_I3200=m CONFIG_EDAC_X38=m CONFIG_EDAC_I5400=m # CONFIG_EDAC_I82860 is not set CONFIG_EDAC_R82600=m # CONFIG_EDAC_I5000 is not set # CONFIG_EDAC_I5100 is not set # CONFIG_EDAC_I7300 is not set CONFIG_RTC_LIB=y CONFIG_RTC_CLASS=y CONFIG_RTC_DEBUG=y # # RTC interfaces # # CONFIG_RTC_INTF_SYSFS is not set CONFIG_RTC_INTF_DEV=y # CONFIG_RTC_INTF_DEV_UIE_EMUL is not set CONFIG_RTC_DRV_TEST=m # # I2C RTC drivers # # CONFIG_RTC_DRV_88PM80X is not set CONFIG_RTC_DRV_DS1307=y # CONFIG_RTC_DRV_DS1374 is not set CONFIG_RTC_DRV_DS1672=y CONFIG_RTC_DRV_DS3232=y CONFIG_RTC_DRV_MAX6900=y CONFIG_RTC_DRV_MAX8998=m CONFIG_RTC_DRV_MAX8997=m # CONFIG_RTC_DRV_MAX77686 is not set # CONFIG_RTC_DRV_RS5C372 is not set # CONFIG_RTC_DRV_ISL1208 is not set CONFIG_RTC_DRV_ISL12022=y CONFIG_RTC_DRV_X1205=m CONFIG_RTC_DRV_PALMAS=m # CONFIG_RTC_DRV_PCF8523 is not set # CONFIG_RTC_DRV_PCF8563 is not set CONFIG_RTC_DRV_PCF8583=m CONFIG_RTC_DRV_M41T80=m CONFIG_RTC_DRV_M41T80_WDT=y # CONFIG_RTC_DRV_BQ32K is not set CONFIG_RTC_DRV_TWL4030=m CONFIG_RTC_DRV_TPS65910=y CONFIG_RTC_DRV_RC5T583=m CONFIG_RTC_DRV_S35390A=y # CONFIG_RTC_DRV_FM3130 is not set # CONFIG_RTC_DRV_RX8581 is not set # CONFIG_RTC_DRV_RX8025 is not set # CONFIG_RTC_DRV_EM3027 is not set # CONFIG_RTC_DRV_RV3029C2 is not set # # SPI RTC drivers # # CONFIG_RTC_DRV_M41T93 is not set # CONFIG_RTC_DRV_M41T94 is not set # CONFIG_RTC_DRV_DS1305 is not set CONFIG_RTC_DRV_DS1390=y CONFIG_RTC_DRV_MAX6902=m # CONFIG_RTC_DRV_R9701 is not set CONFIG_RTC_DRV_RS5C348=y CONFIG_RTC_DRV_DS3234=y CONFIG_RTC_DRV_PCF2123=y CONFIG_RTC_DRV_RX4581=m # # Platform RTC drivers # CONFIG_RTC_DRV_CMOS=y CONFIG_RTC_DRV_DS1286=y CONFIG_RTC_DRV_DS1511=y CONFIG_RTC_DRV_DS1553=m CONFIG_RTC_DRV_DS1742=y CONFIG_RTC_DRV_STK17TA8=m CONFIG_RTC_DRV_M48T86=m CONFIG_RTC_DRV_M48T35=y CONFIG_RTC_DRV_M48T59=y # CONFIG_RTC_DRV_MSM6242 is not set CONFIG_RTC_DRV_BQ4802=y # CONFIG_RTC_DRV_RP5C01 is not set CONFIG_RTC_DRV_V3020=y CONFIG_RTC_DRV_DS2404=m CONFIG_RTC_DRV_WM831X=y # CONFIG_RTC_DRV_WM8350 is not set # CONFIG_RTC_DRV_PCF50633 is not set # # on-CPU RTC drivers # # CONFIG_RTC_DRV_PCAP is not set # CONFIG_RTC_DRV_MC13XXX is not set # # HID Sensor RTC drivers # CONFIG_RTC_DRV_HID_SENSOR_TIME=m # CONFIG_DMADEVICES is not set CONFIG_AUXDISPLAY=y # CONFIG_KS0108 is not set CONFIG_UIO=m # CONFIG_UIO_CIF is not set CONFIG_UIO_PDRV=m CONFIG_UIO_PDRV_GENIRQ=m # CONFIG_UIO_DMEM_GENIRQ is not set CONFIG_UIO_AEC=m CONFIG_UIO_SERCOS3=m CONFIG_UIO_PCI_GENERIC=m CONFIG_UIO_NETX=m CONFIG_VIRTIO=y # # Virtio drivers # CONFIG_VIRTIO_PCI=m # CONFIG_VIRTIO_BALLOON is not set # CONFIG_VIRTIO_MMIO is not set # # Microsoft Hyper-V guest support # CONFIG_STAGING=y CONFIG_ECHO=m CONFIG_COMEDI=m # CONFIG_COMEDI_DEBUG is not set CONFIG_COMEDI_DEFAULT_BUF_SIZE_KB=2048 CONFIG_COMEDI_DEFAULT_BUF_MAXSIZE_KB=20480 CONFIG_COMEDI_MISC_DRIVERS=y CONFIG_COMEDI_KCOMEDILIB=m # CONFIG_COMEDI_BOND is not set CONFIG_COMEDI_TEST=m CONFIG_COMEDI_PARPORT=m # CONFIG_COMEDI_SERIAL2002 is not set CONFIG_COMEDI_SKEL=m CONFIG_COMEDI_PCI_DRIVERS=y CONFIG_COMEDI_8255_PCI=m CONFIG_COMEDI_ADDI_WATCHDOG=m # CONFIG_COMEDI_ADDI_APCI_035 is not set # CONFIG_COMEDI_ADDI_APCI_1032 is not set # CONFIG_COMEDI_ADDI_APCI_1500 is not set # CONFIG_COMEDI_ADDI_APCI_1516 is not set CONFIG_COMEDI_ADDI_APCI_1564=m CONFIG_COMEDI_ADDI_APCI_16XX=m CONFIG_COMEDI_ADDI_APCI_2032=m CONFIG_COMEDI_ADDI_APCI_2200=m CONFIG_COMEDI_ADDI_APCI_3120=m CONFIG_COMEDI_ADDI_APCI_3501=m CONFIG_COMEDI_ADDI_APCI_3XXX=m CONFIG_COMEDI_ADL_PCI6208=m CONFIG_COMEDI_ADL_PCI7X3X=m CONFIG_COMEDI_ADL_PCI8164=m CONFIG_COMEDI_ADL_PCI9111=m # CONFIG_COMEDI_ADL_PCI9118 is not set CONFIG_COMEDI_ADV_PCI1710=m # CONFIG_COMEDI_ADV_PCI1723 is not set CONFIG_COMEDI_ADV_PCI_DIO=m # CONFIG_COMEDI_AMPLC_DIO200_PCI is not set CONFIG_COMEDI_AMPLC_PC236_PCI=m # CONFIG_COMEDI_AMPLC_PC263_PCI is not set CONFIG_COMEDI_AMPLC_PCI224=m # CONFIG_COMEDI_AMPLC_PCI230 is not set # CONFIG_COMEDI_CONTEC_PCI_DIO is not set CONFIG_COMEDI_DAS08_PCI=m # CONFIG_COMEDI_DT3000 is not set CONFIG_COMEDI_DYNA_PCI10XX=m # CONFIG_COMEDI_UNIOXX5 is not set CONFIG_COMEDI_GSC_HPDI=m CONFIG_COMEDI_ICP_MULTI=m CONFIG_COMEDI_II_PCI20KC=m # CONFIG_COMEDI_DAQBOARD2000 is not set CONFIG_COMEDI_JR3_PCI=m CONFIG_COMEDI_KE_COUNTER=m CONFIG_COMEDI_CB_PCIDAS64=m CONFIG_COMEDI_CB_PCIDAS=m # CONFIG_COMEDI_CB_PCIDDA is not set CONFIG_COMEDI_CB_PCIMDAS=m CONFIG_COMEDI_CB_PCIMDDA=m # CONFIG_COMEDI_ME4000 is not set CONFIG_COMEDI_ME_DAQ=m CONFIG_COMEDI_NI_6527=m CONFIG_COMEDI_NI_65XX=m CONFIG_COMEDI_NI_660X=m CONFIG_COMEDI_NI_670X=m # CONFIG_COMEDI_NI_LABPC is not set # CONFIG_COMEDI_NI_PCIDIO is not set CONFIG_COMEDI_NI_PCIMIO=m CONFIG_COMEDI_RTD520=m # CONFIG_COMEDI_S526 is not set CONFIG_COMEDI_S626=m CONFIG_COMEDI_SSV_DNP=m CONFIG_COMEDI_MITE=m CONFIG_COMEDI_NI_TIOCMD=m CONFIG_COMEDI_PCMCIA_DRIVERS=y # CONFIG_COMEDI_CB_DAS16_CS is not set # CONFIG_COMEDI_DAS08_CS is not set # CONFIG_COMEDI_NI_DAQ_700_CS is not set CONFIG_COMEDI_NI_DAQ_DIO24_CS=m CONFIG_COMEDI_NI_MIO_CS=m # CONFIG_COMEDI_QUATECH_DAQP_CS is not set # CONFIG_COMEDI_USB_DRIVERS is not set CONFIG_COMEDI_8255=m CONFIG_COMEDI_FC=m CONFIG_COMEDI_AMPLC_PC236=m CONFIG_COMEDI_DAS08=m CONFIG_COMEDI_NI_TIO=m # CONFIG_ASUS_OLED is not set CONFIG_PANEL=y CONFIG_PANEL_PARPORT=0 CONFIG_PANEL_PROFILE=5 CONFIG_PANEL_CHANGE_MESSAGE=y CONFIG_PANEL_BOOT_MESSAGE="" CONFIG_TRANZPORT=m CONFIG_LINE6_USB=m CONFIG_LINE6_USB_IMPULSE_RESPONSE=y CONFIG_USB_SERIAL_QUATECH2=m CONFIG_DX_SEP=m # # IIO staging drivers # # CONFIG_IIO_ST_HWMON is not set # # Accelerometers # # CONFIG_ADIS16201 is not set CONFIG_ADIS16203=m CONFIG_ADIS16204=m CONFIG_ADIS16209=y CONFIG_ADIS16220=m # CONFIG_ADIS16240 is not set # CONFIG_LIS3L02DQ is not set # CONFIG_SCA3000 is not set # # Analog to digital converters # CONFIG_AD7291=y CONFIG_AD7606=m CONFIG_AD7606_IFACE_PARALLEL=m # CONFIG_AD7606_IFACE_SPI is not set CONFIG_AD799X=m CONFIG_AD799X_RING_BUFFER=y CONFIG_AD7780=m # CONFIG_AD7816 is not set # CONFIG_AD7192 is not set # CONFIG_ADT7410 is not set CONFIG_AD7280=m # # Analog digital bi-direction converters # # CONFIG_ADT7316 is not set # # Capacitance to digital converters # CONFIG_AD7150=m CONFIG_AD7152=y CONFIG_AD7746=y # # Direct Digital Synthesis # # CONFIG_AD5930 is not set CONFIG_AD9832=m CONFIG_AD9834=y # CONFIG_AD9850 is not set # CONFIG_AD9852 is not set CONFIG_AD9910=m # CONFIG_AD9951 is not set # # Digital gyroscope sensors # # CONFIG_ADIS16060 is not set # CONFIG_ADIS16130 is not set # CONFIG_ADIS16260 is not set # # Network Analyzer, Impedance Converters # CONFIG_AD5933=m # # Light sensors # CONFIG_SENSORS_ISL29018=y CONFIG_SENSORS_ISL29028=y CONFIG_TSL2583=y # CONFIG_TSL2x7x is not set # # Magnetometer sensors # # CONFIG_SENSORS_AK8975 is not set CONFIG_SENSORS_HMC5843=y # # Active energy metering IC # # CONFIG_ADE7753 is not set CONFIG_ADE7754=y CONFIG_ADE7758=m # CONFIG_ADE7759 is not set CONFIG_ADE7854=m CONFIG_ADE7854_I2C=m # CONFIG_ADE7854_SPI is not set # # Resolver to digital converters # CONFIG_AD2S90=y CONFIG_AD2S1200=y CONFIG_AD2S1210=y # # Triggers - standalone # CONFIG_IIO_PERIODIC_RTC_TRIGGER=y # CONFIG_IIO_GPIO_TRIGGER is not set # CONFIG_IIO_SYSFS_TRIGGER is not set CONFIG_IIO_DUMMY_EVGEN=y CONFIG_IIO_SIMPLE_DUMMY=y CONFIG_IIO_SIMPLE_DUMMY_EVENTS=y CONFIG_IIO_SIMPLE_DUMMY_BUFFER=y # CONFIG_ZSMALLOC is not set CONFIG_CRYSTALHD=y # CONFIG_FT1000 is not set # # Speakup console speech # CONFIG_TOUCHSCREEN_CLEARPAD_TM1217=m CONFIG_TOUCHSCREEN_SYNAPTICS_I2C_RMI4=m CONFIG_STAGING_MEDIA=y CONFIG_VIDEO_DT3155=m CONFIG_DT3155_CCIR=y # CONFIG_DT3155_STREAMING is not set CONFIG_SOLO6X10=m # # Android # # CONFIG_ANDROID is not set # CONFIG_NET_VENDOR_SILICOM is not set CONFIG_CED1401=m CONFIG_DGRP=m CONFIG_X86_PLATFORM_DEVICES=y CONFIG_DELL_LAPTOP=m CONFIG_SENSORS_HDAPS=m # CONFIG_IBM_RTL is not set # CONFIG_SAMSUNG_LAPTOP is not set CONFIG_SAMSUNG_Q10=m # # Hardware Spinlock drivers # CONFIG_CLKSRC_I8253=y CONFIG_CLKEVT_I8253=y CONFIG_CLKBLD_I8253=y # CONFIG_MAILBOX is not set # CONFIG_IOMMU_SUPPORT is not set # # Remoteproc drivers # CONFIG_REMOTEPROC=y CONFIG_STE_MODEM_RPROC=y # # Rpmsg drivers # CONFIG_VIRT_DRIVERS=y CONFIG_PM_DEVFREQ=y # # DEVFREQ Governors # CONFIG_DEVFREQ_GOV_SIMPLE_ONDEMAND=y CONFIG_DEVFREQ_GOV_PERFORMANCE=y CONFIG_DEVFREQ_GOV_POWERSAVE=y CONFIG_DEVFREQ_GOV_USERSPACE=y # # DEVFREQ Drivers # CONFIG_EXTCON=y # # Extcon Device Drivers # CONFIG_EXTCON_GPIO=y CONFIG_EXTCON_ADC_JACK=m # CONFIG_EXTCON_MAX8997 is not set CONFIG_EXTCON_ARIZONA=m CONFIG_MEMORY=y CONFIG_IIO=y CONFIG_IIO_BUFFER=y # CONFIG_IIO_BUFFER_CB is not set CONFIG_IIO_KFIFO_BUF=y CONFIG_IIO_TRIGGERED_BUFFER=y CONFIG_IIO_TRIGGER=y CONFIG_IIO_CONSUMERS_PER_TRIGGER=2 # # Accelerometers # CONFIG_HID_SENSOR_ACCEL_3D=m CONFIG_KXSD9=y # CONFIG_IIO_ST_ACCEL_3AXIS is not set # # Analog to digital converters # CONFIG_AD_SIGMA_DELTA=m CONFIG_AD7266=m CONFIG_AD7298=m CONFIG_AD7791=m # CONFIG_AD7793 is not set CONFIG_AD7476=m CONFIG_AD7887=y CONFIG_MAX1363=m CONFIG_TI_ADC081C=m CONFIG_TI_AM335X_ADC=m # # Amplifiers # CONFIG_AD8366=m # # Hid Sensor IIO Common # CONFIG_HID_SENSOR_IIO_COMMON=m CONFIG_HID_SENSOR_IIO_TRIGGER=m CONFIG_HID_SENSOR_ENUM_BASE_QUIRKS=y CONFIG_IIO_ST_SENSORS_I2C=y CONFIG_IIO_ST_SENSORS_SPI=y CONFIG_IIO_ST_SENSORS_CORE=y # # Digital to analog converters # CONFIG_AD5064=m # CONFIG_AD5360 is not set CONFIG_AD5380=y # CONFIG_AD5421 is not set CONFIG_AD5624R_SPI=y # CONFIG_AD5446 is not set CONFIG_AD5449=y # CONFIG_AD5504 is not set CONFIG_AD5755=y # CONFIG_AD5764 is not set CONFIG_AD5791=m CONFIG_AD5686=m CONFIG_MAX517=m CONFIG_MCP4725=m # # Frequency Synthesizers DDS/PLL # # # Clock Generator/Distribution # CONFIG_AD9523=m # # Phase-Locked Loop (PLL) frequency synthesizers # CONFIG_ADF4350=y # # Digital gyroscope sensors # # CONFIG_ADIS16080 is not set CONFIG_ADIS16136=m # CONFIG_ADXRS450 is not set # CONFIG_HID_SENSOR_GYRO_3D is not set CONFIG_IIO_ST_GYRO_3AXIS=y CONFIG_IIO_ST_GYRO_I2C_3AXIS=y CONFIG_IIO_ST_GYRO_SPI_3AXIS=y CONFIG_ITG3200=y # # Inertial measurement units # # CONFIG_ADIS16400 is not set # CONFIG_ADIS16480 is not set CONFIG_IIO_ADIS_LIB=y CONFIG_IIO_ADIS_LIB_BUFFER=y # CONFIG_INV_MPU6050_IIO is not set # # Light sensors # # CONFIG_ADJD_S311 is not set # CONFIG_SENSORS_TSL2563 is not set # CONFIG_VCNL4000 is not set CONFIG_HID_SENSOR_ALS=m # # Magnetometer sensors # # CONFIG_HID_SENSOR_MAGNETOMETER_3D is not set CONFIG_IIO_ST_MAGN_3AXIS=m CONFIG_IIO_ST_MAGN_I2C_3AXIS=m CONFIG_IIO_ST_MAGN_SPI_3AXIS=m CONFIG_VME_BUS=y # # VME Bridge Drivers # # CONFIG_VME_CA91CX42 is not set # CONFIG_VME_TSI148 is not set # # VME Board Drivers # # CONFIG_VMIVME_7805 is not set # # VME Device Drivers # CONFIG_VME_USER=y CONFIG_VME_PIO2=m CONFIG_PWM=y CONFIG_PWM_TWL=m CONFIG_PWM_TWL_LED=m CONFIG_IPACK_BUS=m CONFIG_BOARD_TPCI200=m # CONFIG_SERIAL_IPOCTAL is not set # # Firmware Drivers # CONFIG_EDD=m CONFIG_EDD_OFF=y CONFIG_FIRMWARE_MEMMAP=y CONFIG_DELL_RBU=y CONFIG_DCDBAS=m # CONFIG_DMIID is not set CONFIG_DMI_SYSFS=y # CONFIG_ISCSI_IBFT_FIND is not set CONFIG_GOOGLE_FIRMWARE=y # # Google Firmware Drivers # # CONFIG_GOOGLE_MEMCONSOLE is not set # # File systems # CONFIG_DCACHE_WORD_ACCESS=y # CONFIG_FS_POSIX_ACL is not set CONFIG_FILE_LOCKING=y # CONFIG_FSNOTIFY is not set # CONFIG_DNOTIFY is not set # CONFIG_INOTIFY_USER is not set # CONFIG_FANOTIFY is not set # CONFIG_QUOTA is not set # CONFIG_QUOTACTL is not set CONFIG_AUTOFS4_FS=m CONFIG_FUSE_FS=m # CONFIG_CUSE is not set # # Caches # CONFIG_FSCACHE=y # CONFIG_FSCACHE_DEBUG is not set # # Pseudo filesystems # # CONFIG_PROC_FS is not set CONFIG_SYSFS=y # CONFIG_TMPFS is not set # CONFIG_HUGETLBFS is not set # CONFIG_HUGETLB_PAGE is not set CONFIG_CONFIGFS_FS=y CONFIG_MISC_FILESYSTEMS=y CONFIG_ECRYPT_FS=y # CONFIG_PSTORE is not set CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=m # CONFIG_NLS_CODEPAGE_737 is not set CONFIG_NLS_CODEPAGE_775=m CONFIG_NLS_CODEPAGE_850=m CONFIG_NLS_CODEPAGE_852=y CONFIG_NLS_CODEPAGE_855=y # CONFIG_NLS_CODEPAGE_857 is not set CONFIG_NLS_CODEPAGE_860=y CONFIG_NLS_CODEPAGE_861=y # CONFIG_NLS_CODEPAGE_862 is not set CONFIG_NLS_CODEPAGE_863=m CONFIG_NLS_CODEPAGE_864=y CONFIG_NLS_CODEPAGE_865=m CONFIG_NLS_CODEPAGE_866=m CONFIG_NLS_CODEPAGE_869=y CONFIG_NLS_CODEPAGE_936=m CONFIG_NLS_CODEPAGE_950=m # CONFIG_NLS_CODEPAGE_932 is not set # CONFIG_NLS_CODEPAGE_949 is not set # CONFIG_NLS_CODEPAGE_874 is not set # CONFIG_NLS_ISO8859_8 is not set CONFIG_NLS_CODEPAGE_1250=m CONFIG_NLS_CODEPAGE_1251=m CONFIG_NLS_ASCII=y # CONFIG_NLS_ISO8859_1 is not set CONFIG_NLS_ISO8859_2=y CONFIG_NLS_ISO8859_3=y # CONFIG_NLS_ISO8859_4 is not set CONFIG_NLS_ISO8859_5=m # CONFIG_NLS_ISO8859_6 is not set CONFIG_NLS_ISO8859_7=m CONFIG_NLS_ISO8859_9=m CONFIG_NLS_ISO8859_13=m CONFIG_NLS_ISO8859_14=y CONFIG_NLS_ISO8859_15=y # CONFIG_NLS_KOI8_R is not set # CONFIG_NLS_KOI8_U is not set CONFIG_NLS_MAC_ROMAN=m CONFIG_NLS_MAC_CELTIC=y CONFIG_NLS_MAC_CENTEURO=y CONFIG_NLS_MAC_CROATIAN=y CONFIG_NLS_MAC_CYRILLIC=y # CONFIG_NLS_MAC_GAELIC is not set # CONFIG_NLS_MAC_GREEK is not set # CONFIG_NLS_MAC_ICELAND is not set # CONFIG_NLS_MAC_INUIT is not set CONFIG_NLS_MAC_ROMANIAN=y # CONFIG_NLS_MAC_TURKISH is not set CONFIG_NLS_UTF8=y # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y CONFIG_PRINTK_TIME=y CONFIG_DEFAULT_MESSAGE_LOGLEVEL=4 # CONFIG_ENABLE_WARN_DEPRECATED is not set # CONFIG_ENABLE_MUST_CHECK is not set CONFIG_FRAME_WARN=1024 CONFIG_MAGIC_SYSRQ=y # CONFIG_STRIP_ASM_SYMS is not set # CONFIG_READABLE_ASM is not set CONFIG_UNUSED_SYMBOLS=y CONFIG_DEBUG_FS=y CONFIG_HEADERS_CHECK=y # CONFIG_DEBUG_SECTION_MISMATCH is not set CONFIG_DEBUG_KERNEL=y CONFIG_DEBUG_SHIRQ=y # CONFIG_LOCKUP_DETECTOR is not set CONFIG_PANIC_ON_OOPS=y CONFIG_PANIC_ON_OOPS_VALUE=1 # CONFIG_DETECT_HUNG_TASK is not set # CONFIG_DEBUG_OBJECTS is not set # CONFIG_DEBUG_SLAB is not set CONFIG_HAVE_DEBUG_KMEMLEAK=y # CONFIG_DEBUG_KMEMLEAK is not set CONFIG_DEBUG_PREEMPT=y CONFIG_DEBUG_RT_MUTEXES=y CONFIG_DEBUG_PI_LIST=y # CONFIG_RT_MUTEX_TESTER is not set CONFIG_DEBUG_SPINLOCK=y CONFIG_DEBUG_MUTEXES=y CONFIG_DEBUG_LOCK_ALLOC=y # CONFIG_PROVE_LOCKING is not set CONFIG_LOCKDEP=y CONFIG_LOCK_STAT=y CONFIG_DEBUG_LOCKDEP=y CONFIG_DEBUG_ATOMIC_SLEEP=y # CONFIG_DEBUG_LOCKING_API_SELFTESTS is not set CONFIG_STACKTRACE=y # CONFIG_DEBUG_STACK_USAGE is not set # CONFIG_DEBUG_KOBJECT is not set # CONFIG_DEBUG_BUGVERBOSE is not set # CONFIG_DEBUG_INFO is not set CONFIG_DEBUG_VM=y # CONFIG_DEBUG_VM_RB is not set CONFIG_DEBUG_VIRTUAL=y # CONFIG_DEBUG_WRITECOUNT is not set # CONFIG_DEBUG_MEMORY_INIT is not set CONFIG_DEBUG_LIST=y # CONFIG_TEST_LIST_SORT is not set CONFIG_DEBUG_SG=y # CONFIG_DEBUG_NOTIFIERS is not set CONFIG_DEBUG_CREDENTIALS=y CONFIG_ARCH_WANT_FRAME_POINTERS=y CONFIG_FRAME_POINTER=y CONFIG_BOOT_PRINTK_DELAY=y # # RCU Debugging # # CONFIG_PROVE_RCU_DELAY is not set CONFIG_SPARSE_RCU_POINTER=y # CONFIG_RCU_TORTURE_TEST is not set CONFIG_RCU_CPU_STALL_TIMEOUT=21 CONFIG_RCU_CPU_STALL_VERBOSE=y CONFIG_RCU_CPU_STALL_INFO=y # CONFIG_RCU_TRACE is not set # CONFIG_BACKTRACE_SELF_TEST is not set # CONFIG_DEBUG_FORCE_WEAK_PER_CPU is not set # CONFIG_NOTIFIER_ERROR_INJECTION is not set CONFIG_FAULT_INJECTION=y CONFIG_FAILSLAB=y CONFIG_FAIL_PAGE_ALLOC=y # CONFIG_FAIL_MMC_REQUEST is not set # CONFIG_FAULT_INJECTION_DEBUG_FS is not set # CONFIG_DEBUG_PAGEALLOC is not set CONFIG_USER_STACKTRACE_SUPPORT=y CONFIG_NOP_TRACER=y CONFIG_HAVE_FUNCTION_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_FP_TEST=y CONFIG_HAVE_FUNCTION_TRACE_MCOUNT_TEST=y CONFIG_HAVE_DYNAMIC_FTRACE=y CONFIG_HAVE_DYNAMIC_FTRACE_WITH_REGS=y CONFIG_HAVE_FTRACE_MCOUNT_RECORD=y CONFIG_HAVE_SYSCALL_TRACEPOINTS=y CONFIG_HAVE_C_RECORDMCOUNT=y CONFIG_TRACER_MAX_TRACE=y CONFIG_TRACE_CLOCK=y CONFIG_RING_BUFFER=y CONFIG_EVENT_TRACING=y CONFIG_CONTEXT_SWITCH_TRACER=y CONFIG_RING_BUFFER_ALLOW_SWAP=y CONFIG_TRACING=y CONFIG_GENERIC_TRACER=y CONFIG_TRACING_SUPPORT=y CONFIG_FTRACE=y CONFIG_FUNCTION_TRACER=y # CONFIG_IRQSOFF_TRACER is not set CONFIG_PREEMPT_TRACER=y CONFIG_SCHED_TRACER=y CONFIG_FTRACE_SYSCALLS=y CONFIG_TRACER_SNAPSHOT=y CONFIG_TRACE_BRANCH_PROFILING=y # CONFIG_BRANCH_PROFILE_NONE is not set # CONFIG_PROFILE_ANNOTATED_BRANCHES is not set CONFIG_PROFILE_ALL_BRANCHES=y # CONFIG_BRANCH_TRACER is not set CONFIG_STACK_TRACER=y CONFIG_UPROBE_EVENT=y CONFIG_PROBE_EVENTS=y CONFIG_DYNAMIC_FTRACE=y CONFIG_DYNAMIC_FTRACE_WITH_REGS=y CONFIG_FUNCTION_PROFILER=y CONFIG_FTRACE_MCOUNT_RECORD=y CONFIG_FTRACE_SELFTEST=y CONFIG_FTRACE_STARTUP_TEST=y # CONFIG_EVENT_TRACE_TEST_SYSCALLS is not set CONFIG_MMIOTRACE=y CONFIG_MMIOTRACE_TEST=m CONFIG_RING_BUFFER_BENCHMARK=y CONFIG_RBTREE_TEST=m CONFIG_INTERVAL_TREE_TEST=m CONFIG_PROVIDE_OHCI1394_DMA_INIT=y CONFIG_BUILD_DOCSRC=y # CONFIG_DYNAMIC_DEBUG is not set # CONFIG_DMA_API_DEBUG is not set # CONFIG_ATOMIC64_SELFTEST is not set # CONFIG_SAMPLES is not set CONFIG_HAVE_ARCH_KGDB=y CONFIG_KGDB=y CONFIG_KGDB_SERIAL_CONSOLE=y CONFIG_KGDB_TESTS=y CONFIG_KGDB_TESTS_ON_BOOT=y CONFIG_KGDB_TESTS_BOOT_STRING="V1F100" # CONFIG_KGDB_LOW_LEVEL_TRAP is not set # CONFIG_KGDB_KDB is not set CONFIG_HAVE_ARCH_KMEMCHECK=y # CONFIG_TEST_KSTRTOX is not set # CONFIG_STRICT_DEVMEM is not set # CONFIG_X86_VERBOSE_BOOTUP is not set CONFIG_EARLY_PRINTK=y CONFIG_EARLY_PRINTK_DBGP=y CONFIG_DEBUG_STACKOVERFLOW=y # CONFIG_X86_PTDUMP is not set # CONFIG_DEBUG_RODATA is not set # CONFIG_DEBUG_SET_MODULE_RONX is not set # CONFIG_DEBUG_NX_TEST is not set CONFIG_DOUBLEFAULT=y CONFIG_IOMMU_STRESS=y CONFIG_HAVE_MMIOTRACE_SUPPORT=y CONFIG_IO_DELAY_TYPE_0X80=0 CONFIG_IO_DELAY_TYPE_0XED=1 CONFIG_IO_DELAY_TYPE_UDELAY=2 CONFIG_IO_DELAY_TYPE_NONE=3 # CONFIG_IO_DELAY_0X80 is not set CONFIG_IO_DELAY_0XED=y # CONFIG_IO_DELAY_UDELAY is not set # CONFIG_IO_DELAY_NONE is not set CONFIG_DEFAULT_IO_DELAY_TYPE=1 CONFIG_DEBUG_BOOT_PARAMS=y # CONFIG_CPA_DEBUG is not set CONFIG_OPTIMIZE_INLINING=y CONFIG_DEBUG_NMI_SELFTEST=y # # Security options # CONFIG_KEYS=y # CONFIG_TRUSTED_KEYS is not set CONFIG_ENCRYPTED_KEYS=y # CONFIG_KEYS_DEBUG_PROC_KEYS is not set # CONFIG_SECURITY_DMESG_RESTRICT is not set CONFIG_SECURITY=y CONFIG_SECURITYFS=y CONFIG_SECURITY_NETWORK=y CONFIG_SECURITY_PATH=y # CONFIG_SECURITY_YAMA is not set CONFIG_INTEGRITY=y # CONFIG_INTEGRITY_SIGNATURE is not set CONFIG_IMA=y CONFIG_IMA_MEASURE_PCR_IDX=10 # CONFIG_IMA_APPRAISE is not set CONFIG_EVM=y CONFIG_EVM_HMAC_VERSION=2 CONFIG_DEFAULT_SECURITY_DAC=y CONFIG_DEFAULT_SECURITY="" CONFIG_CRYPTO=y # # Crypto core or helper # CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_ALGAPI2=y CONFIG_CRYPTO_AEAD=y CONFIG_CRYPTO_AEAD2=y CONFIG_CRYPTO_BLKCIPHER=y CONFIG_CRYPTO_BLKCIPHER2=y CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_HASH2=y CONFIG_CRYPTO_RNG=y CONFIG_CRYPTO_RNG2=y CONFIG_CRYPTO_PCOMP2=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_MANAGER2=y CONFIG_CRYPTO_MANAGER_DISABLE_TESTS=y CONFIG_CRYPTO_GF128MUL=y CONFIG_CRYPTO_NULL=m CONFIG_CRYPTO_WORKQUEUE=y CONFIG_CRYPTO_CRYPTD=y CONFIG_CRYPTO_AUTHENC=y CONFIG_CRYPTO_TEST=m CONFIG_CRYPTO_ABLK_HELPER_X86=y CONFIG_CRYPTO_GLUE_HELPER_X86=m # # Authenticated Encryption with Associated Data # # CONFIG_CRYPTO_CCM is not set CONFIG_CRYPTO_GCM=m CONFIG_CRYPTO_SEQIV=y # # Block modes # CONFIG_CRYPTO_CBC=y CONFIG_CRYPTO_CTR=m CONFIG_CRYPTO_CTS=m CONFIG_CRYPTO_ECB=y CONFIG_CRYPTO_LRW=y CONFIG_CRYPTO_PCBC=y CONFIG_CRYPTO_XTS=y # # Hash modes # CONFIG_CRYPTO_HMAC=y # CONFIG_CRYPTO_XCBC is not set # CONFIG_CRYPTO_VMAC is not set # # Digest # CONFIG_CRYPTO_CRC32C=y # CONFIG_CRYPTO_CRC32C_INTEL is not set CONFIG_CRYPTO_GHASH=m CONFIG_CRYPTO_MD4=y CONFIG_CRYPTO_MD5=y CONFIG_CRYPTO_MICHAEL_MIC=m CONFIG_CRYPTO_RMD128=y # CONFIG_CRYPTO_RMD160 is not set CONFIG_CRYPTO_RMD256=m # CONFIG_CRYPTO_RMD320 is not set CONFIG_CRYPTO_SHA1=y CONFIG_CRYPTO_SHA256=y CONFIG_CRYPTO_SHA512=y CONFIG_CRYPTO_TGR192=y # CONFIG_CRYPTO_WP512 is not set # # Ciphers # CONFIG_CRYPTO_AES=y CONFIG_CRYPTO_AES_586=y CONFIG_CRYPTO_AES_NI_INTEL=y # CONFIG_CRYPTO_ANUBIS is not set CONFIG_CRYPTO_ARC4=m CONFIG_CRYPTO_BLOWFISH=y CONFIG_CRYPTO_BLOWFISH_COMMON=y CONFIG_CRYPTO_CAMELLIA=m CONFIG_CRYPTO_CAST_COMMON=y CONFIG_CRYPTO_CAST5=y CONFIG_CRYPTO_CAST6=m CONFIG_CRYPTO_DES=y # CONFIG_CRYPTO_FCRYPT is not set CONFIG_CRYPTO_KHAZAD=m CONFIG_CRYPTO_SALSA20=y CONFIG_CRYPTO_SALSA20_586=y CONFIG_CRYPTO_SEED=m CONFIG_CRYPTO_SERPENT=m CONFIG_CRYPTO_SERPENT_SSE2_586=m CONFIG_CRYPTO_TEA=y CONFIG_CRYPTO_TWOFISH=m CONFIG_CRYPTO_TWOFISH_COMMON=m CONFIG_CRYPTO_TWOFISH_586=m # # Compression # CONFIG_CRYPTO_DEFLATE=m # CONFIG_CRYPTO_ZLIB is not set CONFIG_CRYPTO_LZO=m # # Random Number Generation # # CONFIG_CRYPTO_ANSI_CPRNG is not set CONFIG_CRYPTO_HW=y CONFIG_CRYPTO_DEV_PADLOCK=m # CONFIG_CRYPTO_DEV_PADLOCK_AES is not set # CONFIG_CRYPTO_DEV_PADLOCK_SHA is not set CONFIG_CRYPTO_DEV_GEODE=m CONFIG_CRYPTO_DEV_HIFN_795X=y # CONFIG_CRYPTO_DEV_HIFN_795X_RNG is not set CONFIG_ASYMMETRIC_KEY_TYPE=y CONFIG_ASYMMETRIC_PUBLIC_KEY_SUBTYPE=y CONFIG_PUBLIC_KEY_ALGO_RSA=y CONFIG_X509_CERTIFICATE_PARSER=y CONFIG_HAVE_KVM=y CONFIG_VIRTUALIZATION=y CONFIG_BINARY_PRINTF=y # # Library routines # CONFIG_BITREVERSE=y CONFIG_GENERIC_STRNCPY_FROM_USER=y CONFIG_GENERIC_STRNLEN_USER=y CONFIG_GENERIC_FIND_FIRST_BIT=y CONFIG_GENERIC_PCI_IOMAP=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_IO=y CONFIG_PERCPU_RWSEM=y CONFIG_CRC_CCITT=y CONFIG_CRC16=y CONFIG_CRC_T10DIF=m CONFIG_CRC_ITU_T=y CONFIG_CRC32=y # CONFIG_CRC32_SELFTEST is not set # CONFIG_CRC32_SLICEBY8 is not set # CONFIG_CRC32_SLICEBY4 is not set # CONFIG_CRC32_SARWATE is not set CONFIG_CRC32_BIT=y CONFIG_CRC7=y CONFIG_LIBCRC32C=y # CONFIG_CRC8 is not set CONFIG_ZLIB_INFLATE=m CONFIG_ZLIB_DEFLATE=m CONFIG_LZO_COMPRESS=y CONFIG_LZO_DECOMPRESS=y CONFIG_XZ_DEC=m # CONFIG_XZ_DEC_X86 is not set # CONFIG_XZ_DEC_POWERPC is not set # CONFIG_XZ_DEC_IA64 is not set CONFIG_XZ_DEC_ARM=y # CONFIG_XZ_DEC_ARMTHUMB is not set CONFIG_XZ_DEC_SPARC=y CONFIG_XZ_DEC_BCJ=y CONFIG_XZ_DEC_TEST=m CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y CONFIG_CHECK_SIGNATURE=y CONFIG_ARCH_HAS_ATOMIC64_DEC_IF_POSITIVE=y # CONFIG_AVERAGE is not set CONFIG_CLZ_TAB=y # CONFIG_CORDIC is not set # CONFIG_DDR is not set CONFIG_MPILIB=y CONFIG_OID_REGISTRY=y

1 0

[alsa-devel] Fwd: Re: Alsa USB Audio regression
by Ulrich Günther 19 Jul '13

19 Jul '13

-------- Original-Nachricht -------- Betreff: Re: [alsa-devel] Alsa USB Audio regression Datum: Tue, 16 Jul 2013 18:27:10 +0200 Von: Ulrich Günther <ulrich.guenther(a)gmx.at> An: Raymond Yau <superquad.vortex2(a)gmail.com> Am 2013-07-16 04:14, schrieb Raymond Yau: > > > >>> > >>> After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't > work at > >>> all, Ubuntu 11.10 and newer have the bug), the headset only plays > sound > >>> at maximum volume level if the volume is turned down just a little no > >>> sound is being played. > >>> > >>> I have filled a Ubuntu bug for 12.04 LTS at: > >>> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . > >>> I was advised to contact this mailing list. > >>> > >>> AlsaInfo for Ubuntu 12.04 LTS with the bug: > >>> > http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 > . > >>> AlsaInfo for Ubuntu 10.04 LTS without the bug: > >>> > http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc > >> > >> > >> > >> It is strange that dB range of your usb audio is 0 to 0.14 (only > small gain and no attenuation) > >> > >> In 10.04 LTS pulseaudio provide software volume control > >> > >> > >> ( 1.095| 0.000) I: alsa-sink.c: Driver does not support hardware > volume control, falling back to software volume control. > >> > >> > >> control.6 { > >> comment.access 'read write' > >> comment.type INTEGER > >> comment.count 1 comment.range '0 - 37' > >> comment.dbmin 0 comment.dbmax 14 > >> iface MIXER > >> name 'Speaker Playback Volume' value 30 } > >> > >> > >> In 12.04 pulseaudio did not provide software volume attenuation > and use the speaker playback volume control > > > > > > So does that mean that the bug is in pulseaudio, or could it be even > deeper like in the kernel or something? > > Are there any significant change in volume when you use "alsamixer > -c0" to change speaker playback volume from min 0dB to 0.14dB ? > > Did pulseaudio provide software attuentation -1.61dB when the range of > hardware volume control is only 0.14dB ? > > * index: 0 > name: <alsa_output.usb-0d8c_USB_Audio-00-Audio.analog-surround-71> > driver: <module-alsa-card.c> > flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY > DYNAMIC_LATENCY > state: SUSPENDED > suspend cause: IDLE > priority: 9049 > volume: 0: 94% 1: 94% 2: 94% 3: 94% 4: 94% 5: 94% 6: 94% 7: 94% > 0: -1,61 dB 1: -1,61 dB 2: -1,61 dB 3: -1,61 dB 4: -1,61 dB 5: > -1,61 dB 6: -1,61 dB 7: -1,61 dB > balance 0,00 > base volume: 99% > -0,14 dB > volume steps: 65537 > muted: no > > It is strange that 5.1 headset only support 8 channels playback > Okay i did try "alsamixer -c0" in Ubuntu 12.04 LTS, i selected the Usb Headset, Speaker was at 0 using the remote control from the headset i could bring the value to 86, so if the volume is turned down the value goes immideatly to 0. With the alsamixer i was able to regulate the value between 0 and 100, db-change is 0,01 at lowest and 0,14 at highest. I could increment/decrement the db-change in steps of 0,01 (with the up/down keys from the keyboard), the volume from the headset-speakers follows accordingly. When using anything besides the "alsamixer -c0" the volume can only be 0 or 86. I was able to play the test sounds for: front center, front left, front right, rear left,rear right and subwoofer every thing came from the right direction, as expected from a 5.1 headset. I was really surprised, thank you for pointing this out, it really didn't occur to me, maybe i saw the alsamixer some years ago but i really couldn't remember that it was there.

3 3

[alsa-devel] [PATCH] ASoC: jack: Use power efficient workqueue
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..7aa26b5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Fix error path
by Fabio Estevam 18 Jul '13

18 Jul '13

If the 'failed to find codec platform device' error path is executed, it should jump to 'fail' label instead of returning an error immediately. 'fail' label will then free the ssi_np and codec_np previously acquired nodes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a9..1d70e27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + ret = -EINVAL; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: tegra: depend on ARCH_TEGRA, not ARCH_TEGRA_*
by Stephen Warren 18 Jul '13

18 Jul '13

From: Stephen Warren <swarren(a)nvidia.com> I'm planning to remove CONFIG_ARCH_TEGRA_*_SOC, leaving just ARCH_TEGRA. This will reduce the number of configurations that need to be supported by core Tegra code, e.g. CPU idle, hotplug. As a precursor, we need to update Kconfig for all Tegra drivers not to reference ARCH_TEGRA_*_SOC. Stop selecting I2S/AC97/SPDIF controller drivers from the machine driver config options; this doesn't really work if we don't know which specific SoC(s) we're building for. However, set their default values based on SND_SOC_TEGRA, since most people will probably want to enable support for all SoCs. This also avoids the need to change any defconfig files. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 36 +++++++++++++++--------------------- 1 file changed, 15 insertions(+), 21 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 68fd6d0..321a6c9 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -7,8 +7,9 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -18,15 +19,16 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2C controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -34,9 +36,9 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC - default m + tristate "Tegra20 SPDIF controller" + default SND_SOC_TEGRA + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -44,15 +46,16 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -62,8 +65,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -72,8 +73,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -82,8 +81,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -92,8 +89,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB - select SND_SOC_TEGRA20_AC97 + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_WM9712 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -102,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -111,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the -- 1.8.1.5

2 8

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i8
by Chris J Arges 18 Jul '13

18 Jul '13

This patch is similar to commit 1762a59d8e8b5e99f6f4a0f292b40f3cacb108ba. The same non-standard interface descriptor causes snd_usb_create_mixer() to fail for the Focusrite Scarlett 18i8 as well. Signed-off-by: Chris J Arges <christopherarges(a)gmail.com> --- sound/usb/quirks-table.h | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7f1585b..b954ec5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3232,6 +3232,52 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Focusrite Scarlett 18i8 + * + * Avoid mixer creation similar to the Scarlett 18i6 interface. + */ + USB_DEVICE(0x1235, 0x8014), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i8", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, { /* -- 1.7.9.5

3 3

[alsa-devel] [PATCH] ASoC: pcm: Use the power efficient workqueue for delayed powerdown
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> There is no need to use a normal per-CPU workqueue for delayed power downs as they're not timing or performance critical and waking up a core for them would defeat some of the point. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c6403..f4f68cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -411,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: wm8904: fix the typo error for LINER Mux
by Bo Shen 18 Jul '13

18 Jul '13

fix the typo error, from "LINEL Mux" to "LINER Mux" Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c..bdd575a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1017,7 +1017,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hdmi - poll eld at resume time
by Wang Xingchao 18 Jul '13

18 Jul '13

Hdmi driver may not receive intrinsic event from gfx side when it's in runtime suspend mode. There's no ELD info when exit from runtime suspend. This patch avoid missing ELD info. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..cb8ac66 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1868,12 +1868,33 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + generic_hdmi_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hdmi_present_sense(per_pin, 1); + } + return 0; +} +#endif + static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; -- 1.8.1.2

5 29

[alsa-devel] [PATCH, RESEND] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 17 Jul '13

17 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there, preventing a build. Cc: Mark Brown <broonie(a)linaro.org> Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1, added Mark to Cc. sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

1 0

[alsa-devel] Sound Card driver for audio codec, mux and dsp.
by Diego Sueiro 17 Jul '13

17 Jul '13

Hi there, I need to develop a solution to expose to userspace a single sound card with mixer controls for: Audio codec : sgtl5000 Audio mux: used to select an audio source on sgtl5000 LINE_IN. Audio dsp :sta309a How can I implement this drivers? Should I have to create a sound-card driver which is going to register and control the other drivers (codec, mux, dsp)? Is there some device driver or tutorial guide that I can use as an example? I'm using the kernel 3.0.15 from Freescale [1]. [1] - http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/log/?h=imx_3.0.… Kind Regards, -- *dS Diego Sueiro

1 1

[alsa-devel] idea: a reserve alsa plugin
by David Henningsson 17 Jul '13

17 Jul '13

Just had an idea which I'll write down here before I forget it again...and I'm not saying I'll implement this anytime soon either, but here goes: There is a device reserve protocol between PulseAudio and JACK2 - when JACK needs the sound card, it'll send a dbus message to PulseAudio and grab a name in D-Bus. However, there are plenty of applications who like to access ALSA directly, without going through JACK2 or PulseAudio. By making a "reserve" plugin, we could have this functionality for those apps too. In practice, if the app usually opens "plughw:0" or "hw:0", it could instead open "reserve:plughw:0" or "reserve:hw:0" to also reserve the device from PulseAudio usage while the device is open. Meanwhile, PulseAudio is free to use other audio devices (which is not the case when using e g pasuspender). How does that sound? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

6 12

[alsa-devel] Help in understanding SND_PCM_TSTAMP_ENABLE mode snd_pcm_htimestamp function
by divya v 17 Jul '13

17 Jul '13

Hi, I am looking for a method to maintain timestamps for every sample read or written. I noticed the *SND_PCM_TSTAMP_ENABLE mode. But the description mentions - "" *pdate timestamp at every hardware position update" Doe this mean that the when snd_pcm_htimestamp() is issued, we will get the last hardware update timestamp or does it mean that it will fetch the timestamp of that particular sample? Thanks, Divya

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: fsl_ssi: Use devm_ functions
by Fabio Estevam 17 Jul '13

17 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_ functions can make the code cleaner and smaller. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 38 ++++++++++---------------------------- 1 file changed, 10 insertions(+), 28 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..ab8d462 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -680,7 +680,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* The DAI name is the last part of the full name of the node. */ p = strrchr(np->full_name, '/') + 1; - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), + ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p), GFP_KERNEL); if (!ssi_private) { dev_err(&pdev->dev, "could not allocate DAI object\n"); @@ -698,26 +698,24 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - goto error_kmalloc; + return ret; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - ret = -ENOMEM; - goto error_kmalloc; + return -ENOMEM; } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); if (ssi_private->irq == NO_IRQ) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - ret = -ENXIO; - goto error_iomap; + return -ENXIO; } /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, - ssi_private); + ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); goto error_irqmap; @@ -739,11 +737,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = clk_get(&pdev->dev, NULL); + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); - goto error_irq; + goto error_irqmap; } clk_prepare_enable(ssi_private->clk); @@ -794,7 +792,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error_irq; + goto error_clk; } /* Register with ASoC */ @@ -857,23 +855,12 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } - -error_irq: - free_irq(ssi_private->irq, ssi_private); error_irqmap: irq_dispose_mapping(ssi_private->irq); -error_iomap: - iounmap(ssi_private->ssi); - -error_kmalloc: - kfree(ssi_private); - return ret; } @@ -886,15 +873,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) { imx_pcm_dma_exit(pdev); clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); } snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - - free_irq(ssi_private->irq, ssi_private); irq_dispose_mapping(ssi_private->irq); - - kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); return 0; -- 1.8.1.2

2 2

[alsa-devel] Connecting two Codecs to the same CPU DAI
by Daniel Mack 17 Jul '13

17 Jul '13

Hi Mark, Liam, I'm trying to find a sane solution for systems that have multiple Codecs connected to the same CPU DAI. An example would be a codec that receives the same I2S stream as a S/PDIF transceiver, and which is in fact connected to the very same bus in hardware. Currently, I'm setting up two DAI links in such cases, and userspace has to open both subdevices and issue hw_params calls to each of them in order to set up the hardware correctly. This is cumbersome of course, and I wonder how the core could be augmented in order to handle such cases correctly. Ideally, it should only expose one PCM interface and handle a list of Codecs in the back. Or am I unaware of an existing feature? Thanks, Daniel

2 7

[alsa-devel] [BUG] 3.10.[01] modprobe snd-... hangs
by Philipp Matthias Hahn 17 Jul '13

17 Jul '13

Hello, My x86_64 systems has some trouble loading some ALSA snd-* modules since the upgrade to 3.10.[01]: Several automatic modprobe calls hang, but loading snd-intel-hda and snd-audio-usb by hand still works. # ps axf 460 ? S 0:00 \_ [kworker/u8:3] 1087 ? S 0:00 | \_ [kworker/u8:3] 1092 ? S 0:00 | \_ /sbin/modprobe -q -- snd-seq-client-14 ... 1071 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd && { /sbin/modprobe --quiet snd 1080 ? D 0:00 \_ /sbin/modprobe --quiet snd-seq 1115 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd-rawmidi && { /sbin/modprobe --q 1154 ? S 0:00 \_ /sbin/modprobe --quiet snd-seq-midi 1119 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd-rawmidi && { /sbin/modprobe --q 1148 ? S 0:00 \_ /sbin/modprobe --quiet snd-seq-midi In /var/log/daemon.log I find the following messages: Jul 15 19:58:44 scout udevd[912]: timeout: killing '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc01ip00in00' [1044] Jul 15 19:58:56 scout udevd[912]: '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc01ip00in00' [1044] terminated by signal 9 (Killed) Jul 15 19:58:56 scout udevd[919]: timeout: killing '/sbin/modprobe -b pci:v000010DEd0000026Csv00001462sd00007350bc04sc03i00' [1051] Jul 15 19:58:56 scout udevd[914]: timeout: killing '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc03ip00in01' [1045] Jul 15 19:58:56 scout udevd[919]: '/sbin/modprobe -b pci:v000010DEd0000026Csv00001462sd00007350bc04sc03i00' [1051] terminated by signal 9 (Killed) Jul 15 19:58:56 scout udevd[914]: '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc03ip00in01' [1045] terminated by signal 9 (Killed) This is a Debian sid system # modprobe -V kmod version 9 # echo t >/proc/sysrq-trigger produces too much output to include the modprobe processes. I already stopped most processes, but that is not enough. # cat /proc/1092/syscall 175 0x7effcad92000 0xafe8 0x7effca9c2f88 0x7effca6ea3aa 0x0 0x0 0x7fff3e3de648 0x7effca6eef8a # cat /proc/1080/syscall 175 0x7fe01b25e000 0xafe8 0x7fe01ae8ef88 0x7fe01abb63aa 0x3 0x0 0x7fffba8cc748 0x7fe01abbaf8a # cat /proc/1154/syscall 175 0x7fc157a7b000 0xafe8 0x7fc1576abf88 0x7fc1573d33aa 0x3 0x0 0x7fff05809e58 0x7fc1573d7f8a # cat /proc/1148/syscall 175 0x7fc7159d5000 0xafe8 0x7fc715605f88 0x7fc71532d3aa 0x3 0x0 0x7fff47a1f3c8 0x7fc715331f8a Is this a know bug or has someone seen similar problems? # cat /proc/modules ip6table_filter 1575 0 - Live 0xffffffffa0893000 ip6_tables 13784 1 ip6table_filter, Live 0xffffffffa088b000 ebtable_nat 1887 0 - Live 0xffffffffa0887000 ebtables 19917 1 ebtable_nat, Live 0xffffffffa087e000 snd_usb_audio 124196 0 - Live 0xffffffffa0851000 snd_usbmidi_lib 16982 1 snd_usb_audio, Live 0xffffffffa0848000 rfcomm 28731 10 - Live 0xffffffffa083a000 bnep 9786 2 - Live 0xffffffffa0833000 snd_hda_codec_realtek 37669 1 - Live 0xffffffffa0823000 snd_hda_intel 31432 0 - Live 0xffffffffa0814000 snd_hda_codec 140669 2 snd_hda_codec_realtek,snd_hda_intel, Live 0xffffffffa07e2000 snd_hwdep 5627 2 snd_usb_audio,snd_hda_codec, Live 0xffffffffa07dd000 iptable_filter 1618 1 - Live 0xffffffffa07d9000 xt_owner 1326 1 - Live 0xffffffffa07d5000 ip_tables 13390 1 iptable_filter, Live 0xffffffffa07cd000 x_tables 18433 6 ip6table_filter,ip6_tables,ebtables,iptable_filter,xt_owner,ip_tables, Live 0xffffffffa07c3000 cpufreq_conservative 7015 0 - Live 0xffffffffa07bd000 cpufreq_ondemand 8945 0 - Live 0xffffffffa07b6000 cpufreq_powersave 1010 0 - Live 0xffffffffa07b2000 cpufreq_userspace 2136 0 - Live 0xffffffffa07ae000 binfmt_misc 6479 1 - Live 0xffffffffa07a9000 bridge 74378 0 - Live 0xffffffffa078d000 stp 1908 1 bridge, Live 0xffffffffa0789000 llc 4485 2 bridge,stp, Live 0xffffffffa0784000 deflate 1873 0 - Live 0xffffffffa0780000 ctr 3633 0 - Live 0xffffffffa077c000 twofish_generic 5715 0 - Live 0xffffffffa0777000 twofish_x86_64_3way 19885 0 - Live 0xffffffffa076f000 twofish_x86_64 5503 1 twofish_x86_64_3way, Live 0xffffffffa076a000 twofish_common 13133 3 twofish_generic,twofish_x86_64_3way,twofish_x86_64, Live 0xffffffffa0763000 camellia_generic 18308 0 - Live 0xffffffffa075b000 camellia_x86_64 45885 0 - Live 0xffffffffa074b000 serpent_sse2_x86_64 44963 0 - Live 0xffffffffa073d000 serpent_generic 21243 1 serpent_sse2_x86_64, Live 0xffffffffa0734000 ablk_helper 2284 1 serpent_sse2_x86_64, Live 0xffffffffa0730000 cryptd 7992 1 ablk_helper, Live 0xffffffffa072a000 xts 2958 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0726000 lrw 3584 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0722000 gf128mul 6580 2 xts,lrw, Live 0xffffffffa071d000 glue_helper 4810 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0718000 blowfish_generic 3194 0 - Live 0xffffffffa0714000 blowfish_x86_64 14084 0 - Live 0xffffffffa070d000 blowfish_common 6425 2 blowfish_generic,blowfish_x86_64, Live 0xffffffffa0708000 cast5_generic 10378 0 - Live 0xffffffffa0702000 cast_common 5351 1 cast5_generic, Live 0xffffffffa06fd000 des_generic 16147 0 - Live 0xffffffffa06f6000 cmac 2460 0 - Live 0xffffffffa06f2000 xcbc 2423 0 - Live 0xffffffffa06ee000 rmd160 7552 0 - Live 0xffffffffa06e9000 nfsd 208660 2 - Live 0xffffffffa06a8000 auth_rpcgss 39179 1 nfsd, Live 0xffffffffa0698000 oid_registry 2490 1 auth_rpcgss, Live 0xffffffffa0694000 nfs_acl 2285 1 nfsd, Live 0xffffffffa0690000 sha512_generic 5023 0 - Live 0xffffffffa068b000 nfs 127233 0 - Live 0xffffffffa065b000 lockd 57962 2 nfsd,nfs, Live 0xffffffffa0645000 fscache 34198 1 nfs, Live 0xffffffffa0636000 sha1_ssse3 9266 0 - Live 0xffffffffa0630000 sha1_generic 2064 1 sha1_ssse3, Live 0xffffffffa062c000 sunrpc 185043 6 nfsd,auth_rpcgss,nfs_acl,nfs,lockd, Live 0xffffffffa05ea000 md5 1803 0 - Live 0xffffffffa05e6000 crypto_null 2624 0 - Live 0xffffffffa05e2000 af_key 25880 0 - Live 0xffffffffa05d6000 xfrm_algo 6101 1 af_key, Live 0xffffffffa05d1000 ves1820 5159 1 - Live 0xffffffffa05cc000 btusb 14810 0 - Live 0xffffffffa05bc000 bluetooth 183572 24 rfcomm,bnep,btusb, Live 0xffffffffa054a000 stv0297 6493 1 - Live 0xffffffffa052f000 snd_rawmidi 17913 1 snd_usbmidi_lib, Live 0xffffffffa0525000 snd_pcm_oss 32083 0 - Live 0xffffffffa0518000 snd_mixer_oss 12485 1 snd_pcm_oss, Live 0xffffffffa0510000 snd_pcm 73594 4 snd_usb_audio,snd_hda_intel,snd_hda_codec,snd_pcm_oss, Live 0xffffffffa04f5000 snd_page_alloc 7172 2 snd_hda_intel,snd_pcm, Live 0xffffffffa04f0000 dvb_ttpci 93699 0 - Live 0xffffffffa04d1000 saa7146_vv 26567 1 dvb_ttpci, Live 0xffffffffa04c5000 videobuf_dma_sg 8015 1 saa7146_vv, Live 0xffffffffa04bf000 videobuf_core 15578 2 saa7146_vv,videobuf_dma_sg, Live 0xffffffffa04b7000 v4l2_common 7907 1 saa7146_vv, Live 0xffffffffa04b1000 budget_ci 21012 0 - Live 0xffffffffa04a7000 videodev 97317 2 saa7146_vv,v4l2_common, Live 0xffffffffa0486000 budget_core 7679 1 budget_ci, Live 0xffffffffa0481000 ttpci_eeprom 1985 2 dvb_ttpci,budget_core, Live 0xffffffffa0479000 saa7146 11356 4 dvb_ttpci,saa7146_vv,budget_ci,budget_core, Live 0xffffffffa046b000 rc_core 13478 2 budget_ci, Live 0xffffffffa044a000 dvb_core 68839 3 dvb_ttpci,budget_ci,budget_core, Live 0xffffffffa0431000 snd_seq_oss 33717 1 - Loading 0xffffffffa041b000 snd_seq_midi_event 5740 1 snd_seq_oss, Live 0xffffffffa0416000 snd_seq 44232 2 snd_seq_oss,snd_seq_midi_event, Live 0xffffffffa0404000 snd_seq_device 5560 3 snd_rawmidi,snd_seq_oss,snd_seq, Live 0xffffffffa03ff000 snd_timer 17417 2 snd_pcm,snd_seq, Live 0xffffffffa03f6000 snd 58793 14 snd_usb_audio,snd_usbmidi_lib,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_seq_device,snd_timer, Live 0xffffffffa03df000 pcspkr 1926 0 - Live 0xffffffffa03db000 psmouse 52563 0 - Live 0xffffffffa03c6000 serio_raw 4517 0 - Live 0xffffffffa03c1000 soundcore 5811 1 snd, Live 0xffffffffa03b7000 evdev 8465 0 - Live 0xffffffffa039d000 kvm_intel 122108 0 - Live 0xffffffffa0351000 kvm 342589 1 kvm_intel, Live 0xffffffffa0271000 usb_storage 46166 0 - Live 0xffffffffa025c000 fuse 65800 1 - Live 0xffffffffa0243000 ecryptfs 80646 0 - Live 0xffffffffa0227000 cbc 2574 0 - Live 0xffffffffa0223000 sha256_ssse3 16172 2 - Live 0xffffffffa021c000 sha256_generic 9882 1 sha256_ssse3, Live 0xffffffffa0216000 hmac 2673 1 - Live 0xffffffffa0212000 encrypted_keys 9378 1 ecryptfs, Live 0xffffffffa020b000 autofs4 20984 2 - Live 0xffffffffa0200000 ext4 434295 5 - Live 0xffffffffa0173000 crc16 1541 2 bluetooth,ext4, Live 0xffffffffa016f000 jbd2 83146 1 ext4, Live 0xffffffffa0150000 btrfs 682625 1 - Live 0xffffffffa008b000 crc32c 1610 1 - Live 0xffffffffa0087000 libcrc32c 1147 1 btrfs, Live 0xffffffffa0083000 xor 11190 1 btrfs, Live 0xffffffffa007d000 zlib_deflate 18739 2 deflate,btrfs, Live 0xffffffffa0075000 raid6_pq 92000 1 btrfs, Live 0xffffffffa005a000 nbd 8786 0 - Live 0xffffffffa0053000 firewire_ohci 27516 0 - Live 0xffffffffa0048000 firewire_core 49041 1 firewire_ohci, Live 0xffffffffa0035000 crc_itu_t 1565 1 firewire_core, Live 0xffffffffa0031000 forcedeth 50647 0 - Live 0xffffffffa001e000 floppy 53437 0 - Live 0xffffffffa0009000 ohci_hcd 18001 0 - Live 0xffffffffa0000000 BYtE Philipp -- / / (_)__ __ ____ __ Philipp Hahn / /__/ / _ \/ // /\ \/ / /____/_/_//_/\_,_/ /_/\_\ pmhahn(a)pmhahn.de

7 10

[alsa-devel] [PATCH v2] ASoC: sgtl5000: defer the probe if clock is not found
by Shawn Guo 16 Jul '13

16 Jul '13

It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000: Let the codec acquire its clock). [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 [ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered [ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517) [ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v1: * Not use ternary operator. Mark, I'm still not going to fix this regression during -rc at clock core level with the risk of breaking clock API users and being yelled by clock maintainer. Shawn sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..f5ec35b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,6 +1527,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + /* Defer the probe to see if the clk will be provided later */ + if (ret == -ENOENT) + return -EPROBE_DEFER; return ret; } -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: tegra: fix compile warning in AC'97 driver
by Stephen Warren 16 Jul '13

16 Jul '13

From: Stephen Warren <swarren(a)nvidia.com> This fixes the following by deleting dead code: sound/soc/tegra/tegra20_ac97.c: In function ‘tegra20_ac97_platform_probe’: sound/soc/tegra/tegra20_ac97.c:435:1: warning: label ‘err_unregister_pcm’ defined but not used [-Wunused-label] Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra20_ac97.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f..87b845f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -432,8 +432,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) return 0; -err_unregister_pcm: - tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_asoc_utils_fini: -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoc: tegra: Remove unneeded mach-type.h incldues
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra_alc5632.c | 2 -- sound/soc/tegra/tegra_wm8753.c | 2 -- sound/soc/tegra/trimslice.c | 2 -- 3 files changed, 6 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9..c61ea3a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f87fc53..8e774d1 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -28,8 +28,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 05c68aa..734bfcd 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -24,8 +24,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: tegra_alc5632: Remove unused mach-types.h include
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra_alc5632.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9..c61ea3a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> -- 1.8.3.2

2 2

[alsa-devel] [PATCH 0/10] snd_pcm_stop() lock fixes
by Takashi Iwai 16 Jul '13

16 Jul '13

Hi, while reviewing another patch series, I stumbled on the snd_pcm_lock() call in an unlocked context. Then I started looking through the whole tree, and found a bunch of drivers doing it wrong, too. So here is a patch series to fix them. [PATCH 01/10] ALSA: asihpi: Fix unlocked snd_pcm_stop() call [PATCH 02/10] ALSA: atiixp: Fix unlocked snd_pcm_stop() call [PATCH 03/10] ALSA: 6fire: Fix unlocked snd_pcm_stop() call [PATCH 04/10] ALSA: ua101: Fix unlocked snd_pcm_stop() call [PATCH 05/10] ALSA: usx2y: Fix unlocked snd_pcm_stop() call [PATCH 06/10] ALSA: pxa2xx: Fix unlocked snd_pcm_stop() call [PATCH 07/10] ASoC: atmel: Fix unlocked snd_pcm_stop() call [PATCH 08/10] ASoC: s6000: Fix unlocked snd_pcm_stop() call [PATCH 09/10] [media] saa7134: Fix unlocked snd_pcm_stop() call [PATCH 10/10] staging: line6: Fix unlocked snd_pcm_stop() call Takashi

6 22

[alsa-devel] snd_pcm_delay issue - How to debug?
by Sander Jansen 16 Jul '13

16 Jul '13

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

1 0

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: fix error return code in imx_sgtl5000_probe()
by Wei Yongjun 16 Jul '13

16 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/imx-sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 7a8bc12..25f310d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,8 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); goto fail; + } data->clk_frequency = clk_get_rate(data->codec_clk);

3 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add compatible strings for specific devices
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> The driver supports a range of devices but currently doesn't allow those device names to be used for enumeration on DT. Add the currently listed I2C IDs as compatible strings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 8 +++++++- sound/soc/codecs/tlv320aic3x.c | 2 ++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index f47c3f5..26f65f9 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -3,7 +3,13 @@ Texas Instruments - tlv320aic3x Codec module The tlv320aic3x serial control bus communicates through I2C protocols Required properties: -- compatible - "string" - "ti,tlv320aic3x" + +- compatible - "string" - One of: + "ti,tlv320aic3x" - Generic TLV320AIC3x device + "ti,tlv320aic33" - TLV320AIC33 + "ti,tlv320aic3007" - TLV320AIC3007 + + - reg - <int> - I2C slave address diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b9268..c9bb760 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1582,6 +1582,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- 1.8.3.2

1 1

[alsa-devel] [PATCH] ASoC: imx: Enable COMPILE_TEST builds
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa43854..87e28bc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -98,7 +98,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. -- 1.8.3.2

2 3

[alsa-devel] [PATCH v2] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function The ALSA subsystem should never gives such a rate because: - the rates with the internal clock are limited to 44.1, 48 and 96 kHz as specified by the kirkwood_i2s_dai structure, - the other rates are proposed in the structure kirkwood_i2s_dai_extclk only when the external clock is present. In case of programming error (bad rate for internal clock and no external clock), the function will simply cause a backtrace. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 - add more explanations --- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..c9daf3b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -105,14 +105,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for supported rates */ + /* use internal dco for the supported rates + * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { - /* use optional external clk for other rates */ + } else { + /* use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate);

2 1

[alsa-devel] [RFC 1/2] ASoC: dmaengine-pcm: Add support for querying DMA capabilities
by Lars-Peter Clausen 16 Jul '13

16 Jul '13

Currently each platform making use the the generic dmaengine PCM driver still needs to provide a custom snd_pcm_hardware struct which specifies the capabilities of the DMA controller, e.g. the maximum period size that can be supported. This patch adds code which uses the newly introduced dma_get_slave_caps() API to query this information from the dmaengine driver. The new code path will only be taken if the 'pcm_hardware' field of the snd_dmaengine_pcm_config struct is NULL. The patch also introduces a new 'fifo_size' field to the snd_dmaengine_dai_dma_data struct which is used to initialize the snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- For review only for now. The dma_get_slave_caps() API has not been merged yet, the patch adding support for it can be found here: http://permalink.gmane.org/gmane.linux.kernel/1524880 --- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-generic-dmaengine-pcm.c | 46 +++++++++++++++++++++++++++++++++-- 2 files changed, 46 insertions(+), 2 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35c..83b2c3e 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +69,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3c..98044a7 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -92,6 +92,43 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct snd_pcm_hardware hw; + struct dma_slave_caps dma_caps; + int ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret) + return ret; + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = dma_caps.max_sg_nr; + hw.period_bytes_min = 16; + hw.period_bytes_max = dma_caps.max_sg_len; + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + if (hw.periods_max == 0) + hw.periods_max = UINT_MAX; + if (hw.period_bytes_max == 0) + hw.period_bytes_max = SIZE_MAX; + + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + + return snd_soc_set_runtime_hwparams(substream, &hw); +} + static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -99,8 +136,13 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) struct dma_chan *chan = pcm->chan[substream->stream]; int ret; - ret = snd_soc_set_runtime_hwparams(substream, - pcm->config->pcm_hardware); + if (pcm->config->pcm_hardware) { + ret = snd_soc_set_runtime_hwparams(substream, + pcm->config->pcm_hardware); + } else { + ret = dmaengine_pcm_set_runtime_hwparams(substream); + } + if (ret) return ret; -- 1.8.0

3 8

[alsa-devel] [PATCH] ALSA: hda - Headphone mic support for an Asus/Conexant device
by David Henningsson 16 Jul '13

16 Jul '13

This Conexant codec has a single jack that can be used as either headphone or mic (but not headset). The existing hp_mic functionality does not apply here, because the mic and the HP are on separate pins. Hence make a lighter version of what has been earlier done for Realtek codecs. BugLink: https://bugs.launchpad.net/bugs/1198030 Tested-by: Franz Hsieh <franz.hsieh(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 79 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 78 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index de00ce1..4edd2d0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -66,6 +66,8 @@ struct conexant_spec { hda_nid_t eapds[4]; bool dynamic_eapd; + unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + #ifdef ENABLE_CXT_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec) snd_hda_gen_init(codec); if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + return 0; } @@ -3224,6 +3229,8 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_HEADPHONE_MIC_PIN, + CXT_FIXUP_HEADPHONE_MIC, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec, (0 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_update_headset_mode(struct hda_codec *codec) +{ + /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */ + int i; + bool mic_mode = false; + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == mux_pin) { + mic_mode = !!cfg->inputs[i].is_headphone_mic; + break; + } + + if (mic_mode) { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */ + spec->gen.hp_jack_present = false; + } else { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */ + spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]); + } + + snd_hda_gen_update_outputs(codec); +} + +static void cxt_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + cxt_update_headset_mode(codec); +} + +static void cxt_fixup_headphone_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; + spec->gen.automute_hook = cxt_update_headset_mode; + break; + case HDA_FIXUP_ACT_INIT: + cxt_update_headset_mode(codec); + break; + } +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_HEADPHONE_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADPHONE_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { } + } + }, + [CXT_FIXUP_HEADPHONE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headphone_mic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), @@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, + spec->parse_flags); if (err < 0) goto error; @@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->bus->allow_bus_reset = 1; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: kirkwood-i2s: Remove empty remove()
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-i2s.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 44412ea..becc082 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -398,11 +398,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) } -static int kirkwood_i2s_remove(struct snd_soc_dai *dai) -{ - return 0; -} - static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, @@ -413,7 +408,6 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { static struct snd_soc_dai_driver kirkwood_i2s_dai = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, @@ -431,7 +425,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.3.2

1 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID to snd-hda
by Aaron Plattner 16 Jul '13

16 Jul '13

Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip. Reviewed-by: Andy Ritger <aritger(a)nvidia.com> Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..486def7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2536,6 +2536,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2588,6 +2589,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- 1.7.12

3 3

Re: [alsa-devel] Alsa USB Audio regression
by Raymond Yau 16 Jul '13

16 Jul '13

>>> >>> After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at >>> all, Ubuntu 11.10 and newer have the bug), the headset only plays sound >>> at maximum volume level if the volume is turned down just a little no >>> sound is being played. >>> >>> I have filled a Ubuntu bug for 12.04 LTS at: >>> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . >>> I was advised to contact this mailing list. >>> >>> AlsaInfo for Ubuntu 12.04 LTS with the bug: >>> http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . >>> AlsaInfo for Ubuntu 10.04 LTS without the bug: >>> http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc >> >> >> >> It is strange that dB range of your usb audio is 0 to 0.14 (only small gain and no attenuation) >> >> In 10.04 LTS pulseaudio provide software volume control >> >> >> ( 1.095| 0.000) I: alsa-sink.c: Driver does not support hardware volume control, falling back to software volume control. >> >> >> control.6 { >> comment.access 'read write' >> comment.type INTEGER >> comment.count 1 comment.range '0 - 37' >> comment.dbmin 0 comment.dbmax 14 >> iface MIXER >> name 'Speaker Playback Volume' value 30 } >> >> >> In 12.04 pulseaudio did not provide software volume attenuation and use the speaker playback volume control > > > So does that mean that the bug is in pulseaudio, or could it be even deeper like in the kernel or something? Are there any significant change in volume when you use "alsamixer -c0" to change speaker playback volume from min 0dB to 0.14dB ? Did pulseaudio provide software attuentation -1.61dB when the range of hardware volume control is only 0.14dB ? * index: 0 name: <alsa_output.usb-0d8c_USB_Audio-00-Audio.analog-surround-71> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 9049 volume: 0: 94% 1: 94% 2: 94% 3: 94% 4: 94% 5: 94% 6: 94% 7: 94% 0: -1,61 dB 1: -1,61 dB 2: -1,61 dB 3: -1,61 dB 4: -1,61 dB 5: -1,61 dB 6: -1,61 dB 7: -1,61 dB balance 0,00 base volume: 99% -0,14 dB volume steps: 65537 muted: no It is strange that 5.1 headset only support 8 channels playback

1 0

[alsa-devel] [PATCH] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function As the ALSA subsystem should never gives such a rate, this patch removes the check of the external clock pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..8da5cdb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -111,7 +111,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { + } else { /* use optional external clk for other rates */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate);

3 7

[alsa-devel] [PATCH] ALSA: hda - Fix error accessing non-existent node for Realtek codecs
by David Henningsson 15 Jul '13

15 Jul '13

A recent commit caused my hda-emu based test tool to complain about accessing nodes that don't exist. This simple test works around this. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) I don't have ALC5505 hardware here, no alsa-info to look at, and I don't know what it is. Therefore it would be good with some confirmation that this patch does not break such hardware, which could be the case if the Realtek codec breaks the HDA spec by having "hidden" nodes. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..ca73f10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3844,7 +3844,8 @@ static int patch_alc269(struct hda_codec *codec) break; } - if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + if (codec->num_nodes > 0x51 && + snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { spec->has_alc5505_dsp = true; spec->init_hook = alc5505_dsp_init; } -- 1.7.9.5

3 5

[alsa-devel] [PATCH RESEND] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

From: Chih-Chung Chang <chihchung(a)chromium.org> The registers of max98088 are 8 bits, not 16 bits. This bug causes the contents of registers to be overwritten with bad values when the codec is suspended and then resumed. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada5..566a367 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) -- 1.8.1.3.605.g02339dd

3 2

[alsa-devel] [PATCH] ASoC: sgtl5000: defer the probe if clock is not found
by Shawn Guo 15 Jul '13

15 Jul '13

It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the following problem on imx28. [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, This is a fix for 3.11. Shawn sound/soc/codecs/sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7ebf923 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,7 +1527,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); - return ret; + /* Defer the probe to see if the clk will be provided later */ + return ret == -ENOENT ? -EPROBE_DEFER : ret; } ret = clk_prepare_enable(sgtl5000->mclk); -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: omap: Enable COMPILE_TEST build for DT platforms
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> The DT platforms don't have any source dependency on any OMAP stuff so allow them to be built when COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e..68dfc0a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && DMA_OMAP + depends on (ARCH_OMAP && DMA_OMAP) || COMPILE_TEST select SND_SOC_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + depends on SND_OMAP_SOC && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 @@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- 1.8.3.2

3 5

Re: [alsa-devel] [PATCH 1/1] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

+Maintainers and Maxim, Chih-Chung, please CC maintainers on patches. On Tue, Jun 25, 2013 at 12:47 AM, <me(a)google.com> wrote: > From: Chih-Chung Chang <chihchung(a)chromium.org> > > The registers of max98088 are 8 bits, not 16 bits. This bug causes the > contents of registers to be overwritten with bad values when the codec > is suspended and then resumed. > > Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> > > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 3eeada5..566a367 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) > > static void max98088_sync_cache(struct snd_soc_codec *codec) > { > - u16 *reg_cache = codec->reg_cache; > + u8 *reg_cache = codec->reg_cache; > int i; > > if (!codec->cache_sync) > -- > 1.8.3 >

2 1

[alsa-devel] [PATCH] ASoC: mxs: Enable COMPILE_TEST builds
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321c..7daf860 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" - depends on ARCH_MXS + depends on ARCH_MXS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

1 0

[alsa-devel] [PATCH 1/3] ASoC: kirkwood: Remove unused headers
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-openrd.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..addbebc 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -16,9 +16,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/cs42l51.h" static int openrd_client_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..4f4cb56 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -15,9 +15,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/alc5623.h" static int t5325_hw_params(struct snd_pcm_substream *substream, -- 1.8.3.2

1 2

[alsa-devel] [PATCH v5 0/7] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 15 Jul '13

15 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130711 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (6): sound: codec: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 41 ++++ arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 24 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 207 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 57 +++++- 8 files changed, 363 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

2 12

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Let the codec acquire its clock
by Fabio Estevam 15 Jul '13

15 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx6qsabrelite board the following error happens on probe: sgtl5000: probe of 0-000a failed with error -5 imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer Prior to reading the codec ID we need to turn the SYS_MCLK clock, so let's enable the codec clock inside sgtl5000_i2c_probe(). Also remove the codec clock enable/disable functions from the machine driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Shawn, I haven't had a chance to convert mx28 to pass the codec clock via device tree yet. It looks like it is a bit trickier there, as it uses the saif clock. Given that mx28 audio playback is already broken in linux-next, this series does not make things worse on mx28 side. sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/fsl/imx-sgtl5000.c | 30 +++++++----------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8f2afb..2e0227b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -114,6 +114,7 @@ struct sgtl5000_priv { struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; struct regmap *regmap; + struct clk *mclk; }; /* @@ -1522,16 +1523,28 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto disable_clk; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto disable_clk; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1542,17 +1555,30 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto disable_clk; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000; + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index a60aaa0..823151b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,20 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); - } + if (IS_ERR(data->codec_clk)) + goto fail; + + data->clk_frequency = clk_get_rate(data->codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,7 +160,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); @@ -179,8 +169,6 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; -clk_fail: - clk_put(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -194,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; -- 1.8.1.2

3 6

[alsa-devel] [PATCH] hdspm - remove unneeded semicolon
by Fengguang Wu 15 Jul '13

15 Jul '13

sound/pci/rme9652/hdspm.c:1110:2-3: Unneeded semicolon Generated by: coccinelle/misc/semicolon.cocci Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cocci-output-13040-790f4c-hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1107,7 +1107,7 @@ static int hdspm_rate_multiplier(struct else if (hdspm->control_register & HDSPM_DoubleSpeed) return rate * 2; - }; + } return rate; }

2 1

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 15 Jul '13

15 Jul '13

The following changes since commit ad81f0545ef01ea651886dddac4bef6cec930092: Linux 3.11-rc1 (2013-07-14 15:18:27 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc1 for you to fetch changes up to 4824b69a928b6609cfb1cf232ef32d4a8125a371: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-07-15 11:13:58 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 The biggest change here is the OMAP change, these are larger than I'd have liked but make the driver actually usable - during the merge window OMAP removed support for non-DT OMAP4 boards but in doing so removed the method of accessing DMA channels used by the ASoC drivers rendering them unusuable. Otherwise nothing exciting, the symmetric rates change for WM8978 is a fix for the information we expose to userspace. ---------------------------------------------------------------- Fabio Estevam (2): ASoC: sglt5000: Fix the default value of CHIP_SSS_CTRL ASoC: sglt5000: Fix SGTL5000_PLL_FRAC_DIV_MASK Kuninori Morimoto (1): ASoC: wm8978: enable symmetric rates Mark Brown (6): ASoC: wm8994: Remove overly noisy debug logging Merge remote-tracking branch 'asoc/fix/omap' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8978' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Padmavathi Venna (1): ASoC: Samsung: Set RFS and BFS in slave mode Peter Ujfalusi (4): ASoC: omap-pcm: Request the DMA channel differently when DT is involved ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/wm8978.c | 1 + sound/soc/codecs/wm8994.c | 4 ---- sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 11 +---------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- sound/soc/samsung/i2s.c | 8 ++++---- 9 files changed, 46 insertions(+), 54 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 15 Jul '13

15 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there preventing a build. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1 sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ARM: EXYNOS: remove references to non-DT SoCs
by Paul Bolle 15 Jul '13

15 Jul '13

Commit 383ffda2fa ("ARM: EXYNOS: no more support non-DT for EXYNOS SoCs") removed the Kconfig symbols MACH_SMDKV310 and MACH_SMDKC210. Now remove the references to both of these symbols from the Kconfig entry for SND_SOC_SAMSUNG_SMDK_WM9713. They're not needed anymore. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9855dfc..3d76659 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -136,7 +136,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot)
by Peter Ujfalusi 15 Jul '13

15 Jul '13

Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. The platform_data header file will be removed at later point to avoid build issues. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- include/linux/platform_data/omap-abe-twl6040.h | 2 + sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 43 insertions(+), 92 deletions(-) diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h index 5d298ac..065abbe 100644 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ b/include/linux/platform_data/omap-abe-twl6040.h @@ -21,6 +21,8 @@ * 02110-1301 USA */ +/* TODO: This file can be removed in 3.12 since it is no longer needed */ + #ifndef _OMAP_ABE_TWL6040_H_ #define _OMAP_ABE_TWL6040_H_ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- 1.8.2.1

2 4

[alsa-devel] ASoC: codec: SPDIF only works with Tx/Rx seperate codec drivers?
by Nicolin Chen 15 Jul '13

15 Jul '13

Hi Mark, I've a little question about SPDIF codec drivers. I found there're spdif_receiver.c and spdif_transmitter.c, two files, in sound/soc/codecs directory. So it looks like SPDIF module would work separately as one IN device and one OUT device, for example: dts/spear1340.dtsi: spdif-in@d0100000 { dts/spear1340.dtsi: compatible = "st,spdif-in"; dts/spear1340.dtsi: spdif-out@d0000000 { dts/spear1340.dtsi: compatible = "st,spdif-out"; But the question is what if a SPDIF module have both Tx and Rx function in one simple CPU DAI. Is that okay to add a new codec driver for this situation? Actually I'm gonna prepare some patches to add Freescale SPDIF drivers. But this needs to create a brand new dummy SPDIF codec driver, which's some kinda same as spdif_receiver/transmitter.c but quite different in the supports of SAMPLE_RATE/FORMAT and IN/OUT-in-one integration. So I just want to make sure if this'll be okay. Thank you. And looking forward to your reply. Nicolin Chen

3 5

[alsa-devel] Alsa USB Audio regression
by Ulrich Günther 15 Jul '13

15 Jul '13

Hello, i encounterd a possible alsa bug with the "ID 0d8c:0006 C-Media Electronics, Inc. Storm HP-USB500 5.1 Headset". After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at all, Ubuntu 11.10 and newer have the bug), the headset only plays sound at maximum volume level if the volume is turned down just a little no sound is being played. I have filled a Ubuntu bug for 12.04 LTS at: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . I was advised to contact this mailing list. AlsaInfo for Ubuntu 12.04 LTS with the bug: http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . AlsaInfo for Ubuntu 10.04 LTS without the bug: http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc . Thank You for your help.

2 1

Re: [alsa-devel] [PATCH 00/50] USB: cleanup spin_lock in URB->complete()
by Andy Walls 14 Jul '13

14 Jul '13

On Thu, 2013-07-11 at 17:05 +0800, Ming Lei wrote: > Hi, > > As we are going to run URB->complete() in tasklet context[1][2], Hi, Please pardon my naivete, but why was it decided to use tasklets to defer work, as opposed to some other deferred work mechanism? It seems to me that getting rid of tasklets has been an objective for years: http://lwn.net/Articles/239633/ http://lwn.net/Articles/520076/ http://lwn.net/Articles/240054/ Regards, Andy > and > hard interrupt may be enabled when running URB completion handler[3], > so we might need to disable interrupt when acquiring one lock in > the completion handler for the below reasons: > > - URB->complete() holds a subsystem wide lock which may be acquired > in another hard irq context, and the subsystem wide lock is acquired > by spin_lock()/read_lock()/write_lock() in complete() > > - URB->complete() holds a private lock with spin_lock()/read_lock()/write_lock() > but driver may export APIs to make other drivers acquire the same private > lock in its interrupt handler. > > For the sake of safety and making the change simple, this patch set > converts all spin_lock()/read_lock()/write_lock() in completion handler > path into their irqsave version mechanically. > > But if you are sure the above two cases do not happen in your driver, > please let me know and I can drop the unnecessary change. > > Also if you find some conversions are missed, also please let me know so > that I can add it in the next round. > > > [1], http://marc.info/?l=linux-usb&m=137286322526312&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > [3], http://marc.info/?l=linux-usb&m=137286330626363&w=2 > [snip] > > > Thanks, > -- > Ming Lei

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: sglt5000: Provide the reg_stride field
by Fabio Estevam 12 Jul '13

12 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 has 16-bit registers, and only even numbers are valid for its registers addresses. Let regmap knows about this feature by specifying the 'reg_stride' field, so that it can access only the valid registers. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the description in the commit log sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7c99f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- 1.8.1.2

2 7

[alsa-devel] [PATCH] ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared
by Nicolin Chen 12 Jul '13

12 Jul '13

The code enabled SSIEN when triggered by SNDRV_PCM_TRIGGER_START, so move the disable code to SNDRV_PCM_TRIGGER_STOP for symmetric. This also allows us to use the SSI driver more flexible so that it can support some use cases like "aplay S16_LE.wav S24_LE.wav" which would call the driver in sequence like: startup()->hw_params(S16_LE)->trigger(START)->tirgger(STOP)-> hw_params(S24_LE)->trigger(START)->tirgger(STOP)->shutdown() If we disable SSIEN in shutdown(), the second hw_params() would bypass the sample bits setting while using symmetric_rate. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6ab341 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -510,6 +510,9 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + + if ((read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; default: @@ -534,15 +537,6 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - } } static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) -- 1.7.1

5 12

[alsa-devel] [PATCH RFC] ALSA: usb-audio: add front jack channel selector for EMU0204
by Vasily Khoruzhick 12 Jul '13

12 Jul '13

Add support for front jack channel selector which is present on EMU0204. It allows to get 4 channels out of this soundcard. Tested-by: Yury Bushmelev <jay(a)jay-tech.ru> Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- Control request looks like generic UAC_SET_CUR request, but with a bit weird values, valid values are { 0x01, 0x01} and { 0x01, 0x02 }, and there's no such control type ATM. I'm not well familiar with UAC so any suggestins how to get rid of custom control req are welcome! sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 78 insertions(+) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df..4a5f001 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -23,6 +23,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u8 emu0204_ch_switch; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..dc9a397 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -430,6 +430,76 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } +/* EMU0204 */ +#define snd_emu0204_ch_switch_info snd_ctl_boolean_mono_info + +static int snd_emu0204_ch_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = mixer->emu0204_ch_switch; + return 0; +} + +static int snd_emu0204_ch_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int value = ucontrol->value.integer.value[0]; + int err, changed; + unsigned char buf[2]; + + if (value > 1) + return -EINVAL; + + buf[0] = 0x01; + buf[1] = value ? 0x02 : 0x01; + + changed = value != mixer->emu0204_ch_switch; + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) { + err = -ENODEV; + goto out; + } + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x1, + 0x21, + 0x0400, 0x0e00, buf, 2); + out: + up_read(&mixer->chip->shutdown_rwsem); + if (err < 0) + return err; + mixer->emu0204_ch_switch = value; + return changed; +} + + +static struct snd_kcontrol_new snd_emu0204_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Route channels 3/4 to front jack", + .info = snd_emu0204_ch_switch_info, + .get = snd_emu0204_ch_switch_get, + .put = snd_emu0204_ch_switch_put, + .private_value = 0, + }, +}; + +static int snd_emu0204_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_emu0204_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_emu0204_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->emu0204_ch_switch = 0; /* Default is 1/2 */ + + return 0; +} /* ASUS Xonar U1 / U3 controls */ static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, @@ -1337,6 +1407,13 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_audigy2nx_proc_read); break; + /* EMU0204 */ + case USB_ID(0x041e, 0x3f19): + err = snd_emu0204_controls_create(mixer); + if (err < 0) + break; + break; + case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); -- 1.8.3.2

2 3

[alsa-devel] Question about WM8904 regcache
by Bo Shen 12 Jul '13

12 Jul '13

Hi Mark, I am sorry to disturb you. On our EK board, it use wm8904 codec. When I enable the audio, and use alsa utils aplay to test, I meet the following issue. The audio can not playback, it show the following error. ---8>--- #aplay 44.1k_16bit_stereo.wav Playing WAVE '44.1k_16bit_stereo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:1737: write error: Input/output error ---<8--- After check the register, they are not be configured correctly. When read from debug file system, found the cache_dirty always Y before playback and also failed playback. I try to add the following two line in wm8904_set_bias_level into "case SND_SOC_BIAS_ON", it doesn't work. ---8>--- @@ -1832,9 +1834,12 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + regcache_cache_only(wm8904->regmap, false); + regcache_sync(wm8904->regmap); break; ---<8--- However, If I comment the regcache_cache_only(wm8904->regmap, true) as following, then it play OK. ---8>--- @@ -2232,7 +2239,7 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, WM8904_POBCTRL, 0); /* Can leave the device powered off until we need it */ - regcache_cache_only(wm8904->regmap, true); + // regcache_cache_only(wm8904->regmap, true); ---<8--- So, any clue for this issue? Thanks. Best Regards, Bo Shen

2 3

[alsa-devel] [PATCH] ASoC: wm8978: enable symmetric rates
by Kuninori Morimoto 12 Jul '13

12 Jul '13

wm8978 needs .symmetric_rates = 1. The playback/capture will be strange without this patch when it used asymmetric rate in same time Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c..d8fc531 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/5] ASoC: add simple-card DT support
by Kuninori Morimoto 12 Jul '13

12 Jul '13

Hi Mark, Simon These patches add DT support on simple-card driver. #1 patch adds new snd_soc_of_parse_daifmt() function on soc-core.c and #5 patch will use it. #2 - #5 patches are based on mark/topic/fsi branch >> Simon #2, #3 patches modify sh-arm platform settings. please check it Kuninori Morimoto (5): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: simple-card: add asoc_simple_dai_set for initializing ASoC: simple-card: remove pointless struct asoc_simple_dai_init_info ASoC: simple-card: add Device Tree support .../devicetree/bindings/sound/simple-card.txt | 53 ++++++++ arch/arm/mach-shmobile/board-ap4evb.c | 19 +-- arch/arm/mach-shmobile/board-armadillo800eva.c | 18 +-- arch/arm/mach-shmobile/board-kzm9g.c | 12 +- arch/arm/mach-shmobile/board-mackerel.c | 19 +-- arch/sh/boards/mach-ecovec24/setup.c | 11 +- arch/sh/boards/mach-se/7724/setup.c | 13 +- include/sound/simple_card.h | 11 +- include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 132 ++++++++++++++++---- sound/soc/soc-core.c | 115 +++++++++++++++++ 11 files changed, 307 insertions(+), 98 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2 0/4] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, Changes since v1: - Drop the change in dmaengine_pcm.h - Use .filter_data to store the DMA channel name - omap-pcm: checks if the DAI is probed via DT and uses decides which method to use to request the DMA channel - Dropped the cleanup patch for omap-dmic from the series I will send another series with cleanups later I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-pcm: Request the DMA channel differently when DT is involved ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 11 +---------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- 4 files changed, 39 insertions(+), 44 deletions(-) -- 1.8.2.1

3 8

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix a trivial typo in audss documentation
by Sachin Kamat 11 Jul '13

11 Jul '13

Fixed a trivial typo. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- .../devicetree/bindings/clock/clk-exynos-audss.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt index a120180..75e2e19 100644 --- a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt +++ b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt @@ -2,7 +2,7 @@ The Samsung Audio Subsystem clock controller generates and supplies clocks to Audio Subsystem block available in the S5PV210 and Exynos SoCs. The clock -binding described here is applicable to all SoC's in Exynos family. +binding described here is applicable to all SoCs in Exynos family. Required Properties: -- 1.7.9.5

2 2

Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 22:13:35 +0800, Ming Lei wrote: > > On Thu, Jul 11, 2013 at 9:50 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 11 Jul 2013 17:08:30 +0400, > > Sergei Shtylyov wrote: > >> > >> On 11-07-2013 13:06, Ming Lei wrote: > >> > >> > Complete() will be run with interrupt enabled, so change to > >> > spin_lock_irqsave(). > >> > >> Changelog doesn't match the patch. > > > > Yep, but moreover... > > > >> > Cc: Jaroslav Kysela <perex(a)perex.cz> > >> > Cc: Takashi Iwai <tiwai(a)suse.de> > >> > Cc: alsa-devel(a)alsa-project.org > >> > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > >> > --- > >> > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > >> > 1 file changed, 4 insertions(+) > >> > >> > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > >> > index 4967fe9..e2ee893 100644 > >> > --- a/sound/usb/usx2y/usbusx2yaudio.c > >> > +++ b/sound/usb/usx2y/usbusx2yaudio.c > >> > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > >> > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > >> > if (subs) { > >> > if (atomic_read(&subs->state) >= state_PRERUNNING) { > >> > + unsigned long flags; > >> > + > >> > + local_irq_save(flags); > >> > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > >> > + local_irq_restore(flags); > >> > } > > > > ... actually this snd_pcm_stop() call should have been covered by > > snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch > > together with the change, i.e. wrapping with > > snd_pcm_stream_lock_irqsave(). > > Please use snd_pcm_stream_lock_irqsave() so that I can avoid sending > out similar patch later, :-) > > > > > I'll prepare the patch for 3.11 independently from your patch series, > > so please drop this one. > > OK, thanks for dealing with that. > > > > > > > BTW, the word "cleanup" in the subject is inappropriate. This is > > rather a fix together with the core change. > > It is a cleanup since the patchset only addresses lock problem which > is caused by the tasklet conversion. Well, the conversion to irqsave() is needed for the future drop of irq_save() in the caller, right? Then this isn't a cleanup but a preparation for movement ahead. > > And, I wonder whether we can take a safer approach. When the caller > > condition changed, we often introduced either a different ops > > (e.g. ioctl case) or a flag for the new condition, at least during the > > transition period. > > Interrupt is't enabled until all current drivers are cleaned up, so it is really > safe, please see patch [2]. OK. > > Last but not least, is a conversion to tasklet really preferred? > > tasklet is rather an obsoleted infrastructure nowadays, and people > > don't recommend to use it any longer, AFAIK. > > We discussed the problem in the below link previously[1], Steven > and Thomas suggested to use threaded irq handler, but which > may degrade USB mass storage performance, so we have to > take tasklet now until we rewrite transport part of USB mass storage > driver. > > Also the conversion[2] has avoided the tasklet spin lock problem > already. OK, good to know. thanks, Takashi > [1], http://marc.info/?t=137079119200001&r=1&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > > Thanks, > -- > Ming Lei >

1 0

Re: [alsa-devel] [PATCH 44/50] sound: usb: caiaq: spin_lock in complete() cleanup
by Daniel Mack 11 Jul '13

11 Jul '13

On 11.07.2013 11:06, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> Sound right to me, thanks. Acked-by: Daniel Mack <zonque(a)gmail.com> > --- > sound/usb/caiaq/audio.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 7103b09..e5675ab 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -672,10 +672,11 @@ static void read_completed(struct urb *urb) > offset += len; > > if (len > 0) { > - spin_lock(&cdev->spinlock); > + unsigned long flags; > + spin_lock_irqsave(&cdev->spinlock, flags); > fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); > read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); > - spin_unlock(&cdev->spinlock); > + spin_unlock_irqrestore(&cdev->spinlock, flags); > check_for_elapsed_periods(cdev, cdev->sub_playback); > check_for_elapsed_periods(cdev, cdev->sub_capture); > send_it = 1; >

1 0

Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 17:08:30 +0400, Sergei Shtylyov wrote: > > On 11-07-2013 13:06, Ming Lei wrote: > > > Complete() will be run with interrupt enabled, so change to > > spin_lock_irqsave(). > > Changelog doesn't match the patch. Yep, but moreover... > > Cc: Jaroslav Kysela <perex(a)perex.cz> > > Cc: Takashi Iwai <tiwai(a)suse.de> > > Cc: alsa-devel(a)alsa-project.org > > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > > --- > > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > > 1 file changed, 4 insertions(+) > > > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > > index 4967fe9..e2ee893 100644 > > --- a/sound/usb/usx2y/usbusx2yaudio.c > > +++ b/sound/usb/usx2y/usbusx2yaudio.c > > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > > if (subs) { > > if (atomic_read(&subs->state) >= state_PRERUNNING) { > > + unsigned long flags; > > + > > + local_irq_save(flags); > > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > > + local_irq_restore(flags); > > } ... actually this snd_pcm_stop() call should have been covered by snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch together with the change, i.e. wrapping with snd_pcm_stream_lock_irqsave(). I'll prepare the patch for 3.11 independently from your patch series, so please drop this one. BTW, the word "cleanup" in the subject is inappropriate. This is rather a fix together with the core change. And, I wonder whether we can take a safer approach. When the caller condition changed, we often introduced either a different ops (e.g. ioctl case) or a flag for the new condition, at least during the transition period. Last but not least, is a conversion to tasklet really preferred? tasklet is rather an obsoleted infrastructure nowadays, and people don't recommend to use it any longer, AFAIK. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 36/50] media: usb: em28xx: spin_lock in complete() cleanup
by Devin Heitmueller 11 Jul '13

11 Jul '13

On Thu, Jul 11, 2013 at 5:05 AM, Ming Lei <ming.lei(a)canonical.com> wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/em28xx/em28xx-core.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/drivers/media/usb/em28xx/em28xx-core.c b/drivers/media/usb/em28xx/em28xx-core.c > index fc157af..0d698f9 100644 > --- a/drivers/media/usb/em28xx/em28xx-core.c > +++ b/drivers/media/usb/em28xx/em28xx-core.c > @@ -941,6 +941,7 @@ static void em28xx_irq_callback(struct urb *urb) > { > struct em28xx *dev = urb->context; > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -956,9 +957,9 @@ static void em28xx_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->slock); > + spin_lock_irqsave(&dev->slock, flags); > dev->usb_ctl.urb_data_copy(dev, urb); > - spin_unlock(&dev->slock); > + spin_unlock_irqrestore(&dev->slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > -- > 1.7.9.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html I actually stumbled across this a couple of weeks ago, and have had an identical patch running in a local dev tree. Reviewed-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> Tested-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

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[alsa-devel] [PATCH 0/6] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (6): ASoC: dmaengine_pcm: Add dma_name to snd_dmaengine_dai_dma_data ASoC: omap-pcm: Request the DMA channel by name when dma_name is set for the dai ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT ASoC: omap-dmic: Cleanup the probe function include/sound/dmaengine_pcm.h | 1 + sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 35 ++++++++++------------------------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 16 +++++++++++++--- 5 files changed, 48 insertions(+), 59 deletions(-) -- 1.8.2.1

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[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

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[alsa-devel] CA0110 analog mic capture flatline
by Ki'Sak 11 Jul '13

11 Jul '13

Good day alsa-devel, I have a "Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG" which I have been unable to capture any content from the analog mic jack. I am a Gentoo Linux user with the 3.8.13 kernel / alsa 1.0.26. This issue has been filed with the wrong devs over at ubuntu's bugtracker and opensuse's forum. Also, I found a report that analog capture was working on this card as recently as September 2009. Thank you for your time, Ki'Sak Note: I have already tried to report this issue, this is a resend without urls due to 2 weeks without a response or it reaching the mailing list.

2 3

[alsa-devel] [PATCH 1/4] ASoC: pcm3008: Remove noisy version print
by Mark Brown 10 Jul '13

10 Jul '13

From: Mark Brown <broonie(a)linaro.org> The version number has never been updated and the printk isn't based on any interaction with the device. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282..32e5a59 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,8 +28,6 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" - #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -64,8 +62,6 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) struct pcm3008_setup_data *setup = codec->dev->platform_data; int ret = 0; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- 1.8.3.2

1 3

[alsa-devel] [PATCH] ALSA: usb-audio: fix regression for fixed stream quirk
by Eldad Zack 10 Jul '13

10 Jul '13

Commit 8f898e92aea2c24c7f379ee265d178f69ebb9c07 removed the redundant reads of bInterfaceProtocol from the descriptors, but introduced a regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT, since fp->protocol is not set in setup process. As a consequence, audio streams would not get initialized, as the following logs show: [ 48.923043] setting usb interface 3:1 [ 48.923056] Creating new capture data endpoint #81 [ 48.923484] 4:3:1: cannot set freq 48000 to ep 0x81 This patch sets fp->protocol in create_fixed_stream_quirk() and resolves the regression. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b01330..1bc45e7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -129,6 +129,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, { struct audioformat *fp; struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; int stream, err; unsigned *rate_table = NULL; @@ -166,6 +167,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + altsd = get_iface_desc(alts); + fp->protocol = altsd->bInterfaceProtocol; + if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2 0/5] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 10 Jul '13

10 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Tested on at91sam9g35ek. patches applies on next-20130708 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] [Sorry if it's too late for 3.11] Changes from V1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (4): ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: add sound configuration ARM: AT91: DTS: sam9x5ek: enable SSC arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 27 +++++ sound/soc/atmel/Kconfig | 10 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 206 +++++++++++++++++++++++++++++++++++ 5 files changed, 248 insertions(+) create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

5 44

[alsa-devel] [PATCH] firewire-speakers: remove not-reused member from structure
by Takashi Sakamoto 09 Jul '13

09 Jul '13

"pcm" member in struct fwspk is used to set pcm operations but is not used again. This commit remove this member and set pcm operations with snd_pcm_set_ops(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/speakers.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 2c63865..fe9e6e2 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -49,7 +49,6 @@ struct fwspk { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; - struct snd_pcm_substream *pcm; struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; @@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk) return err; pcm->private_data = fwspk; strcpy(pcm->name, fwspk->device_info->short_name); - fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - fwspk->pcm->ops = &ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); return 0; } -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH v2 1/5] sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards
by Richard Genoud 09 Jul '13

09 Jul '13

2013/7/8 Mark Brown <broonie(a)kernel.org>: > On Mon, Jul 08, 2013 at 03:29:49PM +0200, Richard Genoud wrote: > >> + * Nicolas Ferre <nicolas.ferre(a)atmel.com> >> + * >> + * Based on sam9g20_wm8731.c by: >> + * Sedji Gaouaou <sedji.gaouaou(a)atmel.com> > > The obvious question here is of course if we can use the same driver for > both of them. I haven't got a g20 to test that, but that's the goal. For now, g20 is still non-DT, so I think it's best to have a DT-only driver like this one for the 9x5 family (9g15, 9g25, 9x25, 9g35, 9x25). When the g20 will move to DT completely, we can drop sam9g20_wm8731.c and adjust sam9x5_wm8731.c (mainly master clock and widgets it seems) By the way, maybe g45 could use that also (and SAMA5 ?) > >> + codec_dai->driver->playback.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; >> + codec_dai->driver->capture.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; > > You definitely shouldn't be fiddling with a driver's constant static > data. You want to be using snd_pcm_hw_constraint() APIs to set > additional constraints intead. Ok, I'll change that. Thanks ! -- for me, ck means con kolivas and not calvin klein... does it mean I'm a geek ?

3 3

Re: [alsa-devel] [PATCH] ASoC: ak4554: add DT support
by Lars-Peter Clausen 08 Jul '13

08 Jul '13

On 07/06/2013 08:58 PM, Sylwester Nawrocki wrote: > On 07/05/2013 07:13 PM, Stephen Warren wrote: >> On 07/05/2013 11:04 AM, Lars-Peter Clausen wrote: >>> On 07/05/2013 06:55 PM, Stephen Warren wrote: >>>> On 07/05/2013 03:37 AM, Lars-Peter Clausen wrote: >>>>>> + - compatible : "asahi-kasei,ak4554" >>>>> >>>>> The vendor prefix for Asahi Kasei is documented as "ak" in >>>>> Documentation/devicetree/bindings/vendor-prefixes.txt. But it doesn;t seem >>>>> to be used in any upstream drivers, Stephen add that prefix maybe he knows >>>>> if it is used. If it is unused you should probably send a patch updating the >>>>> Documentation. >>>> >>>> It's used at: >>>> >>>> arch/arm/boot/dts/tegra20-seaboard.dts:499: compatible = "ak,ak8975"; >>>> >>>> ... although I see that somebody has incorrectly modified >>>> ./drivers/iio/magnetometer/ak8975.c to support the "asahi-kasei," vendor >>>> prefix:-( >>> >>> It looks as if the ak prefix was never in an upstream kernel and the commit >>> date that added the dt table predates the commit date for your devicetree. >> >> The usage I quoted above is in the mainline kernel, along with the entry >> in vendor-prefixes.txt. >> >> When I fixed vendor-prefixes.txt to define a prefix for AK, and updated >> the Tegra DT to use them[1], I guess I didn't even look at the driver, >> since all I2C device matching uses the i2c_device_id table rather than >> the of_device_id, and the i2c_device_id table doesn't have a concept of >> a vendor prefix. Hence, I didn't notice the conflict. >> >> So, even though the asahi-kasei prefix is in the of_device_id table in >> the driver, I'd say the ak prefix trumps it, since it's actually in-use >> in device trees, and is registered in vendor-prefixes.txt. > > I'm going to update this [2] patch to use "ak,ak8975" compatible string > then. Still there are some audio codec drivers that use "asahi-kasei" > vendor prefix: > > Documentation/devicetree/bindings/sound/ak4104.txt: - compatible : > "asahi-kasei,ak4104" > Documentation/devicetree/bindings/sound/ak4104.txt: compatible = > "asahi-kasei,ak4104"; > Documentation/devicetree/bindings/sound/ak4642.txt: - compatible : > "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" > Documentation/devicetree/bindings/sound/ak4642.txt: compatible = > "asahi-kasei,ak4642"; > Documentation/devicetree/bindings/sound/ak5386.txt: - compatible : > "asahi-kasei,ak5386" > Documentation/devicetree/bindings/sound/ak5386.txt: compatible = > "asahi-kasei,ak5386"; > ... > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = > &soc_codec_dev_ak4648}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > It's not in any dts file though, I'm not sure if that makes any > difference. > Well strictly speaking this is all public ABI, so I guess the best is to add a ak prefixed compatible string besides the asahi-kasei one and stick to ak only fore future drivers. - Lars

2 1

Re: [alsa-devel] No sound with Sony VAIO VPCZ1 (ALC889)
by Raymond Yau 08 Jul '13

08 Jul '13

>>> >>> Actually... there should be three mic channels: >>> The built-in (internal) microphone >>> + >>> One mic jack beside the headphone jack >>> + >>> One other hidden within the headphone jack: >>> (a) the Notebook came with Noise-cancelling headsets, but they are small in-ear plugs so there is no place for noise-cancelling logic in the plugs >>> (b) I think I remember to have read somewhere that this thingy actually needs the windows driver installed to be noise-cancelling! >>> (c) the PLUG of these headphones have another channel (visible through another ring on the jack) >>> >>> This explains the 2nd mic channel :) >> >> >> Try hda-jack-sense-test to find out which node is used by the headset mic jack (TRRS) and mic jack. > > > Thanks for the hint. > > Found and installed: snd-hda-tools_0.20111207_amd64.deb > But it gave me: > > /usr/src % hda-jack-sense-test $1 #28540 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 908, in __init__ > self.fd = os.open("/dev/snd/hwC%sD%s" % (card, device), os.O_RDWR) > OSError: [Errno 13] Permission denied: '/dev/snd/hwC0D0' You need root privilege and need 0.20130207 hda-tools > > /usr/src % sudo hda-jack-sense-test $1 #28541 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 921, in __init__ > self.parse_proc() > File "/usr/share/snd-hda-tools/hda_codec.py", line 1003, in parse_proc > self.proc_codec = HDACodecProc(self.card, self.device, file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 474, in __init__ > self.parse(proc_file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 647, in parse > line, nid = self.decodeintw(line, 'Node ') > File "/usr/share/snd-hda-tools/hda_proc.py", line 140, in decodeintw > self.wrongfile('integer decode %s (%s)' % (repr(str), repr(prefix))) > File "/usr/share/snd-hda-tools/hda_proc.py", line 143, in wrongfile > raise ValueError, "wrong proc file format (%s)" % msg > ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node ')) > http://git.alsa-project.org/?p=alsa.git;a=commitdiff;h=7859cb050f2cbeba54ce…

1 0

[alsa-devel] [PATCH] ASoC: rt5640: add headset detection function
by bardliao＠realtek.com 08 Jul '13

08 Jul '13

From: Bard Liao <bardliao(a)realtek.com> Add headset detection function to rt5640 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 62 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..657efb7 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1911,6 +1911,66 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define rt5640_resume NULL #endif +/** + * rt5640_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int jack_type; + int sclk_src = 0; + int reg63, reg64; + + if (jack_insert) { + reg63 = snd_soc_read(codec, RT5640_PWR_ANLG1); + reg64 = snd_soc_read(codec, RT5640_PWR_ANLG2); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_write(codec, RT5640_PWR_ANLG1, 0xa814); + sclk_src = snd_soc_read(codec, RT5640_GLB_CLK) & + RT5640_SCLK_SRC_MASK; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, + 0x2 << RT5640_SCLK_SRC_SFT); + } + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LDO2, RT5640_PWR_LDO2); + snd_soc_update_bits(codec, RT5640_PWR_ANLG2, + RT5640_PWR_MB1, RT5640_PWR_MB1); + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK | RT5640_MIC1_OVTH_MASK | + RT5640_PWR_CLK25M_MASK | RT5640_PWR_MB_MASK, + RT5640_MIC1_OVCD_EN | RT5640_MIC1_OVTH_600UA | + RT5640_PWR_MB_PU | RT5640_PWR_CLK25M_PU); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0x1); + msleep(100); + if (snd_soc_read(codec, RT5640_IRQ_CTRL2) & 0x8) + jack_type = RT5640_HEADPHO_DET; + else + jack_type = RT5640_HEADSET_DET; + snd_soc_update_bits(codec, RT5640_IRQ_CTRL2, + RT5640_MB1_OC_CLR, 0); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, sclk_src); + snd_soc_write(codec, RT5640_PWR_ANLG1, reg63); + snd_soc_write(codec, RT5640_PWR_ANLG2, reg64); + } else { + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK, + RT5640_MIC1_OVCD_DIS); + + jack_type = RT5640_NO_JACK; + } + + return jack_type; +} +EXPORT_SYMBOL_GPL(rt5640_headset_detect); + #define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..c3856c8 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2089,4 +2089,6 @@ struct rt5640_priv { int dmic_en; }; +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert); + #endif -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH - oss plugin update revised 1/2] oss: Add support of 32-bit and other formats
by alex.d.wiggins＠gmail.com 08 Jul '13

08 Jul '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added support of 32-bit and other audio formats via 3 changes: - Additional audio format definitions at the top of the file (if required); - Additional audio formats in oss_hw_params (to allow as input to pass to OSS); - Additional audio formats in oss_hw_constraint (to allow to report to ALSA). Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d43b44c..d5b5bc4 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -24,6 +24,32 @@ #include <alsa/pcm_external.h> #include <linux/soundcard.h> +/* definitions for additional OSS formats if required */ +#ifndef AFMT_S32_LE +#define AFMT_S32_LE 0x00001000 +#endif +#ifndef AFMT_S32_BE +#define AFMT_S32_BE 0x00002000 +#endif +#ifndef AFMT_S24_LE +#define AFMT_S24_LE 0x00008000 +#endif +#ifndef AFMT_S24_BE +#define AFMT_S24_BE 0x00010000 +#endif +#ifndef AFMT_S24_PACKED +#define AFMT_S24_PACKED 0x00040000 +#endif +#ifndef AFMT_FLOAT +#define AFMT_FLOAT 0x00004000 +#endif +#ifndef AFMT_SPDIF_RAW +#define AFMT_SPDIF_RAW 0x00020000 +#endif +#ifndef AFMT_VORBIS +#define AFMT_VORBIS 0x00000800 +#endif + typedef struct snd_pcm_oss { snd_pcm_ioplug_t io; char *device; @@ -165,6 +191,145 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, case SND_PCM_FORMAT_S16_BE: oss->format = AFMT_S16_BE; break; + /* allow additional data formats (such as 32 bit) as input + to pass to OSS (set in OSS driver) */ + case SND_PCM_FORMAT_S32_LE: + oss->format = AFMT_S32_LE; + break; + case SND_PCM_FORMAT_S32_BE: + oss->format = AFMT_S32_BE; + break; + case SND_PCM_FORMAT_S8: + oss->format = AFMT_S8; + break; + case SND_PCM_FORMAT_U16_LE: + oss->format = AFMT_U16_LE; + break; + case SND_PCM_FORMAT_U16_BE: + oss->format = AFMT_U16_BE; + break; + case SND_PCM_FORMAT_S24_LE: + oss->format = AFMT_S24_LE; + break; + case SND_PCM_FORMAT_S24_BE: + oss->format = AFMT_S24_BE; + break; + case SND_PCM_FORMAT_MU_LAW: + oss->format = AFMT_MU_LAW; + break; + case SND_PCM_FORMAT_A_LAW: + oss->format = AFMT_A_LAW; + break; + case SND_PCM_FORMAT_IMA_ADPCM: + oss->format = AFMT_IMA_ADPCM; + break; + case SND_PCM_FORMAT_MPEG: + oss->format = AFMT_MPEG; + break; + case SND_PCM_FORMAT_S24_3LE: + oss->format = AFMT_S24_PACKED; + break; + case SND_PCM_FORMAT_FLOAT: + oss->format = AFMT_FLOAT; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME: + oss->format = AFMT_SPDIF_RAW; + break; + /* following formats need defining checking native endian: */ + #if 0 + case SND_PCM_FORMAT_S16: + oss->format = AFMT_S16_NE; + break; + case SND_PCM_FORMAT_U16: + oss->format = AFMT_U16_NE; + break; + case SND_PCM_FORMAT_S24: + oss->format = AFMT_S24_NE; + break; + case SND_PCM_FORMAT_S32: + oss->format = AFMT_S32_NE; + #endif + /* following formats found in ALSA but not in OSS: */ + #if 0 + case SND_PCM_FORMAT_U24_LE: + oss->format = AFMT_U24_LE; + break; + case SND_PCM_FORMAT_U24_BE: + oss->format = AFMT_U24_BE; + break; + case SND_PCM_FORMAT_U32_LE: + oss->format = AFMT_U32_LE; + break; + case SND_PCM_FORMAT_U32_BE: + oss->format = AFMT_U32_BE; + break; + case SND_PCM_FORMAT_FLOAT_LE: + oss->format = AFMT_FLOAT_LE; + break; + case SND_PCM_FORMAT_FLOAT_BE: + oss->format = AFMT_FLOAT_BE; + break; + case SND_PCM_FORMAT_FLOAT64_LE: + oss->format = AFMT_FLOAT64_LE; + break; + case SND_PCM_FORMAT_FLOAT64_BE: + oss->format = AFMT_FLOAT64_BE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: + oss->format = AFMT_SPDIF_RAW_LE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: + oss->format = AFMT_SPDIF_RAW_BE; + break; + case SND_PCM_FORMAT_GSM: + oss->format = AFMT_GSM; + break; + case SND_PCM_FORMAT_SPECIAL: + oss->format = AFMT_SPECIAL; + break; + case SND_PCM_FORMAT_S24_3BE: + oss->format = AFMT_S24_PACKED_BE; + break; + case SND_PCM_FORMAT_U24_3LE: + oss->format = AFMT_U24_PACKED; + break; + case SND_PCM_FORMAT_U24_3BE: + oss->format = AFMT_U24_PACKED_BE; + break; + case SND_PCM_FORMAT_S20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U24: + oss->format = AFMT_U24_NE; + break; + case SND_PCM_FORMAT_U32: + oss->format = AFMT_U32_NE; + break; + case SND_PCM_FORMAT_FLOAT64: + oss->format = AFMT_FLOAT64; + break; + #endif default: fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); return -EINVAL; @@ -239,7 +404,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) SND_PCM_ACCESS_MMAP_INTERLEAVED }; unsigned int nformats; - unsigned int format[5]; + unsigned int format[20]; unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ @@ -273,6 +438,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) format[nformats++] = SND_PCM_FORMAT_S16_BE; if (tmp & AFMT_MU_LAW) format[nformats++] = SND_PCM_FORMAT_MU_LAW; + /* allow additional data formats (such as 32 bit) + to be reported from OSS to ALSA below */ + if (tmp & AFMT_S32_LE) + format[nformats++] = SND_PCM_FORMAT_S32_LE; + if (tmp & AFMT_S32_BE) + format[nformats++] = SND_PCM_FORMAT_S32_BE; + if (tmp & AFMT_A_LAW) + format[nformats++] = SND_PCM_FORMAT_A_LAW; + if (tmp & AFMT_IMA_ADPCM) + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; + if (tmp & AFMT_S8) + format[nformats++] = SND_PCM_FORMAT_S8; + if (tmp & AFMT_U16_LE) + format[nformats++] = SND_PCM_FORMAT_U16_LE; + if (tmp & AFMT_U16_BE) + format[nformats++] = SND_PCM_FORMAT_U16_BE ; + if (tmp & AFMT_MPEG) + format[nformats++] = SND_PCM_FORMAT_MPEG; + if (tmp & AFMT_FLOAT) + format[nformats++] = SND_PCM_FORMAT_FLOAT; + if (tmp & AFMT_S24_LE) + format[nformats++] = SND_PCM_FORMAT_S24_LE; + if (tmp & AFMT_S24_BE) + format[nformats++] = SND_PCM_FORMAT_S24_BE; + if (tmp & AFMT_SPDIF_RAW) + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; + if (tmp & AFMT_S24_PACKED) + format[nformats++] = SND_PCM_FORMAT_S24_3LE; + /* following formats found in OSS but not in ALSA */ + if (tmp & AFMT_AC3) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + if (tmp & AFMT_VORBIS) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + /* assume signed 16 bit (native CPU endian) if not found above */ if (! nformats) format[nformats++] = SND_PCM_FORMAT_S16; if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, -- 1.8.3.1

4 5

[alsa-devel] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 08 Jul '13

08 Jul '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Version 9 contains some style fixes. Regards, Markus Changes in v9: - Style and compile fixes. - Removed "ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97". This patch is not necessary with the seperated phycore-ac97-dt driver. Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 451 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 242 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 704 insertions(+), 115 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

4 25

[alsa-devel] Digital Mute DAI control
by Vishal Thanki 08 Jul '13

08 Jul '13

Hi All, I am new to ALSA development. I want to understand the what is the purpose of ".digital_mute" DAI control. Also, why during ".prepare" routine, the ".digital_mute" is invoked? Thanks in advance, Vishal

1 0

[alsa-devel] ROG Xonar Phoebus support planned?
by Andreas Allacher 06 Jul '13

06 Jul '13

Hello, are there any plans to fully (and officially) support the ROG Xonar Phoebus - with good audio quality (last time there was a discussion regarding this it apparantly was possible to use the card but sound quality was bad)

2 1

[alsa-devel] [usb-audio] VIA/Audinst HUD-mx2: One of two PCM device is available for playback
by Takashi Sakamoto 06 Jul '13

06 Jul '13

Hi, This sound card has two PCM devices. If the sound card is realized "card 0", there are "hw:0,0" and "hw:0,1". But former seems not to generate sound and latter seems to generate. This issue is reported to Ubuntu-jp and LinuxMint forum. The reporter said this happens in Fedora18 so I think this does not depends on these desktop invironment. https://forums.ubuntulinux.jp/viewtopic.php?pid=98940#p98940 http://forums.linuxmint.com/viewtopic.php?f=49&t=128478 Please see attached lsusb.log to know usb information and alsainfo.sh in detail. Audinst HUD-mx2 applies VIA Technologies chip internally. http://www.audinst.com/en/bbs/board.php?bo_table=HUDmx2&wr_id=4 The sound card has a lamp which turn on when data comes. When "hw:0,0" is used, the lamp turns on but no sound. When "hw:0,1" is used, the lamp turns on and sound. My question is: 1. does hw:0,0 seem to be available when some control are changed? 2. are there some ways to use hw:0,1 as a default? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 3

[alsa-devel] [PATCH] ASoC: Allow imx-pcm-{dma, fiq}.c to be modules
by Arnd Bergmann 05 Jul '13

05 Jul '13

This is required so we can build the imx sound support when ALSA itself is a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa43854..805b83e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -109,11 +109,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 67f656c..0e3619e 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -32,7 +32,7 @@ imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, dma_data->peripheral_type = IMX_DMATYPE_SSI; } -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -46,7 +46,7 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev); void imx_pcm_fiq_exit(struct platform_device *pdev); #else

2 4

[alsa-devel] [PATCH] ASoC: ak4554: add DT support
by Kuninori Morimoto 05 Jul '13

05 Jul '13

Support for loading the ak4554 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is for mark/topic/ak4554 branch Documentation/devicetree/bindings/sound/ak4554.c | 11 +++++++++++ sound/soc/codecs/ak4554.c | 7 +++++++ 2 files changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c diff --git a/Documentation/devicetree/bindings/sound/ak4554.c b/Documentation/devicetree/bindings/sound/ak4554.c new file mode 100644 index 0000000..934fa02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4554.c @@ -0,0 +1,11 @@ +AK4554 ADC/DAC + +Required properties: + + - compatible : "asahi-kasei,ak4554" + +Example: + +ak4554-adc-dac { + compatible = "asahi-kasei,ak4554"; +}; diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index c1a1733..6aed9c4 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -64,10 +64,17 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } +static struct of_device_id ak4554_of_match[] = { + { .compatible = "asahi-kasei,ak4554" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4554_of_match); + static struct platform_driver ak4554_driver = { .driver = { .name = "ak4554-adc-dac", .owner = THIS_MODULE, + .of_match_table = ak4554_of_match, }, .probe = ak4554_soc_probe, .remove = ak4554_soc_remove, -- 1.7.9.5

4 5

[alsa-devel] ASoC updates for v3.11
by Mark Brown 05 Jul '13

05 Jul '13

The following changes since commit 8bb495e3f02401ee6f76d1b1d77f3ac9f079e376: Linux 3.10 (2013-06-30 15:13:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-4 for you to fetch changes up to 70a9f8797ce298544cfd07328ff59f00488bc491: Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus (2013-07-04 18:30:03 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 A few small fixes, all driver specific. The removal of the GPIO based pinmuxing is a bug fix, since the obsolete nodes had been removed from the DT it stopped the driver loading. ---------------------------------------------------------------- Arnaud Patard (1): ASoC: imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found Mark Brown (5): ASoC: samsung: Remove obsolete GPIO based DT pinmuxing Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/mxs' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Sachin Kamat (1): ASoC: Samsung: Remove redundant comment Shawn Guo (2): ASoC: mxs: register saif mclk to clock framework ASoC: wm8962: fix NULL pdata pointer Documentation/devicetree/bindings/mfd/arizona.txt | 62 + .../devicetree/bindings/sound/adi,adau1701.txt | 23 + .../devicetree/bindings/sound/imx-audio-wm8962.txt | 46 + .../devicetree/bindings/sound/mxs-saif.txt | 17 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 71 + Documentation/devicetree/bindings/sound/rt5640.txt | 30 + .../devicetree/bindings/sound/sgtl5000.txt | 3 + .../devicetree/bindings/sound/spdif-receiver.txt | 10 + .../bindings/sound/spdif-transmitter.txt | 10 + .../devicetree/bindings/sound/ssm2518.txt | 20 + Documentation/devicetree/bindings/sound/wm8962.txt | 23 + arch/blackfin/mach-bf527/boards/ad7160eval.c | 12 - arch/blackfin/mach-bf527/boards/ezkit.c | 32 +- arch/blackfin/mach-bf533/boards/ezkit.c | 12 - arch/blackfin/mach-bf533/boards/stamp.c | 35 +- arch/blackfin/mach-bf537/boards/stamp.c | 30 +- arch/blackfin/mach-bf548/boards/ezkit.c | 28 +- arch/blackfin/mach-bf561/boards/ezkit.c | 14 +- arch/blackfin/mach-bf609/boards/ezkit.c | 2 +- drivers/mfd/arizona-core.c | 234 ++- drivers/mfd/arizona-i2c.c | 10 +- drivers/mfd/arizona-spi.c | 10 +- drivers/mfd/arizona.h | 12 + drivers/mfd/wm5102-tables.c | 9 +- drivers/mfd/wm5110-tables.c | 8 + drivers/misc/atmel-ssc.c | 8 +- include/linux/mfd/arizona/core.h | 2 + include/linux/mfd/arizona/pdata.h | 2 +- include/linux/mfd/arizona/registers.h | 47 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/mfd/wm8994/registers.h | 8 + include/linux/platform_data/ssm2518.h | 22 + include/sound/rt5640.h | 22 + include/sound/soc-dapm.h | 3 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/sam9g20_wm8731.c | 9 - sound/soc/blackfin/Kconfig | 47 +- sound/soc/blackfin/Makefile | 4 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 - sound/soc/blackfin/bf5xx-ac97.c | 8 +- sound/soc/blackfin/bf5xx-ad1836.c | 19 +- sound/soc/blackfin/bf5xx-ad193x.c | 40 +- sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 183 +- sound/soc/blackfin/bf5xx-i2s-pcm.h | 21 +- sound/soc/blackfin/bf5xx-i2s.c | 129 +- sound/soc/blackfin/bf5xx-sport.c | 10 +- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 345 ---- sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 - sound/soc/blackfin/bf5xx-tdm.c | 328 --- sound/soc/blackfin/bf5xx-tdm.h | 23 - sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 16 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 138 +- sound/soc/codecs/88pm860x-codec.c | 9 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 14 +- sound/soc/codecs/ab8500-codec.c | 85 +- sound/soc/codecs/ab8500-codec.h | 42 +- sound/soc/codecs/adau1701.c | 69 +- sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 3 +- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +- sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/max98090.c | 24 +- sound/soc/codecs/rt5640.c | 2128 ++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2092 +++++++++++++++++++ sound/soc/codecs/sgtl5000.c | 267 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 10 + .../{spdif_transciever.c => spdif_transmitter.c} | 10 + sound/soc/codecs/ssm2518.c | 856 ++++++++ sound/soc/codecs/ssm2518.h | 20 + sound/soc/codecs/wm0010.c | 10 + sound/soc/codecs/wm5102.c | 205 +- sound/soc/codecs/wm5110.c | 192 ++ sound/soc/codecs/wm8962.c | 145 +- sound/soc/codecs/wm8994.c | 188 +- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_adsp.c | 461 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/davinci/Kconfig | 10 - sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/davinci/davinci-sffsdr.c | 181 -- sound/soc/dwc/designware_i2s.c | 6 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 13 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 +- sound/soc/fsl/imx-audmux.c | 8 - sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 + sound/soc/fsl/imx-pcm.c | 145 -- sound/soc/fsl/imx-pcm.h | 10 +- sound/soc/fsl/imx-sgtl5000.c | 43 +- sound/soc/fsl/imx-ssi.c | 44 +- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/imx-wm8962.c | 323 +++ sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 17 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 1 - sound/soc/mxs/mxs-pcm.c | 18 - sound/soc/mxs/mxs-pcm.h | 7 - sound/soc/mxs/mxs-saif.c | 72 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 10 +- sound/soc/omap/Makefile | 1 - sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 - sound/soc/pxa/Kconfig | 20 - sound/soc/pxa/Makefile | 4 - sound/soc/pxa/mmp-pcm.c | 6 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/saarb.c | 190 -- sound/soc/pxa/tavorevb3.c | 189 -- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 14 + sound/soc/samsung/i2s.c | 66 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 4 +- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/sh/fsi.c | 10 +- sound/soc/soc-core.c | 106 +- sound/soc/soc-dapm.c | 110 +- sound/soc/soc-pcm.c | 91 +- sound/soc/soc-utils.c | 13 +- sound/soc/spear/Kconfig | 9 + sound/soc/spear/Makefile | 8 + sound/soc/spear/spdif_in.c | 31 +- sound/soc/spear/spdif_out.c | 43 +- sound/soc/spear/spear_pcm.c | 152 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra30_ahub.c | 25 + sound/soc/tegra/tegra30_i2s.c | 26 + sound/soc/tegra/tegra_asoc_utils.c | 23 +- sound/soc/tegra/tegra_rt5640.c | 257 +++ sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/mop500_ab8500.c | 62 +- sound/soc/ux500/ux500_msp_dai.c | 11 +- sound/soc/ux500/ux500_msp_dai.h | 4 - sound/soc/ux500/ux500_msp_i2s.c | 88 +- sound/soc/ux500/ux500_msp_i2s.h | 74 +- sound/soc/ux500/ux500_pcm.c | 30 + 159 files changed, 8829 insertions(+), 3121 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/arizona.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 include/sound/rt5640.h delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.h rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) create mode 100644 sound/soc/codecs/rt5640.c create mode 100644 sound/soc/codecs/rt5640.h rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h delete mode 100644 sound/soc/davinci/davinci-sffsdr.c delete mode 100644 sound/soc/fsl/imx-pcm.c create mode 100644 sound/soc/fsl/imx-wm8962.c delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/tegra/tegra_rt5640.c

2 1

[alsa-devel] [PATCHv2 00/31] HDSPM: A lot
by Adrian Knoth 05 Jul '13

05 Jul '13

Hi! Here's a revised series of the two patchsets I've sent earlier this week. Features by card: AIO: - Support Analog Expansion Boards - Support all card settings like gain, coax/optical switching etc. - Fixed S/PDIF Sync status and frequency reporting RayDAT: - Controls for S/PDIF and WordClock AES(32): - Support for TCO module (originally by Martin Dausel) All cards: - Code cleanups - Documentation Takashi: As agreed, I've split the big "TCO patch" into smaller chunks to ease reviewing. Adrian Knoth (30): ALSA: hdspm - Add missing defines for RME AIO and RayDAT ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ALSA: hdspm - Introduce generic AIO tristate control ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO ALSA: hdspm - Add support for AEBs on RME AIO ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO ALSA: hdspm - Create TCO readout function ALSA: hdspm - AES32: Fix TCO sync check reporting ALSA: hdspm - Cosmetics, no real change ALSA: hdspm - AIO: Drop superfluous HDSPM_AUTOSYNC_REF ALSA: hdspm - AES32: Add TCO and Sync-In text entries ALSA: hdspm - Introduce hdspm_get_aes_sample_rate() ALSA: hdspm - Add prototype declarations ALSA: hdspm - Enable AES32 in hdspm_get_wc_sample_rate ALSA: hdspm - Enable AES32 in hdspm_get_tco_sample_rate ALSA: hdspm - AES32: Ignore float/int format bit ALSA: hdspm - AES32: Enable TCO input in hdspm_external_sample_rate() ALSA: hdspm - AES32: Enable TCO/Sync-In in snd_hdspm_put_sync_ref() ALSA: hdspm - AES32: Include TCO and Sync-In in proc output ALSA: hdspm - Introduce hdspm_external_rate_to_enum() helper function ALSA: hdspm - AES32: Report external sample rate to userspace ALSA: hdspm - AES32: Enable TCO support ALSA: hdspm - Use snd_ctl_enum_info for most text arrays ALSA: hdspm - Use snd_ctl_enum_info() for texts_autosync ALSA: hdspm - Use snd_ctl_enum_info() in snd_hdspm_info_autosync_ref Martin Dausel (1): ALSA: hdspm - Added some comments and control register documentation sound/pci/rme9652/hdspm.c | 765 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 538 insertions(+), 227 deletions(-) -- 1.8.3.1

2 32

[alsa-devel] [PATCH 1/3] ASoC: sglt5000: Provide the reg_stride field
by Fabio Estevam 05 Jul '13

05 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 has 16-bit registers, so we need to tell regmap to use a reg_stride of two bytes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7c99f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- 1.8.1.2

2 3

[alsa-devel] [PATCH] drm/i915: switch disable_power_well default value to 1
by Paulo Zanoni 04 Jul '13

04 Jul '13

From: Paulo Zanoni <paulo.r.zanoni(a)intel.com> Now that the audio driver is using our power well API, everything should be working correctly, so let's give it a try. Signed-off-by: Paulo Zanoni <paulo.r.zanoni(a)intel.com> --- drivers/gpu/drm/i915/i915_drv.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.c b/drivers/gpu/drm/i915/i915_drv.c index 33cb973..e6dc81c 100644 --- a/drivers/gpu/drm/i915/i915_drv.c +++ b/drivers/gpu/drm/i915/i915_drv.c @@ -123,10 +123,10 @@ module_param_named(preliminary_hw_support, i915_preliminary_hw_support, int, 060 MODULE_PARM_DESC(preliminary_hw_support, "Enable preliminary hardware support. (default: false)"); -int i915_disable_power_well __read_mostly = 0; +int i915_disable_power_well __read_mostly = 1; module_param_named(disable_power_well, i915_disable_power_well, int, 0600); MODULE_PARM_DESC(disable_power_well, - "Disable the power well when possible (default: false)"); + "Disable the power well when possible (default: true)"); int i915_enable_ips __read_mostly = 1; module_param_named(enable_ips, i915_enable_ips, int, 0600); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: arizona: Add default case to silence build warning
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb5c80d..bcf739d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -233,6 +233,9 @@ int arizona_init_gpio(struct snd_soc_codec *codec) switch (arizona->type) { case WM5110: snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; } snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8962: fix NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

There is an error in merge commit 384b834 on conflict resolution which causes the following NULL pdata pointer bug. wm8962 0-001a: customer id 0 revision D Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0+ #1 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x134/0x6c8 LR is at regmap_unlock_mutex+0x10/0x14 pc : [<80452100>] lr : [<80304cf4>] psr: a0000113 sp : bf875c98 ip : 00000000 fp : bf875cd4 r10: 00000000 r9 : bfb1830c r8 : 80779bc4 r7 : 00000000 r6 : 00000001 r5 : bfbac010 r4 : bfb33e00 r3 : 80304ce4 r2 : 00000000 r1 : 00000001 r0 : fffffffb Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 1000404a DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0xbf874238) Stack: (0xbf875c98 to 0xbf876000) ... Fix the error by assigning pdata a correct pointer. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 00/11] HDSPM: RME AIO/RayDAT extensions and cleanups
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here's a series of patches that most importantly add support for AIO's Analog Expansion Boards (AEBs) and all the missing controls to adjust card settings (output levels, optical/coaxial etc.). I also refactored a couple of functions to clean up the code and fixed the S/PDIF sync/frequency reporting. Cheers Adrian Knoth (11): ALSA: hdspm - Add missing defines for RME AIO and RayDAT ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ALSA: hdspm - Introduce generic AIO tristate control ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO ALSA: hdspm - Add support for AEBs on RME AIO ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO sound/pci/rme9652/hdspm.c | 254 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 199 insertions(+), 55 deletions(-) -- 1.8.3.1

2 13

[alsa-devel] [RFC] [patch 1/1] imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
by Arnaud Patard 04 Jul '13

04 Jul '13

If the ssi or codec drivers are not loaded (for instance, because spi or i2c bus drivers are not loaded), returning -EINVAL will for people to unload and then reload the module to get sound working. Returning E_PROBE_DEFER will mitigate this. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux/sound/soc/fsl/imx-sgtl5000.c =================================================================== --- linux.orig/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:33:01.000000000 +0200 +++ linux/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:35:14.000000000 +0200 @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct pla ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);

3 4

[alsa-devel] [PATCH 10/11] ALSA: hdspm - Add support for AEBs on RME AIO
by Adrian Knoth 04 Jul '13

04 Jul '13

AIO cards allow to use AEB (Analogue Expansion Boards) to add four input and/or output channels. This patch adds the necessary code to detect and enable the additional I/O channels. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 111713c..585397b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -648,7 +648,8 @@ static char *texts_ports_aio_in_ss[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", - "ADAT.7", "ADAT.8" + "ADAT.7", "ADAT.8", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ss[] = { @@ -657,14 +658,16 @@ static char *texts_ports_aio_out_ss[] = { "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", "ADAT.7", "ADAT.8", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_ds[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ds[] = { @@ -672,14 +675,16 @@ static char *texts_ports_aio_out_ds[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_qs[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_qs[] = { @@ -687,7 +692,8 @@ static char *texts_ports_aio_out_qs[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aes32[] = { @@ -764,8 +770,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in, */ 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -779,7 +785,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -792,7 +799,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 14, 16, 18, /* adat in */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -807,7 +815,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -821,7 +829,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 16, /* adat in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -836,7 +845,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -6602,10 +6612,6 @@ static int snd_hdspm_create(struct snd_card *card, break; case AIO: - if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { - snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n"); - } - hdspm->ss_in_channels = AIO_IN_SS_CHANNELS; hdspm->ds_in_channels = AIO_IN_DS_CHANNELS; hdspm->qs_in_channels = AIO_IN_QS_CHANNELS; @@ -6613,6 +6619,20 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS; hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS; + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { + snd_printk(KERN_INFO "HDSPM: AEB input board found\n"); + hdspm->ss_in_channels += 4; + hdspm->ds_in_channels += 4; + hdspm->qs_in_channels += 4; + } + + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) { + snd_printk(KERN_INFO "HDSPM: AEB output board found\n"); + hdspm->ss_out_channels += 4; + hdspm->ds_out_channels += 4; + hdspm->qs_out_channels += 4; + } + hdspm->channel_map_out_ss = channel_map_aio_out_ss; hdspm->channel_map_out_ds = channel_map_aio_out_ds; hdspm->channel_map_out_qs = channel_map_aio_out_qs; -- 1.8.3.1

1 1

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Remove redundant comment
by Sachin Kamat 04 Jul '13

04 Jul '13

There is a typo in the filename (i2c mentioned instead of i2s). However, this is a redundant piece of information. Delete it altogether. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/s3c-i2s-v2.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 20e98d1..e5e81b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,6 +1,4 @@ -/* sound/soc/samsung/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. * * Copyright (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory(a)wolfsonmicro.com -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: add ak4554 driver
by Kuninori Morimoto 04 Jul '13

04 Jul '13

ak4554 is very simple DA/AD converter which has no setting register. But, playback format is SND_SOC_DAIFMT_RIGHT_J, and, capture format is SND_SOC_DAIFMT_LEFT_J on same bit clock, LR clock. Because of that, snd_soc_dai_driver consists from "playback only" and "capture only" here, and it has software base symmetric_rates check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4554.c | 171 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 176 insertions(+) create mode 100644 sound/soc/codecs/ak4554.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..b4ec4ea 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -196,6 +196,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..e601a47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -136,6 +137,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 0000000..277f216 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,171 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <sound/soc.h> + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * But, playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * Because of that, snd_soc_dai_driver consists from + * "playback only" and "capture only" here. + */ + +struct ak4554_priv { + int rate; + int usrcnt; +}; + +static int ak4554_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + int dai_is_play = (dai->id == 0); + int fmt_is_play = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + dev_err(dai->dev, "can't be clock master\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + fmt_is_play = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt_is_play = 0; + break; + } + + if (dai_is_play != fmt_is_play) { + dev_err(dai->dev, "format error\n"); + return -EINVAL; + } + + return 0; +} + +static int ak4554_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + if ((priv->usrcnt > 0) && + (priv->rate != params_rate(params))) { + dev_err(dai->dev, "asymmetric rate\n"); + return -EIO; + } + + priv->usrcnt++; + priv->rate = params_rate(params); + + return 0; +} + +static void ak4554_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + priv->usrcnt--; + + if (priv->usrcnt <= 0) + priv->rate = 0; +} + +static const struct snd_soc_dai_ops ak4554_dai_ops = { + .set_fmt = ak4554_dai_set_fmt, + .hw_params = ak4554_dai_hw_params, + .shutdown = ak4554_dai_shutdown, +}; + +struct snd_soc_dai_driver ak4554_dai[] = { + { + .name = "ak4554-hifi-playback", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + }, { + .name = "ak4554-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + } +}; +EXPORT_SYMBOL_GPL(ak4554_dai); + +static int ak4554_probe(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "probed\n"); + return 0; +} + +static int ak4554_remove(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "removed\n"); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .probe = ak4554_probe, + .remove = ak4554_remove, +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + struct ak4554_priv *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(struct ak4554_priv), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + ak4554_dai, ARRAY_SIZE(ak4554_dai)); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 1.7.9.5

3 8

[alsa-devel] [PATCH 0/2] HDSPM - Documentation and TCO on AES(32)
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here are two patches that I've received from Martin. The first only adds comments to the driver to document the various register settings. Might be useful for other developers. The second patch adds TCO support on AES(32) cards. I took the liberty to polish Martin's work to reflect the kernel coding style. It still contains plenty of long-line warnings from checkpatch.pl, but I could live with this. There are also a few (five?) whitespace fixes and a typo fix which are technically not related to TCO support. I could manually separate them into two commits, but maybe it's acceptable as it is? Cheers Adrian Knoth (1): ALSA: hdspm - Add TCO support for RME AES(32) Martin Dausel (1): ALSA: hdspm - Added some comments and control register documentation sound/pci/rme9652/hdspm.c | 438 +++++++++++++++++++++++++++++++++------------- 1 file changed, 317 insertions(+), 121 deletions(-) -- 1.8.3.1

2 3

[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/gpio.h | 96 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/arizona/gpio.h diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h new file mode 100644 index 0000000..d2146bb --- /dev/null +++ b/include/linux/mfd/arizona/gpio.h @@ -0,0 +1,96 @@ +/* + * GPIO configuration for Arizona devices + * + * Copyright 2013 Wolfson Microelectronics. PLC. + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ARIZONA_GPIO_H +#define _ARIZONA_GPIO_H + +#define ARIZONA_GP_FN_TXLRCLK 0x00 +#define ARIZONA_GP_FN_GPIO 0x01 +#define ARIZONA_GP_FN_IRQ1 0x02 +#define ARIZONA_GP_FN_IRQ2 0x03 +#define ARIZONA_GP_FN_OPCLK 0x04 +#define ARIZONA_GP_FN_FLL1_OUT 0x05 +#define ARIZONA_GP_FN_FLL2_OUT 0x06 +#define ARIZONA_GP_FN_PWM1 0x08 +#define ARIZONA_GP_FN_PWM2 0x09 +#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A +#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B +#define ARIZONA_GP_FN_FLL1_LOCK 0x0C +#define ARIZONA_GP_FN_FLL2_LOCK 0x0D +#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F +#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10 +#define ARIZONA_GP_FN_HEADPHONE_DET 0x12 +#define ARIZONA_GP_FN_MIC_DET 0x13 +#define ARIZONA_GP_FN_WSEQ_STATUS 0x15 +#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16 +#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A +#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B +#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C +#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D +#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E +#define ARIZONA_GP_FN_DRC1_DECAY 0x1F +#define ARIZONA_GP_FN_DRC1_NOISE 0x20 +#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21 +#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22 +#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23 +#define ARIZONA_GP_FN_DRC2_DECAY 0x24 +#define ARIZONA_GP_FN_DRC2_NOISE 0x25 +#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26 +#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27 +#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28 +#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29 +#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A +#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B +#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C +#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D +#define ARIZONA_GP_FN_OVERCLOCKED 0x2E +#define ARIZONA_GP_FN_DSP_IRQ1 0x35 +#define ARIZONA_GP_FN_DSP_IRQ2 0x36 +#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D +#define ARIZONA_GP_FN_BOOT_DONE 0x44 +#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45 +#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B +#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C + +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +#endif -- 1.7.2.5

2 6

[alsa-devel] ASoC: wm8962: NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

Mark, I'm running next-20130703 and seeing the following NULL pointer bug. Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0-next-20130703 #6 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x140/0x6b4 ... It seems to me that it's caused by a merge error in 384b834. And the change below can remove the problem. Shawn ---8<---- diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat;

3 2

[alsa-devel] [PATCH] Add -lm to test programs to fix build failure.
by Jordi Mallach 04 Jul '13

04 Jul '13

Add -lm to test programs that use sin(), to fix a build failure with recent binutils. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- test/Makefile.am | 2 ++ 1 file changed, 2 insertions(+) diff --git a/test/Makefile.am b/test/Makefile.am index 8705402..c1896f8 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -7,8 +7,10 @@ check_PROGRAMS=control pcm pcm_min latency seq \ control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la +pcm_LDFLAGS= -lm pcm_min_LDADD=../src/libasound.la latency_LDADD=../src/libasound.la +latency_LDFLAGS= -lm seq_LDADD=../src/libasound.la playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la -- 1.8.3.1

1 1

[alsa-devel] [PATCH v2 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 03 Jul '13

03 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/gpio.h | 96 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/arizona/gpio.h diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h new file mode 100644 index 0000000..d2146bb --- /dev/null +++ b/include/linux/mfd/arizona/gpio.h @@ -0,0 +1,96 @@ +/* + * GPIO configuration for Arizona devices + * + * Copyright 2013 Wolfson Microelectronics. PLC. + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ARIZONA_GPIO_H +#define _ARIZONA_GPIO_H + +#define ARIZONA_GP_FN_TXLRCLK 0x00 +#define ARIZONA_GP_FN_GPIO 0x01 +#define ARIZONA_GP_FN_IRQ1 0x02 +#define ARIZONA_GP_FN_IRQ2 0x03 +#define ARIZONA_GP_FN_OPCLK 0x04 +#define ARIZONA_GP_FN_FLL1_OUT 0x05 +#define ARIZONA_GP_FN_FLL2_OUT 0x06 +#define ARIZONA_GP_FN_PWM1 0x08 +#define ARIZONA_GP_FN_PWM2 0x09 +#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A +#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B +#define ARIZONA_GP_FN_FLL1_LOCK 0x0C +#define ARIZONA_GP_FN_FLL2_LOCK 0x0D +#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F +#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10 +#define ARIZONA_GP_FN_HEADPHONE_DET 0x12 +#define ARIZONA_GP_FN_MIC_DET 0x13 +#define ARIZONA_GP_FN_WSEQ_STATUS 0x15 +#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16 +#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A +#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B +#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C +#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D +#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E +#define ARIZONA_GP_FN_DRC1_DECAY 0x1F +#define ARIZONA_GP_FN_DRC1_NOISE 0x20 +#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21 +#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22 +#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23 +#define ARIZONA_GP_FN_DRC2_DECAY 0x24 +#define ARIZONA_GP_FN_DRC2_NOISE 0x25 +#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26 +#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27 +#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28 +#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29 +#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A +#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B +#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C +#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D +#define ARIZONA_GP_FN_OVERCLOCKED 0x2E +#define ARIZONA_GP_FN_DSP_IRQ1 0x35 +#define ARIZONA_GP_FN_DSP_IRQ2 0x36 +#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D +#define ARIZONA_GP_FN_BOOT_DONE 0x44 +#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45 +#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B +#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C + +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +#endif -- 1.7.2.5

2 3

[alsa-devel] [PATCH] mfd: arizona: Move regulator disable to after marking cache only
by Charles Keepax 03 Jul '13

03 Jul '13

If we disable DCVDD before we mark the cache as cache only, we might attempt to write to the chip whilst it is powered down and lose a write. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 6ab0304..aa47709 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -388,9 +388,9 @@ static int arizona_runtime_suspend(struct device *dev) dev_dbg(arizona->dev, "Entering AoD mode\n"); - regulator_disable(arizona->dcvdd); regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); + regulator_disable(arizona->dcvdd); return 0; } -- 1.7.2.5

1 0

[alsa-devel] [v2 0/5] ASoC: atmel-pcm: align with generic dmaengine framwork
by Bo Shen 03 Jul '13

03 Jul '13

This patch set is based on linux next-20130701 mainly implement - move set dma data from hw_params to startup callback to avoid memory leak - align atmel pcm with generic dmaenigne framwork Bo Shen (5): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel_ssc_dai: add error mask define ASoC: atmel-pcm-dma: move prepare for dma to dai prepare ARM: atmel-ssc: change phybase type to dma_addr_t ASoC: atmel-pcm: use generic dmaengine framework include/linux/atmel-ssc.h | 2 +- sound/soc/atmel/Kconfig | 1 + sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 36 ++++++------ 4 files changed, 37 insertions(+), 120 deletions(-) -- 1.7.9.5

3 7

[alsa-devel] [PATCH 0/2] ASoC: atmel-pcm: use generic dmaengine framework
by Bo Shen 03 Jul '13

03 Jul '13

this patch series is based on next-20130701 Bo Shen (2): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel-pcm: use generic dmaengine framework sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 33 ++++++----- 2 files changed, 32 insertions(+), 119 deletions(-) -- 1.7.9.5

3 11

[alsa-devel] no analog playback device for HDA STAC9221 inside VirtualBox 4.2.14
by Raymond Yau 03 Jul '13

03 Jul '13

HDA STAC9221 audio work for (64bit) Fedora 16, Ubuntu 11.10 and Opensuse 11.4 Live CD inside virtualbox 4.2.14 no sound for this emulated HDA sound card since Ubuntu 12.04 Live CD No analog playback device for Saucy or Fedora 19 Live CD

2 7

[alsa-devel] [PATCH] ASoC: mop500: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/ux500/mop500.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 8f5cd00..178d1ba 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -52,6 +52,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { static struct snd_soc_card mop500_card = { .name = "MOP500-card", + .owner = THIS_MODULE, .probe = NULL, .dai_link = mop500_dai_links, .num_links = ARRAY_SIZE(mop500_dai_links),

2 1

[alsa-devel] [PATCH] ASoC: db1200: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/au1x/db1200.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a497a0c..2d2407d 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = { static struct snd_soc_card db1300_ac97_machine = { .name = "DB1300_AC97", + .owner = THIS_MODULE, .dai_link = &db1300_ac97_dai, .num_links = 1, }; static struct snd_soc_card db1550_ac97_machine = { .name = "DB1550_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { static struct snd_soc_card db1300_i2s_machine = { .name = "DB1300_I2S", + .owner = THIS_MODULE, .dai_link = &db1300_i2s_dai, .num_links = 1, }; @@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { static struct snd_soc_card db1550_i2s_machine = { .name = "DB1550_I2S", + .owner = THIS_MODULE, .dai_link = &db1550_i2s_dai, .num_links = 1, };

2 1

[alsa-devel] [PATCH] ASoC: imx_mc13783: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/imx-mc13783.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9df173c..55742a1 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -90,6 +90,7 @@ static const struct snd_soc_dapm_route imx_mc13783_routes[] = { static struct snd_soc_card imx_mc13783 = { .name = "imx_mc13783", + .owner = THIS_MODULE, .dai_link = imx_mc13783_dai_mc13783, .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), .dapm_widgets = imx_mc13783_widget,

2 1

[alsa-devel] [PATCH] ASoC: ttc_dkb: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/pxa/ttc-dkb.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index f4ea4f6..13c9ee0 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -122,6 +122,7 @@ static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { /* ttc/td audio machine driver */ static struct snd_soc_card ttc_dkb_card = { .name = "ttc-dkb-hifi", + .owner = THIS_MODULE, .dai_link = ttc_pm860x_hifi_dai, .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),

2 1

[alsa-devel] [PATCH] ASoC: brownstone: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/pxa/brownstone.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 4ad7609..e582c05 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -129,6 +129,7 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { /* audio machine driver */ static struct snd_soc_card brownstone = { .name = "brownstone", + .owner = THIS_MODULE, .dai_link = brownstone_wm8994_dai, .num_links = ARRAY_SIZE(brownstone_wm8994_dai),

2 1

[alsa-devel] [PATCH 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 02 Jul '13

02 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/registers.h | 35 +++++++++++++++++++++++++++++++++ 1 files changed, 35 insertions(+), 0 deletions(-) diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 715b6ba..522798d 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -4131,6 +4131,41 @@ #define ARIZONA_SLIMTX1_PORT_STS_WIDTH 1 /* SLIMTX1_PORT_STS */ /* + * R3072 - R3076 (0xC00 - 0xC04) ARIZONA_GPIO1_CTRL - ARIZONA_GPIO5_CTRL + */ +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +/* * R3087 (0xC0F) - IRQ CTRL 1 */ #define ARIZONA_IRQ_POL 0x0400 /* IRQ_POL */ -- 1.7.2.5

2 4

[alsa-devel] [PATCH] ASoC: samsung: Remove obsolete GPIO based DT pinmuxing
by Mark Brown 02 Jul '13

02 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since the Samsung platforms have moved to pinctrl for pin muxing and that is handled in the core the old GPIO based muxing code can just be removed. Something similar had been submitted by Thomas Abraham back in March but a resubmission following review never happened. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/i2s.c | 66 +++---------------------------------------------- 1 file changed, 4 insertions(+), 62 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 82ebb1a..7a17346 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1016,52 +1016,6 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) @@ -1235,18 +1189,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, @@ -1267,14 +1213,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; struct resource *res; - struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; -- 1.8.3.1

1 0

[alsa-devel] Multiple DA9055 chips
by Mark Brown 02 Jul '13

02 Jul '13

It seems we have two drivers in mainline for da9055 on I2C. We've got one in sound/soc/codecs/da9055.c and one in drivers/mfd/da9055-i2c.c, both registering themselves identically. What's going on here? Is this a combined CODEC and PMIC, has someone decided to release two chips with the same name or is one of the drivers misnamed?

2 5

[alsa-devel] [PATCH] ASoC: adav80x: Add module device table for adav801
by Axel Lin 02 Jul '13

02 Jul '13

This driver can be built as module, thus add module device table for adav801 to support module auto loading. To make the naming consistent, also rename adav80x_id to adav80x_i2c_id. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc..15b012d0 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + static int adav80x_spi_probe(struct spi_device *spi) { return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); @@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = { }, .probe = adav80x_spi_probe, .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, }; #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static const struct i2c_device_id adav80x_id[] = { +static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, adav80x_id); +MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) @@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = { }, .probe = adav80x_i2c_probe, .remove = adav80x_i2c_remove, - .id_table = adav80x_id, + .id_table = adav80x_i2c_id, }; #endif -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASOC: pcm: support fe dapm widget derived from codec_dai
by Nenghua Cao 02 Jul '13

02 Jul '13

From: nhcao <nhcao(a)marvell.com> In some chips, the codec_dai in cpu is abstracted as one dapm widget. Enhance the framework to use codec_dai as fe widget. Change-Id: I1b545bd1e2f24288f4508a7c6d4efbcfe1e03c5b Signed-off-by: nhcao <nhcao(a)marvell.com> --- sound/soc/soc-pcm.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..6ca23d7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -856,17 +856,37 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *dai = fe->cpu_dai ; struct snd_soc_dapm_widget_list *list; int paths; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (fe->cpu_dai->playback_widget) { + dai = fe->cpu_dai; + } else if (fe->codec_dai->playback_widget) { + dai = fe->codec_dai; + } else { + dev_err(fe->dev, "ASoC: can't find the correspond playback_widget for the dai\n"); + return -EINVAL; + } + } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + if (fe->cpu_dai->capture_widget) { + dai = fe->cpu_dai; + } else if (fe->codec_dai->capture_widget) { + dai = fe->codec_dai; + } else { + dev_err(fe->dev, "ASoC: can't find the correspond capture_widget for the dai\n"); + return -EINVAL; + } + } + list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); if (list == NULL) return -ENOMEM; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + paths = snd_soc_dapm_dai_get_connected_widgets(dai, stream, &list); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/2] ASoC: atmel-pcm: use generic dmaengine framework
by Bo Shen 02 Jul '13

02 Jul '13

this patch series is based on next-20130701 Bo Shen (2): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel-pcm: use generic dmaengine framework sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 33 ++++++----- 2 files changed, 32 insertions(+), 119 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [PATCH] ASoC: wm8994: Move runtime PM init to platform device init
by Mark Brown 01 Jul '13

01 Jul '13

From: Mark Brown <broonie(a)linaro.org> As well as being better style this allows the device to idle when there is no audio card instantaited which is probably what we want. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1d4b1ec..02c320f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4014,9 +4014,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->micdet_irq = control->pdata.micdet_irq; - pm_runtime_enable(codec->dev); - pm_runtime_idle(codec->dev); - /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; @@ -4389,8 +4386,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - pm_runtime_disable(codec->dev); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); @@ -4449,6 +4444,9 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } @@ -4456,6 +4454,8 @@ static int wm8994_probe(struct platform_device *pdev) static int wm8994_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 0/4] Fix mxs audio regressions
by Shawn Guo 01 Jul '13

01 Jul '13

The mxs audio got terribly broken during 3.11 development cycle due to the following two commits. af8ee11 ASoC: sgtl5000: Fix driver probe after reset 9e13f34 ASoC: sgtl5000: Let the codec acquire its clock The series fixes the regressions and bring mxs audio back to work. Mark, Please consider to take the whole series as fixes through your tree during 3.11-rc, so that we do not get a regression in the final v3.11 release. Shawn Shawn Guo (4): ASoC: sgtl5000: give it a ramping time before writting ASoC: sgtl5000: defer the probe if clock is not found ASoC: mxs: register saif mclk to clock framework ARM: mxs: saif0 is the clock provider to sgtl5000 arch/arm/boot/dts/imx28-apx4devkit.dts | 2 +- arch/arm/boot/dts/imx28-evk.dts | 2 +- arch/arm/boot/dts/imx28-m28evk.dts | 2 +- arch/arm/boot/dts/imx28.dtsi | 1 + sound/soc/codecs/sgtl5000.c | 10 ++++++++- sound/soc/mxs/mxs-saif.c | 35 ++++++++++++++++++++++++++++++++ 6 files changed, 48 insertions(+), 4 deletions(-) -- 1.7.9.5

7 24

[alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Jul '13

01 Jul '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ 4 files changed, 349 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..e2daf82 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..4a068d2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..6122fd1 --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,328 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, ret; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + /* enable choosen frequency */ + if (val != -1) + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + + /* enable/disable deemphasis functionality */ + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, priv->deemph); + + return ret; +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static int pcm1681_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static int pcm1681_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .probe = pcm1681_probe, + .remove = pcm1681_remove, + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

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[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 01 Jul '13

01 Jul '13

The following changes since commit a975873a9acc0788c1aee5ca183deb420b5c00e5: ASoC: tas5086: fix Mid-Z implementation (2013-06-28 12:15:30 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-3 for you to fetch changes up to 370887f105025ed52acc9cbb0a15b6567d1550ff: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-30 12:42:25 +0100) ---------------------------------------------------------------- ASoC: Final updates for v3.11 A few final updates: - A couple of additional bug fixes for the AC'97 refactoring. - Some fixes for the ADAU1701 driver. ---------------------------------------------------------------- Daniel Mack (2): ASoC: adau1701: more direct regmap usage ASoC: adau1701: remove control_data assignment Kevin Hilman (2): ASoC: pxa2xx: fixup multi-platform AC'97 build failures ASoC: ac97: fixup multi-platform AC'97 module build failure Mark Brown (6): Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ext' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Thierry Reding (2): ASoC: tegra20-ac97: Remove duplicate error message ASoC: tegra20-ac97: Remove unused variable .../devicetree/bindings/sound/adi,adau1701.txt | 35 ++ include/linux/mfd/twl6040.h | 7 + include/sound/soc.h | 4 +- sound/soc/au1x/ac97c.c | 21 +- sound/soc/au1x/psc-ac97.c | 33 +- sound/soc/blackfin/bf5xx-ac97.c | 29 +- sound/soc/cirrus/ep93xx-ac97.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 6 +- sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/adau1701.c | 321 +++++++++++--- sound/soc/codecs/stac9766.c | 26 +- sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/twl6040.c | 109 ++++- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8990.c | 11 +- sound/soc/codecs/wm8991.h | 9 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/codecs/wm8995.h | 7 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 18 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_adsp.c | 459 ++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +- sound/soc/codecs/wm_hubs.c | 6 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/mpc5200_psc_ac97.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 60 +-- sound/soc/pxa/pxa2xx-ac97.c | 10 +- sound/soc/pxa/pxa2xx-ac97.h | 3 - sound/soc/samsung/ac97.c | 42 +- sound/soc/sh/hac.c | 8 +- sound/soc/soc-core.c | 17 + sound/soc/tegra/tegra20_ac97.c | 67 ++- sound/soc/txx9/txx9aclc-ac97.c | 17 +- 38 files changed, 1124 insertions(+), 374 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau1701: more direct regmap usage
by Daniel Mack 30 Jun '13

30 Jun '13

Replace calls to snd_soc_update_bits() with regmap_update_bits(). Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ac8b2c4..ce56065 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -334,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -362,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -403,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -489,6 +489,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -497,11 +498,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -513,6 +516,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -520,7 +524,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -542,7 +546,8 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); adau1701->sysclk = freq; return 0; -- 1.8.1.4

3 4

[alsa-devel] [PATCH - oss plugin update revised 2/2] oss: Add user variables to set period and buffer config
by alex.d.wiggins＠gmail.com 30 Jun '13

30 Jun '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added 3 user variables to control the OSS period and buffer config: - period_size (integer): if not set then queries OSS for period size (by using SNDCTL_DSP_GETBLKSIZE), if positive then sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES), if negative then allows any power of 2 period size. - periods (integer): sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); - buffer_size (integer): sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). Note that ALSA performs sanity checks to prevent conflicts. Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d5b5bc4..a92c3e9 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -60,6 +60,12 @@ typedef struct snd_pcm_oss { unsigned int period_shift; unsigned int periods; unsigned int frame_bytes; + /* user variables set using ALSA config files */ + /* included here as this snd_pcm_oss_t struct + is stored in io->private_data */ + long int user_period_size; + long int user_periods; + long int user_buffer_size; } snd_pcm_oss_t; static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, @@ -408,6 +414,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ static const unsigned int bytes_list[] = { 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 @@ -500,20 +507,97 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) if (err < 0) return err; - /* period size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; - /* periods */ - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); - if (err < 0) - return err; - /* buffer size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; + /* period and buffer sizes: default behaviour below is to + query period size from OSS */ + + /* consider user set paramaters + (period_size; periods; buffer_size) */ + + /* note: ALSA performs sanity checks + to prevent conflicting parameters */ + + /* period_size */ + if (oss->user_period_size < 0) { + /* period_size set to negative value: + allow any power of 2 period size */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } else { + if (oss->user_period_size > 0) { + /* period_size set to positive value: + use user set period size */ + const unsigned int user_period_size + = oss->user_period_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, + &user_period_size); + } else { + /* query OSS for period size (called fragment size) */ + unsigned int fragsize; + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) + < 0) + fprintf(stderr, +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); + + /* note: second query of SNDCTL_DSP_GETBLKSIZE + can give different result + (i.e. 4096 instead of 2048) */ + /* in oss_audio_core.c getblksize function, + 4096 is the default return value: + so second time probably not reliable */ + + /* period size (in power of two) */ + unsigned int x = fragsize; + /* check if fragsize is a power of two */ + if ((x != 0) && ((x & (~x + 1)) == x)) + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + 1, &fragsize); + else + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } + } + + /* periods (number of periods per buffer) */ + if (oss->user_periods > 0) { + /* force number of periods to user set periods */ + const unsigned int user_periods = oss->user_periods; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); + if (err < 0) + return err; + } else { + /* allow any number of periods between 2 and 1024 inclusive + (ALSA can choose) */ + err = snd_pcm_ioplug_set_param_minmax(io, + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); + if (err < 0) + return err; + } + + /* buffer_size */ + if (oss->user_buffer_size > 0) { + /* force buffer size to user set buffer_size */ + const unsigned int user_buffer_size = oss->user_buffer_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); + if (err < 0) + return err; + } else { + /* buffer size (in power of two) */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } return 0; } @@ -558,6 +642,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) const char *device = "/dev/dsp"; int err; snd_pcm_oss_t *oss; + /* default values for user variables*/ + long int user_period_size = 0; + long int user_periods = 0; + long int user_buffer_size = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -573,6 +661,30 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) } continue; } + + /* user variables */ + if (strcmp(id, "period_size") == 0) { + if (snd_config_get_integer(n, &user_period_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "periods") == 0) { + if (snd_config_get_integer(n, &user_periods) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "buffer_size") == 0) { + if (snd_config_get_integer(n, &user_buffer_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + SNDERR("Unknown field %s", id); return -EINVAL; } @@ -597,6 +709,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) goto error; } + /* save user variables from ALSA config files */ + oss->user_period_size = user_period_size; + oss->user_periods = user_periods; + oss->user_buffer_size = user_buffer_size; + oss->io.version = SND_PCM_IOPLUG_VERSION; oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; oss->io.poll_fd = oss->fd; -- 1.8.3.1

1 0

Re: [alsa-devel] [PATCH 0/2] ASoc: ac97: fix multi-platform build failures
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 11:17:47AM -0700, Kevin Hilman wrote: > Hi Mark, > > When building next-20130628, I noticed a few build errors for > cm_x300_defconfig that were introduced by the new multi-platform AC'97 > support. Applied both, thanks.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Remove duplicate error message
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 09:53:24PM +0200, Thierry Reding wrote: > devm_ioremap_resource() already outputs an error message when any of the > operations it performs fails, so the duplicate in the caller can be > removed. Applied both, thanks.

1 0

[alsa-devel] ASoC updates for v3.11
by Mark Brown 28 Jun '13

28 Jun '13

The following changes since commit 9e895ace5d82df8929b16f58e9f515f6d54ab82d: Linux 3.10-rc7 (2013-06-22 09:47:31 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-2 for you to fetch changes up to 27516080b21cbcb936440d2a3171867860b9a881: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-28 12:17:07 +0100) The diffstat is a bit odd I think due to the synthetic tree I generated in an effort to cut out the fixes from v3.10-rc7 from it :/ ---------------------------------------------------------------- ASoC: More updates for v3.11 Some more fixes and enhancements, and also a bunch of refectoring for AC'97 support which enables more than one AC'97 controller driver to be built in. ---------------------------------------------------------------- Daniel Mack (10): ASoC: tas5086: open-code I2C transfer routines ASoC: tas5086: add more register defines ASoC: tas5086: add DAPM mux controls ASoC: tas5086: add support for pwm start mode config ASoC: adau1701: move firmware download to adau1701_reset() ASoC: adau1701: allow configuration of PLL mode pins ASoC: adau1701: switch to direct regmap API usage ASoC: adau1701: add support for pin muxing ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size ASoC: tas5086: fix Mid-Z implementation Lars-Peter Clausen (11): ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8400: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8903: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8904: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8990: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8991: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8995: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_hubs: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_adsp: Use SND_SOC_DAPM_PGA_E() instead of open-coding it ASoC: 88pm860x: Use SND_SOC_DAPM_PGA_E() instead of open-coding it Mark Brown (25): ASoC: stac9766: Remove version number ASoC: wm9705: Remove noisy print on boot ASoC: samsung-ac97: Use devm_clk_get() ASoC: samsung-ac97: Convert to devm_ioremap_resource() ASoC: psc-ac97: Use devm_ioremap_resource() ASoC: psc-ac97: Convert to module_platform_driver() ASoC: ac97c: Use module_platform_driver() ASoC: bf5xx-ac97: Convert to devm_gpio_request_one() ASoC: ep93xx: Remove redundant dev_set_drvdata() calls ASoC: nuc900-ac97: Convert to use devm_ APIs ASoC: tegra20-ac97: Convert to devm_clk_get() ASoC: tegra20-ac97: Convert to devm_ioremap_resource() ASoC: tegra-ac97: Do common and clock init prior to component registration ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource() ASoC: wm9705: Remove noisy print on boot Merge branch 'topic/stac9766' of git://git.kernel.org/.../broonie/sound into asoc-ac97 Merge branch 'topic/wm9705' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: ac97: Support multi-platform AC'97 ASoC: mfld: Remove unused variable Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ext' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Peter Ujfalusi (4): ASoC: twl6040: Drop using devm_request_threaded_irq() mfd: twl6040: Update register bit definitions ASoC: twl6040: Assign id for each DAI ASoC: twl6040: Add digital mute support Wei Yongjun (1): ASoC: mid-x86: Convert to use devm_* APIs .../devicetree/bindings/sound/adi,adau1701.txt | 35 ++ .../devicetree/bindings/sound/ti,tas5086.txt | 11 + include/linux/mfd/twl6040.h | 7 + include/sound/soc.h | 4 +- sound/soc/au1x/ac97c.c | 21 +- sound/soc/au1x/psc-ac97.c | 33 +- sound/soc/blackfin/bf5xx-ac97.c | 29 +- sound/soc/cirrus/ep93xx-ac97.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 6 +- sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/adau1701.c | 302 ++++++++++++-- sound/soc/codecs/stac9766.c | 26 +- sound/soc/codecs/tas5086.c | 330 ++++++++++++++- sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/twl6040.c | 109 ++++- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8990.c | 11 +- sound/soc/codecs/wm8991.h | 9 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/codecs/wm8995.h | 7 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 18 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_adsp.c | 459 ++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +- sound/soc/codecs/wm_hubs.c | 6 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/mpc5200_psc_ac97.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 60 +-- sound/soc/pxa/pxa2xx-ac97.c | 8 +- sound/soc/samsung/ac97.c | 42 +- sound/soc/sh/hac.c | 8 +- sound/soc/soc-core.c | 16 + sound/soc/tegra/tegra20_ac97.c | 66 ++- sound/soc/txx9/txx9aclc-ac97.c | 17 +- 39 files changed, 1447 insertions(+), 366 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt

2 1

[alsa-devel] [PATCH 1/2] ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
by Daniel Mack 28 Jun '13

28 Jun '13

The TAS5086_CLOCK_CONTROL also has a size of 1 byte. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index fec9a28..84c3f27 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -130,7 +130,7 @@ static const struct reg_default tas5086_reg_defaults[] = { static int tas5086_register_size(struct device *dev, unsigned int reg) { switch (reg) { - case TAS5086_DEV_ID ... TAS5086_BKNDERR: + case TAS5086_CLOCK_CONTROL ... TAS5086_BKNDERR: return 1; case TAS5086_INPUT_MUX: case TAS5086_PWM_OUTPUT_MUX: -- 1.8.1.4

2 2

[alsa-devel] [PATCH] ASoC: mfld: Remove unused variable
by Mark Brown 28 Jun '13

28 Jun '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index aec29a8..ee36384 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -412,8 +412,6 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) static int snd_mfld_mc_remove(struct platform_device *pdev) { - struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); snd_soc_unregister_card(&snd_soc_card_mfld); return 0; -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix the max length of control name in generic parser
by Takashi Iwai 28 Jun '13

28 Jun '13

add_control_with_pfx() in hda_generic.c assumes a shorter name string for the control element, and this resulted in the truncation of the long but valid string like "Headphone Surround Switch" in the middle. This patch aligns the max size to the actual limit of snd_ctl_elem_id, 44. Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1485d87..6460fc5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -854,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[44]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Guess what, it's two more Dell headset mic quirks
by David Henningsson 28 Jun '13

28 Jun '13

Add two more machines that need quirks for headset mics to work. Tested-by: Shawn Wang <shawn.wang(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1195636 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d7041d..86c2406 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3506,6 +3506,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Yet another Dell headset mic quirk
by David Henningsson 28 Jun '13

28 Jun '13

This quirk is needed for the headset mic to work on this Dell machine. BugLink: https://bugs.launchpad.net/bugs/1195597 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae12111..6d7041d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3509,6 +3509,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] C-Media 6632A unsupported (asus xonar u7)
by Julius von kohout 28 Jun '13

28 Jun '13

Hi ive got a new usb soundcard with the brand new C-Media 6632A chip and want to help with debugging. xonar u7 in usb 2.0 mode C-Media 6632A Ubuntu 13.04 julius@R850:~$ alsactl init 1 Found hardware: "USB-Audio" "USB Mixer" "USB1043:857c" "" "" Hardware is initialized using a generic method julius@R850:~$ alsamixer -c 1 cannot load mixer controls: Broken pipe alsa hardware volume (numid=2,iface=MIXER,name='Speaker Playback Volume') level on start is way to low and can only be adjusted via QasHctl. The volume knob on soundcard works for pulseaudio volume. spdif can only be used via direct alsa access without pulseaudio. alsa-info script results http://pastebin.ca/2409660 julius@R850:~$ amixer -c 1 controls numid=7,iface=MIXER,name='PCM Capture Source' numid=5,iface=MIXER,name='PCM Capture Switch' numid=6,iface=MIXER,name='PCM Capture Switch',index=1 numid=4,iface=MIXER,name='IEC958 In Playback Switch' numid=3,iface=MIXER,name='Input Gain Pad Control' numid=1,iface=MIXER,name='Speaker Playback Switch' numid=2,iface=MIXER,name='Speaker Playback Volume' [ 8219.399637] usb 2-1.2.3: new high-speed USB device number 22 using ehci-pci [ 8219.513315] usb 2-1.2.3: New USB device found, idVendor=1043, idProduct=857c [ 8219.513327] usb 2-1.2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 8219.513333] usb 2-1.2.3: Product: Xonar U7 [ 8219.513338] usb 2-1.2.3: Manufacturer: ASUS [ 8219.609328] input: ASUS Xonar U7 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2.3/2-1.2.3:1.4/input/input29 [ 8219.609743] hid-generic 0003:1043:857C.0016: input,hidraw4: USB HID v1.00 Device [ASUS Xonar U7] on usb-0000:00:1d.0-1.2.3/input4 [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.697280] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.718770] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.738130] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.757999] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.779861] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.800222] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.823964] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8520.766440] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8539.332352] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 julius@R850:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.8.0-25-generic. julius@R850:~$ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc4820000 irq 50 1 [U7 ]: USB-Audio - Xonar U7 ASUS Xonar U7 at usb-0000:00:1d.0-1.2.3, high speed julius@R850:~$ lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_realtek 78399 1 snd_usb_audio 140685 0 snd_usbmidi_lib 24938 1 snd_usb_audio snd_hda_intel 39619 3 snd_hda_codec 136453 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13602 2 snd_usb_audio,snd_hda_codec snd_pcm 97451 4 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 2 snd_usbmidi_lib,snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 18 snd_hda_codec_realtek,snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd julius@R850:~$ lsusb Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 002 Device 021: ID 18d1:4ee1 Google Inc. Bus 002 Device 015: ID 0424:2514 Standard Microsystems Corp. USB 2.0 Hub Bus 002 Device 016: ID 046d:c245 Logitech, Inc. Bus 002 Device 017: ID 05a4:9881 Ortek Technology, Inc. Bus 002 Device 022: ID 1043:857c iCreate Technologies Corp. Bus 002 Device 020: ID 046a:0180 Cherry GmbH julius@R850:~$ dpkg -l | grep alsa ii alsa-base 1.0.25+dfsg-0ubuntu4 all ALSA driver configuration files ii alsa-utils 1.0.25-4ubuntu2 amd64 Utilities for configuring and using ALSA ii bluez-alsa:amd64 4.101-0ubuntu8b1 amd64 Bluetooth ALSA support ii gstreamer0.10-alsa:amd64 0.10.36-1.1ubuntu1 amd64 GStreamer plugin for ALSA ii gstreamer1.0-alsa:amd64 1.0.6-1 amd64 GStreamer plugin for ALSA

2 1

[alsa-devel] How to clear the ALSA audio settings remembered by system?
by Lin, Mengdong 27 Jun '13

27 Jun '13

Hi, Could anyone tell me how to clear the ALSA audio settings remembered by system? I change some audio codec settings by amixer. And it seems the ALSA has remembered these settings and apply on next boot. And these will change my test result. How can I let system forgot these changes? Many thanks Mengdong

2 2

Re: [alsa-devel] [PATCH 3/6] sound/pci:trivial: replace numeric with standard PM state macros
by Takashi Iwai 27 Jun '13

27 Jun '13

At Thu, 27 Jun 2013 20:55:11 +0800, Yijing Wang wrote: > > Use standard PM state macros PCI_Dx instead of numeric 0/1/2.. > > Signed-off-by: Yijing Wang <wangyijing(a)huawei.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Jiri Kosina <trivial(a)kernel.org> > Cc: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: alsa-devel(a)alsa-project.org > Cc: linux-kernel(a)vger.kernel.org Thanks, applied. Takashi > --- > sound/pci/cs4281.c | 2 +- > sound/pci/ens1370.c | 2 +- > sound/pci/ymfpci/ymfpci_main.c | 2 +- > 3 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c > index 6a86950..2f6a4b6 100644 > --- a/sound/pci/cs4281.c > +++ b/sound/pci/cs4281.c > @@ -1312,7 +1312,7 @@ static int snd_cs4281_free(struct cs4281 *chip) > /* Sound System Power Management - Turn Everything OFF */ > snd_cs4281_pokeBA0(chip, BA0_SSPM, 0); > /* PCI interface - D3 state */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > > if (chip->irq >= 0) > free_irq(chip->irq, chip); > diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c > index db2dc83..b30afd1 100644 > --- a/sound/pci/ens1370.c > +++ b/sound/pci/ens1370.c > @@ -1939,7 +1939,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) > #endif > if (ensoniq->irq >= 0) > synchronize_irq(ensoniq->irq); > - pci_set_power_state(ensoniq->pci, 3); > + pci_set_power_state(ensoniq->pci, PCI_D3hot); > __hw_end: > #ifdef CHIP1370 > if (ensoniq->dma_bug.area) > diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c > index 22056c5..d591c15 100644 > --- a/sound/pci/ymfpci/ymfpci_main.c > +++ b/sound/pci/ymfpci/ymfpci_main.c > @@ -2258,7 +2258,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) > /* FIXME: temporarily disabled, otherwise we cannot fire up > * the chip again unless reboot. ACPI bug? > */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > #endif > > #ifdef CONFIG_PM_SLEEP > -- > 1.7.1 > >

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: tegra20-ac97: Convert to devm_ioremap_resource()
by Mark Brown 27 Jun '13

27 Jun '13

On Thu, Jun 27, 2013 at 12:22:12PM +0200, Thierry Reding wrote: > Can you please remove this error message as well, please? A message to > the same effect is already output by devm_ioremap_resource(). Other than > that, looks good: Too late, send an incremental patch please.

1 0

[alsa-devel] [RFC PATCH v2 00/12] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 27 Jun '13

27 Jun '13

This is another version of my patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. For reference, here is the original description of the series: One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (10): dmaengine: PL08x: Refactor pl08x_getbytes_chan() to lower indentation dmaengine: PL08x: Add support for different offset of CONFIG register dmaengine: PL08x: Rework LLI handling to be less fragile dmaengine: PL08x: Add support for PL080S variant dmaengine: PL08x: Add support for different maximum transfer size ASoC: Samsung: Do not queue cyclic buffers multiple times clk: samsung: s3c64xx: Add aliases for DMA clocks spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 8 +- arch/arm/mach-s3c64xx/Makefile | 1 + arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/include/mach/dma.h | 65 +++++ arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++++++++++ arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 2 + drivers/dma/amba-pl08x.c | 426 ++++++++++++++++++++++--------- drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 13 files changed, 651 insertions(+), 126 deletions(-) create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.2.1

5 29

[alsa-devel] [PATCH 0/4] ASoC: twl6040: Fixes and updates
by Peter Ujfalusi 27 Jun '13

27 Jun '13

Hi, Updates for the twl6040 codec: - Fix soc probe when the machine driver is removed and loaded again without removing the codec driver (was failing since the IRQ has been already requested) - Playback pop noise reduction on stream start and stop with implementing the digital_mute callback and related tracking of the mute for different paths. Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Drop using devm_request_threaded_irq() MFD: twl6040: Update register bit definitions ASoC: twl6040: Assign id for each DAI ASoC: twl6040: Add digital mute support include/linux/mfd/twl6040.h | 7 +++ sound/soc/codecs/twl6040.c | 109 +++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 114 insertions(+), 2 deletions(-) -- 1.8.2.1

3 8

[alsa-devel] [PATCH] ASoC: ac97: Support multi-platform AC'97
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Currently we can only have a single platform built in with AC'97 support due to the use of a global variable to provide the bus operations. Fix this by making that variable a pointer and having the bus drivers set the operations prior to registering. This is not a particularly good or nice approach but it avoids blocking multiplatform and a real fix involves fixing the fairly deep problems with AC'97 support - we should be converting it to a real bus. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 4 +++- sound/soc/au1x/ac97c.c | 7 +++++-- sound/soc/au1x/psc-ac97.c | 7 +++++-- sound/soc/blackfin/bf5xx-ac97.c | 11 +++++++++-- sound/soc/cirrus/ep93xx-ac97.c | 10 ++++++++-- sound/soc/codecs/ac97.c | 7 ++++--- sound/soc/codecs/ad1980.c | 12 ++++++------ sound/soc/codecs/stac9766.c | 22 +++++++++++----------- sound/soc/codecs/wm9705.c | 14 +++++++------- sound/soc/codecs/wm9712.c | 18 +++++++++--------- sound/soc/codecs/wm9713.c | 18 +++++++++--------- sound/soc/fsl/imx-ssi.c | 11 +++++++++-- sound/soc/fsl/mpc5200_psc_ac97.c | 10 ++++++++-- sound/soc/nuc900/nuc900-ac97.c | 11 ++++++++--- sound/soc/pxa/pxa2xx-ac97.c | 8 ++++++-- sound/soc/samsung/ac97.c | 12 +++++++++--- sound/soc/sh/hac.c | 8 ++++++-- sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 12 ++++++++++-- sound/soc/txx9/txx9aclc-ac97.c | 8 ++++++-- 20 files changed, 154 insertions(+), 72 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 85c1522..6eabee7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -340,7 +340,7 @@ struct snd_soc_jack_gpio; typedef int (*hw_write_t)(void *,const char* ,int); -extern struct snd_ac97_bus_ops soc_ac97_ops; +extern struct snd_ac97_bus_ops *soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, @@ -467,6 +467,8 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); + /* *Controls */ diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index a51dabe..d6f7694 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -179,13 +179,12 @@ static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ac97c_bus_ops = { .read = au1xac97c_ac97_read, .write = au1xac97c_ac97_write, .reset = au1xac97c_ac97_cold_reset, .warm_reset = au1xac97c_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -272,6 +271,10 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); + ret = snd_soc_set_ac97_ops(&ac97c_bus_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, &au1xac97c_dai_driver, 1); if (ret) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a97ba13..a822ab8 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -201,13 +201,12 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = au1xpsc_ac97_read, .write = au1xpsc_ac97_write, .reset = au1xpsc_ac97_cold_reset, .warm_reset = au1xpsc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -417,6 +416,10 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); + ret = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, &wd->dai_drv, 1); if (ret) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 8010f32..efb1dae 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -198,13 +198,12 @@ static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) #endif } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops bf5xx_ac97_ops = { .read = bf5xx_ac97_read, .write = bf5xx_ac97_write, .warm_reset = bf5xx_ac97_warm_reset, .reset = bf5xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) @@ -336,6 +335,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) goto sport_config_err; } + ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto sport_config_err; + } + ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, &bfin_ac97_dai, 1); if (ret) { @@ -350,6 +355,7 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -360,6 +366,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index f193701..ac73c60 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -237,13 +237,12 @@ static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ep93xx_ac97_ops = { .read = ep93xx_ac97_read, .write = ep93xx_ac97_write, .reset = ep93xx_ac97_cold_reset, .warm_reset = ep93xx_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -389,6 +388,10 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); + ret = snd_soc_set_ac97_ops(&ep93xx_ac97_ops); + if (ret) + goto fail; + ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, &ep93xx_ac97_dai, 1); if (ret) @@ -398,6 +401,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); return ret; } @@ -412,6 +416,8 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ef2ae32..ec73518 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -62,13 +62,13 @@ static struct snd_soc_dai_driver ac97_dai = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); return 0; } @@ -79,7 +79,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, &soc_ac97_ops, NULL, &ac97_bus); + ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, + &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index f385342..89fcf7d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -108,7 +108,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -124,7 +124,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; @@ -154,13 +154,13 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) u16 retry_cnt = 0; retry: - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } - soc_ac97_ops.reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ @@ -186,7 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index cbc7ae3..a5455c1 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -143,14 +143,14 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } @@ -162,7 +162,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } @@ -173,7 +173,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { - val = soc_ac97_ops.read(codec->ac97, reg); + val = soc_ac97_ops->read(codec->ac97, reg); return val; } return cache[reg / 2]; @@ -240,15 +240,15 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; @@ -272,7 +272,7 @@ reset: return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); - id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { stac9766_reset(codec, 0); reset++; @@ -336,7 +336,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index a5fc61d..70ce6793 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -209,7 +209,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) case AC97_RESET: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -225,7 +225,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9705_reg))) cache[reg] = val; @@ -294,8 +294,8 @@ static struct snd_soc_dai_driver wm9705_dai[] = { static int wm9705_reset(struct snd_soc_codec *codec) { - if (soc_ac97_ops.reset) { - soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops->reset) { + soc_ac97_ops->reset(codec->ac97); if (ac97_read(codec, 0) == wm9705_reg[0]) return 0; /* Success */ } @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff); return 0; } @@ -323,7 +323,7 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) } for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } return 0; @@ -337,7 +337,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 8e9a6a3..c5eb746 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -455,7 +455,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_REC_GAIN) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -472,7 +472,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -581,15 +581,15 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -624,7 +624,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || (i > 0x58 && i != 0x5c)) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -635,7 +635,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f7afa68..a53e175 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -652,7 +652,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_CD) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -668,7 +668,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1095,15 +1095,15 @@ static struct snd_soc_dai_driver wm9713_dai[] = { int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1180,7 +1180,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || i == AC97_EXTENDED_MSTATUS || i > 0x66) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -1197,7 +1197,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index a8362be..51be377 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -501,13 +501,12 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) imx_ssi_ac97_read(ac97, 0); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops imx_ssi_ac97_ops = { .read = imx_ssi_ac97_read, .write = imx_ssi_ac97_write, .reset = imx_ssi_ac97_reset, .warm_reset = imx_ssi_ac97_warm_reset }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int imx_ssi_probe(struct platform_device *pdev) { @@ -583,6 +582,12 @@ static int imx_ssi_probe(struct platform_device *pdev) platform_set_drvdata(pdev, ssi); + ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto failed_register; + } + ret = snd_soc_register_component(&pdev->dev, &imx_component, dai, 1); if (ret) { @@ -608,6 +613,7 @@ failed_register: release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -627,6 +633,7 @@ static int imx_ssi_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 4141b35..3ef7a0c 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -131,13 +131,12 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) psc_ac97_warm_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -290,6 +289,12 @@ static int psc_ac97_of_probe(struct platform_device *op) if (rc != 0) return rc; + rc = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (rc != 0) { + dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", ret); + return rc; + } + rc = snd_soc_register_component(&op->dev, &psc_ac97_component, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { @@ -318,6 +323,7 @@ static int psc_ac97_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); snd_soc_unregister_component(&op->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 8dea4c1..f4c2417 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -197,13 +197,12 @@ static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops nuc900_ac97_ops = { .read = nuc900_ac97_read, .write = nuc900_ac97_write, .reset = nuc900_ac97_cold_reset, .warm_reset = nuc900_ac97_warm_reset, -} -EXPORT_SYMBOL_GPL(soc_ac97_ops); +}; static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -356,6 +355,10 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_ac97_data = nuc900_audio; + ret = snd_soc_set_ac97_ops(&nuc900_ac97_ops); + if (ret) + goto out; + ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) @@ -367,6 +370,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) return 0; out: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -375,6 +379,7 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); nuc900_ac97_data = NULL; + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 57ea8e6..a3c22ba 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -41,13 +41,12 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa2xx_ac97_finish_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .read = pxa2xx_ac97_read, .write = pxa2xx_ac97_write, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", @@ -244,6 +243,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); + if (ret != 0) + return ret; + /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -255,6 +258,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 04d7fd4..2dd623f 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -214,13 +214,12 @@ static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops s3c_ac97_ops = { .read = s3c_ac97_read, .write = s3c_ac97_write, .warm_reset = s3c_ac97_warm_reset, .reset = s3c_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -452,6 +451,12 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err4; } + ret = snd_soc_set_ac97_ops(&s3c_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err4; + } + ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component, s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); if (ret) @@ -472,7 +477,7 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - + snd_soc_set_ac97_ops(NULL); return ret; } @@ -488,6 +493,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index af19f77..0af2e4d 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -227,13 +227,12 @@ static void hac_ac97_coldrst(struct snd_ac97 *ac97) hac_ac97_warmrst(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops hac_ac97_ops = { .read = hac_ac97_read, .write = hac_ac97_write, .reset = hac_ac97_coldrst, .warm_reset = hac_ac97_warmrst, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -316,6 +315,10 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); + if (ret != 0) + return ret; + return snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } @@ -323,6 +326,7 @@ static int hac_soc_platform_probe(struct platform_device *pdev) static int hac_soc_platform_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4489c5b..309e5c9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2080,6 +2080,22 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +struct snd_ac97_bus_ops *soc_ac97_ops; + +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + if (ops == soc_ac97_ops) + return 0; + + if (soc_ac97_ops && ops) + return -EBUSY; + + soc_ac97_ops = ops; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); + /** * snd_soc_free_ac97_codec - free AC97 codec device * @codec: audio codec diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 48037f7..f52eab6 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -142,13 +142,12 @@ static void tegra20_ac97_codec_write(struct snd_ac97 *ac97_snd, } while (!time_after(jiffies, timeout)); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops tegra20_ac97_ops = { .read = tegra20_ac97_codec_read, .write = tegra20_ac97_codec_write, .reset = tegra20_ac97_codec_reset, .warm_reset = tegra20_ac97_codec_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) { @@ -409,6 +408,12 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + 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; + } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, &tegra20_ac97_dai, 1); if (ret) { @@ -436,6 +441,7 @@ err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -450,6 +456,8 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) clk_disable_unprepare(ac97->clk_ac97); + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8ee8d42..4bcce8a 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -119,12 +119,11 @@ static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops txx9aclc_ac97_ops = { .read = txx9aclc_ac97_read, .write = txx9aclc_ac97_write, .reset = txx9aclc_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) { @@ -206,6 +205,10 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; + err = snd_soc_set_ac97_ops(&txx9aclc_ac97_ops); + if (err < 0) + return err; + return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } @@ -213,6 +216,7 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } -- 1.8.3.1

3 3

[alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Convert to devm_clk_get()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra20_ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 2f70ea7..d8c142d 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -327,7 +327,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); @@ -443,7 +443,6 @@ err_unregister_pcm: err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_clk_put: - clk_put(ac97->clk_ac97); err: return ret; } @@ -458,7 +457,6 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&ac97->util_data); clk_disable_unprepare(ac97->clk_ac97); - clk_put(ac97->clk_ac97); return 0; } -- 1.8.3.1

2 2

[alsa-devel] [PATCH] ASoC: tegra-ac97: Do common and clock init prior to component registration
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Otherwise we may instantiate and hence have something try to access the device while it is still completing initialisation. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra20_ac97.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 9043626d..48037f7 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -395,23 +395,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.maxburst = 4; ac97->capture_dma_data.slave_id = of_dma[0]; - ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, - &tegra20_ac97_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; - } - ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) - goto err_unregister_pcm; + goto err_clk_put; ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); if (ret) @@ -423,17 +409,31 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, + &tegra20_ac97_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_asoc_utils_fini; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_component; + } + /* XXX: crufty ASoC AC97 API - only one AC97 codec allowed */ workdata = ac97; return 0; -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); err_unregister_pcm: tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_asoc_utils_fini: + tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: return ret; -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH -next] ASoC: mid-x86: Convert to use devm_* APIs
by Mark Brown 26 Jun '13

26 Jun '13

On Tue, Jun 25, 2013 at 10:16:56AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > devm_* APIs are device managed and make code simpler. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Daniel Mack 26 Jun '13

26 Jun '13

On 26.06.2013 15:46, Marek Belisko wrote: > Hi Daniel, > On 06/26/2013 03:16 PM, Daniel Mack wrote: >> Hi Marek, >> >> On 26.06.2013 15:05, Marek Belisko wrote: >>> PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add >>> support only for I2C mode. >>> >>> Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> >>> --- >>> .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + >>> sound/soc/codecs/Kconfig | 4 + >>> sound/soc/codecs/Makefile | 2 + >>> sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ >>> 4 files changed, 349 insertions(+) >>> create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> create mode 100644 sound/soc/codecs/pcm1681.c >>> >>> diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> new file mode 100644 >>> index 0000000..4df1718 >>> --- /dev/null >>> +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> @@ -0,0 +1,15 @@ >>> +Texas Instruments PCM1681 8-channel PWM Processor >>> + >>> +Required properties: >>> + >>> + - compatible: Should contain "ti,pcm1681". >>> + - reg: The i2c address. Should contain <0x4c>. >>> + >>> +Examples: >>> + >>> + i2c_bus { >>> + pcm1681@4c { >>> + compatible = "ti,pcm1681"; >>> + reg = <0x4c>; >>> + }; >>> + }; >>> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig >>> index badb6fb..e2daf82 100644 >>> --- a/sound/soc/codecs/Kconfig >>> +++ b/sound/soc/codecs/Kconfig >>> @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS >>> select SND_SOC_MC13783 if MFD_MC13XXX >>> select SND_SOC_ML26124 if I2C >>> select SND_SOC_HDMI_CODEC >>> + select SND_SOC_PCM1681 if I2C >>> select SND_SOC_PCM3008 >>> select SND_SOC_RT5631 if I2C >>> select SND_SOC_RT5640 if I2C >>> @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 >>> config SND_SOC_HDMI_CODEC >>> tristate >>> >>> +config SND_SOC_PCM1681 >>> + tristate >>> + >>> config SND_SOC_PCM3008 >>> tristate >>> >>> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile >>> index 70fd806..4a068d2 100644 >>> --- a/sound/soc/codecs/Makefile >>> +++ b/sound/soc/codecs/Makefile >>> @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o >>> snd-soc-mc13783-objs := mc13783.o >>> snd-soc-ml26124-objs := ml26124.o >>> snd-soc-hdmi-codec-objs := hdmi.o >>> +snd-soc-pcm1681-objs := pcm1681.o >>> snd-soc-pcm3008-objs := pcm3008.o >>> snd-soc-rt5631-objs := rt5631.o >>> snd-soc-rt5640-objs := rt5640.o >>> @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o >>> obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o >>> obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o >>> obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o >>> +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o >>> obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o >>> obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o >>> obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o >>> diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c >>> new file mode 100644 >>> index 0000000..6122fd1 >>> --- /dev/null >>> +++ b/sound/soc/codecs/pcm1681.c >>> @@ -0,0 +1,328 @@ >>> +/* >>> + * PCM1681 ASoC codec driver >>> + * >>> + * Copyright (c) StreamUnlimited GmbH 2013 >>> + * Marek Belisko <marek.belisko(a)streamunlimited.com> >>> + * >>> + * This program is free software; you can redistribute it and/or >>> + * modify it under the terms of the GNU General Public License >>> + * as published by the Free Software Foundation; either version 2 >>> + * of the License, or (at your option) any later version. >>> + * >>> + * This program is distributed in the hope that it will be useful, >>> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >>> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the >>> + * GNU General Public License for more details. >>> + */ >>> + >>> +#include <linux/module.h> >>> +#include <linux/slab.h> >>> +#include <linux/delay.h> >>> +#include <linux/gpio.h> >>> +#include <linux/i2c.h> >>> +#include <linux/regmap.h> >>> +#include <linux/of_device.h> >>> +#include <linux/of_gpio.h> >>> +#include <sound/pcm.h> >>> +#include <sound/pcm_params.h> >>> +#include <sound/soc.h> >>> +#include <sound/tlv.h> >>> + >>> +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ >>> + SNDRV_PCM_FMTBIT_S24_LE) >>> + >>> +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ >>> + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ >>> + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ >>> + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) >>> + >>> +#define PCM1681_SOFT_MUTE_ALL 0xff >>> +#define PCM1681_DEEMPH_RATE_MASK 0x18 >>> +#define PCM1681_DEEMPH_MASK 0x01 >>> + >>> +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ >>> +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ >>> +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ >>> +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ >>> +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ >>> +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ >>> + >>> +static const struct reg_default pcm1681_reg_defaults[] = { >>> + { 0x01, 0xff }, >>> + { 0x02, 0xff }, >>> + { 0x03, 0xff }, >>> + { 0x04, 0xff }, >>> + { 0x05, 0xff }, >>> + { 0x06, 0xff }, >>> + { 0x07, 0x00 }, >>> + { 0x08, 0x00 }, >>> + { 0x09, 0x06 }, >>> + { 0x0A, 0x00 }, >>> + { 0x0B, 0xff }, >>> + { 0x0C, 0x0f }, >>> + { 0x0D, 0x00 }, >>> + { 0x10, 0xff }, >>> + { 0x11, 0xff }, >>> + { 0x12, 0x00 }, >>> + { 0x13, 0x00 }, >>> +}; >>> + >>> +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) >>> +{ >>> + return !((reg == 0x00) || (reg == 0x0f)); >>> +} >>> + >>> +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) >>> +{ >>> + return pcm1681_accessible_reg(dev, reg) && >>> + (reg != PCM1681_ZERO_DETECT_STATUS); >>> +} >>> + >>> +struct pcm1681_private { >>> + struct regmap *regmap; >>> + unsigned int format; >>> + /* Current deemphasis status */ >>> + unsigned int deemph; >>> + /* Current rate for deemphasis control */ >>> + unsigned int rate; >>> +}; >>> + >>> +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; >>> + >>> +static int pcm1681_set_deemph(struct snd_soc_codec *codec) >>> +{ >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int i = 0, val = -1, ret; >>> + >>> + if (priv->deemph) >>> + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) >>> + if (pcm1681_deemph[i] == priv->rate) >>> + val = i; >>> + >>> + /* enable choosen frequency */ >>> + if (val != -1) >>> + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, >>> + PCM1681_DEEMPH_RATE_MASK, val); >>> + >>> + /* enable/disable deemphasis functionality */ >>> + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, >>> + PCM1681_DEEMPH_MASK, priv->deemph); >>> + >>> + return ret; >>> +} >>> + >>> +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, >>> + struct snd_ctl_elem_value *ucontrol) >>> +{ >>> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + ucontrol->value.enumerated.item[0] = priv->deemph; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, >>> + struct snd_ctl_elem_value *ucontrol) >>> +{ >>> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + priv->deemph = ucontrol->value.enumerated.item[0]; >>> + >>> + return pcm1681_set_deemph(codec); >>> +} >>> + >>> +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, >>> + unsigned int format) >>> +{ >>> + struct snd_soc_codec *codec = codec_dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + /* The PCM1681 can only be slave to all clocks */ >>> + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { >>> + dev_err(codec->dev, "Invalid clocking mode\n"); >>> + return -EINVAL; >>> + } >>> + >>> + priv->format = format; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) >>> +{ >>> + struct snd_soc_codec *codec = dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int ret, val = 0; >>> + >>> + if (mute) >>> + val = PCM1681_SOFT_MUTE_ALL; >>> + >>> + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); >>> + if (ret < 0) >>> + return ret; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_hw_params(struct snd_pcm_substream *substream, >>> + struct snd_pcm_hw_params *params, >>> + struct snd_soc_dai *dai) >>> +{ >>> + struct snd_soc_codec *codec = dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int val = 0; >>> + int pcm_format = params_format(params); >>> + >>> + priv->rate = params_rate(params); >>> + >>> + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { >>> + case SND_SOC_DAIFMT_RIGHT_J: >>> + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) >>> + val = 0x00; >>> + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) >>> + val = 0x03; >>> + break; >>> + case SND_SOC_DAIFMT_I2S: >>> + val = 0x04; >>> + break; >>> + case SND_SOC_DAIFMT_LEFT_J: >>> + val = 0x05; >>> + break; >>> + default: >>> + dev_err(codec->dev, "Invalid DAI format\n"); >>> + return -EINVAL; >>> + } >>> + >>> + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); >>> + >>> + return 0; >>> +} >>> + >>> +static const struct snd_soc_dai_ops pcm1681_dai_ops = { >>> + .set_fmt = pcm1681_set_dai_fmt, >>> + .hw_params = pcm1681_hw_params, >>> + .digital_mute = pcm1681_digital_mute, >>> +}; >>> + >>> +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); >>> + >>> +static const struct snd_kcontrol_new pcm1681_controls[] = { >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", >>> + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", >>> + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", >>> + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", >>> + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, >>> + pcm1681_get_deemph, pcm1681_put_deemph), >>> +}; >>> + >>> +struct snd_soc_dai_driver pcm1681_dai = { >>> + .name = "pcm1681-hifi", >>> + .playback = { >>> + .stream_name = "Playback", >>> + .channels_min = 2, >>> + .channels_max = 8, >>> + .rates = PCM1681_PCM_RATES, >>> + .formats = PCM1681_PCM_FORMATS, >>> + }, >>> + .ops = &pcm1681_dai_ops, >>> +}; >>> + >>> +#ifdef CONFIG_OF >>> +static const struct of_device_id pcm1681_dt_ids[] = { >>> + { .compatible = "ti,pcm1681", }, >>> + { } >>> +}; >>> +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); >>> +#endif >>> + >>> +static const struct regmap_config pcm1681_regmap = { >>> + .reg_bits = 8, >>> + .val_bits = 8, >>> + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), >>> + .reg_defaults = pcm1681_reg_defaults, >>> + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), >>> + .writeable_reg = pcm1681_writeable_reg, >>> + .readable_reg = pcm1681_accessible_reg, >>> +}; >>> + >>> +static int pcm1681_probe(struct snd_soc_codec *codec) >>> +{ >>> + return 0; >> >> Is there really nothing the driver has to do in order to bring the codec >> into full operation mode? > This codec have only POWER-ON-RESET function. It means codec need system > clock (normally MCLK) for some amount of time to get out of reset and > then you can communicate with it. >> >>> +} >>> + >>> +static int pcm1681_remove(struct snd_soc_codec *codec) >>> +{ >>> + return 0; >>> +} >>> + >>> +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { >>> + .probe = pcm1681_probe, >>> + .remove = pcm1681_remove, >>> + .controls = pcm1681_controls, >>> + .num_controls = ARRAY_SIZE(pcm1681_controls), >>> +}; >>> + >>> +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) >>> +static const struct i2c_device_id pcm1681_i2c_id[] = { >>> + {"pcm1681", 0}, >>> + {} >>> +}; >>> +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); >>> + >>> +static int pcm1681_i2c_probe(struct i2c_client *client, >>> + const struct i2c_device_id *id) >>> +{ >>> + int ret; >>> + struct pcm1681_private *priv; >>> + >>> + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); >>> + if (!priv) >>> + return -ENOMEM; >>> + >>> + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); >>> + if (IS_ERR(priv->regmap)) { >>> + ret = PTR_ERR(priv->regmap); >>> + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); >>> + return ret; >>> + } >>> + >>> + i2c_set_clientdata(client, priv); >>> + >> >> Is there any way to probe the device is there at all? I'd like to bail >> early in case the bus or i2c address doesn't match for example. > Probing is done here because we assume that codec is out of reset. Yes, I know. I wondered whether there is any register that the driver can read back in order to check the presence of the device. But I quickly checked the datasheet, and there does in fact not seem to be such a register. >> Also, I wonder whether there are any reset lines that might be connected >> to GPIOs of the MCU. If so, it would be good idea to add support for >> that in this driver. But that can be done at some later point as well. > No other pins which can be handled. I see. Well ok then, I have no further remarks :) Daniel

1 0

[alsa-devel] [PATCH] ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/txx9/txx9aclc-ac97.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8a28403..8ee8d42 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -188,9 +188,9 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!r) return -EBUSY; - if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), - dev_name(&pdev->dev))) - return -EBUSY; + drvdata->base = devm_ioremap_resource(&pdev->dev, r); + if (IS_ERR(drvdata->base)) + return PTR_ERR(drvdata->base); drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); if (!drvdata) @@ -201,9 +201,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) r->start >= TXX9_DIRECTMAP_BASE && r->start < TXX9_DIRECTMAP_BASE + 0x400000) drvdata->physbase |= 0xf00000000ull; - drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); - if (!drvdata->base) - return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: nuc900-ac97: Convert to use devm_ APIs
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/nuc900/nuc900-ac97.c | 43 ++++++++++-------------------------------- 1 file changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index fe3285c..08e4c6b 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -326,32 +326,23 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (nuc900_ac97_data) return -EBUSY; - nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); + nuc900_audio = devm_kzalloc(&pdev->dev, sizeof(struct nuc900_audio), + GFP_KERNEL); if (!nuc900_audio) return -ENOMEM; spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) { - ret = -ENODEV; - goto out0; - } + if (!nuc900_audio->res) + return ret; - if (!request_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res), pdev->name)) { - ret = -EBUSY; - goto out0; - } + nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, + nuc900_audio->res); + if (IS_ERR(nuc900_audio->mmio)) + return PTR_ERR(nuc900_audio->mmio); - nuc900_audio->mmio = ioremap(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); - if (!nuc900_audio->mmio) { - ret = -ENOMEM; - goto out1; - } - - nuc900_audio->clk = clk_get(&pdev->dev, NULL); + nuc900_audio->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(nuc900_audio->clk)) { ret = PTR_ERR(nuc900_audio->clk); goto out2; @@ -375,15 +366,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) return 0; -out3: - clk_put(nuc900_audio->clk); -out2: - iounmap(nuc900_audio->mmio); -out1: - release_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); -out0: - kfree(nuc900_audio); +out: return ret; } @@ -391,12 +374,6 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); - clk_put(nuc900_ac97_data->clk); - iounmap(nuc900_ac97_data->mmio); - release_mem_region(nuc900_ac97_data->res->start, - resource_size(nuc900_ac97_data->res)); - - kfree(nuc900_ac97_data); nuc900_ac97_data = NULL; return 0; -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: ep93xx: Remove redundant dev_set_drvdata() calls
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> The driver core does this and it's never legal to rely on the value of drvdata if not set in probe() anyway. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/cirrus/ep93xx-ac97.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 3f4f888..f193701 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -398,7 +398,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -412,7 +411,6 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) ep93xx_ac97_write_reg(info, AC97GCR, 0); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Also clean up the error reporting from failed requests while we're at it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/blackfin/bf5xx-ac97.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c66bef8..8010f32 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -293,13 +293,14 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ - if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { - pr_err("Failed to request GPIO_%d for reset\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM); - ret = -1; + ret = devm_gpio_request_one(&pdev->dev, + CONFIG_SND_BF5XX_RESET_GPIO_NUM, + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); goto gpio_err; } - gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), @@ -349,10 +350,6 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -gpio_err: -#endif return ret; } @@ -363,9 +360,6 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: psc-ac97: Use devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/au1x/psc-ac97.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862a..f5a3921 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -383,15 +383,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!iores) return -ENODEV; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 1.8.3.1

1 2

[alsa-devel] [PATCH 1/3] ASoC: psc-ac97: Use devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/au1x/psc-ac97.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862a..f5a3921 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -383,15 +383,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!iores) return -ENODEV; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 1.8.3.1

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung-ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

On Wed, Jun 26, 2013 at 12:10:15PM +0200, Tomasz Figa wrote: > By the way, Mark, does this driver work for you? I have tried it on both > mini6410 and smdk6410 (configured to use the onboard AC97 codec), with both > old and new DMA drivers and I couldn't get it to play a single sound. I've no access to any hardware which might use this driver, though IIRC it didn't work last time I did have such access.

1 0

[alsa-devel] [PATCH] ASoC: samsung-ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

Reviewed-by: Jingoo Han <jg1.han(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 1c85999..04d7fd4 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -417,11 +417,9 @@ static int s3c_ac97_probe(struct platform_device *pdev) return -ENXIO; } - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } + s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(s3c_ac97.regs)) + return PTR_ERR(s3c_ac97.regs); s3c_ac97_pcm_out.channel = dmatx_res->start; s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; @@ -433,13 +431,6 @@ static int s3c_ac97_probe(struct platform_device *pdev) init_completion(&s3c_ac97.done); mutex_init(&s3c_ac97.lock); - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); @@ -481,16 +472,13 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); return ret; } static int s3c_ac97_remove(struct platform_device *pdev) { - struct resource *mem_res, *irq_res; + struct resource *irq_res; asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); @@ -501,12 +489,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) clk_disable_unprepare(s3c_ac97.ac97_clk); - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: samsung-ac97: Use devm_clk_get()
by Mark Brown 26 Jun '13

26 Jun '13

Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index cb88ead..1c85999 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -440,7 +440,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err1; } - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); ret = -ENODEV; @@ -480,7 +480,6 @@ err5: err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); err2: iounmap(s3c_ac97.regs); err1: @@ -501,7 +500,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); iounmap(s3c_ac97.regs); -- 1.8.3.1

1 1

[alsa-devel] [PATCH alsa-lib 1/2] pcm_hw: Remove unused fields in struct
by David Henningsson 26 Jun '13

26 Jun '13

These fields are not used, and their name similarity to other fields are quite confusing when trying to debug alsa-lib. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 0f3226d..196393d 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -95,8 +95,6 @@ typedef struct { volatile struct snd_pcm_mmap_status * mmap_status; struct snd_pcm_mmap_control *mmap_control; struct snd_pcm_sync_ptr *sync_ptr; - snd_pcm_uframes_t hw_ptr; - snd_pcm_uframes_t appl_ptr; int period_event; snd_timer_t *period_timer; struct pollfd period_timer_pfd; -- 1.7.9.5

3 4

[alsa-devel] *.alsa-project.org web services down?
by Arun Raghavan 26 Jun '13

26 Jun '13

Hello, I can't seem to get to alsa-project.org or git.alsa-project.org. Is the web server down? -- Arun

2 1

[alsa-devel] [patch] ALSA: ak4xx-adda: info leak in ak4xxx_capture_source_info()
by Dan Carpenter 26 Jun '13

26 Jun '13

"idx" is controled by the user and can be a negative offset into the input_names[] array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index cef813d2..ed726d1 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -571,7 +571,7 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); const char **input_names; - int num_names, idx; + unsigned int num_names, idx; num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); if (!num_names)

2 1

[alsa-devel] PulseAudio and softvol
by Arun Raghavan 26 Jun '13

26 Jun '13

Hello, A number of users have intermittently(?) been hitting a crash in alsa-lib 1.0.27 [1, 2] related to the softvol plugin. I'm not able to reproduce this reliably, so can't find an easy way to debug/fix. However, this raises a tangential question - why do we need softvol to be plugged for 'front' at all? David explained to me that this is to guarantee the existence of a PCM control. Perhaps I don't fully understand this, because I'm unconvinced by the reason. Could someone explain/refute? This is especially bad for us, from PulseAudio's perspective, because we aren't getting a zero-copy path. Cheers, Arun [1] https://bugzilla.redhat.com/show_bug.cgi?id=953352 [2] https://bugs.freedesktop.org/show_bug.cgi?id=64299

5 25

[alsa-devel] [PATCH 1/3] clk: mxs: clk-imx28: Provide a dummy clock
by Fabio Estevam 26 Jun '13

26 Jun '13

Sometimes it is useful to be able to pass a dummy clock for a driver. Add support for a dummy clock. Cc: Mike Turquette <mturquette(a)linaro.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/clock/imx28-clock.txt | 1 + drivers/clk/mxs/clk-imx28.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/clock/imx28-clock.txt b/Documentation/devicetree/bindings/clock/imx28-clock.txt index e6587af..5a153ef 100644 --- a/Documentation/devicetree/bindings/clock/imx28-clock.txt +++ b/Documentation/devicetree/bindings/clock/imx28-clock.txt @@ -76,6 +76,7 @@ clocks and IDs. usb0_phy 62 usb1_phy 63 enet_out 64 + dummy 65 Examples: diff --git a/drivers/clk/mxs/clk-imx28.c b/drivers/clk/mxs/clk-imx28.c index 4faf0af..6b55211 100644 --- a/drivers/clk/mxs/clk-imx28.c +++ b/drivers/clk/mxs/clk-imx28.c @@ -143,7 +143,7 @@ enum imx28_clk { emi_xtal, lcdif_div, etm_div, ptp, saif0_div, saif1_div, clk32k_div, rtc, lradc, spdif_div, clk32k, pwm, uart, ssp0, ssp1, ssp2, ssp3, gpmi, spdif, emi, saif0, saif1, lcdif, etm, - fec, can0, can1, usb0, usb1, usb0_phy, usb1_phy, enet_out, + fec, can0, can1, usb0, usb1, usb0_phy, usb1_phy, enet_out, dummy, clk_max }; @@ -234,6 +234,7 @@ int __init mx28_clocks_init(void) clks[usb0_phy] = clk_register_gate(NULL, "usb0_phy", "pll0", 0, PLL0CTRL0, 18, 0, &mxs_lock); clks[usb1_phy] = clk_register_gate(NULL, "usb1_phy", "pll1", 0, PLL1CTRL0, 18, 0, &mxs_lock); clks[enet_out] = clk_register_gate(NULL, "enet_out", "pll2", 0, ENET, 18, 0, &mxs_lock); + clks[dummy] = mxs_clk_fixed("dummy", 0); for (i = 0; i < ARRAY_SIZE(clks); i++) if (IS_ERR(clks[i])) { -- 1.8.1.2

3 4

Re: [alsa-devel] [ALSA] Enable capture from line-in and CD on Revolution 5.1
by Dan Carpenter 26 Jun '13

26 Jun '13

[ This warning is very old, but I was working on some related code and didn't know how to fix it ]. Hello Jochen Voss, This is a semi-automatic email about new static checker warnings. The patch a58e7cb16dfa: "[ALSA] Enable capture from line-in and CD on Revolution 5.1" from Oct 4, 2006, leads to the following Smatch complaint: sound/i2c/other/ak4xxx-adda.c:808 build_adc_controls() error: we previously assumed 'ak->adc_info' could be null (see line 789) sound/i2c/other/ak4xxx-adda.c 788 if (ak->type == SND_AK5365 && (idx % 2) == 0) { 789 if (! ak->adc_info || ^^^^^^^^^^^^ Old check. 790 ! ak->adc_info[mixer_ch].switch_name) { 791 knew.name = "Capture Switch"; 792 knew.index = mixer_ch + ak->idx_offset * 2; 793 } else 794 knew.name = ak->adc_info[mixer_ch].switch_name; 795 knew.info = ak4xxx_switch_info; 796 knew.get = ak4xxx_switch_get; 797 knew.put = ak4xxx_switch_put; 798 knew.access = 0; 799 /* register 2, bit 0 (SMUTE): 0 = normal operation, 800 1 = mute */ 801 knew.private_value = 802 AK_COMPOSE(idx/2, 2, 0, 0) | AK_INVERT; 803 err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); 804 if (err < 0) 805 return err; 806 807 memset(&knew, 0, sizeof(knew)); 808 knew.name = ak->adc_info[mixer_ch].selector_name; ^^^^^^^^^^^^^^^^^^^^^ New dereference. 809 if (!knew.name) { 810 knew.name = "Capture Channel"; regards, dan carpenter

1 0

[alsa-devel] [PATCH] ASoC: wm9705: Remove noisy print on boot
by Mark Brown 25 Jun '13

25 Jun '13

There's no content in the announcement. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm9705.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 05b1f34..a5fc61d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -337,8 +337,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9705 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: stac9766: Remove version number
by Mark Brown 25 Jun '13

25 Jun '13

There is no need to have versioning beyond that for the kernel, especially when the version number never gets updated. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/stac9766.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2eda85ba..cbc7ae3 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,8 +28,6 @@ #include "stac9766.h" -#define STAC9766_VERSION "0.10" - /* * STAC9766 register cache */ @@ -338,8 +336,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; -- 1.8.3.1

1 0

Re: [alsa-devel] [RFC PATCH v2 00/12] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 25 Jun '13

25 Jun '13

On Tue, Jun 25, 2013 at 01:22:20PM +0200, Tomasz Figa wrote: > On Tuesday 25 of June 2013 11:28:18 Mark Brown wrote: > > On my non-DT s3c64xx system I get this when trying to use SPI for the > > first time: > Could you clarify first time? Does it work on further retries? No, the system appears to lock up fairly solidly - I suspect poor error handling in the SPI driver. > > [ 1.739385] s3c64xx-spi s3c6410-spi.0: Failed to get RX DMA channel > > [ 1.743437] spi_master spi0: failed to prepare transfer hardware > > though I haven't yet tried your clock patches (there were some conficts > > that I didn't get round to picking up yet). > This shouldn't really matter, as long as the pl080 drivers probes > correctly. Indeed, I was wondering if it was failing to probe due to not getting its clocks though since that's another area you've been actively working on.

1 0

[alsa-devel] [PATCH 1/5] ASoC: codecs: tas5086: fix documentation
by Daniel Mack 25 Jun '13

25 Jun '13

The property 'ti,charge-period' must be given in nanoseconds, not microseconds. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ti,tas5086.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index 8ea4f5b..e8310f1 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -11,7 +11,7 @@ Optional properties: chip's !RESET pin. If specified, the driver will assert a hardware reset at probe time. - - ti,charge-period: This property should contain the time in microseconds + - ti,charge-period: This property should contain the time in nanoseconds that closely matches the external single-ended split-capacitor charge period. The hardware chip waits for this period of time before starting the -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: Tegra and i.MX are mutually exclusive
by Arnd Bergmann 25 Jun '13

25 Jun '13

Both Tegra and i.MX provide ac97 operations, which can only exist once in the kernel. They can both be built as loadable modules (only one of them needs to be loaded anyway), but we have to disallow enabling them both as built-in. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3843a18..5958948 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -99,6 +99,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC + depends on SND_SOC_TEGRA=n || (SND_SOC_TEGRA=m && m) help Say Y or M if you want to add support for codecs attached to the i.MX CPUs.

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[alsa-devel] [PATCH] snd-firewire-lib: remove unused header inclusion
by Takashi Sakamoto 25 Jun '13

25 Jun '13

spinlock is not used in amdtp.h. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index b680c5e..f6103d6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -3,7 +3,6 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <linux/spinlock.h> #include "packets-buffer.h" /** -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda - clean up code to reset hda link
by mengdong.lin＠intel.com 25 Jun '13

25 Jun '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a cleanup to the previous patch "reset hda link during system/ runtime suspend". In this patch - azx_enter_link_reset() and azx_exit_link_reset() are defined for entering and exiting the link reset respectively. azx_link_reset() is no longer used and replaced by azx_enter_link_reset(). - azx_reset() reuses the above two new functions for a link reset cycle Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f110c7..f39de90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1121,7 +1121,7 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, #endif /* enter link reset */ -static void azx_reset_link(struct azx *chip) +static void azx_enter_link_reset(struct azx *chip) { unsigned long timeout; @@ -1134,11 +1134,22 @@ static void azx_reset_link(struct azx *chip) usleep_range(500, 1000); } -/* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +/* exit link reset */ +static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + +/* reset codec link */ +static int azx_reset(struct azx *chip, int full_reset) +{ if (!full_reset) goto __skip; @@ -1146,12 +1157,7 @@ static int azx_reset(struct azx *chip, int full_reset) azx_writeb(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_enter_link_reset(chip); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 @@ -1159,12 +1165,7 @@ static int azx_reset(struct azx *chip, int full_reset) usleep_range(500, 1000); /* Bring controller out of reset */ - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_exit_link_reset(chip); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ usleep_range(1000, 1200); @@ -2908,7 +2909,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2961,7 +2962,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); return 0; } -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda - Add In-driver connection info
by Wang Xingchao 25 Jun '13

25 Jun '13

Pin's connection list may change dynamically with hot-plug event on Intel Haswell chip. Users would see connections be "0" in codec#. when play audio on this pin, software driver choose converter from cache connections. So add "In-driver connection" info to avoid confuse when raw connections are different with cache connection. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_proc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fa..9760f00 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -504,6 +504,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, int conn_len) { int c, curr = -1; + const hda_nid_t *list; + int cache_len; if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && @@ -521,6 +523,19 @@ static void print_conn_list(struct snd_info_buffer *buffer, } snd_iprintf(buffer, "\n"); } + + /* Get Cache connections info */ + cache_len = snd_hda_get_conn_list(codec, nid, &list); + if (cache_len != conn_len + || memcmp(list, conn, conn_len)) { + snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); + if (cache_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < cache_len; c++) + snd_iprintf(buffer, " 0x%02x", list[c]); + snd_iprintf(buffer, "\n"); + } + } } static void print_gpio(struct snd_info_buffer *buffer, -- 1.8.1.2

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