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[alsa-devel] [PATCH] ASoC: wm8903: disable IRQs during BIAS_OFF
by Stephen Warren 11 Sep '13

11 Sep '13

From: Stephen Warren <swarren(a)nvidia.com> If the WM8903 IRQ is triggered after the device is suspended, or before it is resumed, the underlying I2C bus that the WM8903 is attached to may not be available. This may cause the IRQ handler to emit the following error: wm8903 0-001a: Failed to read IRQ status: -16 ... or perhaps even cause a failure inside the I2C driver. To prevent this, explicitly mask all interrupt sources when entering BIAS_OFF. Also, modify the IRQ handler to immediately return if the device is in BIAS_OFF, so that shared interrupt lines don't cause a similar problem. This requires that WM8903 interrupts be disabled to avoid interrupt storms. During resume (actually, the transition back to BIAS_STANDBY), restore the desired IRQ mask. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 38 ++++++++++++++++++++++++++++---------- 1 file changed, 28 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1d..e847bbe 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1091,6 +1091,22 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Right Line Output PGA", NULL, "Charge Pump" }, }; +static void wm8903_setup_mic_det_irqs(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT; + + /* Enable interrupts we've got a report configured for */ + if (wm8903->mic_det) + irq_mask &= ~WM8903_MICDET_EINT; + if (wm8903->mic_short) + irq_mask &= ~WM8903_MICSHRT_EINT; + + snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, + irq_mask); +} + static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1169,6 +1185,8 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, WM8903_CP_DYN_V, WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V); + + wm8903_setup_mic_det_irqs(codec); } snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, @@ -1177,6 +1195,8 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8903_INTERRUPT_STATUS_1_MASK, 0xffff); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, WM8903_BIAS_ENA, 0); @@ -1600,7 +1620,6 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int det, int shrt) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT; dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n", det, shrt); @@ -1610,15 +1629,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8903->mic_det = det; wm8903->mic_short = shrt; - /* Enable interrupts we've got a report configured for */ - if (det) - irq_mask &= ~WM8903_MICDET_EINT; - if (shrt) - irq_mask &= ~WM8903_MICSHRT_EINT; - - snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, - WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, - irq_mask); + wm8903_setup_mic_det_irqs(codec); if (det || shrt) { /* Enable mic detection, this may not have been set through @@ -1642,6 +1653,13 @@ static irqreturn_t wm8903_irq(int irq, void *data) int mic_report, ret; unsigned int int_val, mask, int_pol; + /* + * During suspend, a shared IRQ may be triggered by some other device. + * If so, just ignore it; WM8903 shouldn't be generating interrupts. + */ + if (wm8903->codec->dapm.bias_level == SND_SOC_BIAS_OFF) + return IRQ_NONE; + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK, &mask); if (ret != 0) { -- 1.8.1.5

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Re: [alsa-devel] alsa crash with “ ALSA sound/usb/urb.c:179 cannot submit urb (err = -27)” on imx6q sabrelite
by Daniel Mack 11 Sep '13

11 Sep '13

On 11.09.2013 14:53, Ankit Agarwal wrote: > > On Sun, Sep 8, 2013 at 8:27 PM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > > On 02.09.2013 12:51, Ankit Agarwal wrote: > > I am using alsa pcm player to listen to audio. The alsa pcm > crashes with > > the error: ALSA sound/usb/urb.c:179 cannot submit urb (err = -27). > I am > > able to hear the data initially but when other transaction begins > on HID > > report endpoint, alsa crashes with the mentioned error. > > -27 == -EFBIG, which seems to relate to a scheduling problem on the bus, > and hence probably rather to a host controller bug than an issue with > snd-usb. > > > I am working on imx6q board with linux kernel-3.0.35. I am using Apple > > device to get the digital audio. > > Please try with a more recent kernel, and see if you can find the > location where EFBIG is initially reported. > > > I have found that EFBIG is being reported by ehci-host controller. It is > coming from file ehci-sched.c under iso_stream_schedule() function. The > error is thrown when the controller tries to schedule the urb too far > into the future, ie, there is overflow of the scheduled URBs. I might be > happening because the URBs are not scheduled properly. Ok. What about trying a newer kernel? The issue might be fixed already. In any case, this does not seem related to snd-usb, so I copied the usb list. Daniel

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[alsa-devel] Scarlett Mixer 8i6
by Abhayadev S 11 Sep '13

11 Sep '13

just checked the ALSA tree and found that the support for 18i6 mixer is enabled but not able to find the same in the kernel mainline tree at kernel.org. When is this planned to be patched in the mainline kernel? Also, i have a 8i6 for which i would like to extend the support for the internal mixer. Can i use the code for 18i6 here? or does it needs any changes? in anyway i could help it to add the support for 8i6 device too? Regards, Abhayadev S

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[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 11 Sep '13

11 Sep '13

The following changes since commit 6e4664525b1db28f8c4e1130957f70a94c19213e: Linux 3.11 (2013-09-02 13:46:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-4 for you to fetch changes up to 2ae2caff8348378c9e464353e9860ee97583c00b: Merge remote-tracking branch 'asoc/fix/rsnd' into asoc-linus (2013-09-11 11:17:17 +0100) No idea what git thinks it's doing with the diffstat here... ---------------------------------------------------------------- ASoC: Fixes for v3.12 A few small fixes, nothing with any broad impact but all useful for the affected systems. The Kirkwood compatible string change is fixing up a string just added in the merge window so that we don't get any changes in released kernels. ---------------------------------------------------------------- Bo Shen (1): ASoC: atmel: disable error interrupt Fabio Estevam (1): ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel Kuninori Morimoto (1): ASoC: rsnd: fixup flag name of rsnd_scu_platform_info Mark Brown (6): ASoC: fsl_spdif: Select regmap-mmio Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/kirkwood' into asoc-linus Merge remote-tracking branch 'asoc/fix/mc13783' into asoc-linus Merge remote-tracking branch 'asoc/fix/rsnd' into asoc-linus Steffen Trumtrar (1): ASoC: mc13783: add spi errata fix Thomas Petazzoni (1): ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver .../devicetree/bindings/misc/atmel-ssc.txt | 23 +- .../devicetree/bindings/serial/mrvl,pxa-ssp.txt | 65 + Documentation/devicetree/bindings/sound/ak4554.c | 11 + .../devicetree/bindings/sound/alc5632.txt | 19 + .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 + .../devicetree/bindings/sound/atmel-wm8904.txt | 55 + .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 12 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 + .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + .../devicetree/bindings/sound/mvebu-audio.txt | 33 + .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 26 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 27 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 +- .../devicetree/bindings/sound/pcm1792a.txt | 18 + Documentation/devicetree/bindings/sound/rt5640.txt | 20 + .../devicetree/bindings/sound/samsung-i2s.txt | 22 +- .../devicetree/bindings/sound/soc-ac97link.txt | 28 + .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + .../devicetree/bindings/sound/tlv320aic3x.txt | 9 +- Documentation/devicetree/bindings/sound/wm8731.txt | 9 + Documentation/devicetree/bindings/sound/wm8753.txt | 24 +- Documentation/devicetree/bindings/sound/wm8903.txt | 19 + Documentation/devicetree/bindings/sound/wm8994.txt | 4 + Documentation/sound/alsa/HD-Audio-Models.txt | 1 + Documentation/sound/alsa/HD-Audio.txt | 2 + MAINTAINERS | 13 + arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 +- arch/arm/plat-pxa/ssp.c | 171 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- include/dt-bindings/sound/fsl-imx-audmux.h | 56 + include/linux/atmel-ssc.h | 2 +- include/linux/mfd/arizona/gpio.h | 96 + include/linux/platform_data/asoc-s3c.h | 1 + include/linux/platform_data/omap-abe-twl6040.h | 49 - include/linux/pxa2xx_ssp.h | 11 + include/sound/core.h | 8 + include/sound/pxa2xx-lib.h | 7 - include/sound/rcar_snd.h | 84 + include/sound/soc-dapm.h | 201 +- include/sound/soc-dpcm.h | 2 +- include/sound/soc.h | 47 +- include/uapi/sound/hdspm.h | 2 +- sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 +- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + .../soc-dmaengine-pcm.c => core/pcm_dmaengine.c} | 0 sound/core/pcm_lib.c | 4 +- sound/drivers/dummy.c | 2 +- sound/firewire/speakers.c | 4 +- sound/isa/gus/interwave.c | 3 +- sound/oss/dmabuf.c | 3 +- sound/pci/hda/Kconfig | 9 +- sound/pci/hda/hda_codec.c | 64 +- sound/pci/hda/hda_codec.h | 21 + sound/pci/hda/hda_generic.c | 79 +- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_hwdep.c | 6 +- sound/pci/hda/hda_intel.c | 34 +- sound/pci/hda/hda_jack.c | 22 +- sound/pci/hda/hda_jack.h | 13 +- sound/pci/hda/hda_proc.c | 33 + sound/pci/hda/patch_analog.c | 4662 +------------------- sound/pci/hda/patch_conexant.c | 79 +- sound/pci/hda/patch_hdmi.c | 9 +- sound/pci/hda/patch_realtek.c | 190 +- sound/pci/hda/patch_sigmatel.c | 14 +- sound/pci/hda/patch_via.c | 2 +- sound/pci/rme96.c | 307 +- sound/pci/rme9652/hdspm.c | 779 +++- sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/atmel/Kconfig | 21 + sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 118 +- sound/soc/atmel/atmel_ssc_dai.c | 36 +- sound/soc/atmel/atmel_wm8904.c | 254 ++ sound/soc/atmel/sam9x5_wm8731.c | 208 + sound/soc/au1x/db1200.c | 4 + sound/soc/au1x/psc-ac97.c | 3 - sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/cirrus/ep93xx-ac97.c | 3 - sound/soc/cirrus/ep93xx-i2s.c | 5 - sound/soc/codecs/Kconfig | 19 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 15 + sound/soc/codecs/ad1980.c | 43 + sound/soc/codecs/ad73311.c | 22 +- sound/soc/codecs/adau1701.c | 25 +- sound/soc/codecs/adav80x.c | 13 +- sound/soc/codecs/ads117x.c | 29 +- sound/soc/codecs/ak4104.c | 34 +- sound/soc/codecs/ak4554.c | 106 + sound/soc/codecs/ak5386.c | 17 +- sound/soc/codecs/arizona.c | 69 +- sound/soc/codecs/arizona.h | 5 +- sound/soc/codecs/bt-sco.c | 22 +- sound/soc/codecs/cs4270.c | 20 + sound/soc/codecs/cs4271.c | 30 +- sound/soc/codecs/dmic.c | 17 +- sound/soc/codecs/hdmi.c | 30 +- sound/soc/codecs/lm4857.c | 107 +- sound/soc/codecs/max9768.c | 16 + sound/soc/codecs/max98090.c | 10 +- sound/soc/codecs/max9877.c | 294 +- sound/soc/codecs/mc13783.c | 5 +- sound/soc/codecs/pcm1681.c | 339 ++ sound/soc/codecs/pcm1792a.c | 257 ++ sound/soc/codecs/pcm1792a.h | 26 + sound/soc/codecs/pcm3008.c | 150 +- sound/soc/codecs/rt5640.c | 235 +- sound/soc/codecs/rt5640.h | 12 + sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 14 + sound/soc/codecs/spdif_receiver.c | 17 +- sound/soc/codecs/spdif_transmitter.c | 18 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/sta32x.c | 10 +- sound/soc/codecs/tlv320aic26.c | 51 +- sound/soc/codecs/tlv320aic32x4.c | 22 +- sound/soc/codecs/tlv320aic3x.c | 56 +- sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 7 +- sound/soc/codecs/uda134x.c | 88 +- sound/soc/codecs/wl1273.c | 17 + sound/soc/codecs/wm0010.c | 12 +- sound/soc/codecs/wm5102.c | 53 +- sound/soc/codecs/wm5110.c | 35 +- sound/soc/codecs/wm8350.c | 6 +- sound/soc/codecs/wm8727.c | 17 +- sound/soc/codecs/wm8731.c | 60 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8782.c | 17 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 3 +- sound/soc/codecs/wm8960.c | 10 +- sound/soc/codecs/wm8962.c | 9 +- sound/soc/codecs/wm8994.c | 39 +- sound/soc/codecs/wm8995.c | 5 +- sound/soc/codecs/wm8997.c | 1175 +++++ sound/soc/codecs/wm8997.h | 23 + sound/soc/codecs/wm_adsp.c | 124 +- sound/soc/codecs/wm_adsp.h | 3 +- sound/soc/codecs/wm_hubs.c | 8 +- sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/Kconfig | 24 +- sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1225 +++++ sound/soc/fsl/fsl_spdif.h | 191 + sound/soc/fsl/fsl_ssi.c | 501 ++- sound/soc/fsl/imx-audmux.c | 79 +- sound/soc/fsl/imx-audmux.h | 52 +- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 20 +- sound/soc/fsl/imx-pcm.h | 26 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 148 + sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/imx-wm8962.c | 3 +- sound/soc/generic/simple-card.c | 2 + sound/soc/kirkwood/Kconfig | 13 +- sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 108 +- sound/soc/kirkwood/kirkwood-i2s.c | 94 +- sound/soc/kirkwood/kirkwood-openrd.c | 6 +- sound/soc/kirkwood/kirkwood-t5325.c | 6 +- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/Kconfig | 3 +- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 3 - sound/soc/omap/Kconfig | 8 +- sound/soc/omap/mcbsp.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 133 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 + sound/soc/omap/omap-mcpdm.c | 3 - sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/mmp-pcm.c | 7 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 76 +- sound/soc/pxa/pxa2xx-ac97.c | 67 +- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/samsung/ac97.c | 11 +- sound/soc/samsung/dma.c | 19 +- sound/soc/samsung/dma.h | 4 +- sound/soc/samsung/i2s-regs.h | 51 +- sound/soc/samsung/i2s.c | 193 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 58 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/fsi.c | 51 +- sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 + sound/soc/sh/rcar/core.c | 861 ++++ sound/soc/sh/rcar/gen.c | 280 ++ sound/soc/sh/rcar/rsnd.h | 302 ++ sound/soc/sh/rcar/scu.c | 236 + sound/soc/sh/rcar/ssi.c | 728 +++ sound/soc/soc-compress.c | 13 +- sound/soc/soc-core.c | 253 +- sound/soc/soc-dapm.c | 819 ++-- sound/soc/soc-jack.c | 4 +- sound/soc/soc-pcm.c | 25 +- sound/soc/spear/Kconfig | 2 +- sound/soc/tegra/Kconfig | 14 +- sound/soc/tegra/tegra20_ac97.c | 8 - sound/soc/tegra/tegra_alc5632.c | 2 - sound/soc/tegra/tegra_rt5640.c | 1 + sound/soc/tegra/tegra_wm8753.c | 2 - sound/soc/tegra/trimslice.c | 2 - sound/soc/txx9/txx9aclc-ac97.c | 3 - sound/soc/ux500/mop500.c | 1 + sound/usb/6fire/firmware.c | 4 +- sound/usb/endpoint.c | 3 + sound/usb/pcm.c | 243 +- sound/usb/usx2y/usbusx2y.c | 8 +- 236 files changed, 12494 insertions(+), 7410 deletions(-) create mode 100644 Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (87%) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mvebu-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h create mode 100644 include/linux/mfd/arizona/gpio.h delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h create mode 100644 include/sound/rcar_snd.h rename sound/{soc/soc-dmaengine-pcm.c => core/pcm_dmaengine.c} (100%) create mode 100644 sound/soc/atmel/atmel_wm8904.c create mode 100644 sound/soc/atmel/sam9x5_wm8731.c create mode 100644 sound/soc/codecs/ak4554.c create mode 100644 sound/soc/codecs/pcm1681.c create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c create mode 100644 sound/soc/sh/rcar/Makefile create mode 100644 sound/soc/sh/rcar/adg.c create mode 100644 sound/soc/sh/rcar/core.c create mode 100644 sound/soc/sh/rcar/gen.c create mode 100644 sound/soc/sh/rcar/rsnd.h create mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/ssi.c

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[alsa-devel] [PATCH v2] codecs: mc13783: add spi errata fix
by Steffen Trumtrar 10 Sep '13

10 Sep '13

The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock and main audio clock speed, the Audio Codec or Stereo DAC do sometimes not start when programmed to do so. This is due to an internal clock timing issue related to the loading of the SPI bits into the audio block. On an i.MX27 based system, this issue lead to switched audio channels under certain circumstances: RTC + Touch + Audio are used and loaded at startup. The mentioned workaround of writing registers 40 and 41 two times is implemented here. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> Cc: stable(a)vger.kernel.org --- Changes in v2: - use correct register defines - fix spelling error in commit description sound/soc/codecs/mc13783.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd..ea141e1 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; -- 1.8.4.rc3

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[alsa-devel] Support for Scarlett Mixer
by Abhayadev S 10 Sep '13

10 Sep '13

just check the ALSA tree and foudn that the support for 18i6 mixer is enabled but not able to find the same in the kernel mainline tree at kernel.org. When is this planned to be patched in the mainline kernel? Also, i have a 8i6 for which i would like to extend the support for the internal mixer. Can i use the same code for 8i6 also? or does it needs any changes? in anyway i could help it to add the support for 8i6 device too? Regards, Abhayadev S

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[alsa-devel] Sound distorted on Intel HDA (Realtek ALC889) (Kernel 3.x)
by Vincent Thiele 10 Sep '13

10 Sep '13

Using Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) Mainboard is: http://www.asus.com/Motherboards/MAXIMUS_IV_GENEZ/ Only fix at the moment is changing "load-module module-udev-detect use_ucm=0" in "/etc/pulse/default.pa" to "load-module module-udev-detect tsched=0". Maybe you should add a specific rule for this device which adds this fix automatically. Best regards alsa-info.sh output: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Mon Jul 22 18:15:08 UTC 2013 !!Linux Distribution !!------------------ Ubuntu 13.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 13.04" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 13.04" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: System manufacturer Product Name: System Product Name Product Version: System Version Firmware Version: 3603 !!Kernel Information !!------------------ Kernel release: 3.11.0-031100rc2-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.11.0-031100rc2-generic Library version: 1.0.25 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7420000 irq 54 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf7080000 irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) 01:00.1 Audio device: NVIDIA Corporation Device 0e0b (rev a1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1c20 (rev 05) Subsystem: 1043:84dc -- 01:00.1 0403: 10de:0e0b (rev a1) Subsystem: 1043:8422 !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,32,-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 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -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,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -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,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -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,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,32,-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 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -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,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -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,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -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,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0889 Subsystem Id: 0x1043846b Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC889 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=8, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=2, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=2, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC889 Digital", type="SPDIF", device=1 Converter: stream=8, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Front Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Surround Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=07, enabled=1 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Side Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=08, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181305f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4007e629: [N/A] Line Out at Ext N/A Conn = Analog, Color = White DefAssociation = 0x2, Sequence = 0x9 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x01451130: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18 0x19* 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x25 0x0b Codec: Nvidia GPU 41 HDMI/DP Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de0041 Subsystem Id: 0x10438422 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x06 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x07 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=9 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Control: name="ELD", index=0, device=9 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x08 [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=8, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x09 [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x0a [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x0b [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T+ 1 root audio 116, 7 Jul 22 20:08 /dev/snd/controlC0 crw-rw---T+ 1 root audio 116, 13 Jul 22 20:08 /dev/snd/controlC1 crw-rw---T+ 1 root audio 116, 6 Jul 22 20:08 /dev/snd/hwC0D0 crw-rw---T+ 1 root audio 116, 12 Jul 22 20:08 /dev/snd/hwC1D0 crw-rw---T+ 1 root audio 116, 5 Jul 22 20:09 /dev/snd/pcmC0D0c crw-rw---T+ 1 root audio 116, 4 Jul 22 20:10 /dev/snd/pcmC0D0p crw-rw---T+ 1 root audio 116, 3 Jul 22 20:09 /dev/snd/pcmC0D1p crw-rw---T+ 1 root audio 116, 2 Jul 22 20:08 /dev/snd/pcmC0D2c crw-rw---T+ 1 root audio 116, 11 Jul 22 20:09 /dev/snd/pcmC1D3p crw-rw---T+ 1 root audio 116, 10 Jul 22 20:09 /dev/snd/pcmC1D7p crw-rw---T+ 1 root audio 116, 9 Jul 22 20:09 /dev/snd/pcmC1D8p crw-rw---T+ 1 root audio 116, 8 Jul 22 20:09 /dev/snd/pcmC1D9p crw-rw---T+ 1 root audio 116, 1 Jul 22 20:08 /dev/snd/seq crw-rw---T+ 1 root audio 116, 33 Jul 22 20:08 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Jul 22 20:08 . drwxr-xr-x 3 root root 340 Jul 22 20:08 .. lrwxrwxrwx 1 root root 12 Jul 22 20:08 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Jul 22 20:08 pci-0000:01:00.1 -> ../controlC1 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC889 Digital [ALC889 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC889 Alt Analog [ALC889 Alt Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xf7420000 irq 54' Mixer name : 'Realtek ALC889' Components : 'HDA:10ec0889,1043846b,00100004' Controls : 57 Simple ctrls : 25 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 59 [92%] [-5.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 254 [100%] [0.20dB] Front Right: Playback 254 [100%] [0.20dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Side',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Line Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [on] Front Right: Capture 0 [0%] [-16.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',2 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!-------Mixer controls for card 1 [NVidia] Card hw:1 'NVidia'/'HDA NVidia at 0xf7080000 irq 17' Mixer name : 'Nvidia GPU 41 HDMI/DP' Components : 'HDA:10de0041,10438422,00100100' Controls : 28 Simple ctrls : 4 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',3 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Front Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Surround Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Center Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.6 { iface MIXER name 'LFE Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.7 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'Side Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.10 { iface MIXER name 'Side Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Headphone Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.12 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.13 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.14 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.16 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.17 { iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.18 { iface MIXER name 'Line Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.19 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.20 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.21 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.22 { iface MIXER name 'Input Source' index 2 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.23 { iface MIXER name 'Capture Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.24 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.25 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.26 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.27 { iface MIXER name 'Capture Volume' index 2 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.28 { iface MIXER name 'Capture Switch' index 2 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.29 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.30 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.31 { iface MIXER name 'Line Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.32 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.33 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.34 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.35 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.36 { iface MIXER name 'IEC958 Default PCM Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.37 { iface MIXER name 'Master Playback Volume' value 59 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -500 } } control.38 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.41 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.42 { iface CARD name 'Line Out Front Jack' value true comment { access read type BOOLEAN count 1 } } control.43 { iface CARD name 'Line Out Surround Jack' value false comment { access read type BOOLEAN count 1 } } control.44 { iface CARD name 'Line Out CLFE Jack' value false comment { access read type BOOLEAN count 1 } } control.45 { iface CARD name 'Line Out Side Jack' value false comment { access read type BOOLEAN count 1 } } control.46 { iface CARD name 'Front Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.47 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.48 { iface CARD name 'SPDIF Phantom Jack' index 1 value true comment { access read type BOOLEAN count 1 } } control.49 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.50 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.51 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access read type INTEGER count 8 range '0 - 36' } } control.52 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.53 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.54 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.55 { iface PCM device 2 name 'Capture Channel Map' index 1 value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.56 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } control.57 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } } state.NVidia { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0482000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface CARD name 'HDMI/DP,pcm=9 Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface MIXER name 'IEC958 Playback Con Mask' index 3 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.21 { iface MIXER name 'IEC958 Playback Pro Mask' index 3 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Default' index 3 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Switch' index 3 value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface PCM device 9 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.25 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.26 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.27 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.28 { iface PCM device 9 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ Module pci_stub vboxpci vboxnetadp vboxnetflt vboxdrv snd_hda_codec_hdmi hid_generic usbhid hid rfcomm bnep parport_pc bluetooth ppdev joydev snd_hda_codec_realtek xpad ff_memless snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_page_alloc nls_iso8859_1 snd_seq_midi eeepc_wmi asus_wmi sparse_keymap snd_seq_midi_event snd_rawmidi snd_seq snd_seq_device nouveau psmouse snd_timer serio_raw snd mxm_wmi ttm drm_kms_helper drm lpc_ich i2c_algo_bit wmi video mac_hid soundcore mei_me mei lp parport usb_storage ahci libahci e1000e ptp pps_core !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x99430140 0x12 0x411111f0 0x14 0x01014010 0x15 0x01011012 0x16 0x01016011 0x17 0x01012014 0x18 0x01a19850 0x19 0x02a19c60 0x1a 0x0181305f 0x1b 0x02214c20 0x1c 0x411111f0 0x1d 0x4007e629 0x1e 0x01451130 0x1f 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x04 0x185600f0 0x05 0x185600f0 0x06 0x185600f0 0x07 0x185600f0 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: !!ALSA/HDA dmesg !!-------------- [ 2.420876] usb 2-1.3: new full-speed USB device number 3 using ehci-pci [ 2.428149] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 2.428271] snd_hda_intel 0000:00:1b.0: irq 54 for MSI/MSI-X [ 2.441009] input: Microsoft X-Box 360 pad as /devices/pci0000:00/0000:00:1c.4/0000:03:00.0/usb3/3-2/3-2:1.0/input/input3 -- [ 2.445106] realtek: Enabling init ASM_ID=0x846b CODEC_ID=10ec0889 [ 2.468021] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input4 [ 2.468197] input: HDA Intel PCH Line Out Side as /devices/pci0000:00/0000:00:1b.0/sound/card0/input5 [ 2.468305] input: HDA Intel PCH Line Out CLFE as /devices/pci0000:00/0000:00:1b.0/sound/card0/input6 [ 2.468419] input: HDA Intel PCH Line Out Surround as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 [ 2.468529] input: HDA Intel PCH Line Out Front as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 2.468644] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 2.468720] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 2.468788] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 2.502287] init: avahi-cups-reload main process (1122) terminated with status 1 -- [ 2.592465] [drm] Initialized nouveau 1.1.1 20120801 for 0000:01:00.0 on minor 0 [ 2.592525] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 2.592581] hda_intel: Disabling MSI [ 2.592587] hda-intel 0000:01:00.1: Handle VGA-switcheroo audio client [ 2.592621] hda-intel 0000:01:00.1: Disabling 64bit DMA [ 2.593469] usb 2-1.4: new low-speed USB device number 4 using ehci-pci [ 2.599949] hda-intel 0000:01:00.1: Enable delay in RIRB handling [ 2.613131] usbcore: registered new interface driver usbhid -- [ 3.060378] sd 9:0:0:0: [sdd] Attached SCSI disk [ 3.093719] input: HDA NVidia HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input15 [ 3.093814] input: HDA NVidia HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input16 [ 3.093882] input: HDA NVidia HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input17 [ 3.093948] input: HDA NVidia HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input18 [ 3.190738] scsi 10:0:0:0: Direct-Access WD Ext HDD 1021 2002 PQ: 0 ANSI: 4 -- [ 4.633575] IPv6: ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [ 4.694338] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. pulseverbose.log: ( 0.000| 0.000) I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted ( 0.000| 0.000) I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted ( 0.000| 0.000) D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us. ( 0.003| 0.003) D: [pulseaudio] core-util.c: RealtimeKit worked. ( 0.003| 0.000) I: [pulseaudio] core-util.c: Successfully gained nice level -11. ( 0.003| 0.000) I: [pulseaudio] main.c: This is PulseAudio 3.0 ( 0.003| 0.000) D: [pulseaudio] main.c: Compilation host: x86_64-pc-linux-gnu ( 0.003| 0.000) D: [pulseaudio] main.c: Compilation CFLAGS: -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -Wall -W -Wextra -pipe -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option ( 0.003| 0.000) D: [pulseaudio] main.c: Running on host: Linux x86_64 3.11.0-031100rc2-generic #201307211535 SMP Sun Jul 21 19:36:42 UTC 2013 ( 0.003| 0.000) D: [pulseaudio] main.c: Found 4 CPUs. ( 0.003| 0.000) I: [pulseaudio] main.c: Page size is 4096 bytes ( 0.003| 0.000) D: [pulseaudio] main.c: Compiled with Valgrind support: no ( 0.003| 0.000) D: [pulseaudio] main.c: Running in valgrind mode: no ( 0.003| 0.000) D: [pulseaudio] main.c: Running in VM: no ( 0.003| 0.000) D: [pulseaudio] main.c: Optimized build: yes ( 0.003| 0.000) D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled. ( 0.003| 0.000) I: [pulseaudio] main.c: Machine ID is 7bd695a64b57f140878196e351d1dc20. ( 0.003| 0.000) I: [pulseaudio] main.c: Using runtime directory /run/user/vincent/pulse. ( 0.003| 0.000) I: [pulseaudio] main.c: Using state directory /home/vincent/.config/pulse. ( 0.003| 0.000) I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-3.0/modules. ( 0.003| 0.000) I: [pulseaudio] main.c: Running in system mode: no ( 0.003| 0.000) I: [pulseaudio] main.c: Fresh high-resolution timers available! Bon appetit! ( 0.004| 0.000) D: [pulseaudio] memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 ( 0.004| 0.000) I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 ( 0.004| 0.000) I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions. ( 0.004| 0.000) I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers. ( 0.004| 0.000) I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions. ( 0.004| 0.000) I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers. ( 0.004| 0.000) I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions. ( 0.004| 0.000) I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions. ( 0.004| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-device-volumes.tdb' ( 0.004| 0.000) I: [pulseaudio] module-device-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-device-volumes'. ( 0.004| 0.000) I: [pulseaudio] module.c: Loaded "module-device-restore" (index: #0; argument: ""). ( 0.004| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-stream-volumes.tdb' ( 0.004| 0.000) I: [pulseaudio] module-stream-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-stream-volumes'. ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 added for object /org/pulseaudio/stream_restore1 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry0 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry1 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry2 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry3 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry4 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry5 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry6 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry7 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry8 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry9 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry10 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry11 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry12 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry13 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry14 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry15 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry16 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry17 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry18 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry19 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry20 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry21 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry22 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry23 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry24 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry25 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry26 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry27 ( 0.005| 0.000) I: [pulseaudio] module.c: Loaded "module-stream-restore" (index: #1; argument: ""). ( 0.005| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-card-database.tdb' ( 0.005| 0.000) I: [pulseaudio] module-card-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-card-database'. ( 0.005| 0.000) I: [pulseaudio] module.c: Loaded "module-card-restore" (index: #2; argument: ""). ( 0.006| 0.000) I: [pulseaudio] module.c: Loaded "module-augment-properties" (index: #3; argument: ""). ( 0.006| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-udev-detect.so': success ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1 is busy: no ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="1" name="pci-0000_01_00.1" card_name="alsa_card.pci-0000_01_00.1" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"' ( 0.008| 0.001) D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus session bus 8923dc8a05563d0ffe2f214c51ed7550 as :1.110 ( 0.008| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio1' ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)utils.c: could not open configuration file /usr/share/alsa/ucm/HDA NVidia/HDA NVidia.conf ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)parser.c: error: could not parse configuration for card HDA NVidia ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)main.c: error: failed to import HDA NVidia use case configuration -2 ( 0.008| 0.000) I: [pulseaudio] alsa-ucm.c: UCM not available for card HDA NVidia ( 0.009| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-mono ( 0.009| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Mono (analog-mono) ( 0.009| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.009| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.009| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-mono ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-stereo ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.010| 0.000) D: [pulseaudio] alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.001) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device front:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:iec958-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D1c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:iec958-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-mono - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-stereo - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:iec958-stereo - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device front:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-mono - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-stereo - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:iec958-stereo - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround40:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround40:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-mono - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-stereo - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:iec958-stereo - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround41:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround41:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-mono - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-stereo - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:iec958-stereo - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround50:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround50:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-mono - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-stereo - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:iec958-stereo - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround51:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround51:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-mono - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-stereo - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:iec958-stereo - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround71:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround71:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-mono - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-stereo - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:iec958-stereo - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D1p' failed (-2) ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-mono - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-stereo - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-mono - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-stereo - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-mono - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-stereo - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/DTS) (iec958-dts-surround-51) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying dca:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM dca:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device dca:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-mono - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-stereo - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:iec958-stereo - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1 ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.029| 0.016) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.045| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo supported. ( 0.045| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1 ( 0.045| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1: No such file or directory ( 0.046| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-0' ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=3 Jack' succeeded (found!) ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc7ab50, direction=1 ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-0 (HDMI / DisplayPort), direction=1, priority=59, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=3, alsa_name='HDMI/DP,pcm=3 Jack', detection possible ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-mono - will not be able to open input:analog-mono ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-stereo - will not be able to open input:analog-stereo ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround) ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1 ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.077| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.093| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround supported. ( 0.093| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1 ( 0.093| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1: No such file or directory ( 0.093| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc8cce0, direction=1 ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-0 (HDMI / DisplayPort), direction=1, priority=59, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=3, alsa_name='HDMI/DP,pcm=3 Jack', detection possible ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-mono - will not be able to open input:analog-mono ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-stereo - will not be able to open input:analog-stereo ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra1 ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra1) ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.125| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.141| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra1 supported. ( 0.141| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.141| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.142| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-1' ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=7 Jack' succeeded (found!) ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc8ef10, direction=1 ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-mono - will not be able to open input:analog-mono ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-stereo - will not be able to open input:analog-stereo ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra1 ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra1) ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.173| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.189| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra1 supported. ( 0.189| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.189| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.189| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb2960, direction=1 ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-mono - will not be able to open input:analog-mono ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-stereo - will not be able to open input:analog-stereo ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra2 ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra2) ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,2 with SND_PCM_NO_AUTO_FORMAT ... ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,2 ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.221| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.237| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra2 supported. ( 0.237| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,2 ( 0.237| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,2: No such file or directory ( 0.238| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-2' ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=8 Jack' succeeded (found!) ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb2d80, direction=1 ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-2 (HDMI / DisplayPort 3), direction=1, priority=57, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=8, alsa_name='HDMI/DP,pcm=8 Jack', detection possible ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-mono - will not be able to open input:analog-mono ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-stereo - will not be able to open input:analog-stereo ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra2 ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra2) ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,2 with SND_PCM_NO_AUTO_FORMAT ... ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,2 ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.269| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.285| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra2 supported. ( 0.285| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,2 ( 0.285| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,2: No such file or directory ( 0.285| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.285| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb4f20, direction=1 ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-2 (HDMI / DisplayPort 3), direction=1, priority=57, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=8, alsa_name='HDMI/DP,pcm=8 Jack', detection possible ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-mono - will not be able to open input:analog-mono ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-stereo - will not be able to open input:analog-stereo ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra3 ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra3) ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,3 with SND_PCM_NO_AUTO_FORMAT ... ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,3 ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.317| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.333| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra3 supported. ( 0.333| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,3 ( 0.333| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,3: No such file or directory ( 0.334| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-3' ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=9 Jack' succeeded (found!) ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc190e0, direction=1 ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-3 (HDMI / DisplayPort 4), direction=1, priority=56, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=9, alsa_name='HDMI/DP,pcm=9 Jack', detection possible ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-mono - will not be able to open input:analog-mono ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-stereo - will not be able to open input:analog-stereo ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra3 ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra3) ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,3 with SND_PCM_NO_AUTO_FORMAT ... ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,3 ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.365| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.381| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra3 supported. ( 0.381| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,3 ( 0.381| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,3: No such file or directory ( 0.381| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb53c0, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-3 (HDMI / DisplayPort 4), direction=1, priority=56, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=9, alsa_name='HDMI/DP,pcm=9 Jack', detection possible ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-mono - will not be able to open input:analog-mono ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-stereo - will not be able to open input:analog-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile set 0xc26dc0, auto_profiles=yes, probed=yes, n_mappings=8, n_profiles=8, n_decibel_fixes=0 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo (Digital Stereo (HDMI)), priority=54, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround (Digital Surround 5.1 (HDMI)), priority=3, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra1 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra1 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra2 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra2 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra3 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra3 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo (Digital Stereo (HDMI) Output), priority=5400, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround (Digital Surround 5.1 (HDMI) Output), priority=300, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra1 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra1 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra2 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra2 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra2 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra2 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra3 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra3 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra3 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra3 ( 0.382| 0.000) I: [pulseaudio] module-card-restore.c: Restoring profile for card alsa_card.pci-0000_01_00.1. ( 0.382| 0.000) I: [pulseaudio] module-card-restore.c: Restoring port latency offsets for card alsa_card.pci-0000_01_00.1. ( 0.382| 0.000) I: [pulseaudio] card.c: Created 0 "alsa_card.pci-0000_01_00.1" ( 0.382| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio1' ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.413| 0.030) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.429| 0.015) I: [pulseaudio] alsa-sink.c: Successfully opened device hdmi:1,1. ( 0.429| 0.000) I: [pulseaudio] alsa-sink.c: Selected mapping 'Digital Stereo (HDMI)' (hdmi-stereo-extra1). ( 0.429| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode. ( 0.429| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.429| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.430| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Added 1 ports ( 0.430| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: sink:alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1:null ( 0.430| 0.000) I: [pulseaudio] sink.c: Created sink 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.resolution_bits = "16" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.api = "alsa" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.class = "sound" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.class = "generic" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subclass = "generic-mix" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.name = "HDMI 1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.id = "HDMI 1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subdevice = "0" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subdevice_name = "subdevice #0" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.device = "7" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.card = "1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.card_name = "HDA NVidia" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 17" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.bus_path = "pci-0000:01:00.1" ( 0.430| 0.000) I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.bus = "pci" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.vendor.id = "10de" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.vendor.name = "NVIDIA Corporation" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.string = "hdmi:1,1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.buffering.buffer_size = "8064" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.buffering.fragment_size = "2688" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.access_mode = "mmap" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.profile.name = "hdmi-stereo-extra1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.profile.description = "Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.description = "HDA NVidia Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.mixer_name = "Nvidia GPU 41 HDMI/DP" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.components = "HDA:10de0041,10438422,00100100" ( 0.430| 0.000) I: [pulseaudio] sink.c: module-udev-detect.discovered = "1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card-pci" ( 0.430| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.430| 0.000) I: [pulseaudio] source.c: Created source 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 0.430| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of HDA NVidia Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.card = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA NVidia" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 17" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 0.430| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:01:00.1" ( 0.430| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1" ( 0.430| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 0.430| 0.000) I: [pulseaudio] source.c: device.vendor.id = "10de" ( 0.430| 0.000) I: [pulseaudio] source.c: device.vendor.name = "NVIDIA Corporation" ( 0.430| 0.000) I: [pulseaudio] source.c: device.string = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Using 3.0 fragments of size 2688 bytes (15.24ms), buffer size is 8064 bytes (45.71ms) ( 0.430| 0.000) D: [pulseaudio] alsa-sink.c: hwbuf_unused=0 ( 0.430| 0.000) D: [pulseaudio] alsa-sink.c: setting avail_min=1 ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path hdmi-output-1 ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Driver does not support hardware volume control, falling back to software volume control. ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Driver does not support hardware mute control, falling back to software mute control. ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Hooks PCM ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2016 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_size : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_time : 15238 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Slave: Hardware PCM card 1 'HDA NVidia' device 7 subdevice 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2016 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_size : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_time : 15238 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 0.431| 0.000) D: [alsa-sink] alsa-sink.c: Thread starting up ( 0.434| 0.003) D: [alsa-sink] core-util.c: RealtimeKit worked. ( 0.434| 0.000) I: [alsa-sink] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 0.434| 0.000) I: [alsa-sink] alsa-sink.c: Starting playback. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Found 4 jacks. ( 0.434| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=3 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-0 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=7 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-1 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=8 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-2 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=9 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-3 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) I: [pulseaudio] module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="1" name="pci-0000_01_00.1" card_name="alsa_card.pci-0000_01_00.1" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). ( 0.435| 0.000) I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1 (alsa_card.pci-0000_01_00.1) module loaded. ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: no ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"' ( 0.435| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio0' ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)utils.c: could not open configuration file /usr/share/alsa/ucm/HDA Intel PCH/HDA Intel PCH.conf ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)parser.c: error: could not parse configuration for card HDA Intel PCH ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)main.c: error: failed to import HDA Intel PCH use case configuration -2 ( 0.435| 0.000) I: [pulseaudio] alsa-ucm.c: UCM not available for card HDA Intel PCH ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-mono ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Mono (analog-mono) ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-mono ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-stereo ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.437| 0.000) D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo supported. ( 0.438| 0.001) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 0.438| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 0.438| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 0.438| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-front' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-rear' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-internal' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Internal Mic Phantom Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone-internal', none of required-any elements preset. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-dock' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone-dock', none of required-any elements preset. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' failed. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Select' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost (+20dB)' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone', none of required-any elements preset. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-linein' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Jack' succeeded (found!) ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-video' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-video' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-radio' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found) ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Auto-Mute Mode' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone', none of required-any elements preset. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc76970, direction=2 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-front (Front Microphone), direction=2, priority=85, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Option Front Mic (analog-input-microphone-front/analog-input-microphone-front) index=0, priority=0 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Mic, alsa_name='Front Mic Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-front (analog-input-microphone-front) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-rear (Rear Microphone), direction=2, priority=82, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Option Rear Mic (analog-input-microphone-rear/analog-input-microphone-rear) index=1, priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Rear Mic, alsa_name='Rear Mic Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-rear (analog-input-microphone-rear) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-linein (Line In), direction=2, priority=81, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=46, min_dB=-16, max_dB=60 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Option Line (analog-input-linein/analog-input-linein) index=2, priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Line, alsa_name='Line Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-linein (analog-input-linein) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:iec958-stereo ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo) ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2) ( 0.441| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:0: No such file or directory ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:iec958-stereo ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-mono ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.442| 0.000) I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-mono - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-stereo - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:iec958-stereo - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo supported. ( 0.443| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 0.443| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 0.443| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-speaker' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Speaker Phantom Jack' succeeded (not found) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-output-speaker', none of required-any elements preset. ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-speaker' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=3, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-mono' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc72640, direction=1 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', detection possible ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', detection unavailable ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', detection unavailable ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-mono - will not be able to open input:analog-mono ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40) ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.445| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround40:0 ( 0.445| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 185 ms ( 2.493| 2.048) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 2.493| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 supported. ( 2.493| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround40:0 ( 2.493| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround40:0: No such file or directory ( 2.494| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc30dd0, direction=1 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-mono - will not be able to open input:analog-mono ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo supported. ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:iec958-stereo - will not be able to open input:iec958-stereo ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41) ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ... ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround41:0 ( 2.495| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.549| 4.054) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.549| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 supported. ( 6.549| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround41:0 ( 6.549| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround41:0: No such file or directory ( 6.550| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xd28070, direction=1 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-mono - will not be able to open input:analog-mono ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo supported. ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50) ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround50:0 ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 supported. ( 6.551| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround50:0 ( 6.551| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround50:0: No such file or directory ( 6.551| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcf8270, direction=1 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-mono - will not be able to open input:analog-mono ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo supported. ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51) ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround51:0 ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 supported. ( 6.552| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround51:0 ( 6.552| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround51:0: No such file or directory ( 6.552| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xd20420, direction=1 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-mono - will not be able to open input:analog-mono ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo supported. ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71) ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.553| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround71:0 ( 6.553| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 92 ms ( 8.597| 2.044) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 8.597| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 supported. ( 8.597| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround71:0 ( 8.597| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround71:0: No such file or directory ( 8.598| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc619a0, direction=1 ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-mono - will not be able to open input:analog-mono ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo supported. ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:iec958-stereo - will not be able to open input:iec958-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo) ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ... ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Managed to open iec958:0 ( 8.599| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 9.617| 1.018) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo supported. ( 9.618| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL iec958:0 ( 9.618| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer iec958:0: No such file or directory ( 9.618| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'iec958-stereo-output' ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=1, enumeration=0). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xce30e0, direction=1 ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Path iec958-stereo-output (Digital Output (S/PDIF)), direction=1, priority=0, probed=yes, supported=yes, has_mute=yes, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-mono - will not be able to open input:analog-mono ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-stereo ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo supported. ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40 ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40) ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.618| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-mono - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-stereo - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-mono - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-stereo - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/DTS) (iec958-dts-surround-51) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying dca:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM dca:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device dca:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-mono - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-stereo - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:iec958-stereo - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D3p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-mono - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-stereo - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:iec958-stereo - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D3p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-mono - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-stereo - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:iec958-stereo - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra1) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,1 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D7p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,1: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-mono - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-stereo - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra1) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,1 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D7p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,1: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-mono - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-stereo - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:iec958-stereo - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra2) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,2 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D8p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,2: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-mono - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-stereo - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra2) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,2 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D8p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,2: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-mono - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-stereo - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:iec958-stereo - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra3) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,3 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D9p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,3: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-mono - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-stereo - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra3) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,3 with SND_PCM_NO_AUTO_FORMAT ... ( 9.621| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D9p' failed (-2) ( 9.621| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,3: No such file or directory ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-mono - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-stereo - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:iec958-stereo - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile set 0xc5d8d0, auto_profiles=yes, probed=yes, n_mappings=7, n_profiles=15, n_decibel_fixes=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-stereo (Analog Stereo), priority=60, channel_map=front-left,front-right, supported=yes, direction=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-40 (Analog Surround 4.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-41 (Analog Surround 4.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,lfe, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-50 (Analog Surround 5.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-51 (Analog Surround 5.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-71 (Analog Surround 7.1), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping iec958-stereo (Digital Stereo (IEC958)), priority=55, channel_map=front-left,front-right, supported=yes, direction=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo (Analog Stereo Input), priority=60, supported=yes n_input_mappings=1, n_output_mappings=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo (Analog Stereo Output), priority=6000, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo (Analog Stereo Duplex), priority=6060, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 (Analog Surround 4.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-40 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo (Analog Surround 4.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-40 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 (Analog Surround 4.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-41 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo (Analog Surround 4.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-41 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 (Analog Surround 5.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-50 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo (Analog Surround 5.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-50 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 (Analog Surround 5.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-51 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo (Analog Surround 5.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-51 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 (Analog Surround 7.1 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-71 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo (Analog Surround 7.1 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-71 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo (Digital Stereo (IEC958) Output), priority=5500, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output iec958-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo (Digital Stereo (IEC958) Output + Analog Stereo Input), priority=5560, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output iec958-stereo ( 9.621| 0.000) I: [pulseaudio] module-card-restore.c: Restoring profile for card alsa_card.pci-0000_00_1b.0. ( 9.621| 0.000) I: [pulseaudio] module-card-restore.c: Restoring port latency offsets for card alsa_card.pci-0000_00_1b.0. ( 9.621| 0.000) I: [pulseaudio] card.c: Created 1 "alsa_card.pci-0000_00_1b.0" ( 9.622| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0' ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 12.681| 3.059) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Successfully opened device front:0. ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode. ( 12.682| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 12.682| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 12.682| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.682| 0.000) D: [pulseaudio] alsa-mixer.c: Added 2 ports ( 12.682| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.682| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: sink:alsa_output.pci-0000_00_1b.0.analog-stereo:null ( 12.682| 0.000) I: [pulseaudio] sink.c: Created sink 1 "alsa_output.pci-0000_00_1b.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.resolution_bits = "16" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.api = "alsa" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.class = "sound" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.class = "generic" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subclass = "generic-mix" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.name = "ALC889 Analog" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.id = "ALC889 Analog" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subdevice = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subdevice_name = "subdevice #0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.device = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.card = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.card_name = "HDA Intel PCH" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0" ( 12.682| 0.000) I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.bus = "pci" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.vendor.id = "8086" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.form_factor = "internal" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.string = "front:0" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.buffering.buffer_size = "8112" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.buffering.fragment_size = "4056" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.access_mode = "mmap" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.profile.name = "analog-stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.profile.description = "Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.description = "Built-in Audio Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.mixer_name = "Realtek ALC889" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.components = "HDA:10ec0889,1043846b,00100004" ( 12.682| 0.000) I: [pulseaudio] sink.c: module-udev-detect.discovered = "1" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card-pci" ( 12.682| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.682| 0.000) I: [pulseaudio] source.c: Created source 1 "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.682| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of Built-in Audio Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.card = "0" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA Intel PCH" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 12.682| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0" ( 12.682| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.682| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 12.682| 0.000) I: [pulseaudio] source.c: device.vendor.id = "8086" ( 12.682| 0.000) I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation" ( 12.682| 0.000) I: [pulseaudio] source.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.682| 0.000) I: [pulseaudio] source.c: device.form_factor = "internal" ( 12.682| 0.000) I: [pulseaudio] source.c: device.string = "0" ( 12.682| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 12.682| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 12.682| 0.000) I: [pulseaudio] alsa-sink.c: Using 2.0 fragments of size 4056 bytes (22.99ms), buffer size is 8112 bytes (45.99ms) ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: hwbuf_unused=0 ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: setting avail_min=1 ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path analog-output ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -177.00 dB to 2.00 dB. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Fixing base volume to -2.00 dB ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Using hardware mute control. ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Soft volume PCM ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Control: PCM Playback Volume ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: min_dB: -51 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: max_dB: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: resolution: 256 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: Read hardware volume: 0: 76% 1: 76% ( 12.683| 0.000) D: [alsa-sink] alsa-sink.c: Thread starting up ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: in dB: 0: -7.20 dB 1: -7.20 dB ( 12.687| 0.003) D: [alsa-sink] core-util.c: RealtimeKit worked. ( 12.687| 0.000) I: [alsa-sink] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 12.687| 0.000) I: [alsa-sink] alsa-sink.c: Starting playback. ( 12.687| 0.000) D: [pulseaudio] module-device-restore.c: Could not set format on sink alsa_output.pci-0000_00_1b.0.analog-stereo ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Successfully opened device front:0. ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Successfully enabled mmap() mode. ( 12.687| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 12.687| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 12.687| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Added 3 ports ( 12.688| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.688| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: source:alsa_input.pci-0000_00_1b.0.analog-stereo:null ( 12.688| 0.000) I: [pulseaudio] source.c: Created source 2 "alsa_input.pci-0000_00_1b.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.resolution_bits = "16" ( 12.688| 0.000) I: [pulseaudio] source.c: device.api = "alsa" ( 12.688| 0.000) I: [pulseaudio] source.c: device.class = "sound" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.class = "generic" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subclass = "generic-mix" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.name = "ALC889 Analog" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.id = "ALC889 Analog" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subdevice = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subdevice_name = "subdevice #0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.device = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.card = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA Intel PCH" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 12.688| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0" ( 12.688| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.688| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 12.688| 0.000) I: [pulseaudio] source.c: device.vendor.id = "8086" ( 12.688| 0.000) I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation" ( 12.688| 0.000) I: [pulseaudio] source.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.688| 0.000) I: [pulseaudio] source.c: device.form_factor = "internal" ( 12.688| 0.000) I: [pulseaudio] source.c: device.string = "front:0" ( 12.688| 0.000) I: [pulseaudio] source.c: device.buffering.buffer_size = "8112" ( 12.688| 0.000) I: [pulseaudio] source.c: device.buffering.fragment_size = "4056" ( 12.688| 0.000) I: [pulseaudio] source.c: device.access_mode = "mmap" ( 12.688| 0.000) I: [pulseaudio] source.c: device.profile.name = "analog-stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: device.profile.description = "Analog Stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: device.description = "Built-in Audio Analog Stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.mixer_name = "Realtek ALC889" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.components = "HDA:10ec0889,1043846b,00100004" ( 12.688| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 12.688| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using 2.0 fragments of size 4056 bytes (22.99ms), buffer size is 8112 bytes (45.99ms) ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: hwbuf_unused=0 ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: setting avail_min=1 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path analog-input-microphone-front ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-front (Front Microphone), direction=2, priority=85, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Option Front Mic (analog-input-microphone-front/analog-input-microphone-front) index=0, priority=0 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Mic, alsa_name='Front Mic Jack', detection possible ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-front (analog-input-microphone-front) priority=0 ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Successfully enabled deferred volume. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Hardware volume ranges from -16.00 dB to 60.00 dB. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Fixing base volume to -60.00 dB ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using hardware volume control. Hardware dB scale supported. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using hardware mute control. ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: stream : CAPTURE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: Read hardware volume: 0: 5% 1: 5% ( 12.688| 0.000) D: [alsa-source] alsa-source.c: Thread starting up ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: in dB: 0: -76.00 dB 1: -76.00 dB ( 12.691| 0.002) D: [alsa-source] core-util.c: RealtimeKit worked. ( 12.691| 0.000) I: [alsa-source] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 12.691| 0.000) I: [alsa-source] alsa-source.c: Starting capture. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Found 4 jacks. ( 12.691| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Front Mic Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-microphone-front to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Rear Mic Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-microphone-rear to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Line Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-linein to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) I: [pulseaudio] module.c: Loaded "module-alsa-card" (index: #5; argument: "device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). ( 12.691| 0.000) I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:1b.0/sound/card0 (alsa_card.pci-0000_00_1b.0) module loaded. ( 12.691| 0.000) I: [pulseaudio] module-udev-detect.c: Found 2 cards. ( 12.691| 0.000) I: [pulseaudio] module.c: Loaded "module-udev-detect" (index: #6; argument: "tsched=0"). ( 12.691| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-jackdbus-detect.so': failure ( 12.691| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-bluetooth-policy.so': success ( 12.692| 0.000) I: [pulseaudio] module.c: Loaded "module-bluetooth-policy" (index: #7; argument: ""). ( 12.692| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-bluetooth-discover.so': success ( 12.692| 0.000) D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus system bus 200a58f6723ca27a0d9b04b351ed7524 as :1.89 ( 12.693| 0.000) D: [pulseaudio] bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired ( 12.693| 0.000) I: [pulseaudio] module.c: Loaded "module-bluetooth-discover" (index: #8; argument: ""). ( 12.693| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-esound-protocol-unix.so': failure ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #9; argument: ""). ( 12.694| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-gconf.so': failure ( 12.694| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.694| 0.000) I: [pulseaudio] module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_1b.0.analog-stereo'. ( 12.694| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.694| 0.000) I: [pulseaudio] module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_1b.0.analog-stereo'. ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-default-device-restore" (index: #10; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-rescue-streams" (index: #11; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-always-sink" (index: #12; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-intended-roles" (index: #13; argument: ""). ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 becomes idle, timeout in 5 seconds. ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1b.0.analog-stereo becomes idle, timeout in 5 seconds. ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1b.0.analog-stereo becomes idle, timeout in 5 seconds. ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). ( 12.695| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-console-kit.so': success ( 12.695| 0.000) I: [pulseaudio] client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" ( 12.695| 0.000) D: [pulseaudio] module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2 ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-console-kit" (index: #15; argument: ""). ( 12.695| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-systemd-login.so': failure ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-filter-heuristics" (index: #17; argument: ""). ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-filter-apply" (index: #18; argument: ""). ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-0 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-1 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-2 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-3 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-microphone-front ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-microphone-rear ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-linein ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-output-headphones ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-switch-on-port-available" (index: #19; argument: ""). ( 12.696| 0.000) D: [pulseaudio] main.c: Got org.PulseAudio1! ( 12.697| 0.000) D: [pulseaudio] main.c: Got org.pulseaudio.Server! ( 12.697| 0.000) I: [pulseaudio] main.c: Daemon startup complete. ( 12.697| 0.000) I: [pulseaudio] client.c: Created 1 "Native client (UNIX socket client)" ( 12.697| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 12.697| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 12.697| 0.000) I: [pulseaudio] client.c: Created 2 "Native client (UNIX socket client)" ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 12.697| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 12.697| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 12.697| 0.000) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for indicator-sound-service ( 12.697| 0.000) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon ( 17.697| 5.000) I: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1b.0.analog-stereo idle for too long, suspending ... ( 17.698| 0.000) D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_1b.0.analog-stereo is 0x0004, suspending ( 17.698| 0.000) I: [alsa-source] alsa-source.c: Device suspended... ( 17.698| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.698| 0.000) I: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1b.0.analog-stereo idle for too long, suspending ... ( 17.698| 0.000) D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_1b.0.analog-stereo is 0x0004, suspending ( 17.699| 0.001) I: [alsa-sink] alsa-sink.c: Device suspended... ( 17.699| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.699| 0.000) I: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 idle for too long, suspending ... ( 17.699| 0.000) D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 is 0x0004, suspending ( 17.700| 0.000) I: [alsa-sink] alsa-sink.c: Device suspended... ( 17.700| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.700| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 17.700| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 19.383| 1.683) I: [pulseaudio] client.c: Created 3 "Native client (UNIX socket client)" ( 19.537| 0.153) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 19.537| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 19.537| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 19.537| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 19.565| 0.028) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for gnome-control-center ( 19.565| 0.000) D: [pulseaudio] module-augment-properties.c: Found /usr/share/applications/gnome-control-center.desktop. ( 30.916| 11.350) I: [pulseaudio] client.c: Freed 3 "GNOME Volume Control Dialog" ( 30.916| 0.000) I: [pulseaudio] protocol-native.c: Connection died. ( 71.860| 40.944) I: [pulseaudio] main.c: Got signal SIGINT. ( 71.860| 0.000) I: [pulseaudio] main.c: Exiting. ( 71.860| 0.000) I: [pulseaudio] main.c: Daemon shutdown initiated. ( 71.860| 0.000) I: [pulseaudio] module.c: Unloading "module-device-restore" (index: #0). ( 71.860| 0.000) I: [pulseaudio] module.c: Unloaded "module-device-restore" (index: #0). ( 71.860| 0.000) I: [pulseaudio] module.c: Unloading "module-stream-restore" (index: #1). ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 removed from object /org/pulseaudio/stream_restore1 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry0 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry1 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry2 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry3 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry4 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry5 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry6 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry7 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry8 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry9 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry10 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry11 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry12 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry13 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry14 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry15 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry16 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry17 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry18 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry19 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry20 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry21 ( 71.860| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry22 ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry23 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry24 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry25 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry26 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry27 ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-stream-restore" (index: #1). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-card-restore" (index: #2). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-card-restore" (index: #2). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-augment-properties" (index: #3). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-augment-properties" (index: #3). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-alsa-card" (index: #4). ( 71.861| 0.000) D: [pulseaudio] module-rescue-streams.c: No sink inputs to move away. ( 71.861| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.861| 0.000) D: [alsa-sink] alsa-sink.c: Thread shutting down ( 71.861| 0.000) I: [pulseaudio] sink.c: Freeing sink 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1" ( 71.861| 0.000) I: [pulseaudio] source.c: Freeing source 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1.monitor" ( 71.861| 0.000) I: [pulseaudio] card.c: Freed 0 "alsa_card.pci-0000_01_00.1" ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-alsa-card" (index: #4). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-alsa-card" (index: #5). ( 71.861| 0.000) D: [pulseaudio] module-always-sink.c: Autoloading null-sink as no other sinks detected. ( 71.861| 0.000) I: [pulseaudio] sink.c: Created sink 2 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 71.861| 0.000) I: [pulseaudio] sink.c: device.description = "Dummy Output" ( 71.861| 0.000) I: [pulseaudio] sink.c: device.class = "abstract" ( 71.861| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card" ( 71.862| 0.000) I: [pulseaudio] source.c: Created source 3 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 71.862| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of Dummy Output" ( 71.862| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 71.862| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-input-microphone" ( 71.862| 0.000) D: [null-sink] module-null-sink.c: Thread starting up ( 71.862| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. ( 71.862| 0.000) I: [pulseaudio] module.c: Loaded "module-null-sink" (index: #20; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No sink inputs to move away. ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.862| 0.000) D: [alsa-sink] alsa-sink.c: Thread shutting down ( 71.862| 0.000) I: [pulseaudio] sink.c: Freeing sink 1 "alsa_output.pci-0000_00_1b.0.analog-stereo" ( 71.862| 0.000) I: [pulseaudio] source.c: Freeing source 1 "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.862| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 71.862| 0.000) D: [alsa-source] alsa-source.c: Thread shutting down ( 71.862| 0.000) I: [pulseaudio] source.c: Freeing source 2 "alsa_input.pci-0000_00_1b.0.analog-stereo" ( 71.862| 0.000) I: [pulseaudio] card.c: Freed 1 "alsa_card.pci-0000_00_1b.0" ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-alsa-card" (index: #5). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-udev-detect" (index: #6). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-udev-detect" (index: #6). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-bluetooth-policy" (index: #7). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-bluetooth-policy" (index: #7). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-bluetooth-discover" (index: #8). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-bluetooth-discover" (index: #8). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-native-protocol-unix" (index: #9). ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 1 "Indicator Sound" ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 2 "GNOME Volume Control Media Keys" ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-native-protocol-unix" (index: #9). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-default-device-restore" (index: #10). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-default-device-restore" (index: #10). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-rescue-streams" (index: #11). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-rescue-streams" (index: #11). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-always-sink" (index: #12). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-always-sink" (index: #12). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-intended-roles" (index: #13). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-intended-roles" (index: #13). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-suspend-on-idle" (index: #14). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-suspend-on-idle" (index: #14). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-console-kit" (index: #15). ( 71.863| 0.000) D: [pulseaudio] module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session2 ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-console-kit" (index: #15). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-position-event-sounds" (index: #16). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-position-event-sounds" (index: #16). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-filter-heuristics" (index: #17). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-filter-heuristics" (index: #17). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-filter-apply" (index: #18). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-filter-apply" (index: #18). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-switch-on-port-available" (index: #19). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-switch-on-port-available" (index: #19). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-null-sink" (index: #20). ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.864| 0.000) D: [null-sink] module-null-sink.c: Thread shutting down ( 71.864| 0.000) I: [pulseaudio] sink.c: Freeing sink 2 "auto_null" ( 71.864| 0.000) I: [pulseaudio] source.c: Freeing source 3 "auto_null.monitor" ( 71.864| 0.000) I: [pulseaudio] module.c: Unloaded "module-null-sink" (index: #20). ( 71.864| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.864| 0.000) I: [pulseaudio] main.c: Daemon terminated.

1 0

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA
by Anssi Hannula 09 Sep '13

09 Sep '13

hdmi_channel_allocation() tries to find a HDMI channel allocation that matches the number channels in the playback stream and contains only speakers that the HDMI sink has reported as available via EDID. If no such allocation is found, 0 (stereo audio) is used. Using CA 0 causes the audio causes the sink to discard everything except the first two channels (front left and front right). However, the sink may be capable of receiving more channels than it has speakers (and then perform downmix or discard the extra channels), in which case it is preferable to use a CA that contains extra channels than to use CA 0 which discards all the non-stereo channels. Additionally, it seems that HBR (HD) passthrough output does not work on Intel HDMI codecs when CA is set to 0 (possibly the codec zeroes channels not present in CA). This happens with all receivers that report a 5.1 speaker mask since a HBR stream is carried on 8 channels to the codec. Add a fallback in the CA selection so that the CA channel count at least matches the stream channel count, even if the stream contains channels not present in the sink speaker descriptor. Thanks to GrimGriefer at OpenELEC forums for discovering that changing the sink speaker mask allowed HBR output. Reported-by: GrimGriefer Reported-by: Ashecrow Reported-by: Frank Zafka <kafkaesque1978(a)gmail.com> Reported-by: Peter Frühberger <fritsch(a)xbmc.org> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- Hopefully this fixes HBR (HD passthrough) for the remaining Intel users who were still experiencing problems. BTW, the hdmi_channel_allocation() logic seems also otherwise somewhat suspect to me. Shouldn't we always select an allocation matching ALSA channel mapping, instead of re-assigning channels received from userspace "randomly" to sink speakers (in case of unusual sink speaker mask)? Anyway, I let it be for now at least. sound/pci/hda/patch_hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..354fc55 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -551,6 +551,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } } + if (!ca) { + /* if there was no match, select the regular ALSA channel + * allocation with the matching number of channels */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels) { + ca = channel_allocations[i].ca_index; + break; + } + } + } + snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); -- 1.8.1.5

3 6

[alsa-devel] [PATCH] codecs: mc13783: add spi errate fix
by Steffen Trumtrar 09 Sep '13

09 Sep '13

The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock and main audio clock speed, the Audio Codec or Stereo DAC do sometimes not start when programmed to do so. This is due to an internal clock timing issue related to the loading of the SPI bits into the audio block. On an i.MX27 based system, this issue lead to switched audio channels under certain circumstances: RTC + Touch + Audio are used and loaded at startup. The mentioned workaround of writing registers 40 and 41 two times is implemented here. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/mc13783.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd..e78353d 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == PMIC_AUDIO_CODEC || reg == PMIC_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; -- 1.8.4.rc3

2 3

[alsa-devel] [PATCH] ASoC: rsnd: fixup flag name of rsnd_scu_platform_info
by Kuninori Morimoto 09 Sep '13

09 Sep '13

it should be *USE*, not *USB* Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 2 +- sound/soc/sh/rcar/scu.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index d35412a..fe66533 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -55,7 +55,7 @@ struct rsnd_ssi_platform_info { /* * flags */ -#define RSND_SCU_USB_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ +#define RSND_SCU_USE_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ struct rsnd_scu_platform_info { u32 flags; diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 184d900..2df2e91 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -157,9 +157,9 @@ static int rsnd_scu_start(struct rsnd_mod *mod, int ret; /* - * SCU will be used if it has RSND_SCU_USB_HPBIF flags + * SCU will be used if it has RSND_SCU_USE_HPBIF flags */ - if (!(flags & RSND_SCU_USB_HPBIF)) { + if (!(flags & RSND_SCU_USE_HPBIF)) { /* it use PIO transter */ dev_dbg(dev, "%s%d is not used\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: ad1980: fix the indentation
by Barry Song 09 Sep '13

09 Sep '13

Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/ad1980.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a88..be5032c 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -61,39 +61,39 @@ static const struct soc_enum ad1980_cap_src = SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { -SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), -SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), -SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), -SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), -SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), -SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), -SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0), -SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), + SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0), + SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), -SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), -SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), + SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), + SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), -SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), -SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), -SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), -SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), + SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), + SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), -SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), -SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), + SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), + SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), -SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), -SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), + SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), + SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), -SOC_ENUM("Capture Source", ad1980_cap_src), + SOC_ENUM("Capture Source", ad1980_cap_src), -SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), + SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { -- 1.8.2.3

2 1

[alsa-devel] [PATCH] aplay - fix two off by one errors
by Jurgen Kramer 09 Sep '13

09 Sep '13

Attached patch for aplay fixes two off-by-one errors preventing usage of the last available sample format. diff -uNrp alsa-utils-1.0.27.2/aplay/aplay.c alsa-utils-1.0.27.2-jk/aplay/aplay.c --- alsa-utils-1.0.27.2/aplay/aplay.c 2013-07-09 17:28:37.000000000 +0200 +++ alsa-utils-1.0.27.2-jk/aplay/aplay.c 2013-09-08 13:46:23.746825429 +0200 @@ -242,7 +242,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" ) , command); printf(_("Recognized sample formats are:")); - for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { + for (k = 0; k <= SND_PCM_FORMAT_LAST; ++k) { const char *s = snd_pcm_format_name(k); if (s) printf(" %s", s); @@ -1114,7 +1114,7 @@ static void show_available_sample_format snd_pcm_format_t format; fprintf(stderr, "Available formats:\n"); - for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { + for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) { if (snd_pcm_hw_params_test_format(handle, params, format) == 0) fprintf(stderr, "- %s\n", snd_pcm_format_name(format)); } Jurgen

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Toshiba Satellite C870 to MSI blacklist
by Takashi Iwai 09 Sep '13

09 Sep '13

Toshiba Satellite C870 shows interrupt problems occasionally when certain mixer controls like "Mic Switch" is toggled. This seems worked around by not using MSI. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=833585 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6c9829..7f88dd6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3400,6 +3400,7 @@ static struct snd_pci_quirk msi_black_list[] = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ + SND_PCI_QUIRK(0x1179, 0xfb44, "Toshiba Satellite C870", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} -- 1.8.4

1 0

[alsa-devel] Help with editing the Raw Midi data with ALSA?
by Ember Autumn Rose Leona 09 Sep '13

09 Sep '13

Dear ALSA team, I just joined the mailing list on after being referred to it. Is it possible for me to edit the Raw Midi data with ALSA... I looked a RtMidi and that opens a port but doesn't help me Midi Poison or Midi Message Map. What I want to do is edit the Midi Messages, re-mapping the notes. I simply want to flip the progression of the keys on a midi piano from high notes to low notes. On the fly. Basically I want to take the note on message byte and use the note number as an index for an array of remapped keys. I want to flip the keyboard on a D note for now... then later allow users to turn this remapping off and on (on the fly.) Also it would be nice to have certain note ranges mapped to different programs/patches. -Ember PS> I wish I could help test the sound drivers... but I only have an inexpensive intel integrated sound chip and I am on a dell which needs to be replaced with something par.

3 5

[alsa-devel] where to get snd_pcm_hardware info when usb audio device plug in?
by loody 09 Sep '13

09 Sep '13

hi all: is it possible to get snd_pcm_hardware information in /proc, /sys or anywhere when usb audio device plug in? if no, is there api I can take in usermode to get snd_pcm_hardware of this usb audio device? -- Thanks for your great help,

2 6

[alsa-devel] [PATCH] ALSA: snd-usb: sample rate detection workaround for Axe-Fx II
by Daniel Mack 08 Sep '13

08 Sep '13

Fractal Audio Systems Axe-Fx II and maybe other devices fail to return their sample rate range descriptor when only asked for the first two bytes. Add a workaround and assume exactly one sample rate in this case, and continue with the full descriptor fetch. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Joachim Gahl <joachim.gahl(a)web.de> --- sound/usb/format.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 3525231..d9de43d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -297,13 +297,17 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, snd_usb_ctrl_intf(chip) | (clock << 8), tmp, sizeof(tmp)); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", - __func__, clock); - goto err; - } + /* + * Some devices, including Fractal Audio Systems Axe-Fx II, fail to + * return only a fraction of this descriptor. If that happens, assume + * we have only one sample rate and continue. In case of any other + * error, we will bail on the full descriptor fetch below. + */ + if (ret < 0) + nr_triplets = 1; + else + nr_triplets = combine_word(tmp); - nr_triplets = (tmp[1] << 8) | tmp[0]; data_size = 2 + 12 * nr_triplets; data = kzalloc(data_size, GFP_KERNEL); if (!data) { @@ -319,8 +323,9 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, data, data_size); if (ret < 0) { - snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", - __func__, clock); + snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range " + "(clock %d, nr_triplets %d)\n", + __func__, clock, nr_triplets); ret = -EINVAL; goto err_free; } -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: Select regmap-mmio
by Mark Brown 08 Sep '13

08 Sep '13

From: Mark Brown <broonie(a)linaro.org> The S/PDIF driver needs regmap so select it to make sure it gets included in the build. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 704e246..b7ab71f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -198,6 +198,7 @@ config SND_SOC_IMX_SPDIF select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF select SND_SOC_SPDIF + select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- 1.8.4.rc3

2 1

[alsa-devel] alsa crash with “ ALSA sound/usb/urb.c:179 cannot submit urb (err = -27)” on imx6q sabrelite
by Ankit Agarwal 08 Sep '13

08 Sep '13

I am using alsa pcm player to listen to audio. The alsa pcm crashes with the error: ALSA sound/usb/urb.c:179 cannot submit urb (err = -27). I am able to hear the data initially but when other transaction begins on HID report endpoint, alsa crashes with the mentioned error. I am working on imx6q board with linux kernel-3.0.35. I am using Apple device to get the digital audio. -- Best Regards, Ankit Agarwal

2 1

[alsa-devel] US-1641 drivers involvment
by Luca Novarini 08 Sep '13

08 Sep '13

Hello, I would participate, if possible, in the ALSA development to add support to the USB card Tascam US-1641. Any reply or help from who tried in this task before is really really appreciated. Thank you! Bye. LL

3 4

[alsa-devel] Request for information
by Michael Jordan 07 Sep '13

07 Sep '13

Hello, I am developing a driver for the PCM3168A. We have created an expansion board with the codec, op-amp buffers, etc. which is designed to connect to the audio expansion connector on an OMAPL138 EXPKIT from LogicPD. I have the rough draft of the pcm3168a.c and pcm3168a.h. Looking through the code, I see that the DA850 is "hard-wired" in the code to expect the TLV320AIC3106, which is the on-board codec on the kit. My question is, would it be proper for me to create entries in davinci-evm.c, mach-types.h, etc. in order to provide a way to register this new driver? Or is there a different way I should proceed? I'll be modifying my code base for testing purposes, but when it comes time for submission, I would like to know the proper way to handle this. Thank you, Michael

1 0

[alsa-devel] [PATCH v2 2/2] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by mengdong.lin＠intel.com 06 Sep '13

06 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_D0(), since the convertor power state is also fixed here. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 974b55c..3d8cd044 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -896,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1035,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1054,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1090,7 +1076,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (is_haswell(codec)) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + haswell_verify_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1361,13 +1347,9 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * changed during the stream playback */ if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock speaker support for ASUS TX300
by Takashi Iwai 06 Sep '13

06 Sep '13

ASUS TX300 has a built-in speaker in the tablet part and in the dock part, and the tablet speaker is supposed to be unused while the machine is docked. The current HD-audio driver, however, doesn't support the dock speaker, partly because BIOS doesn't set up the pin for the corresponding output. But, not only the missing pin config, also the missing unsol event handling is another issue. Otherwise the automatic switching via dock/undock won't work. Through debugging sessions, we found out that the dock speaker pin is NID 0x1b, and it generates an unsol event at docking/undocking, the docking state can be inquired via the normal pin detection verb. Also, it's turned out that GPIO 2 is needed as an amp. So, all materials are ready to cook. This patch provides the basic dock speaker support with TX300: - The dock speaker is turned on/off via "Dock Speaker" mixer mute. - The dock speaker is automatically muted when docked. This is independently from the mixer mute switch, just like the headphone auto-mute function. The implementation is a bit tricky. Since we want to handle it as a secondary speaker, we set it up a pin as a speaker with a jack detection. Then, the fixup function registers the own unsol callback for this pin because the standard automute can't handle the thing like a "speaker jack". In the own automute hook, we apply the mute of the tablet speaker in addition by checking the dock state. Also, the speaker control names are slightly shuffled because the generic parser doesn't give good names but blindly assumes a bass speaker as a secondary speaker. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59791 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 56 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4a90917..d38d6a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3443,6 +3443,56 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, } } +/* mute tablet speaker pin (0x14) via dock plugging in addition */ +static void asus_tx300_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); + if (snd_hda_jack_detect(codec, 0x1b)) + spec->gen.mute_bits |= (1ULL << 0x14); +} + +static void alc282_fixup_asus_tx300(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + /* TX300 needs to set up GPIO2 for the speaker amp */ + static const struct hda_verb gpio2_verbs[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, + {} + }; + static const struct hda_pintbl dock_pins[] = { + { 0x1b, 0x21114000 }, /* dock speaker pin */ + {} + }; + struct snd_kcontrol *kctl; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, gpio2_verbs); + snd_hda_apply_pincfgs(codec, dock_pins); + spec->gen.auto_mute_via_amp = 1; + spec->gen.automute_hook = asus_tx300_automute; + snd_hda_jack_detect_enable_callback(codec, 0x1b, + HDA_GEN_HP_EVENT, + snd_hda_gen_hp_automute); + break; + case HDA_FIXUP_ACT_BUILD: + /* this is a bit tricky; give more sane names for the main + * (tablet) speaker and the dock speaker, respectively + */ + kctl = snd_hda_find_mixer_ctl(codec, "Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Dock Speaker Playback Switch"); + kctl = snd_hda_find_mixer_ctl(codec, "Bass Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Speaker Playback Switch"); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3480,6 +3530,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, + ALC282_FIXUP_ASUS_TX300, }; static const struct hda_fixup alc269_fixups[] = { @@ -3735,6 +3786,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc283_fixup_chromebook, }, + [ALC282_FIXUP_ASUS_TX300] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc282_fixup_asus_tx300, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3784,6 +3839,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), -- 1.8.4

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[alsa-devel] [PATCH] ASoC: core: Remove dependencies on GENERIC_HARDIRQS
by Mark Brown 06 Sep '13

06 Sep '13

From: Mark Brown <broonie(a)linaro.org> All architectures now use the generic IRQ subsystem so this is redundant. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15106c0..b33b45d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -107,7 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C && GENERIC_HARDIRQS + select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1b0cccd..d446c6b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver
by Thomas Petazzoni 06 Sep '13

06 Sep '13

The compatible string of the kirkwood-i2s driver was chosen as "marvell,mvebu-audio". Using such a compatible string is not a good idea, since "mvebu" is the name of a large family of SOCs, in which new, unknown SOCs will be coming in the future. It is therefore impossible to know what will be evolutions of this hardware block in the next generations of the SOCs. For this reason, the recommandation for compatible strings of on-SOCs devices has always been to use the name of the oldest SOC that has the hardware block. New SOCs that have an exactly compatible hardware block can reference it using the same compatible string. See [1], [2] and [3] for various cases were this suggestion was made, including from Rob Herring, a Device Tree binding maintainer. As an example, there are already small differences between current generations: * On Kirkwood, only one interrupt is used for audio. * On Dove, two interrupts are used, one for audio data and one for error reporting. In the near future, I'll be adding audio support to Armada 370, which allows has the same hardware block (but maybe with minor variants). Therefore, this patch changes the driver to accept "marvell,kirkwood-audio" and "marvell,dove-audio" as compatible strings instead of the too-generic "marvell,mvebu-audio". The reason for the two different compatible strings is the difference in the number of interrupts used by the two SOCs for audio. This Device Tree binding has never been part of a Linux kernel stable release so far, so it can be changed now without breaking backward compatibility. [1] http://lists.infradead.org/pipermail/linux-mtd/2012-March/040417.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/2013-April/161065.html [3] http://lists.infradead.org/pipermail/linux-arm-kernel/2012-March/087702.html Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/mvebu-audio.txt | 12 ++++++++---- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 2 files changed, 10 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index 7e5fd37..f0062c5 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -2,13 +2,17 @@ Required properties: -- compatible: "marvell,mvebu-audio" +- compatible: + "marvell,kirkwood-audio" for Kirkwood platforms + "marvell,dove-audio" for Dove platforms - reg: physical base address of the controller and length of memory mapped region. -- interrupts: list of two irq numbers. - The first irq is used for data flow and the second one is used for errors. +- interrupts: + with "marvell,kirkwood-audio", the audio interrupt + with "marvell,dove-audio", a list of two interrupts, the first for + the data flow, and the second for errors. - clocks: one or two phandles. The first one is mandatory and defines the internal clock. @@ -21,7 +25,7 @@ Required properties: Example: i2s1: audio-controller@b4000 { - compatible = "marvell,mvebu-audio"; + compatible = "marvell,dove-audio"; reg = <0xb4000 0x2210>; interrupts = <21>, <22>; clocks = <&gate_clk 13>; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 7fce340..0f3d73d 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,7 +559,8 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) #ifdef CONFIG_OF static struct of_device_id mvebu_audio_of_match[] = { - { .compatible = "marvell,mvebu-audio" }, + { .compatible = "marvell,kirkwood-audio" }, + { .compatible = "marvell,dove-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); -- 1.8.1.2

3 5

Re: [alsa-devel] [PATCH 4/5 v2] ARM: shmobile: bockw: add R-Car sound support (PIO)
by Simon Horman 06 Sep '13

06 Sep '13

[ CC Mark Brown, alsa-devel ] On Thu, Sep 05, 2013 at 02:57:51PM +0900, Simon Horman wrote: > On Mon, Aug 26, 2013 at 01:52:35AM -0700, Kuninori Morimoto wrote: > > This patch enables R-Car sound, > > AK4643 (CN19) and AK4554 (CN20/CN21) codec chip > > on Bock-W. > > > > But, it supports PIO transfer only at this point. > > User can check sound settings (Dip-switch/PFC etc) > > via this patch, but will get under/over flow error > > when playback/capture. > > Because PIO transfer via SSI will be interrupted > > "sampling rate" times per 1 second. > > > > DMA transfer will be supported when HPB-DMAC was > > enabled on r8a7778. > > > > You will notice strange ALSA sound card HW > > numbering on Bock-W board. > > This came from AK4554 strange format on playback/capture. > > The format on playback/capture is same on "normal" codec chip, > > but AK4554 was different. > > Because of that, AK4554 playback/capture are > > registered as a different sound card. > > > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > --- > > v1 -> v2 > > Thanks, I have queued this up for v3.13. > > There was a bit of diff-noise, please make sure I have applied it > correctly. The patch I applied is below: Unfortunately I didn't read your cover email carefully before queuing this up. I now see that it has dependencies that aren't present in v3.11-rc and thus I have temporarily dropped it. It seems to me that the dependencies could be provided by the topic/car branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. Morimoto-san, is this correct? If so, Mark, would it be possible for you to provide topic/car or similar as stable branch for me to use as a base? My intention is to queue these up for v3.13.

3 4

[alsa-devel] Roland Sound Canvas SD-50
by Keith A. Milner 05 Sep '13

05 Sep '13

I saw a recent discussion on Alsa-users about the Roland SD-50 which seemed to tail off inconclusively. Does anyone know if this was found to work or not? I ask because I have the option of picking one up (relatively) cheaply. It looks like an interesting little box. But if there isn't a working ALSA solution yet, I might be tempted to pick it up anyway and try to get it working, although I may need some help with that. Cheers, Keith

2 2

[alsa-devel] Documentation Request
by Carl Canuck 05 Sep '13

05 Sep '13

Hello, I've recently begun the work of porting a Windows application to LInux/ALSA. I think ALSA and it's drivers are quite good, but the documentation and example code are terribly lacking, especially for one of the most important use-cases of all: the full on digital audio workstation. I humbly request that a full_duplex_audio_with_midi.c file be added to the example code with the following parameters: 1. Full duplex audio, just simply memcpy'ing the input to the output would suffice 2. Accepts MIDI input; it doesn't have to do anything useful with the data, just printing the event parameters to stdout would suffice. 3. At run time, the application lists all available audio devices and allows the user to select the device, then make settings such as buffer size, sample rate, bit depth and whether to use mmap based on the capabilities of the chosen device 4. At run time, allows optionally selecting a MIDI device. 5. Copious code comments that makes no assumption that the person reading knows anything about the ALSA API or it's constructs, invocations and methodologies. This could either be an interactive CLI application written in C, or even a full GUI application using Qt or GTK. This could then be used as a shell for writing more complex applications, instead of each developer having to reinvent the wheel by trying to figure out how to combine about 4 or so of the other example applications into a single app. /Carl

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[alsa-devel] Constraints and hw_params with highly configurable HW
by Uli Franke 05 Sep '13

05 Sep '13

Hi everybody We're currently working on some neat enhancements for the still experimental snd_dice driver originally initiated by Clemens Ladisch. To list some of the new and already working features: * firmware loading, backed by a user-space command line utility * generic ALSA control support for mostly all device specific settings for all our (i.e. Weiss Engineering) DICE based products and last but not least * capture support * slave mode (external clock source support). I'm currently debugging and fine-tuning things and that's where some questions arose. The initial implementation of Clemens tried to fetch all possible configurations of a device during probe in order to have all combinations of rates and channels at hand when it comes to formulate the constraints set within the ALSA pcm substream open callback. That's a nice approach but will fail with a lot of devices as their configurations can (and mostly will) be volatile. Our devices for instance feature several switches, options and alike which have heavy influence on the channel layout and sample rates and combination of both. I consulted Takashi's documentation on constraints which suggests to "use the source" but I couldn't find any similar situation within other drivers (in case I should have missed something, just let me know of course). Therefore I had the idea to define some maximum constraints and let the ALSA hw_params callback fail in case some settings or combinations are determined to be not supported during its invocation. Is this viable? Could this be/is this consistent with the specs? If yes, would EINVAL an appropriate error code to return? Regards Uli

2 3

[alsa-devel] Recall: [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by Lin, Mengdong 05 Sep '13

05 Sep '13

Lin, Mengdong would like to recall the message, "[PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell".

1 0

[alsa-devel] [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by mengdong.lin＠intel.com 05 Sep '13

05 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_pin_cvt_D0(), since the convertor power state is also fixed here. - define is_haswell() to replace checking Haswell vendor ID everywhere. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..471c158 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -894,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1033,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_pin_cvt_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1052,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1087,8 +1075,8 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + if (is_haswell(codec)) + haswell_verify_pin_cvt_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1227,7 +1215,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,14 +1346,10 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + if (is_haswell(codec) && + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); @@ -1405,7 +1389,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +1998,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2009,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

4 3

[alsa-devel] [PATCH v2 1/2] ALSA: hda - define is_haswell() to check if a display audio codec is Haswell
by mengdong.lin＠intel.com 04 Sep '13

04 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> To apply Haswell specific fixings, this patch defines is_haswell() to check whether a display audio codec is Haswell, to avoid explicitly checking Haswell vendor ID everywhere. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..974b55c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1087,7 +1089,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { @@ -1227,7 +1229,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,7 +1360,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && + if (is_haswell(codec) && eld->eld_valid && !old_eld_valid && per_pin->setup) { snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1405,7 +1407,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +2016,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2027,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: core: Implement devm_snd_soc_register_component()
by Mark Brown 04 Sep '13

04 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 3 +++ sound/soc/Makefile | 2 +- sound/soc/soc-devres.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 56 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-devres.c diff --git a/include/sound/soc.h b/include/sound/soc.h index a72af63..f126676 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -381,6 +381,9 @@ void snd_soc_unregister_codec(struct device *dev); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d2..8b9e701 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c new file mode 100644 index 0000000..13fe86f --- /dev/null +++ b/sound/soc/soc-devres.c @@ -0,0 +1,52 @@ +/* + * soc-devres.c -- ALSA SoC Audio Layer devres functions + * + * Copyright (C) 2013 Linaro Ltd + * + * 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 <linux/moduleparam.h> +#include <sound/soc.h> + +static void devm_component_release(struct device *dev, void *res) +{ + snd_soc_unregister_component(*(struct device **)res); +} + +/** + * devm_snd_soc_register_component - resource managed component registration + * @dev: Device used to manage component + * @cmpnt_drv: Component driver + * @dai_drv: DAI driver + * @num_dai: Number of DAIs to register + * + * Register a component with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 00/14] SPDIF support
by Russell King - ARM Linux 04 Sep '13

04 Sep '13

This is my latest set of patches to get SPDIF working on the Dove Cubox platform. As per the last time, I've left out the "card" part of this, because that is only useful for non-DT setups, which aren't supported in mainline. The first seven patches have already been applied by Mark, and are included here for completeness by anyone who wishes to test this series. There is very little change between this and the previous set of patches. The HACK patch is gone, as I've found out through much digging and experimentation with DAPM (inspite of the complete lack of assistance from ASoC developers) how to use the stream widgets without the widget overwriting bug getting in the way. The CPU DAI widgets are attached to the CPU DAI context, and their access to the driver private data updated accordingly. Why attaching them to the CPU DAI context previously didn't work, I'm not entirely sure, but that certainly previously failed in testing (it may have been due to the duplicated DAI routes and/or the lack of output pin on the SPDIF tranceiver.) I've added three patches: one to add IEC958 channel status setting, another to prefer the external clock over the internal clock, and removal of the IEC958 subframe support (see the commit message for that patch why this can not be supported.) This is a non-DPCM driver, and is the absolute minimum solution which can provide correct hardware behaviour while still being conformant with single zImage kernels and not causing any known regressions. The same conditions apply as per my previous posting - the DAI link needs to be setup and the associated DAPM routes to tell the CPU DAI which outputs are in use, like this: DAI link: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, }; When DPCM is eventually implemented, the DAPM route remains, the DAI link becomes two, as previously illustrated in other discussions. Full diffstat for this series: arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 ++--- sound/soc/codecs/spdif_transciever.c | 15 +++- sound/soc/kirkwood/Kconfig | 5 - sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 114 ++++-------------- sound/soc/kirkwood/kirkwood-i2s.c | 213 +++++++++++++++++++++++++++++----- sound/soc/kirkwood/kirkwood-openrd.c | 12 ++- sound/soc/kirkwood/kirkwood-t5325.c | 8 +- sound/soc/kirkwood/kirkwood.h | 45 +++++++- sound/soc/soc-dapm.c | 2 +- 11 files changed, 295 insertions(+), 151 deletions(-) Diffstat for the patches remaining for mainline: sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 170 +++++++++++++++++++++++++++++++--- sound/soc/kirkwood/kirkwood-openrd.c | 8 ++ sound/soc/kirkwood/kirkwood-t5325.c | 4 + sound/soc/kirkwood/kirkwood.h | 34 +++++++ 5 files changed, 205 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: generic simple sound card DT bindings
by Markus Pargmann 04 Sep '13

04 Sep '13

Hi, The patches were already in the fsl-ssi ac97 series. I moved the seperated simple sound card DT driver into the existing simple card driver. Regards, Markus Markus Pargmann (3): ASoC: generic: simple card, use private data ASoC: generic simple sound card DT bindings ASoC: fsl: Kconfig, visible config items for simple card .../bindings/sound/generic,simple-dt-card.txt | 28 +++ include/sound/simple_card.h | 4 - sound/soc/fsl/Kconfig | 23 ++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card.c | 225 ++++++++++++++++++--- 5 files changed, 246 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt

4 8

[alsa-devel] [PATCH] ASoC: atmel: disable error interrupt
by Bo Shen 04 Sep '13

04 Sep '13

As once the error interrupt is triggered, it can not be cleared. So, disable it. No side effect found while testing on sama5d3xek and at91sam9x5ek boards. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0ecf356..bb53dea 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -649,7 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); - ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error); + ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", -- 1.7.9.5

2 1

Re: [alsa-devel] Documentation Request
by Carl Canuck 04 Sep '13

04 Sep '13

Sorry, I only sent that last reply to Grant, the reply-to-anybody-but-sender options are a little counter-intuitive in Gmail. On Tue, Sep 3, 2013 at 11:10 PM, Carl Canuck <carl.canuck.official(a)gmail.com > wrote: > Hi Grant, > > I have in fact spent a little time with Jack and even conjured up a > working prototype using Jack audio with ALSA MIDI (that will never see the > light of day). However, you're ignoring the fact that: > > 1. Some people don't like having 20 separate audio applications running > and having to save all of their states individually > 2. If you believe that any existing Jack session manager is a stable, > reliable application that adequately solves problem #1, then it's obvious > that you've not spent much time with Jack session managers > 3. Jack is something like 15 years old now, and they still haven't even > managed to solve #1 and #2 yet, among other long-standing issues? > 4. With points #1, #2 and #3 all being common knowledge, why should I put > my faith in Jack competently wrapping ALSA for me instead of taking matters > into my own hands and going straight to ALSA? > 5. If you visit Windows/Mac audio forums, there are legions of people who > will tell you they tried Linux once, found Jack to be buggy and terrible in > general, never intend to try Linux again. In fact, you can register a user > name on any non-Linux music forum, and ask "i'm thinking of trying Linux > for audio, thoughts?", and people will say exactly the things I described > to you, not because they have any unfounded bias against Linux or Jack, but > because that was truly their experience. > 6. Jack's configuration settings and error logging are atrocious, it is > far too often excessively difficult to understand why Jack won't start and > how to fix it. > 7. Is it really so unreasonable to ask for a basic example of a common > use-case for ALSA without being told that I should just abandon the quest > for that knowledge and put my faith in the Jack developers? Should > deciphering the intended use of any API be an exercise left to the would-be > developer? > > If you like Jack, then by all means use it, but it's not ideal for my > use-case, and I won't be spending any more time evaluating it until the > Jack developers address these long-standing issues. > > Carl > > <http://users.suse.com/~mana/alsa090_howto.html> Also, I have in fact seen the link you provided Grant, but that once again is only bits and pieces of the use-case. I'd like to know that I'm using ALSA as it's creators intended, rather than stringing together bits and pieces and hoping that I'm doing it right. Regards, Carl

2 1

[alsa-devel] [PATCH] ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
by Takashi Iwai 04 Sep '13

04 Sep '13

When the transcoder:port mapping on Haswell HDMI/DP audio is changed during the stream playback, the sound gets lost. Typically this problem is seen when the user switches the graphics mode from eDP+DP to DP-only configuration, where CRTC 1 is used for DP in the former while CRTC 0 is used for the latter. The graphics controller notifies the change via the normal ELD update procedure, so we get the intrinsic event. For enabling the sound again, the HDMI audio driver needs to reset the pin and set up the audio infoframe again. This patch achieves it by: - keep the current status of channels and info frame setup in per_pin struct, - check the reconnection in the intrinsic event handler, - reset the pin and the re-invoke hdmi_setup_audio_infoframe() accordingly. The hdmi_setup_audio_infoframe() function has been changed, too, so that it can be invoked without passing the substream instance. The patch is mostly based on the work by Mengdong Lin. Cc: Mengdong Lin <mengdong.lin(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 41 +++++++++++++++++++++++++++++++---------- 1 file changed, 31 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b83b14f..22b5089 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; + bool setup; /* the stream has been set up by prepare callback */ + int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ @@ -879,18 +881,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - bool non_pcm, - struct snd_pcm_substream *substream) +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool non_pcm) { - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; - int channels = substream->runtime->channels; + int channels = per_pin->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; + if (!channels) + return; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1341,6 +1344,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld_changed = true; } if (update_eld) { + bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; eld_changed = pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, @@ -1350,6 +1354,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_size); pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + + /* Haswell-specific workaround: re-setup when the transcoder is + * changed during the stream playback + */ + if (codec->vendor_id == 0x80862807 && + eld->eld_valid && !old_eld_valid && per_pin->setup) { + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); + } } mutex_unlock(&pin_eld->lock); @@ -1522,14 +1538,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); - hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + per_pin->channels = substream->runtime->channels; + per_pin->setup = true; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); + hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1569,6 +1588,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); + + per_pin->setup = false; + per_pin->channels = 0; } return 0; @@ -1704,8 +1726,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) - hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, - substream); + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); return 0; } -- 1.8.3.4

3 7

[alsa-devel] [PATCH v2] ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel
by Fabio Estevam 04 Sep '13

04 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) This is caused by commit 8548a464b9 (ASoC: imx-audmux: Read default configuration from devicetree). In order to fix this, add a check for 'of_id' so that imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Only call imx_audmux_parse_dt_defaults for dt kernel. sound/soc/fsl/imx-audmux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..d3bf71a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Check if we are running a dt kernel
by Fabio Estevam 03 Sep '13

03 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) In order to fix it, return an error when 'of_id' is NULL, which happens for the non-dt case. Also checked that audio is still functional when running the dt kernel for this board after applying this patch. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..2b29b48 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -317,6 +317,9 @@ static int imx_audmux_probe(struct platform_device *pdev) const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); + if (!of_id) + return -EINVAL; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audmux_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(audmux_base)) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: mxs: mxs-sgtl5000: Simplify probe function
by Fabio Estevam 03 Sep '13

03 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> mxs is a device tree only platform, which allows us to simplify a bit mxs_sgtl5000_probe(), because there is no need to check whether device tree is supported or not. Remove mxs_sgtl5000_probe_dt() and place its content inside mxs_sgtl5000_probe() for making the code simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4bb2737..61822cc 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -122,14 +122,12 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) +static int mxs_sgtl5000_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &mxs_sgtl5000; + int ret, i; struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i; - - if (!np) - return 1; /* no device tree */ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); @@ -152,18 +150,6 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return 0; -} - -static int mxs_sgtl5000_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &mxs_sgtl5000; - int ret; - - ret = mxs_sgtl5000_probe_dt(pdev); - if (ret < 0) - return ret; - /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range -- 1.8.1.2

2 1

[alsa-devel] ASoC updates for v3.12
by Mark Brown 02 Sep '13

02 Sep '13

The following changes since commit 3fd8a8041b3775840c8f85b6338b61beb5057160: Merge tag 'asoc-v3.12-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next (2013-08-30 12:35:49 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-3 for you to fetch changes up to 5df498a27ab385294fdb7cad19c623d4da94a780: Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp (2013-09-01 21:15:59 +0100) These haven't all been in a -next build yet, they work for me though. ---------------------------------------------------------------- ASoC: Final updates for v3.12 A few final updates for v3.12 - some cleanups, a bug fix for ssm2602, pop removal for rt5640 and fixes for the reporting of unidirectional links in the MXS SGTL5000 driver. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5640: change widget sequence for depop Fabio Estevam (2): ASoC: soc-pcm: Allow to specify unidirectional dai_link ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional Lars-Peter Clausen (5): ASoC: dapm: Fix auto-disable for inverted controls ASoC: Remove unused control_type field from snd_soc_codec struct ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct ASoC: Remove unused sysfs_registered field from snd_soc_codec struct ASoC: ssm2602: Fix cache sync Mark Brown (6): Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-dapm ASoC: fsl_spdif: Staticse non-exported symbols Merge remote-tracking branch 'asoc/topic/core' into tmp Merge remote-tracking branch 'asoc/topic/dapm' into tmp Merge remote-tracking branch 'asoc/topic/fsl' into tmp Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp Nicolin Chen (2): ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c ASoC: fsl: Add one blank space after ':=' in Makefile include/sound/soc.h | 8 +- sound/soc/codecs/rt5640.c | 217 ++++++++++++++++++++++++++++++++----------- sound/soc/codecs/rt5640.h | 12 +++ sound/soc/codecs/ssm2602.c | 3 +- sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_spdif.c | 12 +-- sound/soc/mxs/mxs-sgtl5000.c | 2 + sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 10 ++ 9 files changed, 202 insertions(+), 66 deletions(-)

2 1

[alsa-devel] [PATCH 1/5] ASoC: Remove support for reg_access_defaults
by Lars-Peter Clausen 02 Sep '13

02 Sep '13

No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 24 -------------------- sound/soc/soc-cache.c | 63 --------------------------------------------------- sound/soc/soc-core.c | 9 -------- 3 files changed, 96 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7184805..ae6efc6d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -542,22 +536,6 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - -/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f554..eaa898f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -275,66 +275,3 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7d13c4f..b78fa93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4222,15 +4222,6 @@ int snd_soc_register_codec(struct device *dev, } } - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- 1.8.0

2 5

[alsa-devel] [PATCH 0/5] Remove ASoC-level I/O functions from cq93vc
by Mark Brown 02 Sep '13

02 Sep '13

This patch series removes the use of the ASoC-level read and write functions from the cq93vc driver as part of a wider push to remove them completely and just use regmap for all register I/O. Since the driver is essentially doing what regmap-mmio is doing this is done by adding a MMIO regmap to the core device and using that. This is compile tested only, I don't have any hardware to run on. Since I anticipate doing more cleanups over the CODEC drivers during the next release cycle it'd be good to merge via ASoC, though there shouldn't be any overlap with the first patch. Mark Brown (5): mfd: davinci_voicecodec: Remove unused read and write functions mfd: davinci_voicecodec: Provide a regmap for register I/O ASoC: cq93vc: Use core I/O functions ASoC: cq93vc: Don't use control data for core driver data ASoC: cq93vc: Use regmap for I/O drivers/mfd/davinci_voicecodec.c | 23 +++++++++++-------- include/linux/mfd/davinci_voicecodec.h | 3 +++ sound/soc/codecs/cq93vc.c | 42 +++++++++++----------------------- 3 files changed, 29 insertions(+), 39 deletions(-)

2 9

[alsa-devel] [PATCH 00/18] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 02 Sep '13

02 Sep '13

This is first non-RFC 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. Due to changes scattered across different areas of kernel, patches are based on merged 3 branches: - for-next of Kgene's Samsung tree, - clk-next of Mike's clock tree, - next of Vinod's slave DMA tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git v3.12-pl080 Dependencies (already applied in my branch): - for patches 14 and 16 - CCF-based clock driver for s3c64xx. Some of the patches not related to the amba-pl08x driver itself can be likely applied into appropriate trees separately, namely: - 09/18 - ASoC: Samsung: Do not queue cyclic buffers multiple times, - 14/18 - clk: samsung: s3c64xx: Add aliases for DMA clocks. After patch 14/18, both old and new DMA drivers can be supported on S3C64xx, depending on Kconfig options. Patches 15-18 remove the old driver leaving support only for the generic pl08x driver. Feel free to drop those patches for now if we want more testing, but I don't suspect any problems. On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. 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 (16): 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: Move LLI dumping code into separate function 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 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 ARM: s3c64xx: clock: Add clkdev lookup for DMA clocks clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/clock.c | 28 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 753 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 6 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/dma/amba-pl08x.c | 501 ++++++++++++++------ drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 16 files changed, 705 insertions(+), 1015 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

5 44

[alsa-devel] [PATCH 1/2] ASoC: soc-pcm: Allow to specify unidirectional dai_link
by Fabio Estevam 01 Sep '13

01 Sep '13

Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 10 ++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..774d3be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -934,6 +934,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb70fbe..5e44863 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- 1.8.1.2

2 2

[alsa-devel] [PATCH v3 00/11] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 01 Sep '13

01 Sep '13

Hi, This patch series attempts to fix a long standing problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD 68538bf2bce557c3b5fe8c59b034d45352500db1 ). v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065649.html v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html * Changed "altset_idx" to "altsetting" thanks to Clemens' clarification. #3: Thanks to Daniel's test, fixed a NULL dereference caused by userspace calling snd_pcm_open() and then snd_pcm_close() immediately. #8: subs->altset_idx was left intact, since it corresponds to the array index. #10: this patch now removed subs->altset_idx, but also changes the check to be performed against fmt->altsetting. #11: New. This changes the name of snd_usb_endpoint member alt_idx to altsetting for the same reason Clemens described. Clemens and Daniel - thank you very much for your help! Cheers, Eldad Eldad Zack (11): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream ALSA: usb-audio: rename alt_idx to altsetting sound/usb/card.h | 4 +- sound/usb/endpoint.c | 82 ++++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 168 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 201 insertions(+), 70 deletions(-) -- 1.8.1.5

2 14

Re: [alsa-devel] [PATCH 2/2] ARM: Dove: Add the audio device to the Cubox DT
by Jean-Francois Moine 01 Sep '13

01 Sep '13

On Sat, 31 Aug 2013 12:24:31 +0100 Russell King - ARM Linux <linux(a)arm.linux.org.uk> wrote: > > According to the Dove specification, the audio controller works with > > the samples rates 44.1, 48 and 96 kHz, so, I don't see the usage of the > > external clock, except when using the two audio controllers with > > different sample rates. > > I don't see what the Dove specification has to do with that statement: > what the Dove spec says is that if you use just the internal DCO, then > only 44.1kHz, 48kHz and 96kHz are supported (with some trimming of that.) > However, the use of an external clock allows further rates to be supported. > If you have an external clock, there is no requirement to use the DCO for > those sample rates - you can if you wish, or you can use the external clock. > > The mainline driver implements the use of the DCO for the standard 44.1, > 48 and 96kHz rates, otherwise it uses the external clock if present. This > is entirely conformant with the Dove spec. Russell, I looked again at the Armada-510-Functional-Spec, and I found: - section 12.1 ... I2S Supported Features An audio sample rate (Fs) of 44.1/48/96 kHz. ... S/PDIF Supported Features The IEC60958-1, IEC60958-3, and IEC61937 specifications. An audio sample rate (Fs) of 44.1/48/96 kHz. ... - Table 436: Clocks Control Register Offset: Unit0: 0x000B1230, Unit1: 0x000B5230 ... 1:0 MCLK source RW MCLK = 256 Fs, where Fs = 44.1 kHz, 48 kHz, 96 kHz. 0x0 Playback should not be active when selecting an external MCLK, or moving from one external MCLK to another MCLK source. 0 = DCO: MCLK is from DCO. 1 = Reserved 2 = Reserved 3 = External: MCLK is from an external source. As they don't work with my TV set, may you confirm that these webradios http://kerne.online.stalig.net/live-ori.ogg http://www.tv-radio.com/station/france_bleu_breiz_izel_mp3/france_bleu_brei… http://plumfm.online.stalig.net/live-ori.ogg work for you with the external clock? (sample rates resp. 33.075, 24 and 22.05 kHz) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH] ASoC: fsl_spdif: Staticse non-exported symbols
by Mark Brown 01 Sep '13

01 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..c5c228a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return 0; } -int fsl_spdif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); @@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, @@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai_driver fsl_spdif_dai = { +static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { .channels_min = 2, -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ASoC: ssm2602: Fix cache sync
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The ssm2602 uses regmap for caching not soc-cache, so we need to use regcache_sync() instead of snd_soc_cache_sync(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5..492644e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec) static int ssm2602_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + regcache_sync(ssm2602->regmap); ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: cq93vc: Use table based control registration
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Saves a little code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cq93vc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 3ce535b..43737a27 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -142,10 +142,6 @@ static int cq93vc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -164,6 +160,8 @@ static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8400: Use supplies to manage input power
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Rather than using a fake register to manage input power create some supply widgets and use those. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8400.c | 73 ++++++++++++----------------------------------- 1 file changed, 18 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index d2a0928..95c33d1 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -32,13 +32,6 @@ #include "wm8400.h" -/* Fake register for internal state */ -#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) -#define WM8400_INMIXL_PWR 0 -#define WM8400_AINLMUX_PWR 1 -#define WM8400_INMIXR_PWR 2 -#define WM8400_AINRMUX_PWR 3 - static struct regulator_bulk_data power[] = { { .supply = "I2S1VDD", @@ -79,10 +72,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) - return wm8400->fake_register; - else - return wm8400_reg_read(wm8400->wm8400, reg); + return wm8400_reg_read(wm8400->wm8400, reg); } /* @@ -93,11 +83,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) { - wm8400->fake_register = value; - return 0; - } else - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); } static void wm8400_codec_reset(struct snd_soc_codec *codec) @@ -352,32 +338,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event (struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); - - if (fakepower & ((1 << WM8400_INMIXL_PWR) | - (1 << WM8400_AINLMUX_PWR))) { - reg |= WM8400_AINL_ENA; - } else { - reg &= ~WM8400_AINL_ENA; - } - - if (fakepower & ((1 << WM8400_INMIXR_PWR) | - (1 << WM8400_AINRMUX_PWR))) { - reg |= WM8400_AINR_ENA; - } else { - reg &= ~WM8400_AINR_ENA; - } - snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { @@ -658,27 +618,26 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, 0, &wm8400_dapm_rin34_pga_controls[0], ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), +SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT, + 0, NULL, 0), + /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, - &wm8400_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, - &wm8400_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -789,11 +748,13 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { {"LIN34 PGA", "LIN3 Switch", "LIN3"}, {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ + {"INMIXL", NULL, "INL"}, {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, /* AILNMUX */ + {"AILNMUX", NULL, "INL"}, {"AILNMUX", "INMIXL Mix", "INMIXL"}, {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, @@ -808,12 +769,14 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, - /* INMIXL */ + /* INMIXR */ + {"INMIXR", NULL, "INR"}, {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, /* AIRNMUX */ + {"AIRNMUX", NULL, "INR"}, {"AIRNMUX", "INMIXR Mix", "INMIXR"}, {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 1/3] ASoC: Remove unused control_type field from snd_soc_codec struct
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The control_type field was used by the core to track which raw IO methods to use, but when switching to regmap this was no longer necessary and so the last user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map API for ASoC generic physical I/O"). The field is now completely unused and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..9d4c2ab 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -705,7 +705,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.8.0

2 3

[alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 Aug '13

31 Aug '13

Hi, this series implements ac97-slave support for fsl-ssi and adds a simple generic DT sound card. Regards, Markus Changes in v11: - Dropped phycard specific sound patches - Added a generic simple DT sound card with support for ac97 using ac97-codec - Some new bugfix patches Markus Pargmann (5): ASoC: imx-pcm-dma: Add missing include ASoC: core: Generic ac97 link reset functions ASoC: generic: simple DT sound card ASoC: fsl: Kconfig, visible config items for simple card ASoC: fsl-ssi: ac97-slave support Michael Grzeschik (2): ASoC: fsl-ssi: imx-pcm-fiq bugfix ASoC: fsl: disable ssi irq for imx Steffen Trumtrar (1): ASoC: fsl-ssi: add SSIEN errata work around .../devicetree/bindings/sound/fsl,ssi.txt | 4 + .../bindings/sound/generic,simple-dt-card.txt | 28 ++ .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++ include/sound/soc.h | 2 + sound/soc/fsl/Kconfig | 23 +- sound/soc/fsl/fsl_ssi.c | 369 ++++++++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 1 + sound/soc/generic/Kconfig | 1 + sound/soc/generic/Makefile | 3 +- sound/soc/generic/simple-dt-card.c | 186 +++++++++++ sound/soc/soc-core.c | 153 +++++++++ 11 files changed, 711 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 sound/soc/generic/simple-dt-card.c

2 16

[alsa-devel] [PATCH 1/2] ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
by Nicolin Chen 31 Aug '13

31 Aug '13

Add wrapping '\n' for dev_dbg() in fsl_spdif.c Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..98741e9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1071,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, break; } - dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate", + dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", spdif_priv->txclk_src[index], rate[index]); - dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate", + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n", spdif_priv->txclk_div[index], rate[index]); return 0; -- 1.7.1

2 2

[alsa-devel] [PATCH RFC 00/13] Adding SPDIF support to kirkwood-i2s
by Russell King - ARM Linux 30 Aug '13

30 Aug '13

This is only a RFC at present due to the need to test on Kirkwood and fix at least one bug in ASoC core code. I can't say that this stuff is fully tested, because I don't have any Kirkwood platforms, but I do have one Dove platform, which as I've said in the past doesn't run mainline. I don't run DT on it, so these patches also don't contain anything DT related. However, this patch set is aimed at sorting out the SPDIF on Dove and Kirkwood, which the Dove-based Cubox uses. The first patch in the series I've already sent, and I think Mark may have already picked up. The second is my implementation of what I think Mark committed to use the devm_* APIs for the external clock. Patches 3 and 4 fix problems which I've found with the ASoC core code, which prevent these patches from working correctly. Patch 3 I've already discussed with Mark over the weekend, and I think that's pretty much settled, but patch 4 is a definite hack. Patch 4 addresses an issue where we end up registering the CPU DAIs stream widgets twice - once against the codec DAPM context and again against the platform DAPM context. When these are registered, pointers to these widgets are saved into the CPU DAI structure, so one set of these widgets gets overwritten by the other set. The set which wins (normally the ones in the CPU DAPM context) ends up being the set which are made active. However, the platform ones remain on the DAI list, and these can end up being bound to by routes. ASoC people need to fix this problem as a pre-requisit to the following patches working. Patches 5 and 6 are just cleanups - I decided it was easier to do my own version of merging the I2S and DMA files rather than using Jean-F's patch (sorry, I just wanted something that worked...) Patch 7 is based on some changes in Rabeeh's kernel - I will have a following patch at some point to adjust these parameters to something a little saner. This just gets mainline closer to the tree which I can test. Patch 8 adds the I2S widgets to kirkwood-i2s and uses them. With this patch applied, kirkwood-i2s will be silent unless at least one of the output widgets is "powered up" - in the case of this patch, that's the "i2sdo" widget - naming following what's in the functional spec. Patch 9 and 10 add the DAPM routes to connect the I2S inputs/outputs to the codec streams, restoring audio output on these platforms. As I don't have these platforms, this should be tested by someone who has. Patch 11, again, discussed with Mark, this adds an always enabled output pin for the SPDIF transceiver, so that the output is powered up by DAPM. Without this, the SPDIF output stream goes into "active" mode during playback, but is never powered up. This results in any connected source widgets also not being powered up. Patch 12 adds the SPDIF output widget to kirkwood-i2s, called "spdifdo", again a name derived from the functional spec. The actual pin is called "AU_SPDIFDO". Patch 13 adds the IEC958 channel status data to patch 12, allowing the channel status to be set by applications using an appropriate ALSA library configuration file via the standard methods. What I haven't included is the board file (like the openrd and t5325 ones already present) for SPDIF support on Dove, as that's based on the pre-DT setups, and I'm not sure anyone in mainline is interested in that. So the above provides the components necessary, hopefully someone with ASoC & DT knowledge can bind these components together via appropriate DT bindings. For reference, the DAI link looks like this: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", and the required route: static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, };

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[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 30 Aug '13

30 Aug '13

The following changes since commit 1c9a341bbdc14051a4d8c74ea67269786c7d3736: ALSA: hda - Simplify CONFIG_SND_HDA_I915 condition (2013-08-27 14:49:59 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-2 for you to fetch changes up to 6d52825fd4e78b251f4844f6662a3bc4e31671c7: Merge remote-tracking branch 'asoc/topic/wm8904' into tmp (2013-08-30 11:04:35 +0100) The diffstat is large as this includes a merge up of v3.11-rc7. ---------------------------------------------------------------- ASoC: Updates for v3.12 A few more updates for v3.12, mostly small cleanups plus the addition of the DT bindings for kirkwood and the new i.MX S/PDIF driver. ---------------------------------------------------------------- Aaro Koskinen (1): ARM: OMAP: rx51: change musb mode to OTG Al Viro (1): cope with potentially long ->d_dname() output for shmem/hugetlb Alan Stern (1): USB: OHCI: add missing PCI PM callbacks to ohci-pci.c Alex Deucher (1): drm/radeon/r7xx: fix copy paste typo in golden register setup Alexey Khoroshilov (1): via-ircc: don't return zero if via_ircc_open() failed Anatolij Gustschin (1): usb: phy: fix build breakage Andrey Vagin (1): tcp: set timestamps for restored skb-s Anthony Foiani (1): sata_fsl: save irqs while coalescing Anton Blanchard (1): [SCSI] lpfc: Don't force CONFIG_GENERIC_CSUM on Ariel Elior (2): bnx2x: Protect against VFs' ndos when SR-IOV is disabled bnx2x: set VF DMAE when first function has 0 supported VFs Baruch Siach (1): sched_clock: Fix integer overflow Ben Hutchings (1): sfc: Fix lookup of default RX MAC filters when steered using ethtool Ben Skeggs (2): drm/nouveau/mc: fix race condition between constructor and request_irq() drm/nv04/disp: fix framebuffer pin refcounting Benjamin Marzinski (1): GFS2: don't overrun reserved revokes Benoit Cousson (1): MAINTAINERS: Update email address for Benoit Cousson Boris BREZILLON (1): ARM: at91: add missing uart clocks DT entries Chen Gang (1): arm64: KVM: use 'int' instead of 'u32' for variable 'target' in kvm_host.h. Chris Wilson (2): drm/i915: Don't deref pipe->cpu_transcoder in the hangcheck code drm/i915: Invalidate TLBs for the rings after a reset Christian König (1): drm/radeon: fix UVD message buffer validation Christoffer Dall (3): ARM: KVM: Fix 64-bit coprocessor handling ARM: KVM: Fix unaligned unmap_range leak KVM: ARM: Squash len warning Chuck Anderson (1): xen/smp: initialize IPI vectors before marking CPU online Dan Carpenter (6): Hostap: copying wrong data prism2_ioctl_giwaplist() nl80211: nl80211hdr_put() doesn't return an ERR_PTR GFS2: alloc_workqueue() doesn't return an ERR_PTR efs: iget_locked() doesn't return an ERR_PTR() bfs: iget_locked() doesn't return an ERR_PTR VFS: collect_mounts() should return an ERR_PTR Daniel Drake (2): x86: Don't clear olpc_ofw_header when sentinel is detected drivers/platform/olpc/olpc-ec.c: initialise earlier Daniel Gimpelevich (2): hso: Earlier catch of error condition hso: Fix stack corruption on some architectures Daniel Mack (1): ARM: OMAP2: fix musb usage for n8x0 Daniel Serpell (1): sony-laptop: Fix reporting of gfx_switch_status Daniel Vetter (1): drm/i915: unpin backing storage in dmabuf_unmap Dave Airlie (4): Merge branch 'drm-fixes-3.11' of git://people.freedesktop.org/~agd5f/linux Merge tag 'drm-intel-fixes-2013-08-15' of git://people.freedesktop.org/~danvet/drm-intel Merge branch 'drm-nouveau-next' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-fixes Merge tag 'drm-intel-fixes-2013-08-23' of git://people.freedesktop.org/~danvet/drm-intel into drm-fixes David Daney (1): MIPS: Handle OCTEON BBIT instructions in FPU emulator. David S. Miller (3): Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'sfc-3.11' of git://git.kernel.org/.../bwh/sfc David Vrabel (3): x86/xen: do not identity map UNUSABLE regions in the machine E820 xen/events: initialize local per-cpu mask for all possible events xen/events: mask events when changing their VCPU binding Dmitry Kravkov (3): bnx2x: dropless flow control not always functional bnx2x: Change DCB context handling bnx2x: Consider DCBX remote error Duan Jiong (1): ipv6: handle Redirect ICMP Message with no Redirected Header option Emmanuel Grumbach (1): iwlwifi: pcie: disable L1 Active after pci_enable_device Fabio Estevam (8): ARM: 7819/1: fiq: Cast the first argument of flush_icache_range() ASoC: simple-card: Provide owner and MODULE_ALIAS() ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata() ASoC: fsl_spdif: Reduce the noise on comments ASoC: fsl_ssi: Remove unnecessary dev_set_drvdata() ASoC: designware_i2s: Remove unnecessary dev_set_drvdata() ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata() ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF Geert Uytterhoeven (1): dm cache: avoid conflicting remove_mapping() in mq policy Giuseppe CAVALLARO (1): net: phy: rtl8211: fix interrupt on status link change Greg Kroah-Hartman (1): Merge tag 'iio-fixes-for-3.11c' of git://git.kernel.org/.../jic23/iio into staging-linus Guillaume Clement (1): gma500: Fix SDVO turning off randomly Guy Cohen (1): Revert "iwlwifi: pcie: clear RFKILL interrupt in AMPG" Hannes Frederic Sowa (2): ipv6: remove max_addresses check from ipv6_create_tempaddr ipv6: drop packets with multiple fragmentation headers Ian Abbott (1): staging: comedi: bug-fix NULL pointer dereference on failed attach Ilia Mirkin (1): drm/nouveau/fb: fix null derefs in nv49 and nv4e init Ingo Molnar (3): Merge branch 'timers/urgent' of git://git.kernel.org/.../frederic/linux-dynticks into timers/urgent Merge branch 'fortglx/3.11/time' of git://git.linaro.org/people/jstultz/linux into timers/urgent Merge tag 'amd_ucode_fixes' of git://git.kernel.org/.../bp/bp into x86/urgent Jean-Francois Moine (2): ASoc: kirkwood: add DT support to the mvebu audio subsystem ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards Joern Rennecke (1): ARC: [lib] strchr breakage in Big-endian configuration Johannes Berg (3): iwlwifi: mvm: disconnect if time event scheduling fails cfg80211: don't request disconnect if not connected Revert "genetlink: fix family dump race" John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Julien Grall (1): xen/arm: missing put_cpu in xen_percpu_init Jussi Kivilinna (1): zd1201: do not use stack as URB transfer_buffer Kukjin Kim (1): ARM: SAMSUNG: fix to support for missing cpu specific map_io Kuninori Morimoto (1): ASoC: fsi: reserve prefetch period on DMA transferring Lars-Peter Clausen (8): ASoC: dmic: Convert table based DAPM setup ASoC: dapm: Fix marking widgets dirty when a route is added ASoC: tlv320aic32x4: Convert table based control and DAPM setup ASoC: wm8904: Remove unnecessary call to snd_soc_dapm_new_widgets() ASoC: jack: Remove unnecessary call to snd_soc_dapm_new_widgets() ASoC: Call snd_soc_dapm_new_widgets() only once during card initialization ASoC: Move call to snd_soc_dapm_new_widgets() after snd_soc_dapm_auto_nc_codec_pins() ASoC: Pass card instead of dapm context to snd_soc_dapm_new_widgets() Li Zefan (1): cpuset: fix a regression in validating config change Li Zhong (1): nohz: fix compile warning in tick_nohz_init() Linus Lüssing (1): batman-adv: check return type of unicast packet preparations Linus Torvalds (27): Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge git://git.kernel.org/.../steve/gfs2-3.0-fixes proc: more readdir conversion bug-fixes Merge tag 'pinctrl-for-v3.11-3' of git://git.kernel.org/.../linusw/linux-pinctrl Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'arm64-stable' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'stable/for-linus-3.11-rc6-tag' of git://git.kernel.org/.../xen/tip Revert "x86 get_unmapped_area(): use proper mmap base for bottom-up direction" Merge tag 'dm-3.11-fixes' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'devicetree-fixes-for-3.11' of git://sources.calxeda.com/kernel/linux Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'linux-next' of git://cavan.codon.org.uk/platform-drivers-x86 Merge branch 'akpm' (patches from Andrew Morton) Merge git://git.kernel.org/.../davem/net Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'for-3.11-fixes' of git://git.kernel.org/.../tj/cgroup Merge branch 'for-3.11-fixes' of git://git.kernel.org/.../tj/libata Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'acpi-3.11-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge tag 'usb-3.11-rc7' of git://git.kernel.org/.../gregkh/usb Merge tag 'staging-3.11-rc7' of git://git.kernel.org/.../gregkh/staging Linux 3.11-rc7 Maarten Lankhorst (2): drm/nouveau/ltcg: fix ltcg memory initialization after suspend drm/nouveau/ltcg: fix allocating memory as free Marc Zyngier (3): arm64: KVM: fix 2-level page tables unmapping arm64: KVM: perform save/restore of PAR_EL1 arm64: KVM: add missing dsb before invalidating Stage-2 TLBs Mark Brown (11): Merge remote-tracking branch 'asoc/topic/dmic' into asoc-core Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-core Merge remote-tracking branch 'asoc/topic/dapm' into asoc-core Merge remote-tracking branch 'asoc/topic/core' into tmp Merge remote-tracking branch 'asoc/topic/designware' into tmp Merge remote-tracking branch 'asoc/topic/ep93xx' into tmp Merge remote-tracking branch 'asoc/topic/fsi' into tmp Merge remote-tracking branch 'asoc/topic/fsl' into tmp Merge remote-tracking branch 'asoc/topic/kirkwood' into tmp Merge remote-tracking branch 'asoc/topic/samsung' into tmp Merge remote-tracking branch 'asoc/topic/wm8904' into tmp Mark Langsdorf (1): sata, highbank: fix ordering of SGPIO signals Martin Peschke (3): [SCSI] zfcp: fix lock imbalance by reworking request queue locking [SCSI] zfcp: fix schedule-inside-lock in scsi_device list loops [SCSI] zfcp: remove access control tables interface (keep sysfs files) Matthew Garrett (1): Revert "hp-wmi: Enable hotkeys on some systems" Maxime Ripard (3): pinctrl: sunxi: Read register before writing to it in irq_set_type pinctrl: sunxi: Fix gpio_set behaviour pinctrl: sunxi: Add spinlocks Michal Hocko (1): memcg: get rid of swapaccount leftovers Nicolas Ferre (2): ARM: at91/DT: at91sam9x5ek: fix USB host property to enable port C ARM: at91/DT: fix at91sam9n12ek memory node Nicolas Pitre (1): ARM: 7816/1: CONFIG_KUSER_HELPERS: fix help text Nicolin Chen (1): ASoC: fsl: Add S/PDIF machine driver Oleg Nesterov (1): proc: kill the extra proc_readfd_common()->dir_emit_dots() Olof Johansson (2): Merge tag 'samsung-fixes-2' of git://git.kernel.org/.../kgene/linux-samsung into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Pali Rohár (1): drm/nouveau: fix reclocking on nv40 Paolo Bonzini (2): Merge branch 'kvm-arm64/fixes-3.11-rc4' of git://git.kernel.org/.../maz/arm-platforms into kvm-master Merge tag 'kvm-arm-fixes-3.11' of git://git.linaro.org/people/cdall/linux-kvm-arm into kvm-master Peter Meerwald (1): iio: adjd_s311: Fix non-scan mode data read Peter Wu (1): r8169: remember WOL preferences on driver load Radu Caragea (1): x86 get_unmapped_area: Access mmap_legacy_base through mm_struct member Rafael J. Wysocki (1): Revert "ACPI / video: Always call acpi_video_init_brightness() on init" Rafał Miłecki (1): drm/radeon: fix WREG32_OR macro setting bits in a register Randy Dunlap (1): kernel: fix new kernel-doc warning in wait.c Richard Genoud (1): proc: return on proc_readdir error Richard Laager (1): lib/lz4: correct the LZ4 license Roland Dreier (1): [SCSI] sg: Fix user memory corruption when SG_IO is interrupted by a signal Russell King (1): Merge branch 'security-fixes' into fixes Sathya Perla (1): be2net: fix disabling TX in be_close() Sekhar Nori (1): ARM: davinci: nand: specify ecc strength Stanislaw Gruszka (1): iwlwifi: dvm: fix calling ieee80211_chswitch_done() with NULL Stefano Stabellini (1): Merging v3.10-rc2 as I need to apply a fix for Stephen Warren (1): ARM: tegra: always enable USB VBUS regulators Steven Rostedt (1): nohz: Do not warn about unstable tsc unless user uses nohz_full Steven Whitehouse (2): GFS2: Fix typo in gfs2_create_inode() GFS2: Check for glock already held in gfs2_getxattr Tejun Heo (1): GFS2: WQ_NON_REENTRANT is meaningless and going away Terry Suereth (1): libata: apply behavioral quirks to sil3826 PMP Tomasz Figa (1): ASoC: Samsung: Do not queue cyclic buffers multiple times Torsten Kaiser (2): x86, microcode, AMD: Make cpu_has_amd_erratum() use the correct struct cpuinfo_x86 x86, microcode, AMD: Fix early microcode loading Toshiaki Makita (1): bridge: Use the correct bit length for bitmap functions in the VLAN code Vijaya Kumar K (1): ARM: 7815/1: kexec: offline non panic CPUs on Kdump panic Vlad Yasevich (3): macvtap: simplify usage of tap_features macvtap: Correctly set tap features when IFF_VNET_HDR is disabled. macvtap: Ignore tap features when VNET_HDR is off Vyacheslav Dubeyko (2): nilfs2: remove double bio_put() in nilfs_end_bio_write() for BIO_EOPNOTSUPP error nilfs2: fix issue with counting number of bio requests for BIO_EOPNOTSUPP error detection Wei Yongjun (2): sony-laptop: return a negative error code in sonypi_compat_init() ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe() Will Deacon (2): arm64: perf: fix array out of bounds access in armpmu_map_hw_event() arm64: perf: fix event validation for software group leaders Willem de Bruijn (1): packet: restore packet statistics tp_packets to include drops Wladislav Wiebe (1): of: fdt: fix memory initialization for expanded DT Yuval Mintz (1): bnx2x: prevent VF benign attentions stephen hemminger (1): ipv4: expose IPV4_DEVCONF .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++ Documentation/kernel-parameters.txt | 2 +- MAINTAINERS | 6 +- Makefile | 2 +- arch/arc/lib/strchr-700.S | 10 ++ arch/arm/boot/dts/at91sam9n12ek.dts | 4 +- arch/arm/boot/dts/at91sam9x5ek.dtsi | 5 +- arch/arm/boot/dts/tegra20-seaboard.dts | 2 + arch/arm/boot/dts/tegra20-trimslice.dts | 2 + arch/arm/boot/dts/tegra20-whistler.dts | 4 + arch/arm/kernel/fiq.c | 3 +- arch/arm/kernel/machine_kexec.c | 1 + arch/arm/kvm/coproc.c | 26 +++- arch/arm/kvm/coproc.h | 3 + arch/arm/kvm/coproc_a15.c | 6 +- arch/arm/kvm/mmio.c | 3 +- arch/arm/kvm/mmu.c | 36 +++-- arch/arm/mach-at91/at91sam9x5.c | 2 + arch/arm/mach-davinci/board-dm355-leopard.c | 1 + arch/arm/mach-davinci/board-dm644x-evm.c | 1 + arch/arm/mach-davinci/board-dm646x-evm.c | 1 + arch/arm/mach-davinci/board-neuros-osd2.c | 1 + arch/arm/mach-omap2/board-n8x0.c | 4 - arch/arm/mach-omap2/board-rx51.c | 2 +- arch/arm/mach-omap2/usb-musb.c | 5 +- arch/arm/mm/Kconfig | 9 +- arch/arm/plat-samsung/init.c | 5 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- arch/arm/xen/enlighten.c | 1 + arch/arm64/include/asm/kvm_asm.h | 17 ++- arch/arm64/include/asm/kvm_host.h | 2 +- arch/arm64/kernel/perf_event.c | 10 +- arch/arm64/kvm/hyp.S | 13 ++ arch/arm64/kvm/sys_regs.c | 3 + arch/mips/math-emu/cp1emu.c | 26 ++++ arch/x86/include/asm/bootparam_utils.h | 4 +- arch/x86/include/asm/microcode_amd.h | 2 +- arch/x86/kernel/cpu/amd.c | 20 +-- arch/x86/kernel/microcode_amd.c | 27 ++-- arch/x86/kernel/microcode_amd_early.c | 27 ++-- arch/x86/kernel/sys_x86_64.c | 2 +- arch/x86/mm/mmap.c | 8 +- arch/x86/xen/setup.c | 22 +++ arch/x86/xen/smp.c | 11 +- drivers/acpi/video.c | 11 +- drivers/ata/libata-pmp.c | 12 +- drivers/ata/sata_fsl.c | 5 +- drivers/ata/sata_highbank.c | 4 +- drivers/gpu/drm/gma500/psb_intel_sdvo.c | 3 +- drivers/gpu/drm/i915/i915_gem_dmabuf.c | 8 ++ drivers/gpu/drm/i915/i915_reg.h | 2 + drivers/gpu/drm/i915/intel_display.c | 86 ++++++++---- drivers/gpu/drm/i915/intel_ringbuffer.c | 12 ++ drivers/gpu/drm/nouveau/core/core/mm.c | 4 + drivers/gpu/drm/nouveau/core/include/subdev/mc.h | 7 +- drivers/gpu/drm/nouveau/core/subdev/fb/ramnv49.c | 12 +- drivers/gpu/drm/nouveau/core/subdev/fb/ramnv4e.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/ltcg/nvc0.c | 34 +++-- drivers/gpu/drm/nouveau/core/subdev/mc/base.c | 6 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv04.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv44.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv50.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv98.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nvc0.c | 3 +- drivers/gpu/drm/nouveau/dispnv04/crtc.c | 58 +++++--- drivers/gpu/drm/nouveau/dispnv04/disp.h | 1 + drivers/gpu/drm/nouveau/nouveau_display.c | 3 + drivers/gpu/drm/nouveau/nv40_pm.c | 2 +- drivers/gpu/drm/radeon/radeon.h | 2 +- drivers/gpu/drm/radeon/radeon_uvd.c | 8 ++ drivers/gpu/drm/radeon/rv770.c | 12 +- drivers/iio/light/adjd_s311.c | 3 +- drivers/md/dm-cache-policy-mq.c | 16 +-- drivers/net/ethernet/broadcom/bnx2x/bnx2x.h | 2 + drivers/net/ethernet/broadcom/bnx2x/bnx2x_dcb.c | 49 +++++-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_dcb.h | 3 + drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 45 +++++-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_sriov.c | 56 ++++---- drivers/net/ethernet/emulex/benet/be_main.c | 2 +- drivers/net/ethernet/realtek/r8169.c | 2 +- drivers/net/ethernet/sfc/filter.c | 2 +- drivers/net/irda/via-ircc.c | 6 +- drivers/net/macvtap.c | 18 +-- drivers/net/phy/realtek.c | 4 +- drivers/net/usb/hso.c | 15 ++- drivers/net/wireless/hostap/hostap_ioctl.c | 4 +- drivers/net/wireless/iwlwifi/dvm/mac80211.c | 5 +- drivers/net/wireless/iwlwifi/iwl-prph.h | 2 - drivers/net/wireless/iwlwifi/mvm/time-event.c | 33 +++-- drivers/net/wireless/iwlwifi/pcie/rx.c | 8 -- drivers/net/wireless/iwlwifi/pcie/trans.c | 10 +- drivers/net/wireless/zd1201.c | 4 +- drivers/of/fdt.c | 2 + drivers/pinctrl/pinctrl-sunxi.c | 66 ++++++++- drivers/pinctrl/pinctrl-sunxi.h | 2 + drivers/platform/olpc/olpc-ec.c | 2 +- drivers/platform/x86/hp-wmi.c | 16 --- drivers/platform/x86/sony-laptop.c | 8 +- drivers/s390/scsi/zfcp_erp.c | 29 +++- drivers/s390/scsi/zfcp_qdio.c | 8 +- drivers/s390/scsi/zfcp_sysfs.c | 14 ++ drivers/scsi/Kconfig | 1 - drivers/staging/comedi/drivers.c | 2 +- drivers/usb/host/ohci-pci.c | 5 + drivers/usb/phy/phy-fsl-usb.h | 2 +- drivers/usb/phy/phy-fsm-usb.c | 2 +- drivers/xen/events.c | 13 +- fs/bfs/inode.c | 2 +- fs/bio.c | 20 ++- fs/dcache.c | 11 ++ fs/efs/inode.c | 2 +- fs/gfs2/glock.c | 8 +- fs/gfs2/glops.c | 18 ++- fs/gfs2/inode.c | 6 +- fs/gfs2/main.c | 2 +- fs/hugetlbfs/inode.c | 8 +- fs/namespace.c | 2 +- fs/nilfs2/segbuf.c | 5 +- fs/proc/fd.c | 2 - fs/proc/generic.c | 2 +- fs/proc/root.c | 4 +- include/linux/dcache.h | 1 + include/linux/inetdevice.h | 34 +---- include/linux/ipv6.h | 1 + include/linux/mm_types.h | 1 + include/linux/sched.h | 1 - include/linux/wait.h | 57 ++++++++ include/net/ip6_route.h | 2 + include/sound/soc-dapm.h | 2 +- include/uapi/linux/ip.h | 34 +++++ init/Kconfig | 2 +- kernel/cpuset.c | 14 +- kernel/time/sched_clock.c | 2 +- kernel/time/tick-sched.c | 5 +- kernel/wait.c | 3 +- lib/lz4/lz4_compress.c | 4 +- lib/lz4/lz4_decompress.c | 6 +- lib/lz4/lz4hc_compress.c | 4 +- mm/memcontrol.c | 1 - mm/shmem.c | 8 +- net/batman-adv/unicast.c | 10 +- net/bridge/br_fdb.c | 10 +- net/bridge/br_netlink.c | 4 +- net/bridge/br_vlan.c | 4 +- net/ipv4/tcp.c | 7 + net/ipv6/addrconf.c | 10 +- net/ipv6/ndisc.c | 4 +- net/ipv6/reassembly.c | 5 + net/ipv6/route.c | 21 +++ net/netlink/genetlink.c | 7 - net/packet/af_packet.c | 2 + net/wireless/nl80211.c | 22 +-- net/wireless/sme.c | 10 +- sound/soc/cirrus/ep93xx-i2s.c | 2 - sound/soc/codecs/dmic.c | 17 +-- sound/soc/codecs/tlv320aic32x4.c | 22 +-- sound/soc/codecs/wm8904.c | 1 - sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/Kconfig | 10 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_spdif.c | 17 +-- sound/soc/fsl/fsl_ssi.c | 1 - sound/soc/fsl/imx-spdif.c | 148 +++++++++++++++++++++ sound/soc/generic/simple-card.c | 2 + sound/soc/kirkwood/Kconfig | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 26 +++- sound/soc/samsung/dma.c | 7 + sound/soc/sh/fsi.c | 51 ++++--- sound/soc/soc-core.c | 9 +- sound/soc/soc-dapm.c | 9 +- sound/soc/soc-jack.c | 2 - 172 files changed, 1316 insertions(+), 598 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/mvebu-audio.txt create mode 100644 sound/soc/fsl/imx-spdif.c

2 1

[alsa-devel] [PATCH] ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF
by Fabio Estevam 30 Aug '13

30 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a708380..704e246 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -197,7 +197,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_FSL_UTILS select SND_SOC_SPDIF help SoC Audio support for i.MX boards with S/PDIF -- 1.8.1.2

3 2

[alsa-devel] [PATCH v12] ASoC: fsl: Add S/PDIF machine driver
by Nicolin Chen 29 Aug '13

29 Aug '13

This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v11->v12: * Dropped unused spdif_pdev. * Register spdif-dit/dir depending on DT bool property. v10->v11: * Use boolean properties for spdif-out/in switch instead of codec phandles. * Accordingly create dummy codec driver in probe() instead of DT nodes. .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-spdif.c | 148 ++++++++++++++++++++ 4 files changed, 195 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/imx-spdif.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 0000000..7d13479 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc..a708380 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e..e2aaff7 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..816013b --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register dit failed: %d\n", ret); + goto end; + } + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register dir failed: %d\n", ret); + goto error_dit; + } + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto error_dir; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto error_dir; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + if (data->rxdev) + platform_device_unregister(data->rxdev); +error_dit: + if (data->txdev) + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + if (data->rxdev) + platform_device_unregister(data->rxdev); + if (data->txdev) + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- 1.7.1

3 2

[alsa-devel] [PATCH] alsa-info: Run without lspci (if there is no pci bus)
by David Henningsson 29 Aug '13

29 Aug '13

On some embedded ARM hardware, there is no pci bus. In that case, do not require lspci so we can still get as much info as we can. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsa/utils/alsa-info.sh | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/alsa/utils/alsa-info.sh b/alsa/utils/alsa-info.sh index 9d996aa..2280af5 100755 --- a/alsa/utils/alsa-info.sh +++ b/alsa/utils/alsa-info.sh @@ -393,8 +393,10 @@ trap cleanup 0 if [ "$PROCEED" = "yes" ]; then if [ -z "$LSPCI" ]; then - echo "This script requires lspci. Please install it, and re-run this script." - exit 0 + if [ -d /sys/bus/pci ]; then + echo "This script requires lspci. Please install it, and re-run this script." + exit 0 + fi fi #Fetch the info and store in temp files/variables @@ -407,8 +409,6 @@ KERNEL_OS=`uname -o` ALSA_DRIVER_VERSION=`cat /proc/asound/version |head -n1|awk {'print $7'} |sed 's/\.$//'` get_alsa_library_version ALSA_UTILS_VERSION=`amixer -v |awk {'print $3'}` -VENDOR_ID=`lspci -vn | grep 040[1-3] | awk -F':' '{print $3}'| awk {'print substr($0, 2);}' >$TEMPDIR/vendor_id.tmp` -DEVICE_ID=`lspci -vn | grep 040[1-3] | awk -F':' '{print $4}'| awk {'print $1'} >$TEMPDIR/device_id.tmp` LAST_CARD=$((`grep "]: " /proc/asound/cards | wc -l` - 1 )) ESDINST=$(which esd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) @@ -434,7 +434,9 @@ fi cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp cat /proc/asound/cards >$TEMPDIR/alsacards.tmp +if [[ ! -z "$LSPCI" ]]; then lspci |grep -i "multi\|audio">$TEMPDIR/lspci.tmp +fi #Check for HDA-Intel cards codec#* cat /proc/asound/card*/codec\#* > $TEMPDIR/alsa-hda-intel.tmp 2> /dev/null @@ -547,6 +549,8 @@ echo "" >> $FILE cat $TEMPDIR/alsacards.tmp >> $FILE echo "" >> $FILE echo "" >> $FILE + +if [[ ! -z "$LSPCI" ]]; then echo "!!PCI Soundcards installed in the system" >> $FILE echo "!!--------------------------------------" >> $FILE echo "" >> $FILE @@ -559,6 +563,7 @@ echo "" >> $FILE lspci -vvn |grep -A1 040[1-3] >> $FILE echo "" >> $FILE echo "" >> $FILE +fi if [ "$SNDOPTIONS" ] then -- 1.7.9.5

2 3

[alsa-devel] Compiling pyalsa with the latest alsalib
by Proyecto MultiK 29 Aug '13

29 Aug '13

Hello, Together with a team, I am working on a multiseat project for which we need to access more than 32 sound cards on a Linux system (not all used simultaneously). We are running Ubuntu with a modified version of the 3.6.3 kernel. We used the code from Takashi Iwai's sound repository to patch the original 3.6.3 source to allow ALSA to recognize and name more than 32 devices (since there was a bug introduced in the 3.8 kernel which threw a kernel panic when connecting more than 28 cards, we could not use the latest kernel). Currently, we can detect more than 36 sound cards, they have ids and names given in /proc/asound/cards. Before applying Takashi's patches, they would stop being named at id = 31. So this seems to be working all right. The problem comes when trying to access the cards using pyalsa. We downloaded the pyalsa 1.0.26 tar from ftp://ftp.alsa-project.org/pub/pyalsa/pyalsa-1.0.26.tar.bz2. We need to know how to compile pyalsa with the latest alsalib that supports more than 32 cards. If we run pyalsa.alsacard.asoundlib_version() from the python interpreter, we get '1.0.25'. We know the latest version is 1.0.27 and we have put the alsa headers in pyalsa-1.0.26/pyalsa/alsa/ dir before compiling, but it doesn't seem to compile using these files and keeps saying version is 1.0.25. Guidance on this would be greatly appreciated. Thanks, Esteban Serrano MultiK team

2 1

[alsa-devel] [PATCH 1/3] ASoC: fsl_ssi: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 29 Aug '13

29 Aug '13

Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff..6f4a4c5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -909,7 +909,6 @@ error_dai: snd_soc_unregister_component(&pdev->dev); error_dev: - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); error_clk: @@ -935,7 +934,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); irq_dispose_mapping(ssi_private->irq); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.1.2

2 5

Re: [alsa-devel] [PATCH -next] ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe()
by Mark Brown 29 Aug '13

29 Aug '13

On Thu, Aug 29, 2013 at 08:00:05AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > There is a error message within devm_ioremap_resource > already, so remove the dev_err call to avoid redundant > error message. Applied, thanks.

1 0

[alsa-devel] [RFC] ASoC: soc-pcm: Add support for unidirectional dai links
by Fabio Estevam 29 Aug '13

29 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- After I get some feedback, I can split the patch in two parts (one for soc-pcm and another one for mxs-sgtl5000). include/sound/soc.h | 4 ++++ sound/soc/mxs/mxs-sgtl5000.c | 2 ++ sound/soc/soc-pcm.c | 10 ++++++++++ 3 files changed, 16 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..774d3be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -934,6 +934,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index ce084eb..4bb2737 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .capture_only = true, }, }; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb70fbe..5e44863 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- 1.8.1.2

3 3

[alsa-devel] [PATCH 1/5] ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
by Lars-Peter Clausen 29 Aug '13

29 Aug '13

The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite different in the analog domain. This patch adds support for the common parts of the ADAU1X61 and ADAU1X81 CODECs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/platform_data/adau17x1.h | 23 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau17x1.c | 895 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau17x1.h | 148 ++++++ 5 files changed, 1072 insertions(+) create mode 100644 include/linux/platform_data/adau17x1.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h diff --git a/include/linux/platform_data/adau17x1.h b/include/linux/platform_data/adau17x1.h new file mode 100644 index 0000000..fdf419a --- /dev/null +++ b/include/linux/platform_data/adau17x1.h @@ -0,0 +1,23 @@ +/* + * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs + * + * Copyright 2011-2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU17X1_H__ +#define __LINUX_PLATFORM_DATA_ADAU17X1_H__ + +/** + * enum adau17x1_micbias_voltage - Microphone bias voltage + * @ADAU17X1_MICBIAS_0_90_AVDD: 0.9 * AVDD + * @ADAU17X1_MICBIAS_0_65_AVDD: 0.65 * AVDD + */ +enum adau17x1_micbias_voltage { + ADAU17X1_MICBIAS_0_90_AVDD = 0, + ADAU17X1_MICBIAS_0_65_AVDD = 1, +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15106c0..56137d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -193,6 +193,10 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU17X1 + select SND_SOC_SIGMADSP + tristate + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b4ecf6c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau17x1-objs := adau17x1.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -138,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o obj-$(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 diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c new file mode 100644 index 0000000..b7d678d --- /dev/null +++ b/sound/soc/codecs/adau17x1.c @@ -0,0 +1,895 @@ +/* + * Common code for ADAU1X61 and ADAU1X81 codecs + * + * Copyright 2011-2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gcd.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include "sigmadsp.h" +#include "adau17x1.h" + +static const char * const adau17x1_capture_mixer_boost_text[] = { + "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", +}; + +static const SOC_ENUM_SINGLE_DECL(adau17x1_capture_boost_enum, + ADAU17X1_REC_POWER_MGMT, 5, adau17x1_capture_mixer_boost_text); + +static const char * const adau17x1_mic_bias_mode_text[] = { + "Normal operation", "High performance", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_mic_bias_mode_enum, + ADAU17X1_MICBIAS, 3, adau17x1_mic_bias_mode_text); + +static const char * const adau17x1_mono_stereo_text[] = { + "Stereo", + "Mono Left Channel (L+R)", + "Mono Right Channel (L+R)", + "Mono (L+R)", +}; + +static const SOC_ENUM_SINGLE_DECL(adau17x1_dac_mode_enum, + ADAU17X1_DAC_CONTROL0, 6, adau17x1_mono_stereo_text); + +static const DECLARE_TLV_DB_MINMAX(adau17x1_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adau17x1_controls[] = { + SOC_DOUBLE_R_TLV("Digital Capture Volume", + ADAU17X1_LEFT_INPUT_DIGITAL_VOL, + ADAU17X1_RIGHT_INPUT_DIGITAL_VOL, + 0, 0xff, 1, adau17x1_digital_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", ADAU17X1_DAC_CONTROL1, + ADAU17X1_DAC_CONTROL2, 0, 0xff, 1, adau17x1_digital_tlv), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAU17X1_ADC_CONTROL, + 5, 1, 0), + SOC_SINGLE("Playback De-emphasis Switch", ADAU17X1_DAC_CONTROL0, + 2, 1, 0), + + SOC_ENUM("Capture Boost", adau17x1_capture_boost_enum), + + SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), + + SOC_ENUM("DAC Mono Stereo", adau17x1_dac_mode_enum), +}; + +static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + int ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + adau->pll_regs[5] = 1; + } else { + adau->pll_regs[5] = 0; + /* Bypass the PLL when disabled, otherwise registers will become + * inaccessible. */ + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, 0); + } + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + mdelay(5); + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL); + } + + return 0; +} + +static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("PLL", 3, SND_SOC_NOPM, 0, 0, adau17x1_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("AIFCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU17X1_MICBIAS, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 1, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Dec Filter Enable", ADAU17X1_ADC_CONTROL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Dec Filter Enable", ADAU17X1_ADC_CONTROL, + 1, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_routes[] = { + { "Left ADC", NULL, "SYSCLK" }, + { "Right ADC", NULL, "SYSCLK" }, + { "Left DAC", NULL, "SYSCLK" }, + { "Right DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + + { "Left ADC", NULL, "Left Dec Filter Enable" }, + { "Right ADC", NULL, "Right Dec Filter Enable" }, + + { "AIFOUT", NULL, "AIFCLK" }, + { "AIFIN", NULL, "AIFCLK" }, +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_pll_route = { + "SYSCLK", NULL, "PLL", +}; + +static const char * const adau17x1_dac_mux_text[] = { + "AIFIN", + "DSP", +}; + +int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int val, change; + + if (ucontrol->value.enumerated.item[0] >= e->max) + return -EINVAL; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + switch (e->reg) { + case ADAU17X1_SERIAL_INPUT_ROUTE: + val = (adau->tdm_dac_slot * 2) + 1; + adau->dsp_playback_bypass = true; + break; + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + val = (adau->tdm_adc_slot * 2) + 1; + adau->dsp_capture_bypass = true; + break; + default: + return -EINVAL; + } + break; + default: + val = 0; + + switch (e->reg) { + case ADAU17X1_SERIAL_INPUT_ROUTE: + adau->dsp_playback_bypass = false; + break; + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + adau->dsp_capture_bypass = false; + break; + default: + return -EINVAL; + } + break; + } + + change = snd_soc_test_bits(codec, e->reg, 0xff, val); + if (change) { + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = 0xff; + update.val = val; + + snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol, + ucontrol->value.enumerated.item[0], e, &update); + } + + return change; +} +EXPORT_SYMBOL_GPL(adau17x1_dsp_mux_enum_put); + +int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + + val = snd_soc_read(codec, e->reg); + ucontrol->value.enumerated.item[0] = val == 0; + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_dsp_mux_enum_get); + +static const SOC_ENUM_SINGLE_DECL(adau17x1_dac_mux_enum, + ADAU17X1_SERIAL_INPUT_ROUTE, 0, + adau17x1_dac_mux_text); + +static const struct snd_kcontrol_new adau17x1_dac_mux = + ADAU17X1_DSP_MUX_ENUM("DAC Playback Mux", adau17x1_dac_mux_enum); + +static const struct snd_soc_dapm_widget adau17x1_dsp_dapm_widgets[] = { + SND_SOC_DAPM_PGA("DSP", ADAU17X1_DSP_RUN, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("DSP Siggen"), + + SND_SOC_DAPM_VIRT_MUX("DAC Playback Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mux), + +}; + +static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { + { "DAC Playback Mux", "DSP", "DSP" }, + { "DAC Playback Mux", "AIFIN", "AIFIN" }, + + { "Left DAC", NULL, "DAC Playback Mux" }, + { "Right DAC", NULL, "DAC Playback Mux" }, + + { "AIFOUT Capture Mux", "DSP", "DSP" }, + + { "AIFOUT", NULL, "AIFOUT Capture Mux" }, + + { "DSP", NULL, "DSP Siggen" }, +}; + +static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { + { "Left DAC", NULL, "AIFIN" }, + { "Right DAC", NULL, "AIFIN" }, +}; + +static void adau17x1_check_aifclk(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + /* If we are in master mode we need to generate bit- and frameclock, + * regardless of whether there is an active path or not */ + if (codec->active && adau->master) + snd_soc_dapm_force_enable_pin(&codec->dapm, "AIFCLK"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "AIFCLK"); + snd_soc_dapm_sync(&codec->dapm); +} + +bool adau17x1_has_dsp(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1381: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_dsp); + +static int adau17x1_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 adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int val, div, dsp_div; + unsigned int freq; + + if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + freq = adau->pll_freq; + else + freq = adau->sysclk / adau->sysclk_div; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* fs */ + div = 0; + dsp_div = 1; + break; + case 6144: /* fs / 6 */ + div = 1; + dsp_div = 6; + break; + case 4096: /* fs / 4 */ + div = 2; + dsp_div = 5; + break; + case 3072: /* fs / 3 */ + div = 3; + dsp_div = 4; + break; + case 2048: /* fs / 2 */ + div = 4; + dsp_div = 3; + break; + case 1536: /* fs / 1.5 */ + div = 5; + dsp_div = 2; + break; + case 512: /* fs / 0.5 */ + div = 6; + dsp_div = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, 7, div); + if (adau17x1_has_dsp(adau)) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_SAMPLING_RATE, div); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); + } + + adau17x1_check_aifclk(codec); + + if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); +} + +static int adau17x1_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + adau17x1_check_aifclk(dai->codec); + + return 0; +} + +static void adau17x1_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + adau17x1_check_aifclk(dai->codec); +} + +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int div; + unsigned int r, n, m, i, j; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + if (!freq_out) { + r = 0; + n = 0; + m = 0; + div = 0; + } else { + if (freq_out % freq_in != 0) { + div = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= div; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + div--; + } else { + r = freq_out / freq_in; + n = 0; + m = 0; + div = 0; + } + if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) + return -EINVAL; + } + + adau->pll_regs[0] = m >> 8; + adau->pll_regs[1] = m & 0xff; + adau->pll_regs[2] = n >> 8; + adau->pll_regs[3] = n & 0xff; + adau->pll_regs[4] = (r << 3) | (div << 1); + if (m != 0) + adau->pll_regs[4] |= 1; /* Fractional mode */ + adau->pll_regs[5] = 0; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + struct snd_soc_dapm_context *dapm = &dai->codec->dapm; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + case ADAU17X1_CLK_SRC_PLL: + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (adau->clk_src != clk_id) { + if (clk_id == ADAU17X1_CLK_SRC_PLL) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_set_dai_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + + if (div < 0 || div > 4) + return -EINVAL; + + adau->sysclk_div = div; + + return regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_INFREQ_MASK, (div - 1) << 1); +} + +static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1; + int lrclk_pol; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl0 = ADAU17X1_SERIAL_PORT0_MASTER; + adau->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl0 = 0; + adau->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + lrclk_pol = 0; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + lrclk_pol = 1; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + case SND_SOC_DAIFMT_DSP_A: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_DSP_B: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk_pol = !lrclk_pol; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + lrclk_pol = !lrclk_pol; + break; + default: + return -EINVAL; + } + + if (lrclk_pol) + ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; + + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + + adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ser_ctrl0, ser_ctrl1; + unsigned int conv_ctrl0, conv_ctrl1; + + /* I2S mode */ + if (slots == 0) { + slots = 2; + rx_mask = 3; + tx_mask = 3; + slot_width = 32; + } + + switch (slots) { + case 2: + ser_ctrl0 = ADUA_SERIAL_PORT0_STEREO; + break; + case 4: + ser_ctrl0 = ADUA_SERIAL_PORT0_TDM4; + break; + case 8: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl0 = ADUA_SERIAL_PORT0_TDM8; + break; + default: + return -EINVAL; + } + + switch (slot_width * slots) { + case 32: + if (adau->type == ADAU1761) + return -EINVAL; + + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK32; + break; + case 64: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK64; + break; + case 48: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK48; + break; + case 128: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK128; + break; + case 256: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK256; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0x03: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(1); + adau->tdm_adc_slot = 0; + break; + case 0x0c: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(2); + adau->tdm_adc_slot = 1; + break; + case 0x30: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(3); + adau->tdm_adc_slot = 2; + break; + case 0xc0: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(4); + adau->tdm_adc_slot = 3; + break; + default: + return -EINVAL; + } + + switch (tx_mask) { + case 0x03: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(1); + adau->tdm_dac_slot = 0; + break; + case 0x0c: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(2); + adau->tdm_dac_slot = 1; + break; + case 0x30: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(3); + adau->tdm_dac_slot = 2; + break; + case 0xc0: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(4); + adau->tdm_dac_slot = 3; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_DAC_PAIR_MASK, conv_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER1, + ADAU17X1_CONVERTER1_ADC_PAIR_MASK, conv_ctrl1); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, + ADAU17X1_SERIAL_PORT0_TDM_MASK, ser_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_BCLK_MASK, ser_ctrl1); + + if (adau17x1_has_dsp(adau)) { + if (adau->dsp_playback_bypass) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_INPUT_ROUTE, + (adau->tdm_dac_slot * 2) + 1); + } + if (adau->dsp_capture_bypass) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_OUTPUT_ROUTE, + (adau->tdm_adc_slot * 2) + 1); + } + } + + return 0; +} + +const struct snd_soc_dai_ops adau17x1_dai_ops = { + .hw_params = adau17x1_hw_params, + .set_sysclk = adau17x1_set_dai_sysclk, + .set_fmt = adau17x1_set_dai_fmt, + .set_pll = adau17x1_set_dai_pll, + .set_clkdiv = adau17x1_set_dai_clkdiv, + .set_tdm_slot = adau17x1_set_dai_tdm_slot, + .startup = adau17x1_dai_startup, + .shutdown = adau17x1_dai_shutdown, +}; +EXPORT_SYMBOL_GPL(adau17x1_dai_ops); + +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + switch (micbias) { + case ADAU17X1_MICBIAS_0_90_AVDD: + case ADAU17X1_MICBIAS_0_65_AVDD: + break; + default: + return -EINVAL; + } + + regmap_write(adau->regmap, ADAU17X1_MICBIAS, micbias << 2); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); + +bool adau17x1_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU17X1_CLOCK_CONTROL: + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_REC_POWER_MGMT: + case ADAU17X1_MICBIAS: + case ADAU17X1_SERIAL_PORT0: + case ADAU17X1_SERIAL_PORT1: + case ADAU17X1_CONVERTER0: + case ADAU17X1_CONVERTER1: + case ADAU17X1_LEFT_INPUT_DIGITAL_VOL: + case ADAU17X1_RIGHT_INPUT_DIGITAL_VOL: + case ADAU17X1_ADC_CONTROL: + case ADAU17X1_PLAY_POWER_MGMT: + case ADAU17X1_DAC_CONTROL0: + case ADAU17X1_DAC_CONTROL1: + case ADAU17X1_DAC_CONTROL2: + case ADAU17X1_SERIAL_PORT_PAD: + case ADAU17X1_CONTROL_PORT_PAD0: + case ADAU17X1_CONTROL_PORT_PAD1: + case ADAU17X1_DSP_SAMPLING_RATE: + case ADAU17X1_SERIAL_INPUT_ROUTE: + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + case ADAU17X1_DSP_ENABLE: + case ADAU17X1_DSP_RUN: + case ADAU17X1_SERIAL_SAMPLING_RATE: + return true; + default: + break; + } + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_readable_register); + +bool adau17x1_volatile_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter and program memory */ + if (reg < 0x4000) + return true; + + switch (reg) { + /* The PLL register is 6 bytes long */ + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: + return true; + default: + break; + } + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_volatile_register); + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware) +{ + int ret; + int dspsr; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); + if (ret) + return ret; + + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + + ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware); + if (ret) { + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); + return ret; + } + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_load_firmware); + +int adau17x1_probe(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_add_codec_controls(codec, adau17x1_controls, + ARRAY_SIZE(adau17x1_controls)); + if (ret) + return ret; + ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets, + ARRAY_SIZE(adau17x1_dapm_widgets)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau17x1_dsp_dapm_widgets, + ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_probe); + +int adau17x1_add_routes(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes, + ARRAY_SIZE(adau17x1_dapm_routes)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_dsp_dapm_routes)); + } else { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_no_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_routes); + +#if IS_ENABLED(CONFIG_SPI_MASTER) +static void adau17x1_spi_mode(struct device *dev) +{ + /* To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. */ + spi_w8r8(to_spi_device(dev), 0x00); + spi_w8r8(to_spi_device(dev), 0x00); + spi_w8r8(to_spi_device(dev), 0x00); +} +#else +static inline void adau17x1_spi_mode(struct device *dev) {} +#endif + +int adau17x1_suspend(struct snd_soc_codec *codec) +{ + codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_suspend); + +int adau17x1_resume(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + if (adau->spi_mode) + adau17x1_spi_mode(codec->dev); + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + regcache_sync(adau->regmap); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_resume); + +int adau17x1_bus_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, bool spi_mode) +{ + struct adau *adau; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau = devm_kzalloc(dev, sizeof(*adau), GFP_KERNEL); + if (!adau) + return -ENOMEM; + + adau->regmap = regmap; + adau->spi_mode = spi_mode; + adau->type = type; + adau->sysclk_div = 1; + + dev_set_drvdata(dev, adau); + + if (spi_mode) + adau17x1_spi_mode(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_bus_probe); + +MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h new file mode 100644 index 0000000..d49442e --- /dev/null +++ b/sound/soc/codecs/adau17x1.h @@ -0,0 +1,148 @@ +#ifndef __ADAU17X1_H__ +#define __ADAU17X1_H__ + +#include <linux/regmap.h> +#include <linux/platform_data/adau17x1.h> + +enum adau17x1_type { + ADAU1361, + ADAU1761, + ADAU1381, + ADAU1781, +}; + +enum adau17x1_pll { + ADAU17X1_PLL, +}; + +enum adau17x1_pll_src { + ADAU17X1_PLL_SRC_MCLK, +}; + +enum adau17x1_clk_src { + ADAU17X1_CLK_SRC_MCLK, + ADAU17X1_CLK_SRC_PLL, +}; + +struct adau { + unsigned int sysclk; + unsigned int sysclk_div; + unsigned int pll_freq; + + enum adau17x1_clk_src clk_src; + enum adau17x1_type type; + bool spi_mode; + + unsigned int dai_fmt; + + uint8_t pll_regs[6]; + + bool master; + + unsigned int tdm_dac_slot; + unsigned int tdm_adc_slot; + + bool dsp_playback_bypass; + bool dsp_capture_bypass; + + struct regmap *regmap; +}; + +int adau17x1_probe(struct snd_soc_codec *codec); +int adau17x1_add_routes(struct snd_soc_codec *codec); +int adau17x1_bus_probe(struct device *dev, + struct regmap *regmap, enum adau17x1_type type, + bool spi_mode); +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias); +bool adau17x1_readable_register(struct device *dev, unsigned int reg); +bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +int adau17x1_suspend(struct snd_soc_codec *codec); +int adau17x1_resume(struct snd_soc_codec *codec); + +extern const struct snd_soc_dai_ops adau17x1_dai_ops; + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware); +bool adau17x1_has_dsp(struct adau *adau); +int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +#define ADAU17X1_DSP_MUX_ENUM(xname, xenum) \ + SOC_DAPM_ENUM_EXT(xname, xenum, \ + adau17x1_dsp_mux_enum_get, adau17x1_dsp_mux_enum_put) + +#define DECLARE_ADAU17X1_AIFOUT_MUX(name, text) \ +static const char * const adau17x1_ ## name ## _aifout_text[] = { \ + text, \ + "DSP", \ +}; \ +static const SOC_ENUM_SINGLE_DECL(adau17x1_ ## name ## _aifout_mux_enum, \ + ADAU17X1_SERIAL_OUTPUT_ROUTE, 0, adau17x1_ ## name ## _aifout_text); \ +static const struct snd_kcontrol_new adau17x1_ ## name ## _aifout_mux = \ + ADAU17X1_DSP_MUX_ENUM("AIFOUT Capture Mux", \ + adau17x1_ ## name ## _aifout_mux_enum); \ +static const struct snd_soc_dapm_widget adau17x1_ ## name ## _dsp_widget = \ + SND_SOC_DAPM_VIRT_MUX("AIFOUT Capture Mux", SND_SOC_NOPM, 0, 0, \ + &adau17x1_ ## name ## _aifout_mux) \ + +#define ADAU17X1_CLOCK_CONTROL 0x4000 +#define ADAU17X1_PLL_CONTROL 0x4002 +#define ADAU17X1_REC_POWER_MGMT 0x4009 +#define ADAU17X1_MICBIAS 0x4010 +#define ADAU17X1_SERIAL_PORT0 0x4015 +#define ADAU17X1_SERIAL_PORT1 0x4016 +#define ADAU17X1_CONVERTER0 0x4017 +#define ADAU17X1_CONVERTER1 0x4018 +#define ADAU17X1_LEFT_INPUT_DIGITAL_VOL 0x401a +#define ADAU17X1_RIGHT_INPUT_DIGITAL_VOL 0x401b +#define ADAU17X1_ADC_CONTROL 0x4019 +#define ADAU17X1_PLAY_POWER_MGMT 0x4029 +#define ADAU17X1_DAC_CONTROL0 0x402a +#define ADAU17X1_DAC_CONTROL1 0x402b +#define ADAU17X1_DAC_CONTROL2 0x402c +#define ADAU17X1_SERIAL_PORT_PAD 0x402d +#define ADAU17X1_CONTROL_PORT_PAD0 0x402f +#define ADAU17X1_CONTROL_PORT_PAD1 0x4030 +#define ADAU17X1_DSP_SAMPLING_RATE 0x40eb +#define ADAU17X1_SERIAL_INPUT_ROUTE 0x40f2 +#define ADAU17X1_SERIAL_OUTPUT_ROUTE 0x40f3 +#define ADAU17X1_DSP_ENABLE 0x40f5 +#define ADAU17X1_DSP_RUN 0x40f6 +#define ADAU17X1_SERIAL_SAMPLING_RATE 0x40f8 + +#define ADAU17X1_SERIAL_PORT0_BCLK_POL BIT(4) +#define ADAU17X1_SERIAL_PORT0_LRCLK_POL BIT(3) +#define ADAU17X1_SERIAL_PORT0_MASTER BIT(0) + +#define ADAU17X1_SERIAL_PORT1_DELAY1 0x00 +#define ADAU17X1_SERIAL_PORT1_DELAY0 0x01 +#define ADAU17X1_SERIAL_PORT1_DELAY8 0x02 +#define ADAU17X1_SERIAL_PORT1_DELAY16 0x03 +#define ADAU17X1_SERIAL_PORT1_DELAY_MASK 0x03 + +#define ADAU17X1_CLOCK_CONTROL_INFREQ_MASK 0x6 +#define ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL BIT(3) +#define ADAU17X1_CLOCK_CONTROL_SYSCLK_EN BIT(0) + +#define ADUA_SERIAL_PORT1_BCLK32 (0x0 << 5) +#define ADUA_SERIAL_PORT1_BCLK48 (0x1 << 5) +#define ADUA_SERIAL_PORT1_BCLK64 (0x2 << 5) +#define ADUA_SERIAL_PORT1_BCLK128 (0x3 << 5) +#define ADUA_SERIAL_PORT1_BCLK256 (0x4 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK_MASK (0x7 << 5) + +#define ADUA_SERIAL_PORT0_STEREO (0x0 << 1) +#define ADUA_SERIAL_PORT0_TDM4 (0x1 << 1) +#define ADUA_SERIAL_PORT0_TDM8 (0x2 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM_MASK (0x3 << 1) +#define ADAU17X1_SERIAL_PORT0_PULSE_MODE BIT(5) + +#define ADAU17X1_CONVERTER0_DAC_PAIR(x) (((x) - 1) << 5) +#define ADAU17X1_CONVERTER0_DAC_PAIR_MASK (0x3 << 5) +#define ADAU17X1_CONVERTER1_ADC_PAIR(x) ((x) - 1) +#define ADAU17X1_CONVERTER1_ADC_PAIR_MASK 0x3 + +#endif -- 1.8.0

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[alsa-devel] 32337ea "ASoC: warn if we will overwrite the DAI stream widgets"
by Stephen Warren 29 Aug '13

29 Aug '13

Russell, if the warning in commit 32337ea "ASoC: warn if we will overwrite the DAI stream widgets" fires, what do I need to do to the ASoC driver to fix it? (I'm deliberately ignoring any other issues with this patch, and looking only to discover what is apparently wrong with my driver that's triggering this...). Steev Klimaszewski reported this warning to me, and I've repro'd it running next-20130827 on a couple of Tegra systems (TrimSlice and Seaboard at least), using sound/soc/tegra/tegra_{trimslice,wm8903}.c, and associated ASoC components. For the record, the commit description is: > ASoC: warn if we will overwrite the DAI stream widgets > > The DAI stream widgets can end up being created multiple times if the > CPU DAI and DMA/platform components share the same struct device. This > bug has been reported in various forms to the ASoC maintainer(s) several > times, yet they refuse to acknowledge this: > > > Essentially all the dmaengine based platforms in mainline use a shared > > device for DMA and DAI; I'm fairly sure someone would have mentioned if > > there were problems. > > > > As you have been repeatedly told the Kirkwood drivers are the first > > drivers submitted to mainline which use DPCM and therefore it is not > > surprising that there are a few issues which need to be worked through, > > there were a few revisions to the framework which went in as a result of > > review during the mainline merge. The problem you are seeing here is > > due to this being the first platform with a *shared* device to use DPCM. > > This is inspite of stating multiple times that this has nothing to do > with DPCM what so ever. So, having tried to convince them over the > last month that this is the case, and them being obstructive and/or > obtuse, this is the only way left open to convince them that there is > a problem here. > > It is only through complete frustration over their inability to recognise > that there could be a problem that I am putting this into linux-next; > this is not something I want to do, but they leave me no other choice. > > If they are correct, then this patch will have no effect what so ever. > If I am correct, there will be warnings produced where a CPU DAI and > DMA/platform component shares the same struct device. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk>

2 1

Re: [alsa-devel] [PATCH V3] i2c: move of helpers into the core
by Grant Likely 28 Aug '13

28 Aug '13

On Thu, 22 Aug 2013 18:00:14 +0200, Wolfram Sang <wsa(a)the-dreams.de> wrote: > I2C of helpers used to live in of_i2c.c but experience (from SPI) shows > that it is much cleaner to have this in the core. This also removes a > circular dependency between the helpers and the core, and so we can > finally register child nodes in the core instead of doing this manually > in each driver. So, fix the drivers and documentation, too. > > Acked-by: Rob Herring <rob.herring(a)calxeda.com> > Reviewed-by: Felipe Balbi <balbi(a)ti.com> > Acked-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> > Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> Acked-by: Grant Likely <grant.likely(a)linaro.org> > --- > > V2->V3: Was trying to be too smart by only fixing includes needed. > Took a more general approach this time, converting of_i2c.h > to i2c.h in case i2c.h was not already there. Otherwise > remove it. Improved my build scripts and no build failures, > no complaints from buildbot as well. > > > Documentation/acpi/enumeration.txt | 1 - > arch/powerpc/platforms/44x/warp.c | 1 - > drivers/gpu/drm/tilcdc/tilcdc_slave.c | 1 - > drivers/gpu/drm/tilcdc/tilcdc_tfp410.c | 1 - > drivers/gpu/host1x/drm/output.c | 2 +- > drivers/i2c/busses/i2c-at91.c | 3 - > drivers/i2c/busses/i2c-cpm.c | 6 -- > drivers/i2c/busses/i2c-davinci.c | 2 - > drivers/i2c/busses/i2c-designware-platdrv.c | 2 - > drivers/i2c/busses/i2c-gpio.c | 3 - > drivers/i2c/busses/i2c-i801.c | 2 - > drivers/i2c/busses/i2c-ibm_iic.c | 4 - > drivers/i2c/busses/i2c-imx.c | 3 - > drivers/i2c/busses/i2c-mpc.c | 2 - > drivers/i2c/busses/i2c-mv64xxx.c | 3 - > drivers/i2c/busses/i2c-mxs.c | 3 - > drivers/i2c/busses/i2c-nomadik.c | 3 - > drivers/i2c/busses/i2c-ocores.c | 3 - > drivers/i2c/busses/i2c-octeon.c | 3 - > drivers/i2c/busses/i2c-omap.c | 3 - > drivers/i2c/busses/i2c-pnx.c | 3 - > drivers/i2c/busses/i2c-powermac.c | 9 +- > drivers/i2c/busses/i2c-pxa.c | 2 - > drivers/i2c/busses/i2c-s3c2410.c | 2 - > drivers/i2c/busses/i2c-sh_mobile.c | 2 - > drivers/i2c/busses/i2c-sirf.c | 3 - > drivers/i2c/busses/i2c-stu300.c | 2 - > drivers/i2c/busses/i2c-tegra.c | 3 - > drivers/i2c/busses/i2c-versatile.c | 2 - > drivers/i2c/busses/i2c-wmt.c | 3 - > drivers/i2c/busses/i2c-xiic.c | 3 - > drivers/i2c/i2c-core.c | 109 +++++++++++++++++++++- > drivers/i2c/i2c-mux.c | 3 - > drivers/i2c/muxes/i2c-arb-gpio-challenge.c | 1 - > drivers/i2c/muxes/i2c-mux-gpio.c | 1 - > drivers/i2c/muxes/i2c-mux-pinctrl.c | 1 - > drivers/media/platform/exynos4-is/fimc-is-i2c.c | 4 +- > drivers/media/platform/exynos4-is/fimc-is.c | 2 +- > drivers/media/platform/exynos4-is/media-dev.c | 1 - > drivers/of/Kconfig | 6 -- > drivers/of/Makefile | 1 - > drivers/of/of_i2c.c | 114 ----------------------- > drivers/staging/imx-drm/imx-tve.c | 2 +- > include/linux/i2c.h | 20 ++++ > include/linux/of_i2c.h | 46 --------- > sound/soc/fsl/imx-sgtl5000.c | 2 +- > sound/soc/fsl/imx-wm8962.c | 2 +- > 47 files changed, 138 insertions(+), 262 deletions(-) > delete mode 100644 drivers/of/of_i2c.c > delete mode 100644 include/linux/of_i2c.h > > diff --git a/Documentation/acpi/enumeration.txt b/Documentation/acpi/enumeration.txt > index d9be7a9..958266e 100644 > --- a/Documentation/acpi/enumeration.txt > +++ b/Documentation/acpi/enumeration.txt > @@ -238,7 +238,6 @@ An I2C bus (controller) driver does: > if (ret) > /* handle error */ > > - of_i2c_register_devices(adapter); > /* Enumerate the slave devices behind this bus via ACPI */ > acpi_i2c_register_devices(adapter); > > diff --git a/arch/powerpc/platforms/44x/warp.c b/arch/powerpc/platforms/44x/warp.c > index 4cfa499..534574a 100644 > --- a/arch/powerpc/platforms/44x/warp.c > +++ b/arch/powerpc/platforms/44x/warp.c > @@ -16,7 +16,6 @@ > #include <linux/interrupt.h> > #include <linux/delay.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > #include <linux/slab.h> > #include <linux/export.h> > > diff --git a/drivers/gpu/drm/tilcdc/tilcdc_slave.c b/drivers/gpu/drm/tilcdc/tilcdc_slave.c > index dfffaf0..a19f657 100644 > --- a/drivers/gpu/drm/tilcdc/tilcdc_slave.c > +++ b/drivers/gpu/drm/tilcdc/tilcdc_slave.c > @@ -16,7 +16,6 @@ > */ > > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/pinmux.h> > #include <linux/pinctrl/consumer.h> > #include <drm/drm_encoder_slave.h> > diff --git a/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c b/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > index 925c7cd..c38b56b 100644 > --- a/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > +++ b/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > @@ -16,7 +16,6 @@ > */ > > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/gpio.h> > #include <linux/of_gpio.h> > #include <linux/pinctrl/pinmux.h> > diff --git a/drivers/gpu/host1x/drm/output.c b/drivers/gpu/host1x/drm/output.c > index 8140fc6..137ae81 100644 > --- a/drivers/gpu/host1x/drm/output.c > +++ b/drivers/gpu/host1x/drm/output.c > @@ -9,7 +9,7 @@ > > #include <linux/module.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > > #include "drm.h" > > diff --git a/drivers/i2c/busses/i2c-at91.c b/drivers/i2c/busses/i2c-at91.c > index 6bb839b..fd05930 100644 > --- a/drivers/i2c/busses/i2c-at91.c > +++ b/drivers/i2c/busses/i2c-at91.c > @@ -28,7 +28,6 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <linux/platform_data/dma-atmel.h> > @@ -775,8 +774,6 @@ static int at91_twi_probe(struct platform_device *pdev) > return rc; > } > > - of_i2c_register_devices(&dev->adapter); > - > dev_info(dev->dev, "AT91 i2c bus driver.\n"); > return 0; > } > diff --git a/drivers/i2c/busses/i2c-cpm.c b/drivers/i2c/busses/i2c-cpm.c > index 2e1f7eb..b2b8aa9 100644 > --- a/drivers/i2c/busses/i2c-cpm.c > +++ b/drivers/i2c/busses/i2c-cpm.c > @@ -42,7 +42,6 @@ > #include <linux/dma-mapping.h> > #include <linux/of_device.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > #include <sysdev/fsl_soc.h> > #include <asm/cpm.h> > > @@ -681,11 +680,6 @@ static int cpm_i2c_probe(struct platform_device *ofdev) > dev_dbg(&ofdev->dev, "hw routines for %s registered.\n", > cpm->adap.name); > > - /* > - * register OF I2C devices > - */ > - of_i2c_register_devices(&cpm->adap); > - > return 0; > out_shut: > cpm_i2c_shutdown(cpm); > diff --git a/drivers/i2c/busses/i2c-davinci.c b/drivers/i2c/busses/i2c-davinci.c > index fa55605..62be3b3 100644 > --- a/drivers/i2c/busses/i2c-davinci.c > +++ b/drivers/i2c/busses/i2c-davinci.c > @@ -38,7 +38,6 @@ > #include <linux/slab.h> > #include <linux/cpufreq.h> > #include <linux/gpio.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > > #include <mach/hardware.h> > @@ -728,7 +727,6 @@ static int davinci_i2c_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "failure adding adapter\n"); > goto err_unuse_clocks; > } > - of_i2c_register_devices(adap); > > return 0; > > diff --git a/drivers/i2c/busses/i2c-designware-platdrv.c b/drivers/i2c/busses/i2c-designware-platdrv.c > index 4c5fada..27ea436 100644 > --- a/drivers/i2c/busses/i2c-designware-platdrv.c > +++ b/drivers/i2c/busses/i2c-designware-platdrv.c > @@ -35,7 +35,6 @@ > #include <linux/err.h> > #include <linux/interrupt.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/pm.h> > #include <linux/pm_runtime.h> > @@ -172,7 +171,6 @@ static int dw_i2c_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "failure adding adapter\n"); > return r; > } > - of_i2c_register_devices(adap); > acpi_i2c_register_devices(adap); > > pm_runtime_set_autosuspend_delay(&pdev->dev, 1000); > diff --git a/drivers/i2c/busses/i2c-gpio.c b/drivers/i2c/busses/i2c-gpio.c > index bc6e139..e5da9fe 100644 > --- a/drivers/i2c/busses/i2c-gpio.c > +++ b/drivers/i2c/busses/i2c-gpio.c > @@ -16,7 +16,6 @@ > #include <linux/platform_device.h> > #include <linux/gpio.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > > struct i2c_gpio_private_data { > struct i2c_adapter adap; > @@ -224,8 +223,6 @@ static int i2c_gpio_probe(struct platform_device *pdev) > if (ret) > goto err_add_bus; > > - of_i2c_register_devices(adap); > - > platform_set_drvdata(pdev, priv); > > dev_info(&pdev->dev, "using pins %u (SDA) and %u (SCL%s)\n", > diff --git a/drivers/i2c/busses/i2c-i801.c b/drivers/i2c/busses/i2c-i801.c > index 4ebceed..4296d17 100644 > --- a/drivers/i2c/busses/i2c-i801.c > +++ b/drivers/i2c/busses/i2c-i801.c > @@ -87,7 +87,6 @@ > #include <linux/slab.h> > #include <linux/wait.h> > #include <linux/err.h> > -#include <linux/of_i2c.h> > > #if (defined CONFIG_I2C_MUX_GPIO || defined CONFIG_I2C_MUX_GPIO_MODULE) && \ > defined CONFIG_DMI > @@ -1230,7 +1229,6 @@ static int i801_probe(struct pci_dev *dev, const struct pci_device_id *id) > goto exit_free_irq; > } > > - of_i2c_register_devices(&priv->adapter); > i801_probe_optional_slaves(priv); > /* We ignore errors - multiplexing is optional */ > i801_add_mux(priv); > diff --git a/drivers/i2c/busses/i2c-ibm_iic.c b/drivers/i2c/busses/i2c-ibm_iic.c > index 973f516..ff3caa0 100644 > --- a/drivers/i2c/busses/i2c-ibm_iic.c > +++ b/drivers/i2c/busses/i2c-ibm_iic.c > @@ -42,7 +42,6 @@ > #include <linux/io.h> > #include <linux/i2c.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > > #include "i2c-ibm_iic.h" > > @@ -759,9 +758,6 @@ static int iic_probe(struct platform_device *ofdev) > dev_info(&ofdev->dev, "using %s mode\n", > dev->fast_mode ? "fast (400 kHz)" : "standard (100 kHz)"); > > - /* Now register all the child nodes */ > - of_i2c_register_devices(adap); > - > return 0; > > error_cleanup: > diff --git a/drivers/i2c/busses/i2c-imx.c b/drivers/i2c/busses/i2c-imx.c > index e242797..bbbea6b 100644 > --- a/drivers/i2c/busses/i2c-imx.c > +++ b/drivers/i2c/busses/i2c-imx.c > @@ -50,7 +50,6 @@ > #include <linux/slab.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_data/i2c-imx.h> > > /** Defines ******************************************************************** > @@ -570,8 +569,6 @@ static int __init i2c_imx_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c_imx->adapter); > - > /* Set up platform driver data */ > platform_set_drvdata(pdev, i2c_imx); > > diff --git a/drivers/i2c/busses/i2c-mpc.c b/drivers/i2c/busses/i2c-mpc.c > index 7607dc0..9f2513d 100644 > --- a/drivers/i2c/busses/i2c-mpc.c > +++ b/drivers/i2c/busses/i2c-mpc.c > @@ -18,7 +18,6 @@ > #include <linux/sched.h> > #include <linux/init.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > #include <linux/slab.h> > > #include <linux/io.h> > @@ -691,7 +690,6 @@ static int fsl_i2c_probe(struct platform_device *op) > dev_err(i2c->dev, "failed to add adapter\n"); > goto fail_add; > } > - of_i2c_register_devices(&i2c->adap); > > return result; > > diff --git a/drivers/i2c/busses/i2c-mv64xxx.c b/drivers/i2c/busses/i2c-mv64xxx.c > index b1f42bf..8220322 100644 > --- a/drivers/i2c/busses/i2c-mv64xxx.c > +++ b/drivers/i2c/busses/i2c-mv64xxx.c > @@ -21,7 +21,6 @@ > #include <linux/of.h> > #include <linux/of_device.h> > #include <linux/of_irq.h> > -#include <linux/of_i2c.h> > #include <linux/clk.h> > #include <linux/err.h> > > @@ -689,8 +688,6 @@ mv64xxx_i2c_probe(struct platform_device *pd) > goto exit_free_irq; > } > > - of_i2c_register_devices(&drv_data->adapter); > - > return 0; > > exit_free_irq: > diff --git a/drivers/i2c/busses/i2c-mxs.c b/drivers/i2c/busses/i2c-mxs.c > index df8ff5a..62ed07d 100644 > --- a/drivers/i2c/busses/i2c-mxs.c > +++ b/drivers/i2c/busses/i2c-mxs.c > @@ -27,7 +27,6 @@ > #include <linux/stmp_device.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/dma-mapping.h> > #include <linux/dmaengine.h> > > @@ -701,8 +700,6 @@ static int mxs_i2c_probe(struct platform_device *pdev) > return err; > } > > - of_i2c_register_devices(adap); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-nomadik.c b/drivers/i2c/busses/i2c-nomadik.c > index 512dfe6..519df17 100644 > --- a/drivers/i2c/busses/i2c-nomadik.c > +++ b/drivers/i2c/busses/i2c-nomadik.c > @@ -24,7 +24,6 @@ > #include <linux/pm_runtime.h> > #include <linux/platform_data/i2c-nomadik.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/consumer.h> > > #define DRIVER_NAME "nmk-i2c" > @@ -1045,8 +1044,6 @@ static int nmk_i2c_probe(struct amba_device *adev, const struct amba_id *id) > goto err_add_adap; > } > > - of_i2c_register_devices(adap); > - > pm_runtime_put(&adev->dev); > > return 0; > diff --git a/drivers/i2c/busses/i2c-ocores.c b/drivers/i2c/busses/i2c-ocores.c > index 0e1f824..0a52b78 100644 > --- a/drivers/i2c/busses/i2c-ocores.c > +++ b/drivers/i2c/busses/i2c-ocores.c > @@ -24,7 +24,6 @@ > #include <linux/i2c-ocores.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > #include <linux/log2.h> > > struct ocores_i2c { > @@ -435,8 +434,6 @@ static int ocores_i2c_probe(struct platform_device *pdev) > if (pdata) { > for (i = 0; i < pdata->num_devices; i++) > i2c_new_device(&i2c->adap, pdata->devices + i); > - } else { > - of_i2c_register_devices(&i2c->adap); > } > > return 0; > diff --git a/drivers/i2c/busses/i2c-octeon.c b/drivers/i2c/busses/i2c-octeon.c > index 956fe32..b929ba2 100644 > --- a/drivers/i2c/busses/i2c-octeon.c > +++ b/drivers/i2c/busses/i2c-octeon.c > @@ -15,7 +15,6 @@ > #include <linux/interrupt.h> > #include <linux/kernel.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/delay.h> > #include <linux/sched.h> > #include <linux/slab.h> > @@ -599,8 +598,6 @@ static int octeon_i2c_probe(struct platform_device *pdev) > } > dev_info(i2c->dev, "version %s\n", DRV_VERSION); > > - of_i2c_register_devices(&i2c->adap); > - > return 0; > > out: > diff --git a/drivers/i2c/busses/i2c-omap.c b/drivers/i2c/busses/i2c-omap.c > index 142b694d..a9f0f80 100644 > --- a/drivers/i2c/busses/i2c-omap.c > +++ b/drivers/i2c/busses/i2c-omap.c > @@ -38,7 +38,6 @@ > #include <linux/clk.h> > #include <linux/io.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > #include <linux/slab.h> > #include <linux/i2c-omap.h> > @@ -1245,8 +1244,6 @@ omap_i2c_probe(struct platform_device *pdev) > dev_info(dev->dev, "bus %d rev%d.%d at %d kHz\n", adap->nr, > major, minor, dev->speed); > > - of_i2c_register_devices(adap); > - > pm_runtime_mark_last_busy(dev->dev); > pm_runtime_put_autosuspend(dev->dev); > > diff --git a/drivers/i2c/busses/i2c-pnx.c b/drivers/i2c/busses/i2c-pnx.c > index 5f39c6d..7b57d67 100644 > --- a/drivers/i2c/busses/i2c-pnx.c > +++ b/drivers/i2c/busses/i2c-pnx.c > @@ -23,7 +23,6 @@ > #include <linux/err.h> > #include <linux/clk.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > #define I2C_PNX_TIMEOUT_DEFAULT 10 /* msec */ > #define I2C_PNX_SPEED_KHZ_DEFAULT 100 > @@ -741,8 +740,6 @@ static int i2c_pnx_probe(struct platform_device *pdev) > goto out_irq; > } > > - of_i2c_register_devices(&alg_data->adapter); > - > dev_dbg(&pdev->dev, "%s: Master at %#8x, irq %d.\n", > alg_data->adapter.name, res->start, alg_data->irq); > > diff --git a/drivers/i2c/busses/i2c-powermac.c b/drivers/i2c/busses/i2c-powermac.c > index 8dc90da..1010f2e 100644 > --- a/drivers/i2c/busses/i2c-powermac.c > +++ b/drivers/i2c/busses/i2c-powermac.c > @@ -440,7 +440,9 @@ static int i2c_powermac_probe(struct platform_device *dev) > adapter->algo = &i2c_powermac_algorithm; > i2c_set_adapdata(adapter, bus); > adapter->dev.parent = &dev->dev; > - adapter->dev.of_node = dev->dev.of_node; > + > + /* Clear of_node to skip automatic registration of i2c child nodes */ > + adapter->dev.of_node = NULL; > rc = i2c_add_adapter(adapter); > if (rc) { > printk(KERN_ERR "i2c-powermac: Adapter %s registration " > @@ -450,9 +452,8 @@ static int i2c_powermac_probe(struct platform_device *dev) > > printk(KERN_INFO "PowerMac i2c bus %s registered\n", adapter->name); > > - /* Cannot use of_i2c_register_devices() due to Apple device-tree > - * funkyness > - */ > + /* Use custom child registration due to Apple device-tree funkyness */ > + adapter->dev.of_node = dev->dev.of_node; > i2c_powermac_register_devices(adapter, bus); > > return rc; > diff --git a/drivers/i2c/busses/i2c-pxa.c b/drivers/i2c/busses/i2c-pxa.c > index fbafed2..bc65014 100644 > --- a/drivers/i2c/busses/i2c-pxa.c > +++ b/drivers/i2c/busses/i2c-pxa.c > @@ -31,7 +31,6 @@ > #include <linux/i2c-pxa.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/err.h> > #include <linux/clk.h> > @@ -1185,7 +1184,6 @@ static int i2c_pxa_probe(struct platform_device *dev) > printk(KERN_INFO "I2C: Failed to add bus\n"); > goto eadapt; > } > - of_i2c_register_devices(&i2c->adap); > > platform_set_drvdata(dev, i2c); > > diff --git a/drivers/i2c/busses/i2c-s3c2410.c b/drivers/i2c/busses/i2c-s3c2410.c > index cab1c91..643426e 100644 > --- a/drivers/i2c/busses/i2c-s3c2410.c > +++ b/drivers/i2c/busses/i2c-s3c2410.c > @@ -36,7 +36,6 @@ > #include <linux/cpufreq.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > #include <linux/pinctrl/consumer.h> > > @@ -1154,7 +1153,6 @@ static int s3c24xx_i2c_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c->adap); > platform_set_drvdata(pdev, i2c); > > pm_runtime_enable(&pdev->dev); > diff --git a/drivers/i2c/busses/i2c-sh_mobile.c b/drivers/i2c/busses/i2c-sh_mobile.c > index debf745..aa1268f 100644 > --- a/drivers/i2c/busses/i2c-sh_mobile.c > +++ b/drivers/i2c/busses/i2c-sh_mobile.c > @@ -27,7 +27,6 @@ > #include <linux/platform_device.h> > #include <linux/interrupt.h> > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/err.h> > #include <linux/pm_runtime.h> > #include <linux/clk.h> > @@ -758,7 +757,6 @@ static int sh_mobile_i2c_probe(struct platform_device *dev) > "I2C adapter %d with bus speed %lu Hz (L/H=%x/%x)\n", > adap->nr, pd->bus_speed, pd->iccl, pd->icch); > > - of_i2c_register_devices(adap); > return 0; > > err_all: > diff --git a/drivers/i2c/busses/i2c-sirf.c b/drivers/i2c/busses/i2c-sirf.c > index a63c7d5..0ff22e2 100644 > --- a/drivers/i2c/busses/i2c-sirf.c > +++ b/drivers/i2c/busses/i2c-sirf.c > @@ -12,7 +12,6 @@ > #include <linux/slab.h> > #include <linux/platform_device.h> > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/clk.h> > #include <linux/err.h> > #include <linux/io.h> > @@ -366,8 +365,6 @@ static int i2c_sirfsoc_probe(struct platform_device *pdev) > > clk_disable(clk); > > - of_i2c_register_devices(adap); > - > dev_info(&pdev->dev, " I2C adapter ready to operate\n"); > > return 0; > diff --git a/drivers/i2c/busses/i2c-stu300.c b/drivers/i2c/busses/i2c-stu300.c > index d1a6b20..047546c 100644 > --- a/drivers/i2c/busses/i2c-stu300.c > +++ b/drivers/i2c/busses/i2c-stu300.c > @@ -17,7 +17,6 @@ > #include <linux/clk.h> > #include <linux/io.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > /* the name of this kernel module */ > #define NAME "stu300" > @@ -936,7 +935,6 @@ stu300_probe(struct platform_device *pdev) > platform_set_drvdata(pdev, dev); > dev_info(&pdev->dev, "ST DDC I2C @ %p, irq %d\n", > dev->virtbase, dev->irq); > - of_i2c_register_devices(adap); > > return 0; > } > diff --git a/drivers/i2c/busses/i2c-tegra.c b/drivers/i2c/busses/i2c-tegra.c > index 9aa1b60..c457cb4 100644 > --- a/drivers/i2c/busses/i2c-tegra.c > +++ b/drivers/i2c/busses/i2c-tegra.c > @@ -25,7 +25,6 @@ > #include <linux/interrupt.h> > #include <linux/delay.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > #include <linux/module.h> > #include <linux/clk/tegra.h> > @@ -802,8 +801,6 @@ static int tegra_i2c_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c_dev->adapter); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-versatile.c b/drivers/i2c/busses/i2c-versatile.c > index f3a8790..6bb3a89 100644 > --- a/drivers/i2c/busses/i2c-versatile.c > +++ b/drivers/i2c/busses/i2c-versatile.c > @@ -16,7 +16,6 @@ > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > > #define I2C_CONTROL 0x00 > #define I2C_CONTROLS 0x00 > @@ -108,7 +107,6 @@ static int i2c_versatile_probe(struct platform_device *dev) > ret = i2c_bit_add_numbered_bus(&i2c->adap); > if (ret >= 0) { > platform_set_drvdata(dev, i2c); > - of_i2c_register_devices(&i2c->adap); > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-wmt.c b/drivers/i2c/busses/i2c-wmt.c > index baaa7d1..c65da3d 100644 > --- a/drivers/i2c/busses/i2c-wmt.c > +++ b/drivers/i2c/busses/i2c-wmt.c > @@ -21,7 +21,6 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_address.h> > -#include <linux/of_i2c.h> > #include <linux/of_irq.h> > #include <linux/platform_device.h> > > @@ -439,8 +438,6 @@ static int wmt_i2c_probe(struct platform_device *pdev) > > platform_set_drvdata(pdev, i2c_dev); > > - of_i2c_register_devices(adap); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-xiic.c b/drivers/i2c/busses/i2c-xiic.c > index 3d0f052..8823db7 100644 > --- a/drivers/i2c/busses/i2c-xiic.c > +++ b/drivers/i2c/busses/i2c-xiic.c > @@ -40,7 +40,6 @@ > #include <linux/i2c-xiic.h> > #include <linux/io.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > #define DRIVER_NAME "xiic-i2c" > > @@ -752,8 +751,6 @@ static int xiic_i2c_probe(struct platform_device *pdev) > i2c_new_device(&i2c->adap, pdata->devices + i); > } > > - of_i2c_register_devices(&i2c->adap); > - > return 0; > > add_adapter_failed: > diff --git a/drivers/i2c/i2c-core.c b/drivers/i2c/i2c-core.c > index f32ca29..08ebd78 100644 > --- a/drivers/i2c/i2c-core.c > +++ b/drivers/i2c/i2c-core.c > @@ -23,7 +23,11 @@ > SMBus 2.0 support by Mark Studebaker <mdsxyz123(a)yahoo.com> and > Jean Delvare <khali(a)linux-fr.org> > Mux support by Rodolfo Giometti <giometti(a)enneenne.com> and > - Michael Lawnick <michael.lawnick.ext(a)nsn.com> */ > + Michael Lawnick <michael.lawnick.ext(a)nsn.com> > + OF support is copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > + (based on a previous patch from Jon Smirl <jonsmirl(a)gmail.com>) and > + (c) 2013 Wolfram Sang <wsa(a)the-dreams.de> > + */ > > #include <linux/module.h> > #include <linux/kernel.h> > @@ -35,7 +39,9 @@ > #include <linux/init.h> > #include <linux/idr.h> > #include <linux/mutex.h> > +#include <linux/of.h> > #include <linux/of_device.h> > +#include <linux/of_irq.h> > #include <linux/completion.h> > #include <linux/hardirq.h> > #include <linux/irqflags.h> > @@ -954,6 +960,104 @@ static void i2c_scan_static_board_info(struct i2c_adapter *adapter) > up_read(&__i2c_board_lock); > } > > +/* OF support code */ > + > +#if IS_ENABLED(CONFIG_OF) > +static void of_i2c_register_devices(struct i2c_adapter *adap) > +{ > + void *result; > + struct device_node *node; > + > + /* Only register child devices if the adapter has a node pointer set */ > + if (!adap->dev.of_node) > + return; > + > + dev_dbg(&adap->dev, "of_i2c: walking child nodes\n"); > + > + for_each_available_child_of_node(adap->dev.of_node, node) { > + struct i2c_board_info info = {}; > + struct dev_archdata dev_ad = {}; > + const __be32 *addr; > + int len; > + > + dev_dbg(&adap->dev, "of_i2c: register %s\n", node->full_name); > + > + if (of_modalias_node(node, info.type, sizeof(info.type)) < 0) { > + dev_err(&adap->dev, "of_i2c: modalias failure on %s\n", > + node->full_name); > + continue; > + } > + > + addr = of_get_property(node, "reg", &len); > + if (!addr || (len < sizeof(int))) { > + dev_err(&adap->dev, "of_i2c: invalid reg on %s\n", > + node->full_name); > + continue; > + } > + > + info.addr = be32_to_cpup(addr); > + if (info.addr > (1 << 10) - 1) { > + dev_err(&adap->dev, "of_i2c: invalid addr=%x on %s\n", > + info.addr, node->full_name); > + continue; > + } > + > + info.irq = irq_of_parse_and_map(node, 0); > + info.of_node = of_node_get(node); > + info.archdata = &dev_ad; > + > + if (of_get_property(node, "wakeup-source", NULL)) > + info.flags |= I2C_CLIENT_WAKE; > + > + request_module("%s%s", I2C_MODULE_PREFIX, info.type); > + > + result = i2c_new_device(adap, &info); > + if (result == NULL) { > + dev_err(&adap->dev, "of_i2c: Failure registering %s\n", > + node->full_name); > + of_node_put(node); > + irq_dispose_mapping(info.irq); > + continue; > + } > + } > +} > + > +static int of_dev_node_match(struct device *dev, void *data) > +{ > + return dev->of_node == data; > +} > + > +/* must call put_device() when done with returned i2c_client device */ > +struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > +{ > + struct device *dev; > + > + dev = bus_find_device(&i2c_bus_type, NULL, node, > + of_dev_node_match); > + if (!dev) > + return NULL; > + > + return i2c_verify_client(dev); > +} > +EXPORT_SYMBOL(of_find_i2c_device_by_node); > + > +/* must call put_device() when done with returned i2c_adapter device */ > +struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > +{ > + struct device *dev; > + > + dev = bus_find_device(&i2c_bus_type, NULL, node, > + of_dev_node_match); > + if (!dev) > + return NULL; > + > + return i2c_verify_adapter(dev); > +} > +EXPORT_SYMBOL(of_find_i2c_adapter_by_node); > +#else > +static void of_i2c_register_devices(struct i2c_adapter *adap) { } > +#endif /* CONFIG_OF */ > + > static int i2c_do_add_adapter(struct i2c_driver *driver, > struct i2c_adapter *adap) > { > @@ -1058,6 +1162,8 @@ static int i2c_register_adapter(struct i2c_adapter *adap) > > exit_recovery: > /* create pre-declared device nodes */ > + of_i2c_register_devices(adap); > + > if (adap->nr < __i2c_first_dynamic_bus_num) > i2c_scan_static_board_info(adap); > > @@ -1282,7 +1388,6 @@ void i2c_del_adapter(struct i2c_adapter *adap) > } > EXPORT_SYMBOL(i2c_del_adapter); > > - > /* ------------------------------------------------------------------------- */ > > int i2c_for_each_dev(void *data, int (*fn)(struct device *, void *)) > diff --git a/drivers/i2c/i2c-mux.c b/drivers/i2c/i2c-mux.c > index 7409ebb..797e311 100644 > --- a/drivers/i2c/i2c-mux.c > +++ b/drivers/i2c/i2c-mux.c > @@ -25,7 +25,6 @@ > #include <linux/i2c.h> > #include <linux/i2c-mux.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > > /* multiplexer per channel data */ > struct i2c_mux_priv { > @@ -185,8 +184,6 @@ struct i2c_adapter *i2c_add_mux_adapter(struct i2c_adapter *parent, > dev_info(&parent->dev, "Added multiplexed i2c bus %d\n", > i2c_adapter_id(&priv->adap)); > > - of_i2c_register_devices(&priv->adap); > - > return &priv->adap; > } > EXPORT_SYMBOL_GPL(i2c_add_mux_adapter); > diff --git a/drivers/i2c/muxes/i2c-arb-gpio-challenge.c b/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > index 210b6f7..b901638 100644 > --- a/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > +++ b/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > @@ -21,7 +21,6 @@ > #include <linux/i2c-mux.h> > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > #include <linux/platform_device.h> > #include <linux/slab.h> > diff --git a/drivers/i2c/muxes/i2c-mux-gpio.c b/drivers/i2c/muxes/i2c-mux-gpio.c > index 5a0ce00..128a981 100644 > --- a/drivers/i2c/muxes/i2c-mux-gpio.c > +++ b/drivers/i2c/muxes/i2c-mux-gpio.c > @@ -16,7 +16,6 @@ > #include <linux/module.h> > #include <linux/slab.h> > #include <linux/gpio.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > > struct gpiomux { > diff --git a/drivers/i2c/muxes/i2c-mux-pinctrl.c b/drivers/i2c/muxes/i2c-mux-pinctrl.c > index a43c0ce..859a6d2 100644 > --- a/drivers/i2c/muxes/i2c-mux-pinctrl.c > +++ b/drivers/i2c/muxes/i2c-mux-pinctrl.c > @@ -20,7 +20,6 @@ > #include <linux/i2c-mux.h> > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/consumer.h> > #include <linux/i2c-mux-pinctrl.h> > #include <linux/platform_device.h> > diff --git a/drivers/media/platform/exynos4-is/fimc-is-i2c.c b/drivers/media/platform/exynos4-is/fimc-is-i2c.c > index 617a798..9930556 100644 > --- a/drivers/media/platform/exynos4-is/fimc-is-i2c.c > +++ b/drivers/media/platform/exynos4-is/fimc-is-i2c.c > @@ -12,7 +12,7 @@ > > #include <linux/clk.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/platform_device.h> > #include <linux/pm_runtime.h> > #include <linux/slab.h> > @@ -67,8 +67,6 @@ static int fimc_is_i2c_probe(struct platform_device *pdev) > pm_runtime_enable(&pdev->dev); > pm_runtime_enable(&i2c_adap->dev); > > - of_i2c_register_devices(i2c_adap); > - > return 0; > } > > diff --git a/drivers/media/platform/exynos4-is/fimc-is.c b/drivers/media/platform/exynos4-is/fimc-is.c > index 967f6a9..2276fdc 100644 > --- a/drivers/media/platform/exynos4-is/fimc-is.c > +++ b/drivers/media/platform/exynos4-is/fimc-is.c > @@ -21,7 +21,7 @@ > #include <linux/interrupt.h> > #include <linux/kernel.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/of_irq.h> > #include <linux/of_address.h> > #include <linux/of_platform.h> > diff --git a/drivers/media/platform/exynos4-is/media-dev.c b/drivers/media/platform/exynos4-is/media-dev.c > index 19f556c..f8c66b4 100644 > --- a/drivers/media/platform/exynos4-is/media-dev.c > +++ b/drivers/media/platform/exynos4-is/media-dev.c > @@ -20,7 +20,6 @@ > #include <linux/of.h> > #include <linux/of_platform.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/pm_runtime.h> > #include <linux/types.h> > diff --git a/drivers/of/Kconfig b/drivers/of/Kconfig > index 80e5c13..78cc760 100644 > --- a/drivers/of/Kconfig > +++ b/drivers/of/Kconfig > @@ -48,12 +48,6 @@ config OF_IRQ > def_bool y > depends on !SPARC > > -config OF_I2C > - def_tristate I2C > - depends on I2C > - help > - OpenFirmware I2C accessors > - > config OF_NET > depends on NETDEVICES > def_bool y > diff --git a/drivers/of/Makefile b/drivers/of/Makefile > index 1f9c0c4..efd0510 100644 > --- a/drivers/of/Makefile > +++ b/drivers/of/Makefile > @@ -3,7 +3,6 @@ obj-$(CONFIG_OF_FLATTREE) += fdt.o > obj-$(CONFIG_OF_PROMTREE) += pdt.o > obj-$(CONFIG_OF_ADDRESS) += address.o > obj-$(CONFIG_OF_IRQ) += irq.o > -obj-$(CONFIG_OF_I2C) += of_i2c.o > obj-$(CONFIG_OF_NET) += of_net.o > obj-$(CONFIG_OF_SELFTEST) += selftest.o > obj-$(CONFIG_OF_MDIO) += of_mdio.o > diff --git a/drivers/of/of_i2c.c b/drivers/of/of_i2c.c > deleted file mode 100644 > index b667264..0000000 > --- a/drivers/of/of_i2c.c > +++ /dev/null > @@ -1,114 +0,0 @@ > -/* > - * OF helpers for the I2C API > - * > - * Copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > - * > - * Based on a previous patch from Jon Smirl <jonsmirl(a)gmail.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/i2c.h> > -#include <linux/irq.h> > -#include <linux/of.h> > -#include <linux/of_i2c.h> > -#include <linux/of_irq.h> > -#include <linux/module.h> > - > -void of_i2c_register_devices(struct i2c_adapter *adap) > -{ > - void *result; > - struct device_node *node; > - > - /* Only register child devices if the adapter has a node pointer set */ > - if (!adap->dev.of_node) > - return; > - > - dev_dbg(&adap->dev, "of_i2c: walking child nodes\n"); > - > - for_each_available_child_of_node(adap->dev.of_node, node) { > - struct i2c_board_info info = {}; > - struct dev_archdata dev_ad = {}; > - const __be32 *addr; > - int len; > - > - dev_dbg(&adap->dev, "of_i2c: register %s\n", node->full_name); > - > - if (of_modalias_node(node, info.type, sizeof(info.type)) < 0) { > - dev_err(&adap->dev, "of_i2c: modalias failure on %s\n", > - node->full_name); > - continue; > - } > - > - addr = of_get_property(node, "reg", &len); > - if (!addr || (len < sizeof(int))) { > - dev_err(&adap->dev, "of_i2c: invalid reg on %s\n", > - node->full_name); > - continue; > - } > - > - info.addr = be32_to_cpup(addr); > - if (info.addr > (1 << 10) - 1) { > - dev_err(&adap->dev, "of_i2c: invalid addr=%x on %s\n", > - info.addr, node->full_name); > - continue; > - } > - > - info.irq = irq_of_parse_and_map(node, 0); > - info.of_node = of_node_get(node); > - info.archdata = &dev_ad; > - > - if (of_get_property(node, "wakeup-source", NULL)) > - info.flags |= I2C_CLIENT_WAKE; > - > - request_module("%s%s", I2C_MODULE_PREFIX, info.type); > - > - result = i2c_new_device(adap, &info); > - if (result == NULL) { > - dev_err(&adap->dev, "of_i2c: Failure registering %s\n", > - node->full_name); > - of_node_put(node); > - irq_dispose_mapping(info.irq); > - continue; > - } > - } > -} > -EXPORT_SYMBOL(of_i2c_register_devices); > - > -static int of_dev_node_match(struct device *dev, void *data) > -{ > - return dev->of_node == data; > -} > - > -/* must call put_device() when done with returned i2c_client device */ > -struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > -{ > - struct device *dev; > - > - dev = bus_find_device(&i2c_bus_type, NULL, node, > - of_dev_node_match); > - if (!dev) > - return NULL; > - > - return i2c_verify_client(dev); > -} > -EXPORT_SYMBOL(of_find_i2c_device_by_node); > - > -/* must call put_device() when done with returned i2c_adapter device */ > -struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > -{ > - struct device *dev; > - > - dev = bus_find_device(&i2c_bus_type, NULL, node, > - of_dev_node_match); > - if (!dev) > - return NULL; > - > - return i2c_verify_adapter(dev); > -} > -EXPORT_SYMBOL(of_find_i2c_adapter_by_node); > - > -MODULE_LICENSE("GPL"); > diff --git a/drivers/staging/imx-drm/imx-tve.c b/drivers/staging/imx-drm/imx-tve.c > index a56797d..2d76fd4 100644 > --- a/drivers/staging/imx-drm/imx-tve.c > +++ b/drivers/staging/imx-drm/imx-tve.c > @@ -21,7 +21,7 @@ > #include <linux/clk.h> > #include <linux/clk-provider.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/regmap.h> > #include <linux/regulator/consumer.h> > #include <linux/spinlock.h> > diff --git a/include/linux/i2c.h b/include/linux/i2c.h > index e988fa9..2189189 100644 > --- a/include/linux/i2c.h > +++ b/include/linux/i2c.h > @@ -542,6 +542,26 @@ static inline int i2c_adapter_id(struct i2c_adapter *adap) > > #endif /* I2C */ > > +#if IS_ENABLED(CONFIG_OF) > +/* must call put_device() when done with returned i2c_client device */ > +extern struct i2c_client *of_find_i2c_device_by_node(struct device_node *node); > + > +/* must call put_device() when done with returned i2c_adapter device */ > +extern struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node); > + > +#else > + > +static inline struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > +{ > + return NULL; > +} > + > +static inline struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > +{ > + return NULL; > +} > +#endif /* CONFIG_OF */ > + > #if IS_ENABLED(CONFIG_ACPI_I2C) > extern void acpi_i2c_register_devices(struct i2c_adapter *adap); > #else > diff --git a/include/linux/of_i2c.h b/include/linux/of_i2c.h > deleted file mode 100644 > index cfb545c..0000000 > --- a/include/linux/of_i2c.h > +++ /dev/null > @@ -1,46 +0,0 @@ > -/* > - * Generic I2C API implementation for PowerPC. > - * > - * Copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > - * > - * 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. > - */ > - > -#ifndef __LINUX_OF_I2C_H > -#define __LINUX_OF_I2C_H > - > -#if defined(CONFIG_OF_I2C) || defined(CONFIG_OF_I2C_MODULE) > -#include <linux/i2c.h> > - > -extern void of_i2c_register_devices(struct i2c_adapter *adap); > - > -/* must call put_device() when done with returned i2c_client device */ > -extern struct i2c_client *of_find_i2c_device_by_node(struct device_node *node); > - > -/* must call put_device() when done with returned i2c_adapter device */ > -extern struct i2c_adapter *of_find_i2c_adapter_by_node( > - struct device_node *node); > - > -#else > -static inline void of_i2c_register_devices(struct i2c_adapter *adap) > -{ > - return; > -} > - > -static inline struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > -{ > - return NULL; > -} > - > -/* must call put_device() when done with returned i2c_adapter device */ > -static inline struct i2c_adapter *of_find_i2c_adapter_by_node( > - struct device_node *node) > -{ > - return NULL; > -} > -#endif /* CONFIG_OF_I2C */ > - > -#endif /* __LINUX_OF_I2C_H */ > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 3f726e4..f2fbde9 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -13,7 +13,7 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/clk.h> > #include <sound/soc.h> > > diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c > index 52a36a9..9fd7a65 100644 > --- a/sound/soc/fsl/imx-wm8962.c > +++ b/sound/soc/fsl/imx-wm8962.c > @@ -15,7 +15,7 @@ > > #include <linux/module.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/slab.h> > #include <linux/clk.h> > #include <sound/soc.h> > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

1 0

[alsa-devel] [PATCH] ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards
by Jean-Francois Moine 28 Aug '13

28 Aug '13

This patch permits the generation of the Kirkwood audio driver which may be used in the Dove boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 9e1970c..78ed4a4 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC - tristate "SoC Audio for the Marvell Kirkwood chip" - depends on ARCH_KIRKWOOD || COMPILE_TEST + tristate "SoC Audio for the Marvell Kirkwood and Dove chips" + depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

Re: [alsa-devel] Buffer size for ALSA USB PCM audio
by Alan Stern 27 Aug '13

27 Aug '13

On Tue, 27 Aug 2013, James Stone wrote: > This does not help at all with running jackd at lower latencies. > > With this patch, I get xruns at 128 frames/period or less with jackd > (accompanied by iso underrun messages), whereas with vanilla 3.10.9 > (which now has both yours and Clemen's patches applied), performance > is extremely good (64 frames per period possible). Was this was both recording and playback, or playback only? If you were doing both, what happens with just playback? Assuming you see the same problem with just playback, please collect a usbmon trace showing one or two of those underruns at say 64 frames/period. And for comparison, it would be nice to have a usbmon trace showing a couple of seconds of normal operation under vanilla 3.10.9. Alan Stern

1 0

Re: [alsa-devel] [PATCH] ASoC: codecs/sgtl5000.c: specify correct reg_stride in regmap_config.
by David Jander 27 Aug '13

27 Aug '13

On Sun, 25 Aug 2013 16:44:26 +0800 zengzhaoming <zengzm.kernel(a)gmail.com> wrote: > On 08-22/08:23, David Jander wrote: > > This makes the RB-Tree more compact and efficient. > > > > Signed-off-by: David Jander <david(a)protonic.nl> > > --- > > 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 760e8bf..ba5ca67 100644 > > --- a/sound/soc/codecs/sgtl5000.c > > +++ b/sound/soc/codecs/sgtl5000.c > > @@ -1480,6 +1480,7 @@ 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 > > Fabio has post same patch for this. > http://comments.gmane.org/gmane.linux.alsa.devel/109903 Oh. I didn't see that one, sorry. Fabio's patch is from July, 5th. It takes quite some time for this to hit mainline then.... Best regards, -- David Jander Protonic Holland.

2 1

[alsa-devel] [PATCH v11] ASoC: fsl: Add S/PDIF machine driver
by Nicolin Chen 27 Aug '13

27 Aug '13

This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v10->v11: * Use boolean properties for spdif-out/in switch instead of codec phandles. * Accordingly create dummy codec driver in probe() instead of DT nodes. .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-spdif.c | 152 ++++++++++++++++++++ 4 files changed, 199 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/imx-spdif.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 0000000..7d13479 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc..a708380 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e..e2aaff7 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..af5b553 --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,152 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct platform_device *spdif_pdev; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + spdif_pdev = of_find_device_by_node(spdif_np); + if (!spdif_pdev) { + dev_err(&pdev->dev, "failed to find S/PDIF device\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto end; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto end; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register spdif-dit failed (%d)\n", ret); + goto end; + } + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register spdif-dir failed (%d)\n", ret); + goto error_dit; + } + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + platform_device_unregister(data->rxdev); +error_dit: + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + platform_device_unregister(data->rxdev); + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- 1.7.1

4 6

[alsa-devel] [PATCH 1/8] ASoC: dapm: Fix marking widgets dirty when a route is added
by Lars-Peter Clausen 27 Aug '13

27 Aug '13

The current calls to dapm_mark_dirty() in snd_soc_dapm_add_path() are on a path that is only reached if the sink widget is either a mixer or a mux. Move the calls further up so they are called for all widget types. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d84bd0f..7e9afbc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2374,6 +2374,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, wsource->ext = 1; } + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + /* connect static paths */ if (control == NULL) { list_add(&path->list, &dapm->card->paths); @@ -2436,9 +2439,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, return 0; } - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); - return 0; err: kfree(path); -- 1.8.0

2 17

[alsa-devel] [PATCH 0/9] ALSA: compress offfload fixes
by Vinod Koul 27 Aug '13

27 Aug '13

Hi Takashi, Here is the patch series which fixes various issues being reported by users (out of tree sadly) The first three and and last one are marked to stable as would like these to be fixed in older kernels as well. It would be good if you can send them as fixes to linus for 3.11. Rest can go in the merge window Fixes: - using lock for all operation was a very bad idead. This is bad as some of the ioctls like drain, partial drain can be time consuming and thus prevent any other operation while these are ongoing like Pause, Stop or timestamp query, so fix this be removing bunch of ioctls not to use device lock. - Now we dont have lock for pointer updates so this maybe racy, so use lock for doing lowest level calculation. - As disscused on our sample rate problem, lets move to use rate values and I will fix the lib too. Since the driver are not upstream the impact of this change wont be huge. - Plus few fix like use snprintf, state chacks for pause, write etc.. Vinod Koul (9): ALSA: Compress - dont use lock for all ioctls ALSA: compress: use mutex in drain ASoC: compress: dont aquire lock for draining states ALSA: compress: use snprint instread of sprintf ALSA: compres: wakeup the poll thread on pause ALSA: compress: dont write when stream is paused ALSA: compress: allow write when stream is setup ALSA: compress: call pointer callback and updates under a lock ALSA: compress: use rate values for passing sampling rates include/sound/compress_driver.h | 2 + include/uapi/sound/compress_offload.h | 2 +- sound/core/compress_offload.c | 140 +++++++++++++++++++++++++------- sound/soc/soc-compress.c | 10 +++ 4 files changed, 122 insertions(+), 32 deletions(-) Thanks ~Vinod

3 39

[alsa-devel] hda: Is it ok to enable SND_HDA_I915 to support Haswell Gfx power well by default?
by Lin, Mengdong 27 Aug '13

27 Aug '13

Hi Takashi and David, Is it okay to enable SND_HDA_I915 by default? This would make it easy to upgrade kernel on Haswell. A user only needs to apply default configurations for the new kernel, and display audio can work automatically. The user need not have any idea about the display power well. Now the user needs to manually enable SND_HDA_I915, otherwise display audio won't work since Gfx driver may turn off the power well which make both the controller and codec lose power. Thanks Mengdong

3 4

[alsa-devel] [PATCH v2 1/4] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..2596e36 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,66 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEVLIST_LEN; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + struct hda_devlist_entry *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices].present_detect = !!(parm & AC_DE_PD); + dev_list[devices].eld_valid = !!(parm & AC_DE_ELDV); + dev_list[devices].inactive = !!(parm & AC_DE_IA); + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..316076e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -460,6 +464,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -795,6 +804,16 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ }; +/* display pin's device list entry */ +struct hda_devlist_entry { + unsigned int present_detect:1; + unsigned int eld_valid:1; + unsigned int inactive:1; +}; + +#define AC_MAX_DEVLIST_LEN 64 +#define AC_DEVLIST_LEN 0x3f + /* codec information */ struct hda_codec { struct hda_bus *bus; @@ -885,6 +904,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +992,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + struct hda_devlist_entry *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

3 5

[alsa-devel] Fw: trouble with AMD Hudson Azalia sound card, no sound, alsamixer installed but not found, no /etc/asound.conf file
by Julie Brunias 27 Aug '13

27 Aug '13

hello i contact the Alsa Development Team because I face a problem with sound under Ubuntu. My sound card AMD Hudson Azalia is not seen by phonon and alsa cannot play any sound. But I do see my sound card in lspci. I tried many stuff found on forums but still not working... Would you be so kind and help me to figure out what's wrong? My ubuntu is 12.04 with alsa-base and alsa-utils. I cannot access to alsamixer (even with root) "not found", and cannot access to cat /etc/asound.conf "not found". Everything in my Bios is enabled. Here is an output I made. http://www.alsa-project.org/db/?f=82c2830fb9e05e618677523a179726de01569667 Until now (since 10 years) I never had such problem. My kernel is ok, i already uninstalled and purged alsa-utils and reinstall the whole thing as described on a ubuntu forum, not solving the issue. That would be very cool from you if you could help. I saw many users on the net with this s**** sound card who face problems. Apparently only on Ubuntu. If I am not at the right place to ask for some help, would you mind to orient me to the right people? I receive no answer to my post on Ubuntu Forums and I hope you could help. Best regards and thx for all the good work! Tomski

1 0

[alsa-devel] [trivial PATCH] treewide: Fix printks with 0x%#
by Joe Perches 27 Aug '13

27 Aug '13

Using 0x%# emits 0x0x. Only one is necessary. Signed-off-by: Joe Perches <joe(a)perches.com> --- arch/parisc/kernel/signal.c | 2 +- drivers/clocksource/acpi_pm.c | 4 ++-- drivers/net/ethernet/sis/sis900.c | 2 +- mm/memory-failure.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/via82xx.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) diff --git a/arch/parisc/kernel/signal.c b/arch/parisc/kernel/signal.c index 940188d..35c5bf1 100644 --- a/arch/parisc/kernel/signal.c +++ b/arch/parisc/kernel/signal.c @@ -85,7 +85,7 @@ restore_sigcontext(struct sigcontext __user *sc, struct pt_regs *regs) err |= __copy_from_user(regs->iaoq, sc->sc_iaoq, sizeof(regs->iaoq)); err |= __copy_from_user(regs->iasq, sc->sc_iasq, sizeof(regs->iasq)); err |= __get_user(regs->sar, &sc->sc_sar); - DBG(2,"restore_sigcontext: iaoq is 0x%#lx / 0x%#lx\n", + DBG(2,"restore_sigcontext: iaoq is %#lx / %#lx\n", regs->iaoq[0],regs->iaoq[1]); DBG(2,"restore_sigcontext: r28 is %ld\n", regs->gr[28]); return err; diff --git a/drivers/clocksource/acpi_pm.c b/drivers/clocksource/acpi_pm.c index 6efe4d1..6eab889 100644 --- a/drivers/clocksource/acpi_pm.c +++ b/drivers/clocksource/acpi_pm.c @@ -200,14 +200,14 @@ static int __init init_acpi_pm_clocksource(void) if ((value2 < value1) && ((value2) < 0xFFF)) break; printk(KERN_INFO "PM-Timer had inconsistent results:" - " 0x%#llx, 0x%#llx - aborting.\n", + " %#llx, %#llx - aborting.\n", value1, value2); pmtmr_ioport = 0; return -EINVAL; } if (i == ACPI_PM_READ_CHECKS) { printk(KERN_INFO "PM-Timer failed consistency check " - " (0x%#llx) - aborting.\n", value1); + " (%#llx) - aborting.\n", value1); pmtmr_ioport = 0; return -ENODEV; } diff --git a/drivers/net/ethernet/sis/sis900.c b/drivers/net/ethernet/sis/sis900.c index eb4aea3..6c1e34c 100644 --- a/drivers/net/ethernet/sis/sis900.c +++ b/drivers/net/ethernet/sis/sis900.c @@ -1723,7 +1723,7 @@ static irqreturn_t sis900_interrupt(int irq, void *dev_instance) if(netif_msg_intr(sis_priv)) printk(KERN_DEBUG "%s: exiting interrupt, " - "interrupt status = 0x%#8.8x.\n", + "interrupt status = %#8.8x\n", net_dev->name, sr32(isr)); spin_unlock (&sis_priv->lock); diff --git a/mm/memory-failure.c b/mm/memory-failure.c index 09ae111..29d3f38 100644 --- a/mm/memory-failure.c +++ b/mm/memory-failure.c @@ -1267,7 +1267,7 @@ void memory_failure_queue(unsigned long pfn, int trapno, int flags) if (kfifo_put(&mf_cpu->fifo, &entry)) schedule_work_on(smp_processor_id(), &mf_cpu->work); else - pr_err("Memory failure: buffer overflow when queuing memory failure at 0x%#lx\n", + pr_err("Memory failure: buffer overflow when queuing memory failure at %#lx\n", pfn); spin_unlock_irqrestore(&mf_cpu->lock, proc_flags); put_cpu_var(memory_failure_cpu); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index ca8929b..61262f3 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1842,7 +1842,7 @@ static int snd_ensoniq_create_gameport(struct ensoniq *ensoniq, int dev) default: if (!request_region(io_port, 8, "ens137x: gameport")) { - printk(KERN_WARNING "ens137x: gameport io port 0x%#x in use\n", + printk(KERN_WARNING "ens137x: gameport io port %#x in use\n", io_port); return -EBUSY; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 3c511d0..5ae6f04 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1940,7 +1940,7 @@ static int snd_via686_create_gameport(struct via82xx *chip, unsigned char *legac r = request_region(JOYSTICK_ADDR, 8, "VIA686 gameport"); if (!r) { - printk(KERN_WARNING "via82xx: cannot reserve joystick port 0x%#x\n", + printk(KERN_WARNING "via82xx: cannot reserve joystick port %#x\n", JOYSTICK_ADDR); return -EBUSY; }

3 3

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 27 Aug '13

27 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also remove dev_set_drvdata from the remove function sound/soc/fsl/fsl_spdif.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42a4382..a8ef46a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1184,7 +1184,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; + return ret; } ret = imx_pcm_dma_init(pdev); @@ -1197,8 +1197,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) error_component: snd_soc_unregister_component(&pdev->dev); -error_dev: - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -1207,7 +1205,6 @@ static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.1.2

3 6

[alsa-devel] [PATCH v3 4/4] ALSA: hda - add device entry and inactive flag to unsolicited response
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds two fields to unsolicited response, according to spec HDA040-A: - Device Entry (bit 20:15) - Inactive (bit 2) and show the info in debug message. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b838c70..7aa9870 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 87ca984..895a0d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 3/4] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 2/4] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..a8cb22e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + u8 dev_list[AC_MAX_DEV_LIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEV_LIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + !!(dev_list[i] & AC_DE_PD), + !!(dev_list[i] & AC_DE_ELDV), + !!(dev_list[i] & AC_DE_IA)); + } +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 1/4] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..5b6c4e3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEV_LIST_LEN_MASK; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices] = (u8)parm; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..b838c70 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -352,6 +356,10 @@ enum { #define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ #define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + /* * Control Parameters */ @@ -460,6 +468,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -885,6 +898,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +986,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

1 0

[alsa-devel] Support new HDA codec
by Jorge Fernandez Monteagudo 26 Aug '13

26 Aug '13

Hi all, This is my first email to the devel mailing list. I'm asking the same question in the user mailing list but I suspect that I'll have to add some code in order to get this hda codec from Wolfson to work. Attached is the log I get from alsa-info.sh. This codec has three output paths: 1) Node 0x06, DAC1, and Node 0x11, Port A, the stereo headphone playback, which is the default one and it seems it works because I can see the stream set to 5 when a sound is played. 2) Node 0x04, SPDIF Tx1, and Node 0x0f, SPDIF OUT, the SPDIF transmit functionality provides an IEC-60958-3 compatible S/PIDF output which I would like to enable but without luck yet. I can enable the node 0x04, with hda-verb, and the spdif amplifier receives spdif frames but without audio info. 3) Node 0x07, DAC2, and Node 0x12, Port G, the stereo line playback with a differential stereo output wich I would like to enable but without luck again. I just can mute/unmute the node but without effect in the outputs. I would like to know if I have to develop a patch for this device or with the attached log info I can expect to use it with the current code in kernel. Now, with alsamixer and other config tools I can't see any control related to the outputs for 2) and 3)... Any hint is welcome! Thanks Jorge --- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Tue Feb 21 09:44:41 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 11.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 11.10" !!DMI Information !!--------------- Manufacturer: nVidia Product Name: MMP9-ION Product Version: To Be Filled By O.E.M. !!Kernel Information !!------------------ Kernel release: 3.0.0-12-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb84 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-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,-1 beep_mode : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -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,-1,-1 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -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,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Wolfson Microelectronics ID 8800 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x1aec8800 Subsystem Id: 0x00000100 Revision Id: 0x100301 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=1 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 Node 0x02 [Audio Input] wcaps 0x1005d1: Stereo Device: name="HDA Generic", type="Audio", device=0 Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x1ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=02, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x03 [Audio Input] wcaps 0x1005d1: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=03, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Output] wcaps 0x2d1: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=04, enabled=0 Processing caps: benign=0, ncoeff=0 Node 0x05 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Copyright Digital category: 0x0 PCM: rates [0x1f0]: 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=05, enabled=0 Connection: 1 0x10 Processing caps: benign=0, ncoeff=0 Node 0x06 [Audio Output] wcaps 0x495: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="HDA Generic", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x57 0x57] Converter: stream=5, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=06, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x07 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=07, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x08 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=08, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x09 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x0b* 0x0e Node 0x0a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x18 0x18] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x98 0x98] Connection: 1 0x0d Node 0x0c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000064: IN Detect Balanced Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=0c, enabled=0 Node 0x0d [Pin Complex] wcaps 0x400083: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001767: IN Detect Balanced Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=0d, enabled=0 Node 0x0e [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000020: IN Pin Default 0x10a600f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x014190f0: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=0f, enabled=0 Connection: 1 0x04 Node 0x10 [Pin Complex] wcaps 0x400281: Stereo Digital Pincap 0x00000020: IN Pin Default 0x01c520f0: [Jack] SPDIF In at Ext Rear Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=10, enabled=0 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=11, enabled=0 Connection: 1 0x06 Node 0x12 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016121: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=12, enabled=0 Connection: 1 0x07 Node 0x13 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011022: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=13, enabled=0 Connection: 1 0x08 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x04, nsteps=0x04, stepsize=0x17, mute=0 Amp-Out vals: [0x04] Node 0x15 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=2 Digital: Digital category: 0x0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=15, enabled=0 Connection: 1 0x16 Processing caps: benign=0, ncoeff=0 Node 0x16 [Pin Complex] wcaps 0x40030b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000030: IN OUT Pin Default 0x10a601f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Connection: 1 0x17 Node 0x17 [Audio Output] wcaps 0x291: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=17, enabled=0 --endcollapse-- !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 20' Mixer name : 'Wolfson Microelectronics ID 8800' Components : 'HDA:1aec8800,00000100,00100301' Controls : 3 Simple ctrls : 2 Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 87 [69%] [-20.00dB] [on] Front Right: Playback 87 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.2 { iface MIXER name 'Master Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -2000 dbvalue.1 -2000 } } control.3 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 5 Feb 21 09:38 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Feb 21 09:38 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Feb 21 09:38 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 2 Feb 21 09:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Feb 21 09:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Feb 21 09:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Feb 21 09:38 . drwxr-xr-x 3 root root 180 Feb 21 09:38 .. lrwxrwxrwx 1 root root 12 Feb 21 09:38 pci-0000:00:08.0 -> ../controlC0 Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. 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4 6

[alsa-devel] [PATCH] ASoC: fsi: reserve prefetch period on DMA transferring
by Kuninori Morimoto 26 Aug '13

26 Aug '13

Current FSI is supporting DMAEngine transfer, but, it needs to use work queue. Therefore, DMA transfer settings might be late if there is heavy task. This patch reserves next period beforehand on DMA transfer function. Android sound will be breaking up without this patch. Tested-by: Tomohito Esaki <etom(a)igel.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 51 +++++++++++++++++++++++++++++++++------------------ 1 file changed, 33 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4fc2a2a..c608c22 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -235,6 +235,8 @@ struct fsi_stream { struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int loop_cnt; + int additional_pos; }; struct fsi_clk { @@ -1383,6 +1385,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ + io->additional_pos = 0; io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -1399,11 +1403,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) { struct snd_pcm_runtime *runtime = io->substream->runtime; + int period = io->period_pos + additional; - return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); + if (period >= runtime->periods) + period = 0; + + return io->dma + samples_to_bytes(runtime, period * io->period_samples); } static void fsi_dma_complete(void *data) @@ -1415,7 +1423,7 @@ static void fsi_dma_complete(void *data) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io), + dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; @@ -1441,7 +1449,7 @@ static void fsi_dma_do_work(struct work_struct *work) struct snd_pcm_runtime *runtime; enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len; + int len, i; dma_addr_t buf; if (!fsi_stream_is_working(fsi, io)) @@ -1451,26 +1459,33 @@ static void fsi_dma_do_work(struct work_struct *work) runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); - buf = fsi_dma_get_area(io); - dma_sync_single_for_device(dai->dev, buf, len, dir); + for (i = 0; i < io->loop_cnt; i++) { + buf = fsi_dma_get_area(io, io->additional_pos); - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } + dma_sync_single_for_device(dai->dev, buf, len, dir); - desc->callback = fsi_dma_complete; - desc->callback_param = io; + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; + desc->callback = fsi_dma_complete; + desc->callback_param = io; + + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + return; + } + + dma_async_issue_pending(io->chan); + + io->additional_pos = 1; } - dma_async_issue_pending(io->chan); + io->loop_cnt = 1; /* * FIXME -- 1.7.9.5

2 1

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] [RFC PATCH 1/3] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 26 Aug '13

26 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..fb58291 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,63 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEVLIST_LEN; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + union hda_devlist_entry *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + devices = 0; + + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices].value = parm & 0x7; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..736b55a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -795,6 +799,19 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ }; +/* display pin's device list entry */ +union hda_devlist_entry { + unsigned int value; + struct { + unsigned int present_detect:1; + unsigned int eld_valid:1; + unsigned int inactive:1; + } bits; +}; + +#define AC_MAX_DEVLIST_LEN 64 +#define AC_DEVLIST_LEN 0x3f + /* codec information */ struct hda_codec { struct hda_bus *bus; @@ -885,6 +902,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +990,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + union hda_devlist_entry *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

3 3

[alsa-devel] [PATCH v2 4/4] ALSA: hda - add device entry and inactive flag to unsolicited response
by mengdong.lin＠intel.com 26 Aug '13

26 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds two fields to unsolicited response, according to spec HDA040-A: - Device Entry (bit 20:15) - Inactive (bit 2) and show the info in debug message. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 316076e..c578f5e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 87ca984..895a0d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v2 3/4] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 26 Aug '13

26 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v2 2/4] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 26 Aug '13

26 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..1806250 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,38 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + struct hda_devlist_entry dev_list[AC_MAX_DEVLIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEVLIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %ld, ELDV = %ld, IA = %ld\n", i, + dev_list[i].present_detect, dev_list[i].eld_valid, + dev_list[i].inactive); + } + + return 0; + +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +783,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v5 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 24 Aug '13

24 Aug '13

Changelog: v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 76 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1336 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1817 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

6 36

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 24 Aug '13

24 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42a4382..bf58086 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1184,7 +1184,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; + return ret; } ret = imx_pcm_dma_init(pdev); @@ -1197,8 +1197,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) error_component: snd_soc_unregister_component(&pdev->dev); -error_dev: - dev_set_drvdata(&pdev->dev, NULL); return ret; } -- 1.8.1.2

1 1

[alsa-devel] [PATCH] ASoC: simple-card: Provide owner and MODULE_ALIAS()
by Fabio Estevam 23 Aug '13

23 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add .owner field and also MODULE_ALIAS(), so that auto module loading can work. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6cf8355..8c49147 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .owner = THIS_MODULE, }, .probe = asoc_simple_card_probe, .remove = asoc_simple_card_remove, @@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = { module_platform_driver(asoc_simple_card); +MODULE_ALIAS("platform:asoc-simple-card"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ASoC Simple Sound Card"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 1.8.1.2

1 0

[alsa-devel] [PATCH V3] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 23 Aug '13

23 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- I removed RT5640_[LR]_MUTE bits control in rt5640_hp_event. And moved the required delay for hp depop to rt5640_hp_post_event. This delay time is for waitting for the codec finish the depop process. You will not haer anything if you play a very short sound without this delay. So I put it in post_event. And I think it is fine if hp is unmute during playback without this delay. I use rt5640->hp_mute to record the "HP Amp"'s status. However, I think it is better to use a dapm API for that purpose. But I don't know if there is a suitable API for it. Can anyone give me some advice? Thanks in advance. --- sound/soc/codecs/rt5640.c | 217 ++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 12 +++ 2 files changed, 175 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..c26a8f8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + rt5640->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(10000, 15000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); @@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, rt5640_dai, ARRAY_SIZE(rt5640_dai)); if (ret < 0) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..5e8df25a 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,7 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool hp_mute; }; #endif -- 1.8.1.1.439.g50a6b54

3 4

Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 23 Aug '13

23 Aug '13

[Please do not drop the mailing list address] On 19.10.2012 15:45, Didier 'Ptitjes' Villevalois wrote: > On ven., 2012-10-19 at 15:20 +0200, Daniel Mack wrote: >> On 19.10.2012 15:09, Didier 'Ptitjes' Villevalois wrote: >>> Hello, >>> >>> I've bought a Reloop Play usb soundcard and I'd like to make it work >>> with ALSA. >>> >>> It is a 4-channels USB soundcard. It is currently correctly recognized >>> by ALSA as a 4-channels soundcard and I can correctly open the channels >>> and feed them with audio. However, no audio is output by the box. >> >> You're most probably affected by a regression in the driver that was >> fixes in 3.6.0 and 3.5.5 (and all later kernels). Could you please try >> one of these and tell us if that works? > > Sure, I'll do that now (I'm currently using 3.5.4 + mbp8,3 patches). > I guess 3.5.7 is OK ? I'm building it now and will reboot on that. Any kernel >= 3.6.0 and 3.5.5 should be ok. >>> It seems to contain a Texas Instrument TUSB3200AC chipset (same chipset >>> as the Maya44 USB and Hercules DJ console). I guess the problem is a >>> firmware problem but I don't know what to do about it. >>> >>> I've got a MacOS X driver that works correctly, if that of any help. (I >>> suppose the Windows driver works too but I don't have any Windows at >>> hand.) >> >> Did you install any vendor driver on OS X or does it just work with the >> one that ships with the OS? > > Yep, I had to install the Vendor-provided driver. What was the effect without that special driver? Could you see the device listed in the AudioMIDISetup? Could you stream audio? IOW, is the effect you see there comparable to how your Linux box behaves? You could try and remove that driver again and see what happens. > But it seems the > built-in USB soundcard support on MacOS is very poor. No it's not. In fact, it's actually very good. Vendors who keep their hardware compliant to the USB audio spec don't have to ship any driver at all. And this is how it should be on Linux, too. > Also the mpkg install file (inside the provided install dmg file) was > called something like "firmware0_and_driver.mpkg". This is why I > suspected a firmware problem. However I tried to look at the package > content and did not find any .bin file or whatever could be a standalone > firmware file. Maybe it is embedded in the driver install executable. Maybe, but they could also write the firmware permanently to the device once with that tool. That would be perfect for Linux users. Daniel

3 10

[alsa-devel] [PATCH 1/2] ASoC: pcm1792a: Remove empty capture DAI stub
by Mark Brown 23 Aug '13

23 Aug '13

From: Mark Brown <broonie(a)linaro.org> These intialisations are just what will be done for static data anyway so remove them. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 72cf835..c57d3a5 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -162,10 +162,6 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .channels_max = 2, .rates = PCM1792A_RATES, .formats = PCM1792A_FORMATS, }, - .capture = { - .channels_min = 0, - .channels_max = 0, - }, .ops = &pcm1792a_dai_ops, }; -- 1.8.4.rc2

2 3

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Fix missing audmux external port setup check
by Andreas Pretzsch 23 Aug '13

23 Aug '13

Return value of imx_audmux_v2_configure_port(ext_port, ...) was not assigned. Following check tested the preceding call (int_port). Signed-off-by: Andreas Pretzsch <apr(a)cn-eng.de> --- Fix not tested (no matching hardware), only noticed during code reading. Patch against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm8962 sound/soc/fsl/imx-wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a9..e018632 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -192,7 +192,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 23 Aug '13

23 Aug '13

The following changes since commit b36f4be3de1b123d8601de062e7dbfc904f305fb: Linux 3.11-rc6 (2013-08-18 14:36:53 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12 for you to fetch changes up to 43bcb402f84fe459102120b4e2d28d7117f16cd0: Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next (2013-08-22 14:29:02 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.12 - DAPM is now mandatory for CODEC drivers in order to avoid the repeated regressions in the special cases for non-DAPM CODECs and make it easier to integrate with other components on boards. All existing drivers have had some level of DAPM support added. - A lot of cleanups in DAPM plus support for maintaining controls in a specific state while a DAPM widget all contributed by Lars-Peter Clausen. - Core helpers for bitbanged AC'97 reset from Markus Pargmann. - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson Microelectronics WM8997. - Support for building drivers that can support it cross-platform for compile test. ---------------------------------------------------------------- Axel Lin (1): ASoC: adav80x: Add module device table for adav801 Bard Liao (1): ASoC: rt5640: remove unused mux Bo Shen (7): 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 ASoC: wm8904: fix the typo error for LINER Mux ASoC: atmel: add wm8904 based audio machine driver Charles Keepax (6): mfd: arizona: Add GPIO control register bit definitions ASoC: arizona: Add signal activity output for DRC ASoC: arizona: Add default case to silence build warning ASoC: wm5110: Correct input OSR bits for wm5110 ASoC: arizona: Add widget<->mux route into mux route macro ASoC: wm8997: Initial CODEC driver Dan Carpenter (2): ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET ASoC: dapm: using freed pointer in dapm_kcontrol_add_widget() Daniel Mack (11): ARM: pxa: ssp: remove unnecessary warning on kzalloc() failure ARM: pxa: ssp: add shortcut for &pdev->dev ARM: pxa: ssp: add DT bindings ARM: pxa: ssp: use devm_ functions ARM: pxa: ssp: add pxa_ssp_request_of() ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm ALSA: core: allow SND_DMAENGINE_PCM use from modules Dimitris Papastamos (4): ASoC: core: Add snd_soc_card_get_kcontrol() ASoC: wm_adsp: Simplify kcontrol handling ASoC: wm0010: Use DMA-safe memory for SPI transfers ASoC: wm_adsp: Sanitize parameter passing Fabio Estevam (11): ASoC: sglt5000: Provide the reg_stride field ASoC: fsl: fsl_ssi: Use devm_ functions ASoC: fsl: fsl_ssi: Check the return value from clk_prepare_enable() ASoC: fsl: imx-wm8962: Fix error path ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk() ASoC: fsl: fsl_ssi: Add MODULE_ALIAS ASoC: fsl: imx-audmux: Check the return value from clk_prepare_enable() ASoC: fsl: Fix module build ASoC: imx-mc13783: Make SND_SOC_IMX_MC13783 visible again ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code ASoC: fsl: fsl_ssi: Fix the order of resources removal Jean-Francois Moine (1): ASoC: kirkwood-i2s: fix a compilation warning Julia Lawall (5): sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument ASoC: samsung-ac97: simplify use of devm_ioremap_resource ASoC: mioa701_wm9713: Remove definition of ARRAY_AND_SIZE() ASoC: tegra20-ac97: simplify use of devm_ioremap_resource ASoC: omap: simplify platform_get_resource_byname/devm_ioremap_resource Kuninori Morimoto (13): ASoC: add ak4554 driver ASoC: ak4554: add DT support ASoC: add Renesas R-Car core feature ASoC: add Renesas R-Car module feature ASoC: add Renesas R-Car Generation feature ASoC: add Renesas R-Car SCU feature ASoC: add Renesas R-Car ADG feature ASoC: add Renesas R-Car SSI feature ASoC: rsnd: remove platform dai and add dai_id on platform setting ASoC: rsnd: add common DMAEngine method ASoC: rsnd: SSI supports DMA transfer ASoC: rsnd: SSI supports DMA transfer via BUSIF ASoC: rsnd: scu: cleanup empty functions Lars-Peter Clausen (28): ASoC: Remove unused dapm_get_snd_card() and dapm_get_soc_card() ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected ASoC: adau1701: Add adau1702 and adau1401(a) device ids ASoC: dapm: Pass snd_soc_card directly to soc_dpcm_runtime_update() ASoC: dapm: Run widget updates for shared controls at the same time ASoC: dapm: Add a update parameter to snd_soc_dapm_{mux,mixer}_update_power ASoC: tlv320aic3x: Use snd_soc_dapm_mixer_update_power ASoC: lm4857: Use table based setup for DAPM and controls ASoC: lm4857: Convert to regmap ASoC: dapm: Move snd_soc_dapm_update from dapm context to card ASoC: dapm: Pass card instead of dapm context to dapm_power_widgets() ASoC: dapm: Add a helper to get the CODEC for DAPM kcontrol ASoC: dapm: Wrap kcontrol widget list access ASoC: dapm: Move 'value' field from widget to control ASoC: dapm: Keep a list of paths per kcontrol ASoC: dapm: Make widget power register settings more flexible ASoC: dapm: Add snd_soc_dapm_add_path() helper function ASoC: dapm: Delay w->power update until the changes are written ASoC: bf5xx-ac97: Remove unused extern declaration ASoC: dapm: Check return value of snd_soc_cnew() ASoC: dapm: Fix kcontrol path list corruption ASoC: dapm: Implement mixer input auto-disable ASoC: tlv320aic26: Fix keyclick feature ASoC: mc13783: Remove embedded snd_soc_codec structs from private data structs ASoC: twl4030: Remove embedded snd_soc_codec structs from private data structs ASoC: Remove unused soc_pm_waitq ASoC: Remove unused snd_soc_info_volsw_ext() ASoC: Use snd_soc_info_enum_double() for SOC_ENUM_EXT controls Lothar Waßmann (1): ASoC: mxs: add some error messages to help identifying problems Ma Haijun (1): ASoC: wm8960: Fix ADC volume bits Marek Belisko (1): ASoC: Add PCM1681 codec driver. Mark Brown (169): ASoC: codecs: Make ALL_CODECS depend on COMPILE_TEST ASoC: pcm3008: Remove noisy version print ASoC: pcm3008: Move gpio allocation to probe ASoC: pcm3008: Check for platform data ASoC: pcm3008: Convert to devm_gpio_request_one() ASoC: wm8994: Move runtime PM init to platform device init ASoC: kirkwood: Remove unused headers ASoC: kirkwood: Enable build on non-Kirkwood platforms ASoC: kirkwood-i2s: Use devm_clk_get() for extclk ASoC: kirkwood-i2s: Remove empty remove() ASoC: kirkwood-i2s: Inline KIRKWOOD_I2S_RATES ASoC: tlv320aic3x: Add compatible strings for specific devices ASoC: tlv320aic3x: List tlv320aic3106 as a supported device ASoC: imx: Enable COMPILE_TEST builds ASoC: samsung-spdif: Convert to devm_clk_get() ASoC: omap: Enable COMPILE_TEST build for DT platforms ASoC: tegra: Remove unneeded mach-type.h incldues ASoC: tegra: Add GPIOLIB dependencies ASoC: tegra: Always use the generic dmaengine helper library ASoC: tegra: Enable COMPILE_TEST builds ASoC: imx-mc13783: Depend on ARCH_ARM ASoC: pcm: Use the power efficient workqueue for delayed powerdown ASoC: jack: Use power efficient workqueue ASoC: mxs: Enable COMPILE_TEST builds ASoC: max98090: Use power efficient workqueue ASoC: sta32x: Use power efficient workqueue ASoC: twl6040: Use power efficient workqueue ASoC: wm8350: Use power efficient workqueue ASoC: wm8753: Use power efficient workqueue ASoC: wm8994: Use power efficient workqueue ASoC: fsl_ssi: Provide register I/O functions by default ASoC: twl6040: Use power efficient workqueue ASoC: mxs: Remove unneeded mach-types.h inclusions ASoC: wm8962: Use power efficient workqueue ASoC: mxs: Depends on COMMON_CLK ASoC: dapm: Use generic power check for everything except DAIs ASoC: wm8994: Add clock bindings to the device tree ASoC: smdk_wm8994: Make driver name more unique ASoC: smdk_wm8994: Configure DAI format at init time ASoC: smdk_wm8994: Configure the MCLK1 rate based on the board Merge branch 'topic/core' of git://git.kernel.org/.../broonie/sound into asoc-dapm ASoC: samsung: Make secondary I2S DAI device a child of primary Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-new-pcm ASoC: pcm1792a: Fix build with !OF ASoC: wm8994: Fix class W controls ASoC: ad1980: Provide stub DAPM support ASoC: ads711x: Add DAPM support ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding ASoC: pcm1681: Staticise DAI driver ASoC: ad73311: Add DAPM support ASoC: ad73311: Add to list of of Analog Devices supported CODECs ASoC: max9768: Add DAPM support ASoC: cs4271: Convert to table based control init ASoC: cs4271: Add DAPM support ASoC: doc: Add documentation to MAINTAINERS patterns ASoC: compress: Use power efficient workqueue ASoC: adau1701: Use gpio_set_value_cansleep() ASoC: ak4104: Add stub DAPM support ASoC: ak4104: Manage TXE using DAPM ASoC: ak5386: Add DAPM support Merge tag 'v3.11-rc5' into asoc-dapm Merge remote-tracking branch 'asoc/fix/dapm' into asoc-dapm ASoC: dapm: Don't create routes when creating kcontrols ASoC: dapm: Ensure kcontrol list is initialised ASoC: dt: Move WM8753 pin list from Tegra board binding to CODEC binding ASoC: dt: Move RT5640 pin list from Tegra board binding to CODEC binding ASoC: dt: Move ALC5632 pin list from Tegra board binding to CODEC binding ASoC: wm8753: Fix typo in DT binding example ASoC: ak4554: Add DAPM support ASoC: max9877: Convert to use regmap API ASoC: max9877: Convert to standard CODEC driver ASoC: max9877: Add basic DAPM support Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-pxa ASoC: pcm1792a: Remove empty capture DAI stub ASoC: pcm1792a: Add DAPM support ASoC: pcm1681: Add DAPM support Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-omap Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-omap ASoC: samsung: Fix DTS breakage from missing dependencies ASoC: tlv320aic26: Remove direct use of internal I/O functions ASoC: tlv320aic26: Remove noisy print ASoC: tlv320aic26: Add basic DAPM support ASoC: si476x: Add DAPM support ASoC: pcm3008: Use gpio_set_value_cansleep() ASoC: pcm3008: Add DAPM support ASoC: pcm3008: Manage DAC and ADC power with DAPM ASoC: wl1273: Add stub DAPM support ASoC: cs4270: Add DAPM support ASoC: wm8727: Add DAPM support ASoC: wm8782: Add DAPM support ASoC: wm5102: Add inputs for noise and mic mixers ASoC: wm5110: Add inputs for noise and mic mixers ASoC: wm8997: Add inputs for noise and mic mixers ASoC: ac97: Provide stub DAPM integration ASoC: hdmi: Provide stub DAPM integration ASoC: bt-sco: Add generic compatible string ASoC: bt-sco: Provide stub DAPM integration ASoC: spdif: Add stub DAPM widgets for Rx ASoC: spdif: Remove duplicate const ASoC: imx: Add MODULE_LICENSE to DMA drivers ASoC: samsung: Check to see if we managed to allocate a channel ASoC: samsung: Rename DMA platform registration functions Merge remote-tracking branch 'asoc/topic/ac97' into asoc-fsl Merge remote-tracking branch 'asoc/fix/wm8960' into asoc-linus Merge remote-tracking branch 'asoc/topic/ac97' into asoc-next Merge remote-tracking branch 'asoc/topic/ad1980' into asoc-next Merge remote-tracking branch 'asoc/topic/ad73311' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-next Merge remote-tracking branch 'asoc/topic/ads711x' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4554' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/au1x' into asoc-next Merge remote-tracking branch 'asoc/topic/blackfin' into asoc-next Merge remote-tracking branch 'asoc/topic/bt' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4270' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm4857' into asoc-next Merge remote-tracking branch 'asoc/topic/max9768' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/max9877' into asoc-next Merge remote-tracking branch 'asoc/topic/mc13783' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/new-pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3008' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/s6000' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spdif' into asoc-next Merge remote-tracking branch 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic26' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/txx9' into asoc-next Merge remote-tracking branch 'asoc/topic/uda134x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wl1273' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8350' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8727' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8731' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8753' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8782' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: imx-audmux: Read default configuration from devicetree ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: imx-audmux: default configuration parser fixups ASoC: imx-audmux: Move definitions to dt-bindings ASoC: core: Generic ac97 link reset functions ASoC: fsl-ssi: ac97-slave support Michael Grzeschik (2): ASoC: fsl-ssi: imx-pcm-fiq bugfix ASoC: fsl: disable ssi irq for imx Michael Trimarchi (2): ASoC: omap-mcbsp: Support SND_SOC_DAIFMT_CBM_CFS for omap3/4 ASoC: Add PCM1792A spi mode codec support Mike Dyer (1): ASoC: wm8960: Fix PLL register writes Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Nicolin Chen (4): ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared ASoC: fsl: Set sdma peripheral type directly ASoC: spdif: Add S20_3LE and S24_LE support for dummy codec drivers ASoC: fsl: Add S/PDIF CPU DAI driver Oskar Schirmer (1): ASoC: sgtl5000: fix codec clock source transition to avoid clockless moment Padmavathi Venna (6): ASoC: Samsung: I2S: Modify driver to give more flexibility ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: exynos5250: move common i2s properties to exynos5 dtsi ARM: dts: Change i2s compatible string on exynos5250 ARM: dts: Change i2s compatible string on exynos5250 Peter Ujfalusi (1): ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot) Richard Fitzgerald (1): ASoC: core: init delayed_work for codec-codec links Richard Genoud (2): ASoC: wm8731: add rates constraints ASoC: atmel: update atmel SSC with DMA binding Rongjun Ying (1): ASoC: hdmi-codec: let the driver support HDMI sink Russell King (8): ASoC: uda134x: fix codec driver by converting to DAPM ASoC: spdif_transceiver: add output pin widget ASoC: kirkwood: merge struct kirkwood_dma_priv with struct kirkwood_dma_data ASoC: kirkwood: Free external clock if it is a duplicate of internal ASoC: avoid duplicated DAI routes ASoC: kirkwood: provide KIRKWOOD_PLAYCTL_ENABLE_MASK ASoC: kirkwood: combine kirkwood-i2s and kirkwood-dma drivers ASoC: kirkwood: move calculation of max buffer size to kirkwood.h Sachin Kamat (2): ASoC: s6105-ipcam: Fix incorrect placement of __initdata ASoC: pxa: Remove duplicate inclusion of dmaengine.h Steffen Trumtrar (1): ASoC: fsl-ssi: add SSIEN errata work around Stephen Warren (2): ASoC: tegra: fix compile warning in AC'97 driver ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver Tushar Behera (1): ASoC: samsung: Fix build error with dma function rename Vinod Koul (1): ASoC: compress: use soc_xxx handlers for metadata Wei Yongjun (7): ASoC: db1200: add .owner to struct snd_soc_card ASoC: imx_mc13783: add .owner to struct snd_soc_card ASoC: brownstone: add .owner to struct snd_soc_card ASoC: ttc_dkb: add .owner to struct snd_soc_card ASoC: mop500: add .owner to struct snd_soc_card ASoC: imx-sgtl5000: fix error return code in imx_sgtl5000_probe() ASoC: rcar: fix return value check in rsnd_gen1_probe() Wolfram Sang (5): ASoC: au1x: don't check resource with devm_ioremap_resource ASoC: ep93xx: don't check resource with devm_ioremap_resource ASoC: nuc900: don't check resource with devm_ioremap_resource ASoC: pxa: don't check resource with devm_ioremap_resource ASoC: txx9: don't check resource with devm_ioremap_resource .../devicetree/bindings/misc/atmel-ssc.txt | 23 +- .../devicetree/bindings/serial/mrvl,pxa-ssp.txt | 65 + Documentation/devicetree/bindings/sound/ak4554.c | 11 + .../devicetree/bindings/sound/alc5632.txt | 19 + .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 + .../devicetree/bindings/sound/atmel-wm8904.txt | 55 + .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 12 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 + .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 26 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 27 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 +- .../devicetree/bindings/sound/pcm1792a.txt | 18 + Documentation/devicetree/bindings/sound/rt5640.txt | 20 + .../devicetree/bindings/sound/samsung-i2s.txt | 22 +- .../devicetree/bindings/sound/soc-ac97link.txt | 28 + .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + .../devicetree/bindings/sound/tlv320aic3x.txt | 9 +- Documentation/devicetree/bindings/sound/wm8731.txt | 9 + Documentation/devicetree/bindings/sound/wm8753.txt | 24 +- Documentation/devicetree/bindings/sound/wm8903.txt | 19 + Documentation/devicetree/bindings/sound/wm8994.txt | 4 + MAINTAINERS | 2 + arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 +- arch/arm/plat-pxa/ssp.c | 171 ++- include/dt-bindings/sound/fsl-imx-audmux.h | 56 + include/linux/atmel-ssc.h | 2 +- include/linux/mfd/arizona/gpio.h | 96 ++ include/linux/platform_data/asoc-s3c.h | 1 + include/linux/platform_data/omap-abe-twl6040.h | 49 - include/linux/pxa2xx_ssp.h | 11 + include/sound/pxa2xx-lib.h | 7 - include/sound/rcar_snd.h | 84 ++ include/sound/soc-dapm.h | 199 ++-- include/sound/soc-dpcm.h | 2 +- include/sound/soc.h | 39 +- sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 +- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + .../soc-dmaengine-pcm.c => core/pcm_dmaengine.c} | 0 sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/atmel/Kconfig | 21 + sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 118 +- sound/soc/atmel/atmel_ssc_dai.c | 36 +- sound/soc/atmel/atmel_wm8904.c | 254 ++++ sound/soc/atmel/sam9x5_wm8731.c | 208 ++++ sound/soc/au1x/db1200.c | 4 + sound/soc/au1x/psc-ac97.c | 3 - sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/cirrus/ep93xx-ac97.c | 3 - sound/soc/cirrus/ep93xx-i2s.c | 3 - sound/soc/codecs/Kconfig | 19 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 15 + sound/soc/codecs/ad1980.c | 43 + sound/soc/codecs/ad73311.c | 22 +- sound/soc/codecs/adau1701.c | 25 +- sound/soc/codecs/adav80x.c | 13 +- sound/soc/codecs/ads117x.c | 29 +- sound/soc/codecs/ak4104.c | 34 +- sound/soc/codecs/ak4554.c | 106 ++ sound/soc/codecs/ak5386.c | 17 +- sound/soc/codecs/arizona.c | 69 +- sound/soc/codecs/arizona.h | 5 +- sound/soc/codecs/bt-sco.c | 22 +- sound/soc/codecs/cs4270.c | 20 + sound/soc/codecs/cs4271.c | 30 +- sound/soc/codecs/hdmi.c | 30 +- sound/soc/codecs/lm4857.c | 107 +- sound/soc/codecs/max9768.c | 16 + sound/soc/codecs/max98090.c | 10 +- sound/soc/codecs/max9877.c | 294 ++--- sound/soc/codecs/mc13783.c | 1 - sound/soc/codecs/pcm1681.c | 339 ++++++ sound/soc/codecs/pcm1792a.c | 257 ++++ sound/soc/codecs/pcm1792a.h | 26 + sound/soc/codecs/pcm3008.c | 150 ++- sound/soc/codecs/rt5640.c | 26 - sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 14 + sound/soc/codecs/spdif_receiver.c | 17 +- sound/soc/codecs/spdif_transmitter.c | 18 +- sound/soc/codecs/sta32x.c | 10 +- sound/soc/codecs/tlv320aic26.c | 51 +- sound/soc/codecs/tlv320aic3x.c | 56 +- sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 7 +- sound/soc/codecs/uda134x.c | 88 +- sound/soc/codecs/wl1273.c | 17 + sound/soc/codecs/wm0010.c | 12 +- sound/soc/codecs/wm5102.c | 53 +- sound/soc/codecs/wm5110.c | 35 +- sound/soc/codecs/wm8350.c | 6 +- sound/soc/codecs/wm8727.c | 17 +- sound/soc/codecs/wm8731.c | 60 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8782.c | 17 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8960.c | 10 +- sound/soc/codecs/wm8962.c | 7 +- sound/soc/codecs/wm8994.c | 39 +- sound/soc/codecs/wm8995.c | 5 +- sound/soc/codecs/wm8997.c | 1175 +++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + sound/soc/codecs/wm_adsp.c | 124 +- sound/soc/codecs/wm_adsp.h | 3 +- sound/soc/codecs/wm_hubs.c | 8 +- sound/soc/fsl/Kconfig | 13 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/fsl_ssi.c | 500 ++++++-- sound/soc/fsl/imx-audmux.c | 78 +- sound/soc/fsl/imx-audmux.h | 52 +- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 20 +- sound/soc/fsl/imx-pcm.h | 26 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/imx-wm8962.c | 3 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 108 +- sound/soc/kirkwood/kirkwood-i2s.c | 67 +- sound/soc/kirkwood/kirkwood-openrd.c | 6 +- sound/soc/kirkwood/kirkwood-t5325.c | 6 +- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/Kconfig | 3 +- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 30 +- sound/soc/nuc900/nuc900-ac97.c | 3 - sound/soc/omap/Kconfig | 8 +- sound/soc/omap/omap-abe-twl6040.c | 133 +-- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 + sound/soc/omap/omap-mcpdm.c | 3 - sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/mmp-pcm.c | 7 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 76 +- sound/soc/pxa/pxa2xx-ac97.c | 67 +- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/samsung/ac97.c | 11 +- sound/soc/samsung/dma.c | 12 +- sound/soc/samsung/dma.h | 4 +- sound/soc/samsung/i2s-regs.h | 51 +- sound/soc/samsung/i2s.c | 193 ++- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 58 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 ++++ sound/soc/sh/rcar/core.c | 861 ++++++++++++++ sound/soc/sh/rcar/gen.c | 280 +++++ sound/soc/sh/rcar/rsnd.h | 302 +++++ sound/soc/sh/rcar/scu.c | 236 ++++ sound/soc/sh/rcar/ssi.c | 728 ++++++++++++ sound/soc/soc-compress.c | 13 +- sound/soc/soc-core.c | 236 +++- sound/soc/soc-dapm.c | 810 +++++++------ sound/soc/soc-jack.c | 2 +- sound/soc/soc-pcm.c | 15 +- sound/soc/spear/Kconfig | 2 +- sound/soc/tegra/Kconfig | 14 +- sound/soc/tegra/tegra20_ac97.c | 8 - sound/soc/tegra/tegra_alc5632.c | 2 - sound/soc/tegra/tegra_rt5640.c | 1 + sound/soc/tegra/tegra_wm8753.c | 2 - sound/soc/tegra/trimslice.c | 2 - sound/soc/txx9/txx9aclc-ac97.c | 3 - sound/soc/ux500/mop500.c | 1 + 193 files changed, 10275 insertions(+), 2357 deletions(-) create mode 100644 Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (87%) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h create mode 100644 include/linux/mfd/arizona/gpio.h delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h create mode 100644 include/sound/rcar_snd.h rename sound/{soc/soc-dmaengine-pcm.c => core/pcm_dmaengine.c} (100%) create mode 100644 sound/soc/atmel/atmel_wm8904.c create mode 100644 sound/soc/atmel/sam9x5_wm8731.c create mode 100644 sound/soc/codecs/ak4554.c create mode 100644 sound/soc/codecs/pcm1681.c create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/sh/rcar/Makefile create mode 100644 sound/soc/sh/rcar/adg.c create mode 100644 sound/soc/sh/rcar/core.c create mode 100644 sound/soc/sh/rcar/gen.c create mode 100644 sound/soc/sh/rcar/rsnd.h create mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/ssi.c

2 1

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 23 Aug '13

23 Aug '13

At Fri, 23 Aug 2013 00:56:01 +0300, Stratos Karafotis wrote: > > On 08/23/2013 12:23 AM, Takashi Iwai wrote: > > At Thu, 22 Aug 2013 19:03:44 +0300, > > Stratos Karafotis wrote: > >> > >> On 08/22/2013 10:59 AM, Takashi Iwai wrote: > >>> At Thu, 22 Aug 2013 00:42:41 +0300, > >>> Stratos Karafotis wrote: > >>>> > >>>> Hi, > >>>> > >>>> I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > >>>> not set (3.11-rc6). > >>>> The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > >>>> as suggested in printk message. > >>>> > >>>> Regards, > >>>> Stratos > >>>> > >>>> > >>> > >>> Could you check the patch below? > >>> Thanks! > >>> > >>> > >>> Takashi > >>> > >>> --- > >>> From: Takashi Iwai <tiwai(a)suse.de> > >>> Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n > >>> > >>> Without the dynamic minor assignment, HDMI codec may have less PCM > >>> instances than the number of pins, which eventually leads to Oops. > >>> > >>> Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> > >>> Cc: <stable(a)vger.kernel.org> > >>> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > >>> --- > >>> sound/pci/hda/patch_hdmi.c | 3 +++ > >>> 1 file changed, 3 insertions(+) > >>> > >>> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > >>> index 030ca86..e2cb92b 100644 > >>> --- a/sound/pci/hda/patch_hdmi.c > >>> +++ b/sound/pci/hda/patch_hdmi.c > >>> @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) > >>> struct snd_pcm_chmap *chmap; > >>> struct snd_kcontrol *kctl; > >>> int i; > >>> + > >>> + if (pin_idx >= codec->num_pcms) > >>> + break; > >>> err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, > >>> SNDRV_PCM_STREAM_PLAYBACK, > >>> NULL, 0, pin_idx, &chmap); > >>> > >> > >> Hi, > >> > >> Unfortunately, still the same problem after applying your patch. > > > > Bah, it's a wrong one. The patch below should work better. > > Sorry for inconvenience. > > No problem! :) > > Yes, no more oops with the latest patch. > Tested with CONFIG_SND_DYNAMIC_MINORS=y and n. > > There is an error -16 now (please see below), but I guess it's normal(?) Yes, it's the expected behavior. Thanks for quick tests! Takashi > Thanks, > Stratos > > > [ 16.392453] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 16.392457] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 16.392460] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 16.392461] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 16.409640] ALSA sound/core/control.c:349 control 0:0:0:HDMI/DP Jack:0 is already present > [ 16.410627] hda_codec: cannot build controls for #3 (error -16) > [ 16.411683] input: HDA NVidia HDMI/DP as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input15 > [ 16.411844] input: HDA NVidia HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input16 > [ 16.411986] input: HDA NVidia HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input17 > [ 19.682495] EXT4-fs (sda2): mounted filesystem with ordered data mode. Opts: (null) > [ 21.367420] FAT-fs (sda1): Volume was not properly unmounted. Some data may be corrupt. Please run fsck. > [ 21.420191] EXT4-fs (dm-2): mounted filesystem with ordered data mode. Opts: (null) > [ 21.847085] type=1305 audit(1377207061.920:3): audit_pid=534 old=0 auid=4294967295 ses=4294967295 > res=1 > [ 22.164707] alsactl[549]: segfault at 1 ip 00007f6c8ad58a7d sp 00007fff19912de0 error 4 in libc-2.17.so[7f6c8ad10000+1b6000] >

1 0

[alsa-devel] [RFC PATCH 3/3] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 23 Aug '13

23 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [RFC PATCH 2/3] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 23 Aug '13

23 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..fd62d8f 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,38 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + union hda_devlist_entry dev_list[AC_MAX_DEVLIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEVLIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %ld, ELDV = %ld, IA = %ld\n", + i, dev_list[i].bits.present_detect, + dev_list[i].bits.eld_valid, dev_list[i].bits.inactive); + } + + return 0; + +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +783,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

Re: [alsa-devel] [PATCH] rme96 Add missing vmalloc.h inclusion
by Takashi Iwai 23 Aug '13

23 Aug '13

At Thu, 22 Aug 2013 15:06:38 +0200, Knut Petersen wrote: > > [1 <text/plain; ISO-8859-1 (7bit)>] > On 22.08.2013 10:53, Takashi Iwai wrote: > > Actually this triggers the double free. It's freed in > > snd_rme96_free() via snd_card_free(). > Yes. Now I see that. > > I applied the patch with removal of this double vfree call. > Thanks. > > Unfortunately there still is a problem with the code: > vmalloc() and vfree() were used without inclusion > of linux/vmalloc.h. Indeed. Applied now. Thanks. Takashi > > cu, > Knut > > > [2 0001-alsa-rme96-Add-missing-inclusion-of-linux-vmalloc.h.patch <text/x-patch (7bit)>] > >From 2a3f98ff69308f3c2ee508ac0887e1f8bf493d64 Mon Sep 17 00:00:00 2001 > From: Knut Petersen <Knut_Petersen(a)t-online.de> > Date: Thu, 22 Aug 2013 14:36:16 +0200 > Subject: [PATCH] alsa/rme96: Add missing inclusion of linux/vmalloc.h > > Signed-off-by: Knut Petersen <Knut_Petersen(a)t-online.de> > --- > sound/pci/rme96.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c > index 2acf565..cc6deca 100644 > --- a/sound/pci/rme96.c > +++ b/sound/pci/rme96.c > @@ -28,6 +28,7 @@ > #include <linux/interrupt.h> > #include <linux/pci.h> > #include <linux/module.h> > +#include <linux/vmalloc.h> > > #include <sound/core.h> > #include <sound/info.h> > -- > 1.8.1.4 >

1 0

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 23 Aug '13

23 Aug '13

At Thu, 22 Aug 2013 19:03:44 +0300, Stratos Karafotis wrote: > > On 08/22/2013 10:59 AM, Takashi Iwai wrote: > > At Thu, 22 Aug 2013 00:42:41 +0300, > > Stratos Karafotis wrote: > >> > >> Hi, > >> > >> I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > >> not set (3.11-rc6). > >> The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > >> as suggested in printk message. > >> > >> Regards, > >> Stratos > >> > >> > > > > Could you check the patch below? > > Thanks! > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n > > > > Without the dynamic minor assignment, HDMI codec may have less PCM > > instances than the number of pins, which eventually leads to Oops. > > > > Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> > > Cc: <stable(a)vger.kernel.org> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/patch_hdmi.c | 3 +++ > > 1 file changed, 3 insertions(+) > > > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > > index 030ca86..e2cb92b 100644 > > --- a/sound/pci/hda/patch_hdmi.c > > +++ b/sound/pci/hda/patch_hdmi.c > > @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) > > struct snd_pcm_chmap *chmap; > > struct snd_kcontrol *kctl; > > int i; > > + > > + if (pin_idx >= codec->num_pcms) > > + break; > > err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, > > SNDRV_PCM_STREAM_PLAYBACK, > > NULL, 0, pin_idx, &chmap); > > > > Hi, > > Unfortunately, still the same problem after applying your patch. Bah, it's a wrong one. The patch below should work better. Sorry for inconvenience. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH v2] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n Without the dynamic minor assignment, HDMI codec may have less PCM instances than the number of pins, which eventually leads to Oops. Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..9f35862 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (!codec->pcm_info[pin_idx].pcm) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); -- 1.8.3.4

1 0

[alsa-devel] [PATCH v2 00/10] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 22 Aug '13

22 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD b43dd416be21bc8ad60984e13def032f01aaaa18 ). v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html Thanks to Clemens I was able to get rid of the get_interface workaround, so I believe it's ready now, unless there's any other issues. Other changes: * Cleaned up and moved the set_param flag patch to #7. * Fixed a NULL dereference in patch #8 for explicit feedback. * Fixed a logic error in patch #9. * Added patch #10. Cheers, Eldad Eldad Zack (10): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream sound/usb/card.h | 2 +- sound/usb/endpoint.c | 76 +++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 161 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 192 insertions(+), 64 deletions(-) -- 1.8.1.5

3 17

[alsa-devel] [PATCH v10 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 22 Aug '13

22 Aug '13

Changelog: v9->v10: * Removed "dummy" words from binding doc in PATCH-2. v8->v9: * Use bool instead of atomic_t. * Use clk_prepare_enable() instead. * Dropped dumpregs(). * Dropped unnecessary clk_round_rate(). * Dropped unused clock source enum. * Revised dev_err() message. v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/imx-spdif.c | 134 +++ 7 files changed, 1662 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 10

[alsa-devel] [PATCH v5 1/1] ASoc: kirkwood: add DT support to the mvebu audio subsystem
by Jean-Francois Moine 22 Aug '13

22 Aug '13

This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This patch does not add any new function as S/PDIF or DPCM support. --- .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 26 +++++++++++++++++----- 2 files changed, 49 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 0000000..7e5fd37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "marvell,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c.next b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a..a4170b4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c.next +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id mvebu_audio_of_match[] = { + { .compatible = "marvell,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mvebu_audio_of_match), }, }; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: samsung: Fix build error with dma function rename
by Tushar Behera 22 Aug '13

22 Aug '13

commit 85ff3c29d720 ("ASoC: samsung: Rename DMA platform registration functions") renames the DMA registration functions. Fix the places where it was left out. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 1566afe..e54256f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -594,7 +594,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) goto err5; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -623,7 +623,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev) struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 47e2386..ea885cb 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -176,7 +176,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the DMA: %d\n", ret); goto err; @@ -190,7 +190,7 @@ err: static int s3c2412_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 8b34145..9c8ebd8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -480,7 +480,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the dma: %d\n", ret); goto err; @@ -494,7 +494,7 @@ err: static int s3c24xx_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5ea70ab..28487dc 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -442,7 +442,7 @@ static int spdif_probe(struct platform_device *pdev) spdif->dma_playback = &spdif_stereo_out; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to register DMA: %d\n", ret); goto err5; @@ -468,7 +468,7 @@ static int spdif_remove(struct platform_device *pdev) struct samsung_spdif_info *spdif = &spdif_info; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); iounmap(spdif->regs); -- 1.7.9.5

2 1

[alsa-devel] ice1712 suspend/resume for M-Audio Audiophile 2496
by Tommi Uimonen 22 Aug '13

22 Aug '13

Hello, the lack of suspend/resume support for ice1712 driver has been bugging me a lot, and I was wondering if it's possible to implement it by following how it's done in 1724.c? I don't really know much about the driver, but after some comparison between 1712.c and 1724.c it seems doable. At least the suspend part seems straightforward, just changing ICEMT1724 with ICEMT and ICEREG1724 with ICEREG. To resume, I should add some new functions like snd_ice1712_chip_reset (I guess it follows the same logic as in 1724.c, just use ICE1712_RESET instead) and maybe some spdif stuff but I think the 1724 has different spdif options. I don't have any 1724 cards so can't be sure, but 1712 does not have spdif rate. Maybe it will work without those, at least for me since I don't use spdif. And then add this to the pci_driver ice1712_driver struct: .driver = { .pm = SND_ICE1712_PM_OPS, } and define the SIMPLE_DEV_PM_OPS accordingly. Or is this already under work by someone? Tommi

1 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix ALC283 headphone pop-noise better
by Takashi Iwai 22 Aug '13

22 Aug '13

From: Kailang Yang <kailang(a)realtek.com> Fixed ALC283 D3 to D0 and D0 to D3 Headphone pop noise. The previous fix [c5177c86: ALSA: hda - Fix the noise after suspend on ALC283 codec] doesn't work sufficiently for some laptops. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 111 +++++++++++++++++++++++++++++------------- 1 file changed, 76 insertions(+), 35 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d00639..4bdccd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2527,6 +2527,7 @@ enum { ALC269_TYPE_ALC269VD, ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, + ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, }; @@ -2552,6 +2553,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: + case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: ssids = alc269_ssids; break; @@ -2586,6 +2588,74 @@ static void alc269_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc283_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) + return; + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ + /* Headphone capless set to high power mode */ + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + if (hp_pin_sense) + msleep(85); + /* Index 0x46 Combo jack auto switch control 2 */ + /* 3k pull low control for Headset jack. */ + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val & ~(3 << 12)); + /* Headphone capless set to normal mode */ + alc_write_coef_idx(codec, 0x43, 0x9614); +} + +static void alc283_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) { + alc269_shutup(codec); + return; + } + + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val | (3 << 12)); + + if (hp_pin_sense) + msleep(85); + snd_hda_shutup_pins(codec); + alc_write_coef_idx(codec, 0x43, 0x9614); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -2715,12 +2785,6 @@ static int alc269_resume(struct hda_codec *codec) if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); - /* clear the power-save mode for ALC283 */ - if (codec->vendor_id == 0x10ec0283) { - alc_write_coef_idx(codec, 0x4, 0xaf01); - alc_write_coef_idx(codec, 0x6, 0x2104); - } - return 0; } #endif /* CONFIG_PM */ @@ -3815,30 +3879,6 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_write_coef_idx(codec, 0x4, val | (1<<11)); } -/* don't clear mic pin; otherwise it results in noise in D3 */ -static void alc283_headset_shutup(struct hda_codec *codec) -{ - int i; - - if (codec->bus->shutdown) - return; - - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - /* use read here for syncing after issuing each verb */ - if (pin->nid != 0x19) - snd_hda_codec_read(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } - - alc_write_coef_idx(codec, 0x4, 0x0f01); /* power save */ - alc_write_coef_idx(codec, 0x6, 0x2100); /* power save */ - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_VREFHIZ); - codec->pins_shutup = 1; -} - /* */ static int patch_alc269(struct hda_codec *codec) @@ -3853,9 +3893,6 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; - if (codec->vendor_id == 0x10ec0283) - spec->shutup = alc283_headset_shutup; - snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3897,11 +3934,15 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; - case 0x10ec0233: case 0x10ec0282: - case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; break; + case 0x10ec0233: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC283; + spec->shutup = alc283_shutup; + spec->init_hook = alc283_init; + break; case 0x10ec0284: case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; -- 1.8.3.4

2 3

Re: [alsa-devel] [PATCH -next] ASoC: mmp-pcm: remove duplicated include from mmp-pcm.c
by Mark Brown 22 Aug '13

22 Aug '13

On Thu, Aug 22, 2013 at 10:37:20AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Remove duplicated include. Sorry, I already applied a patch for this which I saw first (though yours was actually sent sooner). Thanks.

1 0

[alsa-devel] [PATCH 1/1] ASoC: pxa: Remove duplicate inclusion of dmaengine.h
by Sachin Kamat 22 Aug '13

22 Aug '13

dmaengine.h header file was included twice. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/pxa/mmp-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 9a97843..8235e23 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -17,7 +17,6 @@ #include <linux/dmaengine.h> #include <linux/platform_data/dma-mmp_tdma.h> #include <linux/platform_data/mmp_audio.h> -#include <linux/dmaengine.h> #include <sound/pxa2xx-lib.h> #include <sound/core.h> -- 1.7.9.5

2 1

[alsa-devel] [PATCH] rme96 synchronization support
by Knut Petersen 22 Aug '13

22 Aug '13

Hi everybody! As I need support for synchronized pcm streams on an RME Digi96/8 PAD I wrote the attached patch. It does work here. My own program, trying to get synchronized capture/playback streams, succeeds. Programs that use either the playback or the capture part of the hardware are not affected by start / pause/ stop operations of applications using the other part of the hardware (tested with aplay, arecord, and ecasound). I wrote the test code and the patch, so it is not impossible that I acted on wrong assumptions in both places. Please have a close look at the patch. cu, knut

3 15

[alsa-devel] [PATCH] Update iatomic.h functions definitions for mips
by jiashuo.kang＠gmail.com 22 Aug '13

22 Aug '13

From: Kai Kang <jiashuo.kang at gmail.com> Functions atomic_add(s) and atomic_sub(s) are defined with 'extern __inline__' that may cause compile fails when cross compile for mips. The error message looks like: | pcm/.libs/libpcm.a(pcm_meter.o): In function `snd_pcm_meter_update_scope': | .../alsa-lib-1.0.27.2/src/pcm/pcm_meter.c:139: undefined reference to `atomic_sub' Replace the 'extern __inline__' with 'static __inline__' to fix this issue. Signed-off-by: Kai Kang <jiashuo.kang at gmail.com> --- include/iatomic.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 5711fe8..2393297 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -720,7 +720,7 @@ typedef struct { volatile int counter; } atomic_t; * Atomically adds @i to @v. Note that the guaranteed useful range * of an atomic_t is only 24 bits. */ -extern __inline__ void atomic_add(int i, atomic_t * v) +static __inline__ void atomic_add(int i, atomic_t * v) { unsigned long temp; @@ -744,7 +744,7 @@ extern __inline__ void atomic_add(int i, atomic_t * v) * Atomically subtracts @i from @v. Note that the guaranteed * useful range of an atomic_t is only 24 bits. */ -extern __inline__ void atomic_sub(int i, atomic_t * v) +static __inline__ void atomic_sub(int i, atomic_t * v) { unsigned long temp; @@ -763,7 +763,7 @@ extern __inline__ void atomic_sub(int i, atomic_t * v) /* * Same as above, but return the result value */ -extern __inline__ int atomic_add_return(int i, atomic_t * v) +static __inline__ int atomic_add_return(int i, atomic_t * v) { unsigned long temp, result; @@ -784,7 +784,7 @@ extern __inline__ int atomic_add_return(int i, atomic_t * v) return result; } -extern __inline__ int atomic_sub_return(int i, atomic_t * v) +static __inline__ int atomic_sub_return(int i, atomic_t * v) { unsigned long temp, result; -- 1.8.1.2

3 2

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 22 Aug '13

22 Aug '13

At Thu, 22 Aug 2013 00:42:41 +0300, Stratos Karafotis wrote: > > Hi, > > I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > not set (3.11-rc6). > The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > as suggested in printk message. > > Regards, > Stratos > > > [ 8.670497] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 8.670500] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 8.670501] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 8.670502] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 8.688015] BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 > [ 8.688609] IP: [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.689191] PGD 0 > [ 8.689762] Oops: 0000 [#1] SMP > [ 8.690326] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek arc4 rt2800pci eeprom_93cx6 rt2x00pci rt2800lib crc_ccitt rt2x00mmio rt2x00lib mac80211 cfg80211 eeepc_wmi asus_wmi sparse_keymap rfkill snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq iTCO_wdt iTCO_vendor_support r8169 mii i2c_i801 i2c_core snd_seq_device snd_pcm serio_raw pcspkr lpc_ich mfd_core snd_page_alloc snd_timer snd soundcore binfmt_misc uinput usb_storage wmi video > [ 8.692808] CPU: 1 PID: 417 Comm: systemd-udevd Not tainted 3.11.0-rc6 #3 > [ 8.693424] Hardware name: ASUSTeK COMPUTER INC. CM6870/CM6870, BIOS 0606 08/27/2012 > [ 8.694044] task: ffff880210091750 ti: ffff880210598000 task.ti: ffff880210598000 > [ 8.694666] RIP: 0010:[<ffffffffa0086992>] [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.695310] RSP: 0018:ffff880210599968 EFLAGS: 00010246 > [ 8.695954] RAX: ffffffffa008c51e RBX: ffff880212d20b80 RCX: 0000000000000000 > [ 8.696607] RDX: ffffffffa008c533 RSI: ffff880212d20b80 RDI: ffff880210599980 > [ 8.697261] RBP: ffff8802105999f8 R08: 0000000000000000 R09: ffff880216003b00 > [ 8.697923] R10: 0000000000000000 R11: ffff8802133d93c0 R12: 0000000000000000 > [ 8.698585] R13: ffff880210599a10 R14: 0000000000000000 R15: ffff880210599980 > [ 8.699255] FS: 00007fe8eb8ea880(0000) GS:ffff88021ec40000(0000) knlGS:0000000000000000 > [ 8.699936] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > [ 8.700619] CR2: 0000000000000018 CR3: 00000002101c2000 CR4: 00000000001407e0 > [ 8.701310] Stack: > [ 8.702005] 0000000000000000 000000000b5a4000 0000000000000000 0000000000000003 > [ 8.702715] 0000000000000000 ffffffffa008c51e 1000001100000000 0000000000000000 > [ 8.703422] ffffffffa0084a80 ffffffffa0086cd0 0000000000000000 ffffffffa0085b30 > [ 8.704132] Call Trace: > [ 8.704836] [<ffffffffa0084a80>] ? snd_pcm_hw_rule_msbits+0x50/0x50 [snd_pcm] > [ 8.705558] [<ffffffffa0086cd0>] ? snd_pcm_hw_rule_ratdens+0x2b0/0x2b0 [snd_pcm] > [ 8.706281] [<ffffffffa0085b30>] ? snd_pcm_hw_param_last+0x240/0x240 [snd_pcm] > [ 8.707008] [<ffffffffa02af7ed>] generic_hdmi_build_controls+0x14d/0x1e0 [snd_hda_codec_hdmi] > [ 8.707748] [<ffffffffa02ae827>] ? generic_hdmi_init+0xb7/0xd0 [snd_hda_codec_hdmi] > [ 8.708493] [<ffffffffa013a812>] snd_hda_codec_build_controls+0x1c2/0x220 [snd_hda_codec] > [ 8.709242] [<ffffffffa0135255>] ? snd_hda_codec_configure+0x295/0x450 [snd_hda_codec] > [ 8.709990] [<ffffffffa013a898>] snd_hda_build_controls+0x28/0x80 [snd_hda_codec] > [ 8.710728] [<ffffffffa00fbbed>] azx_probe_continue+0x84d/0xcc0 [snd_hda_intel] > [ 8.711456] [<ffffffffa00fb060>] ? perf_trace_azx_pcm_trigger+0xe0/0xe0 [snd_hda_intel] > [ 8.712187] [<ffffffffa00f9ee0>] ? azx_resume+0x130/0x130 [snd_hda_intel] > [ 8.712916] [<ffffffffa00fac20>] ? azx_pcm_prepare+0x5f0/0x5f0 [snd_hda_intel] > [ 8.713646] [<ffffffffa00f98f0>] ? azx_runtime_suspend+0x40/0x40 [snd_hda_intel] > [ 8.714377] [<ffffffffa00f8800>] ? azx_remove+0x30/0x30 [snd_hda_intel] > [ 8.715111] [<ffffffffa00fc4bf>] azx_probe+0x3bf/0x7e0 [snd_hda_intel] > [ 8.715845] [<ffffffff8130b3ee>] local_pci_probe+0x3e/0x70 > [ 8.716574] [<ffffffff8130c6d1>] pci_device_probe+0x121/0x130 > [ 8.717303] [<ffffffff813bf3c7>] driver_probe_device+0x87/0x390 > [ 8.718033] [<ffffffff813bf7a3>] __driver_attach+0x93/0xa0 > [ 8.718762] [<ffffffff813bf710>] ? __device_attach+0x40/0x40 > [ 8.719481] [<ffffffff813bd303>] bus_for_each_dev+0x63/0xa0 > [ 8.720189] [<ffffffff813bee1e>] driver_attach+0x1e/0x20 > [ 8.720886] [<ffffffff813be9b8>] bus_add_driver+0x1e8/0x2a0 > [ 8.721578] [<ffffffffa016a169>] ? ftrace_define_fields_azx_get_position+0xcd/0xcd [snd_hda_intel] > [ 8.722277] [<ffffffff813bfdc4>] driver_register+0x74/0x150 > [ 8.722980] [<ffffffffa016a169>] ? ftrace_define_fields_azx_get_position+0xcd/0xcd [snd_hda_intel] > [ 8.723682] [<ffffffff8130b27b>] __pci_register_driver+0x4b/0x50 > [ 8.724368] [<ffffffffa016a187>] azx_driver_init+0x1e/0xe97 [snd_hda_intel] > [ 8.725043] [<ffffffff810002c2>] do_one_initcall+0xf2/0x1a0 > [ 8.725706] [<ffffffff8103f183>] ? set_memory_nx+0x43/0x50 > [ 8.726360] [<ffffffff810b9f3d>] load_module+0x1b9d/0x2640 > [ 8.726999] [<ffffffff810b6270>] ? store_uevent+0x40/0x40 > [ 8.727631] [<ffffffff810bab56>] SyS_finit_module+0x86/0xb0 > [ 8.728250] [<ffffffff81621782>] system_call_fastpath+0x16/0x1b > [ 8.728856] Code: ff 48 89 de 4c 89 ff 48 89 43 18 8b 85 7c ff ff ff 89 43 20 48 c7 c0 1e c5 08 a0 4c 89 45 d0 48 0f 45 c2 49 c1 e2 07 48 89 45 98 <41> 8b 44 24 18 4f 8d 34 14 89 45 8c 41 8b 86 c8 00 00 00 89 45 > [ 8.730268] RIP [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.730932] RSP <ffff880210599968> > [ 8.731585] CR2: 0000000000000018 > [ 8.732240] ---[ end trace 92e1db8c3a6c8fab ]--- Could you check the patch below? Thanks! Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n Without the dynamic minor assignment, HDMI codec may have less PCM instances than the number of pins, which eventually leads to Oops. Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..e2cb92b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (pin_idx >= codec->num_pcms) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); -- 1.8.3.4

1 0

[alsa-devel] alsa-lib bug tracking
by Cornel Spinu 22 Aug '13

22 Aug '13

Hi, I'm looking for a list of bugs for alsa-lib. It seems that https://bugtrack.alsa-project.org/alsa-bug is not working. Do you have any clue where I can find this list? Thank you, Cornel

2 1

[alsa-devel] CM108 head set problem
by Nikolay Martynov 22 Aug '13

22 Aug '13

Hi. I'm experiencing a problem with CM108 based headset. The problem manifests itself in 'phone' software. When I connect the headset and run pavucontrol I can see the headset's mic in input devices and when I say something the activity bar moves - I sort of see that mic is producing some input for the OS. Then I try to make a Skype call (I've set skype to use proper input and output devices). When I make Skype call the activity bar from pavucontrol disappears and mic stops working - the other side of the call doesn't hear me. Similar thing happens with Google hangout. The only difference is that with Skype it happens immediately and with Google hangout it looks like it only happens when I do something 'heavy'. The thing that often triggers the problem with Google hangout is switching desktops (I have compiz running). I'm not entirely sure that problem is alsa related but I have this in dmesg: [72856.844289] usb 3-1: USB disconnect, device number 10 [72858.632109] usb 3-1: new full-speed USB device number 11 using uhci_hcd [72858.792090] usb 3-1: New USB device found, idVendor=045e, idProduct=070f [72858.792098] usb 3-1: New USB device strings: Mfr=0, Product=1, SerialNumber=0 [72858.792103] usb 3-1: Product: Microsoft LifeChat LX-3000 [72858.875205] input: Microsoft LifeChat LX-3000 as /devices/pci0000:00/0000:00:1d.1/usb3/3-1/3-1:1.3/input/input19 [72858.875399] hid-generic 0003:045E:070F.000B: input,hidraw0: USB HID v1.00 Device [Microsoft LifeChat LX-3000 ] on usb-0000:00:1d.1-1/input3 [72876.015919] retire_capture_urb: 22 callbacks suppressed [72876.411354] delay: estimated 133, actual 1 [72876.496315] cannot submit urb (err = -18) [72876.496330] cannot submit urb (err = -18) [72876.496341] cannot submit urb (err = -18) [72876.496349] cannot submit urb (err = -18) [72876.496357] cannot submit urb (err = -18) [72876.496366] cannot submit urb (err = -18) [72876.496375] cannot submit urb (err = -18) [72876.496383] cannot submit urb (err = -18) [72876.499076] delay: estimated 222, actual 1 [72876.587212] delay: estimated 90, actual 1 [72876.627072] delay: estimated 266, actual 1 [72876.675644] delay: estimated 353, actual 0 [72876.747102] delay: estimated 354, actual 1 [72876.851132] delay: estimated 353, actual 1 [72877.059072] delay: estimated 90, actual 1 [72877.171110] delay: estimated 353, actual 1 [72877.219082] delay: estimated 90, actual 1 [72877.299493] delay: estimated 310, actual 1 [72877.547100] delay: estimated 354, actual 1 It looks like once I get error like message 'cannot submit urb (err = -18)' mic stops working. I would really appreciate any help solving this problem and I would be happy to do any testing required or provide any additional information. I'm using Linux Mint 15 (Ubuntu 13.04) with '3.10.5' kernel from Ubuntu mainline kernel builds. The very same problem also happens with 'standard' kernel that comes with Ubuntu. I have a dell inspiron 6400 laptop with core2duo CPU and ati video. Thanks! -- Martynov Nikolay. Email: mar.kolya(a)gmail.com

1 0

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 21 Aug '13

21 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 200 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/rt5640.h | 11 +++ 2 files changed, 157 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..218aae2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,103 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(30000, 35000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val = 0, pow; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + if (!(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (!(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(35000, 40000); + break; + + case SND_SOC_DAPM_PRE_PMD: + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1212,27 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1473,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1490,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1508,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1890,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1940,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..4d2cfe7 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) -- 1.8.1.1.439.g50a6b54

4 4

[alsa-devel] [PATCH v8 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 21 Aug '13

21 Aug '13

Changelog: v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1281 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1740 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 15

[alsa-devel] [PATCH v9 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v8->v9: * Use bool instead of atomic_t. * Use clk_prepare_enable() instead. * Dropped dumpregs(). * Dropped unnecessary clk_round_rate(). * Dropped unused clock source enum. * Revised dev_err() message. v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/imx-spdif.c | 134 +++ 7 files changed, 1662 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 7

[alsa-devel] [PATCH v7 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1277 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1738 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

6 21

[alsa-devel] [PATCH 0/6] simplify platform_get_resource_byname/devm_ioremap_resource
by Julia Lawall 20 Aug '13

20 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource_byname as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call to platform_get_resource_byname. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l,f; expression list es; @@ ( res = f(...); - if (res == NULL) { ... $goto l;\|return ret;$ } e = devm_ioremap_resource(e1, res); | - res = f(es); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = f(es); e = devm_ioremap_resource(e1, res); ) // </smpl> In practice, f is always platform_get_resource_byname (or platform_get_resource, which was handled by a previous patch series). And the call to platform_get_resource_byname always immediately precedes the call to devm_ioremap_resource.

3 3

[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix the order of resources removal
by Fabio Estevam 20 Aug '13

20 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> In fsl_ssi_remove() we need to remove the resources in the opposite order that they were acquired in probe. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff..c8aee4e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -928,14 +928,14 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); - clk_disable_unprepare(ssi_private->clk); - } snd_soc_unregister_component(&pdev->dev); - device_remove_file(&pdev->dev, &ssi_private->dev_attr); - irq_dispose_mapping(ssi_private->irq); dev_set_drvdata(&pdev->dev, NULL); + device_remove_file(&pdev->dev, &ssi_private->dev_attr); + if (ssi_private->ssi_on_imx) + clk_disable_unprepare(ssi_private->clk); + irq_dispose_mapping(ssi_private->irq); return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: samsung: Rename DMA platform registration functions
by Mark Brown 20 Aug '13

20 Aug '13

From: Mark Brown <broonie(a)linaro.org> The current naming with a simple asoc_ prefix is too generic for use in multiplatform kernels. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/dma.c | 8 ++++---- sound/soc/samsung/dma.h | 4 ++-- sound/soc/samsung/i2s.c | 6 +++--- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index c732df9..2acf987 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -457,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) if (ret) goto err5; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -480,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) { struct resource *irq_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index d54df85..9338d11 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -444,17 +444,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; -int asoc_dma_platform_register(struct device *dev) +int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_register); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -void asoc_dma_platform_unregister(struct device *dev) +void samsung_asoc_dma_platform_unregister(struct device *dev) { snd_soc_unregister_platform(dev); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..0e86315 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,7 +22,7 @@ struct s3c_dma_params { char *ch_name; }; -int asoc_dma_platform_register(struct device *dev); -void asoc_dma_platform_unregister(struct device *dev); +int samsung_asoc_dma_platform_register(struct device *dev); +void samsung_asoc_dma_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ce487c2..b302f3b 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1146,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_component(&sec_dai->pdev->dev, &samsung_i2s_component, &sec_dai->i2s_dai_drv, 1); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1263,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; err: @@ -1293,7 +1293,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ASoC: samsung: Check to see if we managed to allocate a channel
by Mark Brown 20 Aug '13

20 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 6e2b2b4..d54df85 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -183,6 +183,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ch = prtd->params->ops->request( prtd->params->channel, &req, rtd->cpu_dai->dev, prtd->params->ch_name); + if (!prtd->params->ch) { + pr_err("Failed to allocate DMA channel\n"); + return -ENXIO; + } prtd->params->ops->config(prtd->params->ch, &config); } -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH RFC 0/9] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 19 Aug '13

19 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. As Clemens found out, when implicit feedback is used, playback and capture cannot be used in the same time - unless started at the same time (e.g., with jack). While a substream is active, attempting to start a substream in the opposite direction will fail and also (on some devices?) cause the currently active stream to stop. There are multiple issues that contribute to this: 1. URBs are unconditionally deactivated in hw_free. On implicit feedback devices, the sync_endpoint (= data_endpoint of the capture) will be deactivated here. Fixed in patch #4. 2. Changing the alternate settings of an interface might break currently running streams. This happens on my device, but I suspect other devices too. Fixed in patch #7. 3. The endpoint use_count has a subtle issue (see patch #9). This causes jack to work fine, while using two separate programs (most probably) fails. In short: set_format is called from hw_params, but use_count is incremented only later (prepare or trigger). 4. Concurrent usage should be allowed but only if the hw_params match up (patch #8) One problem with this series is that I query the device for the altsettting. This might not be reliable for all devices, so do I plan to remove this. Of course, I've only tested this on my device, but it seems to work good and I couldn't break it so far, so hopefully I got this right. Can someone give it a try on other devices (also on non-implicit-feedback)? Cheers, Eldad Eldad Zack (9): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: correct ep use_count semantics (add set_param flag) sound/usb/card.h | 1 + sound/usb/endpoint.c | 65 +++++++++++++++------ sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 155 +++++++++++++++++++++++++++++++++++++++------------ 4 files changed, 176 insertions(+), 56 deletions(-) -- 1.8.1.5

2 11

[alsa-devel] [PATCH 0/3] HDSPM: Fix LTC ioctl
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! This series fixes the broken LTC ioctl to query the TCO. Apparently, nobody ever attempted to use the code, otherwise, this rather obvious problem would have been spotted way earlier. This is maybe because not too many people happen to have a TV studio around to make use of this feature. ;) I'll also send a patch to fix alsa-lib. Cheers Adrian Knoth (3): ALSA: hdspm - Fix default value in SNDRV_HDSPM_IOCTL_GET_LTC ALSA: hdspm - Use enums in hdspm_tco_ltc_frames() ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC include/uapi/sound/hdspm.h | 2 +- sound/pci/rme9652/hdspm.c | 12 ++++++------ 2 files changed, 7 insertions(+), 7 deletions(-) -- 1.8.3.2

2 4

[alsa-devel] [PATCH 0/1] alsa-lib: update hdspm.h
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! Here's the patch for alsa-lib to fix the broken LTC ioctl. Cheers Adrian Knoth (1): hdspm.h: Update LTC ioctl to use struct hdspm_ltc include/sound/hdspm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 1.8.3.2

1 1

Re: [alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 19 Aug '13

19 Aug '13

On Mon, Aug 19, 2013 at 05:13:57PM +0200, Marc Kleine-Budde wrote: > On 08/19/2013 05:13 PM, Marc Kleine-Budde wrote: > > On 08/19/2013 05:05 PM, Markus Pargmann wrote: > >> Michael Grzeschik (2): > >> ASoC: fsl-ssi: imx-pcm-fiq bugfix > >> ASoC: fsl: disable ssi irq for imx > >> > >> Steffen Trumtrar (1): > >> ASoC: fsl-ssi: add SSIEN errata work around > > > > You should add you S-o-b to patches from other develops. > developers yes, I forgot that, thanks Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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Re: [alsa-devel] [PATCH 3.11-rc5 1/2] ALSA: pcm: Add snd_printd_ratelimit()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Fri, 16 Aug 2013 11:18:59 -0600, Tim Gardner wrote: > > Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not > defined. Add a wrapper macro around printk_limit() that is conditionally defined by > CONFIG_SND_DEBUG. > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Yacine Belkadi <yacine.belkadi.1(a)gmail.com> > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> > --- > > checkpatch is moaning about the use of printk_ratelimit(), but > printk_ratelimited() seems like overkill for a debug feature. > > WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit > #38: FILE: include/sound/core.h:382: > +#define snd_printd_ratelimit() printk_ratelimit() Thanks, applied both patches now. But it's no real bug, so I postpone this for 3.12 inclusion. Takashi > > include/sound/core.h | 8 ++++++++ > 1 file changed, 8 insertions(+) > > diff --git a/include/sound/core.h b/include/sound/core.h > index c586617..2a14f1f 100644 > --- a/include/sound/core.h > +++ b/include/sound/core.h > @@ -27,6 +27,7 @@ > #include <linux/rwsem.h> /* struct rw_semaphore */ > #include <linux/pm.h> /* pm_message_t */ > #include <linux/stringify.h> > +#include <linux/printk.h> > > /* number of supported soundcards */ > #ifdef CONFIG_SND_DYNAMIC_MINORS > @@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line, > #define snd_BUG() WARN(1, "BUG?\n") > > /** > + * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. > + */ > +#define snd_printd_ratelimit() printk_ratelimit() > + > +/** > * snd_BUG_ON - debugging check macro > * @cond: condition to evaluate > * > @@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {} > unlikely(__ret_warn_on); \ > }) > > +static inline bool snd_printd_ratelimit(void) { return false; } > + > #endif /* CONFIG_SND_DEBUG */ > > #ifdef CONFIG_SND_DEBUG_VERBOSE > -- > 1.7.9.5 >

1 0

Re: [alsa-devel] [PATCH v1 45/49] sound: usb: caiaq: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:10 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.e> thanks, Takashi > Cc: alsa-devel(a)alsa-project.org > Acked-by: Daniel Mack <zonque(a)gmail.com> > Signed-off-by: Ming Lei <ming.lei(a)canonical.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; > -- > 1.7.9.5 >

1 0

Re: [alsa-devel] [PATCH v1 44/49] sound: usb: midi: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:09 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > sound/usb/midi.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/midi.c b/sound/usb/midi.c > index b901f46..86af276 100644 > --- a/sound/usb/midi.c > +++ b/sound/usb/midi.c > @@ -279,15 +279,16 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb) > struct out_urb_context *context = urb->context; > struct snd_usb_midi_out_endpoint* ep = context->ep; > unsigned int urb_index; > + unsigned long flags; > > - spin_lock(&ep->buffer_lock); > + spin_lock_irqsave(&ep->buffer_lock, flags); > urb_index = context - ep->urbs; > ep->active_urbs &= ~(1 << urb_index); > if (unlikely(ep->drain_urbs)) { > ep->drain_urbs &= ~(1 << urb_index); > wake_up(&ep->drain_wait); > } > - spin_unlock(&ep->buffer_lock); > + spin_unlock_irqrestore(&ep->buffer_lock, flags); > if (urb->status < 0) { > int err = snd_usbmidi_urb_error(urb->status); > if (err < 0) { > -- > 1.7.9.5 >

1 0

[alsa-devel] [PATCH] ALSA: hda - Try to allow haswell HDMI audio even without powerwell
by David Henningsson 19 Aug '13

19 Aug '13

If compiled without CONFIG_SND_HDA_I915, the audio driver cannot request power well. However, if the power well is on for other reasons, maybe audio can still work. Therefore, do not skip the card completely if compiled without CONFIG_SND_HDA_I915. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) I'm going to apply this on my local DKMS tree, it's important for our backports as we combine different versions of the kernel with backports for haswell etc. I'm not sure if this is beneficial to have on 3.11+ kernels as well, but I guess it could be. At least it shouldn't harm. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f9e406..c6c9829 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3855,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip) /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { +#ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); goto out_free; } +#endif hda_display_power(true); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Limit internal mic boost for a few more Thinkpad machines
by David Henningsson 19 Aug '13

19 Aug '13

The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d67a48c..00d00b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,9 +3704,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ac97: Provide stub DAPM integration
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ac97.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ec73518..8d9ba4ba 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: hdmi: Provide stub DAPM integration
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/hdmi.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index f0986b9..68342b1 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver hdmi_codec; +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; static struct snd_soc_dai_driver hdmi_codec_dai = { .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -38,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_FMTBIT_S24_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | @@ -50,6 +60,13 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + static int hdmi_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &hdmi_codec, -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/2] ASoC: bt-sco: Add generic compatible string
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Provide a common compatible string for device trees to list as a fallback for simplicity. We don't currently have a binding document but let's not fix that right now... Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/bt-sco.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9f..5c040ce 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -50,6 +50,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", }, + { + .name = "bt-sco", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH] ASoC: spdif: Add stub DAPM widgets for Rx
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure that the driver continues to work with mandatory DAPM. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/spdif_receiver.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 26d3474..d5463c0 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -23,13 +23,26 @@ #include <sound/initval.h> #include <linux/of.h> +static const struct snd_soc_dapm_widget dir_widgets[] = { + SND_SOC_DAPM_INPUT("spdif-in"), +}; + +static const const struct snd_soc_dapm_route dir_routes[] = { + { "Capture", NULL, "spdif-in" }, +}; + #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir; +static struct snd_soc_codec_driver soc_codec_spdif_dir = { + .dapm_widgets = dir_widgets, + .num_dapm_widgets = ARRAY_SIZE(dir_widgets), + .dapm_routes = dir_routes, + .num_dapm_routes = ARRAY_SIZE(dir_routes), +}; static struct snd_soc_dai_driver dir_stub_dai = { .name = "dir-hifi", -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ALSA: hda - Limit internal mic boost for another Thinkpad machine
by David Henningsson 19 Aug '13

19 Aug '13

The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 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 d67a48c..ef09fbc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,6 +3704,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.7.9.5

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

On Mon, Aug 19, 2013 at 02:31:47PM +0800, Bhushan Bharat-R65777 wrote: > > > > We here suppose the reset bit would be cleared -- "The software > > > > reset will last > > > > 8 cycles." from RM, so if this happened to be a failure, the whole > > > > IP module won't be normally working as well. > > > > > > Also add a comment describing this against why cycle = 1000 is selected. > > > > If it is done in 8 cycles, 1000-cycle will be surely a safe value for it. > > As long as it finished in 8 cycles, it would quit anyway. Why against? > > I am not against, I am saying why it was not 200 or 50 or 20 etc. I am saying that write a comment saying this much is sufficient as per specification and so keep 1000/etc as preservative. I did't mean that. The 'against' is from "Also add a comment describing this 'against' why cycle = 1000 is selected." Well, if you insist this extra comment for easy-understand, I'll add them Thank you. Nicolin Chen

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

Hi Bhushan, I'll revise some as you suggest. Just a few replies here. On Mon, Aug 19, 2013 at 12:38:11PM +0800, Bhushan Bharat-R65777 wrote: > > We here suppose the reset bit would be cleared -- "The software reset will last > > 8 cycles." from RM, so if this happened to be a failure, the whole IP module > > won't be normally working as well. > > Also add a comment describing this against why cycle = 1000 is selected. If it is done in 8 cycles, 1000-cycle will be surely a safe value for it. As long as it finished in 8 cycles, it would quit anyway. Why against? > > > > +static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) > > > > +{ > > > > + /* Sync all registers after reset */ > > > > > > Where us sync :) ? > > > > The "return true" would do that. For volatile registers, if no "return true" > > here, the whole regmap would use the value in cache, while for some bits > > we need to trace its true value from the physical registers not from cache. > > Where will be device registers cached? Do not we program them to be non-cacheable in core? regmap has a regcache for all the mapped registers. Set the regsiters as volatile will allow the driver to sync the regcache with physical memory each time when using regmap_read/write/update_bits(). But I think I can try to use the regcache_bypass instead. Thank you, Nicolin Chen

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

Hi Bhushan, Thank you for the comments :) I'll fix some in v7. Here is my some replies to you. On Sat, Aug 17, 2013 at 02:24:19AM +0800, Bhushan Bharat-R65777 wrote: > > This patch add S/PDIF controller driver for Freescale SoC. > > Please give some more description of the driver? I've referred some ASoC drivers, all of them seem to be brief as mine. So I'm not sure what else information I should provide here. It's already kinda okay to me. > > +struct spdif_mixer_control { > > + /* buffer ptrs for writer */ > > + u32 upos; > > + u32 qpos; > > They does not look like pointer? They are more like offsets to get the correspond pointer. But I'll change the confusing comments. > > +/* U/Q Channel receive register full */ > > +static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) > > +{ > > + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; > > + struct regmap *regmap = spdif_priv->regmap; > > + struct platform_device *pdev = spdif_priv->pdev; > > + u32 *pos, size, val, reg; > > + > > + switch (name) { > > + case 'U': > > + pos = &ctrl->upos; > > + size = SPDIF_UBITS_SIZE; > > + reg = REG_SPDIF_SRU; > > + break; > > + case 'Q': > > + pos = &ctrl->qpos; > > + size = SPDIF_QSUB_SIZE; > > + reg = REG_SPDIF_SRQ; > > + break; > > + default: > > + return; > > Should return error. IMHO, this should be fine. It's a void type function and being used in the isr(). The params 'name' is totally controlled by driver itself, so basically we don't need to worry about the default path. > > + if (*pos >= size * 2) { > > + *pos = 0; > > + } else if (unlikely((*pos % size) + 3 > size)) { > > + dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); > > + return; > > Should return error. Ditto, it's being used in isr(), we don't need to detect the return value, just use dev_err() to warn users and let the driver clear the irq. > > +/* U/Q Channel framing error */ > > +static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv) > > +{ > > + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; > > + struct regmap *regmap = spdif_priv->regmap; > > + struct platform_device *pdev = spdif_priv->pdev; > > + u32 val; > > + > > + dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n"); > > + > > + /* read U/Q data and do buffer reset */ > > + regmap_read(regmap, REG_SPDIF_SRU, &val); > > + regmap_read(regmap, REG_SPDIF_SRQ, &val); > > Above prints says read u/q data and buffer reset, what is buffer reset? Is that read on clear? That's the behavior needed by IP, according to the reference manual: "U Channel receive register full, can't be cleared with reg. IntClear. To clear it, read from U Rx reg." and "Q Channel receive register full, can't be cleared with reg. IntClear. To clear it, read from Q Rx reg." > > +static void spdif_softreset(struct fsl_spdif_priv *spdif_priv) > > +{ > > + struct regmap *regmap = spdif_priv->regmap; > > + u32 val, cycle = 1000; > > + > > + regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET); > > + regcache_sync(regmap); > > + > > + /* RESET bit would be cleared after finishing its reset procedure */ > > + do { > > + regmap_read(regmap, REG_SPDIF_SCR, &val); > > + } while ((val & SCR_SOFT_RESET) && cycle--); > > What if reset is not cleared and timeout happen? We here suppose the reset bit would be cleared -- "The software reset will last 8 cycles." from RM, so if this happened to be a failure, the whole IP module won't be normally working as well. Well, but I don't mind to put here an extra failed return to make it clear. > > +static u8 reverse_bits(u8 input) > > +{ > > + u8 tmp = input; > > + > > + tmp = ((tmp & 0b10101010) >> 1) | ((tmp << 1) & 0b10101010); > > + tmp = ((tmp & 0b11001100) >> 2) | ((tmp << 2) & 0b11001100); > > + tmp = ((tmp & 0b11110000) >> 4) | ((tmp << 4) & 0b11110000); > > What is this logic, can the hardcoding be removed and some description on above calculation? This was provided by Philipp Zabel in his review at patch v3. It's pretty clear to me that it just reverses the bits for u8. I don't think this logic has any problem and the mask here doesn't look like any hardcode to me. > > +static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) > > +{ > > + /* Sync all registers after reset */ > > Where us sync :) ? The "return true" would do that. For volatile registers, if no "return true" here, the whole regmap would use the value in cache, while for some bits we need to trace its true value from the physical registers not from cache. Best regards, Nicolin Chen

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[alsa-devel] [PATCH 0/4] clk: Samsung: audss: Add support for Exynos5420
by Padmavathi Venna 18 Aug '13

18 Aug '13

This patch set adds support for audio subsystem clks on Exynos5420. Exynos5420 audio subsystem has a gate bit for ADMA controller and the some of parent clks for mout_i2s are also different from Exynos5250. So this patch adds provision for supporting both the platforms by passing the parent clk names through device tree. Andrew Bresticker (3): clk: exynos-audss: add support for Exynos 5420 clk: exynos-audss: allow input clocks to be specified in device tree ARM: dts: exynos5420: add audio clock controller Padmavathi Venna (1): ARM: dts: Correct the /include entry on exynos5420 dtsi file .../devicetree/bindings/clock/clk-exynos-audss.txt | 38 +++++++++++++++++-- arch/arm/boot/dts/exynos5420.dtsi | 13 ++++++- drivers/clk/samsung/clk-exynos-audss.c | 36 ++++++++++++++++-- include/dt-bindings/clk/exynos-audss-clk.h | 3 +- 4 files changed, 80 insertions(+), 10 deletions(-) -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: wm8782: Add DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8782.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f1fdbf6..8092495 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -26,6 +26,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782", .capture = { @@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782; +static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { + .dapm_widgets = wm8782_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), + .dapm_routes = wm8782_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), +}; static int wm8782_probe(struct platform_device *pdev) { -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: wm8727: Add DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8727.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4..7b1a6d5 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route wm8727_dapm_routes[] = { + { "VOUTL", NULL, "Playback" }, + { "VOUTR", NULL, "Playback" }, +}; + /* * Note this is a simple chip with no configuration interface, sample rate is * determined automatically by examining the Master clock and Bit clock ratios @@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727; +static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { + .dapm_widgets = wm8727_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), + .dapm_routes = wm8727_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes), +}; static int wm8727_probe(struct platform_device *pdev) { -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: wl1273: Add stub DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to ensure that the device continues to work with DAPM support being mandatory provide stub DAPM widgets and routes. Note that the public information on the device appears to make no mention of the FM support the driver appears to have. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wl1273.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 54cd3da..b7ab2ef 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = { snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), }; +static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route wl1273_dapm_routes[] = { + { "Capture", NULL, "RX" }, + + { "TX", NULL, "Playback" }, +}; + static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + + .dapm_widgets = wl1273_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), + .dapm_routes = wl1273_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes), }; static int wl1273_platform_probe(struct platform_device *pdev) -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/1] ASoC: wm8960: Fix PLL register writes
by Mike Dyer 18 Aug '13

18 Aug '13

Bit 9 of PLL2,3 and 4 is reserved as '0'. The 24bit fractional part should be split across each register in 8bit chunks. Signed-off-by: Mike Dyer <mike.dyer(a)md-soft.co.uk> --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..5e5af89 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code
by Fabio Estevam 18 Aug '13

18 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e15f771..3a4808d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_IMX_WM8962 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -187,7 +186,6 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. -- 1.8.1.2

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[alsa-devel] [PATCH v4 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 17 Aug '13

17 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH v6 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 16 Aug '13

16 Aug '13

Changelog: v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1272 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1733 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH 1/2] ALSA: hda - Fix internal mic boost on three Thinkpad machines
by David Henningsson 16 Aug '13

16 Aug '13

The internal mic boost is so noisy on boosts 2 and 3 so they are unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213055 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae9504..6ec96f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,6 +3704,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [git pull] [PATCH 0/7] add support for many Roland and Yamaha devices
by Clemens Ladisch 16 Aug '13

16 Aug '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git full-roland-support for you to fetch changes up to b7f33917bcd993ff81f3f80b9dc1890fb7410c6d: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE (2013-06-27 21:59:50 +0200) ---------------------------------------------------------------- Clemens Ladisch (7): ALSA: usb-audio: store protocol version in struct audioformat ALSA: usb-audio: detect implicit feedback on Roland devices ALSA: usb-audio: add support for many Roland/Yamaha devices ALSA: usb-audio: add MIDI port names for some Roland devices ALSA: usb-audio: remove superfluous Roland quirks ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE sound/usb/card.h | 1 + sound/usb/clock.c | 4 +- sound/usb/format.c | 34 +--- sound/usb/format.h | 2 +- sound/usb/midi.c | 74 +++++++ sound/usb/pcm.c | 45 ++++- sound/usb/quirks-table.h | 497 +++++++++-------------------------------------- sound/usb/quirks.c | 209 ++++++++++++++++++++ sound/usb/stream.c | 18 +- sound/usb/usbaudio.h | 2 + 10 files changed, 445 insertions(+), 441 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Remove direct use of internal I/O functions
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Use the core to do I/O rather than directly calling the driver operations in order to support further refactoring. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic26.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b192cd4..a4f9360 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -174,9 +174,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +185,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +212,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -348,7 +348,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -368,20 +368,20 @@ static int aic26_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: imx: Add MODULE_LICENSE to DMA drivers
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Ben Hutchings <ben(a)decadent.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-pcm-dma.c | 2 ++ sound/soc/fsl/imx-pcm-fiq.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f323ce0..aa6de5c 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -73,3 +73,5 @@ void imx_pcm_dma_exit(struct platform_device *pdev) snd_dmaengine_pcm_unregister(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 3b2ba99..34043c5 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -408,3 +408,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev) snd_soc_unregister_platform(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); + +MODULE_LICENSE("GPL"); -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH] ASoC: si476x: Add DAPM support
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> This ensures the driver continues to work with DAPM mandatory and makes it easier to connect the device up to other components in the subsystem. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c..38f3b10 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH] ASoC: imx-audmux: Move definitions to dt-bindings
by Markus Pargmann 16 Aug '13

16 Aug '13

Move imx-audmux macro definitions to include/dt-bindings, so they can be used for devicetree. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 56 ++++++++++++++++++++++++++++++ sound/soc/fsl/imx-audmux.h | 52 +-------------------------- 2 files changed, 57 insertions(+), 51 deletions(-) create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h new file mode 100644 index 0000000..50b09e9 --- /dev/null +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -0,0 +1,56 @@ +#ifndef __DT_FSL_IMX_AUDMUX_H +#define __DT_FSL_IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +#endif /* __DT_FSL_IMX_AUDMUX_H */ diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index b8ff44b..38a4209 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -1,57 +1,7 @@ #ifndef __IMX_AUDMUX_H #define __IMX_AUDMUX_H -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 -#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) +#include <dt-bindings/sound/fsl-imx-audmux.h> int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); -- 1.8.3.2

2 1

Re: [alsa-devel] RealTek Inverted Mic Patch
by Takashi Iwai 16 Aug '13

16 Aug '13

At Thu, 15 Aug 2013 14:27:07 -0600, Nathanael D. Noblet wrote: > > Hello, > > I have hardware that suffers from an internal phase inverted mic that > isn't properly detected. Below is a patch that fixes the issue for my > particular hardware. > > Pre patch alsa-info: > > http://www.alsa-project.org/db/?f=a922c77ddf473a8f478ec334fa9dca684f213b59 > > Post patch alsa-info: > http://www.alsa-project.org/db/?f=14cd25b323d3bbb02fabbac78e62851569893c3f > > Incase you care. The patch is small. I don't know if you need any kind > of signed off/written by etc for something so small. In any case there > is no intellectual property in this at all. Feel free to do as you wish > with it. However including it in the kernel would make my life easier. Thanks, I fixed it in sound git tree now. Takashi > > Thanks, > -- > Nathanael d. Noblet > t 403.875.4613 > [2 alsa-realtek-inverted-mic.patch <text/x-patch (7bit)>] > --- a/sound/pci/hda/patch_realtek.c.org 2013-08-15 13:17:49.804492932 -0600 > +++ b/sound/pci/hda/patch_realtek.c 2013-08-15 13:18:40.582364931 -0600 > @@ -4208,6 +4208,7 @@ > SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), > SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), > SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), > + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), > > #if 0 > /* Below is a quirk table taken from the old code.

1 0

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

15 Aug '13

The following changes since commit d4e4ab86bcba5a72779c43dc1459f71fea3d89c8: Linux 3.11-rc5 (2013-08-11 18:04:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc5 for you to fetch changes up to 44ffb69ec617f64cd83182e8b64d93b84758d636: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-08-15 11:37:54 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A few driver specific fixes here plus one core fix for a memory corruption issue in DAPM initialisation which could lead to crashes. ---------------------------------------------------------------- Brian Austin (2): ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume ASoC: cs42l52: Add new TLV for Beep Volume Lars-Peter Clausen (1): ASoC: dapm: Fix empty list check in dapm_new_mux() Lothar Waßmann (2): ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control Mark Brown (4): Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Stephen Warren (1): ASoC: tegra: fix Tegra30 I2S capture parameter setup sound/soc/codecs/cs42l52.c | 5 ++++- sound/soc/codecs/sgtl5000.c | 18 ++++++++++++++---- sound/soc/soc-dapm.c | 7 ++++--- sound/soc/tegra/tegra30_i2s.c | 2 +- 4 files changed, 23 insertions(+), 9 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: core: allow SND_DMAENGINE_PCM use from modules
by Daniel Mack 15 Aug '13

15 Aug '13

When users of SND_DMAENGINE_PCM are built as module, the config symbol SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/core/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 94ce1c4..313f22e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -7,7 +7,7 @@ config SND_PCM select SND_TIMER config SND_DMAENGINE_PCM - bool + tristate config SND_HWDEP tristate -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH 0/2] add sysfs for acpi interfaces of thinkpad hardware mute
by David Henningsson 15 Aug '13

15 Aug '13

(Added alsa-devel to CC) On 08/14/2013 09:16 AM, Alex Hung wrote: > Added in David and Tim who also has some ideas to share. > > On Wed, Aug 14, 2013 at 3:00 PM, Matthew Garrett > <matthew.garrett(a)nebula.com> wrote: >> On Wed, 2013-08-14 at 14:41 +0800, Alex Hung wrote: >>> Hi Matthew, >>> >>> The problem is that thinkpad-acpi does not aware of mute is changed by >>> either hotkey presses or in desktop. >> >> Ok, how about this. Register the LED with the kernel LED subsystem. Add >> a new LED trigger for volume-mute. Have the HDA driver call that trigger >> whenever the mute state changes. >> >> -- >> Matthew Garrett <matthew.garrett(a)nebula.com> > > > So, to take a wider grip on this issue. First, the hardware mute. Because this is directly connected to the internal speaker (and headphones?), it needs to turn into a ALSA kcontrol on the sound card that controls speaker and/or headphones. And it needs to named accordingly, e g "Speaker Playback Switch" if it controls the speaker only. Exactly how this is going to work out given the arbitrary module load order of snd-hda-intel and thinkpad-acpi, I'm not sure, but the way of creating another sound card (which I've seen on some thinkpads) just isn't working out for userspace. Second, about the mute LEDs and mic-mute LEDs, we currently have several interfaces to choose from: a) HP LEDs currently uses a ALSA kcontrol to control them. For Mute LED there is also a limited automatic control of this LED from kernel space (by default), so it follows the status of "Master Playback Switch" IIRC. b) thinkpad-acpi currently uses custom sysfs nodes to control the LEDs. c) and we also have the /sys/class/leds interface. There might be even more solutions. It's important we get a consistent interfaces towards userspace for the LEDs. And we should also make sure we don't have a permissions problem: if we want the mute/mic-mute LED to follow the status of the currently selected sound card in PulseAudio, which IMO should be our goal, writes cannot be root-only. To have the HDA driver call into the LED subsystem is better than nothing, but in the end we would want the LEDs to follow whatever sound input/output the user currently uses, whether that is bluetooth, USB, HDMI, or internal audio. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 11

[alsa-devel] [PATCH 0/5] ASoC: pxa dmaengine preparation patches
by Daniel Mack 15 Aug '13

15 Aug '13

Mark, This series is an excerpt of my pxa-dma rework that can be merged independently in the next merge window. The rest of the series will take some time to get sorted out, and given that -rc5 is released already, I don't think the entire series will make it in time. Thanks, Daniel Daniel Mack (5): ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 ++ .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + include/sound/pxa2xx-lib.h | 7 - sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 ++- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + sound/core/pcm_dmaengine.c | 367 +++++++++++++++++++++ sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/omap/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 8 +- sound/soc/pxa/mmp-sspa.c | 12 +- sound/soc/pxa/pxa-ssp.c | 76 +++-- sound/soc/pxa/pxa2xx-ac97.c | 67 ++-- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/soc-dmaengine-pcm.c | 367 --------------------- sound/soc/spear/Kconfig | 2 +- 22 files changed, 607 insertions(+), 499 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 sound/core/pcm_dmaengine.c delete mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.8.3.1

3 14

[alsa-devel] [PATCH 1/2] ASoC: pcm3008: Add DAPM support
by Mark Brown 15 Aug '13

15 Aug '13

From: Mark Brown <broonie(a)linaro.org> Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378..19f5028 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- 1.8.4.rc2

1 1

[alsa-devel] [PATCH] ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document this in the DT binding. This enables the DT to include the Mic Jack in the audio routing table, and hence enables capture of audio, in addition to the previously-working playback. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt | 1 + sound/soc/tegra/tegra_rt5640.c | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt index cba4f88..dc62249 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -16,6 +16,7 @@ Required properties: * Headphones * Speakers + * Mic Jack - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f9..4511c5a 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), }; static const struct snd_kcontrol_new tegra_rt5640_controls[] = { -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: tegra: fix Tegra30 I2S capture parameter setup
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable(a)vger.kernel.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146c..47565fd04 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- 1.8.1.5

2 1

[alsa-devel] [PATCH v4 resent 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 15 Aug '13

15 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

3 10

[alsa-devel] [PATCH 1/16 v2] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 14 Aug '13

14 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. At the suggestion of Robert Jarzmik, this additionally removes the definition of ARRAY_AND_SIZE, which is not used elsewhere in the file. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. v2 removes the definition of ARRAY_AND_SIZE. diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..bbea778 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -56,8 +56,6 @@ #include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" -#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) - #define AC97_GPIO_PULL 0x58 /* Use GPIO8 for rear speaker amplifier */ @@ -133,10 +131,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

2 4

[alsa-devel] [PATCH 2/2] ASoC: wm8960: Fix shared LRCLK support
by Ma Haijun 14 Aug '13

14 Aug '13

Shared LRCLK initialization does not survive wm8960_reset, so place it after the reset. Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 368d39f..b606dd3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -990,6 +990,9 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pdata && pdata->shared_lrclk) + snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1041,7 +1044,6 @@ static const struct regmap_config wm8960_regmap = { static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1054,16 +1056,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } - i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8960: Fix ADC volume bits
by Ma Haijun 14 Aug '13

14 Aug '13

Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..368d39f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 0/29] simplify use of devm_ioremap_resource
by Julia Lawall 14 Aug '13

14 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call site. To make the connection between the call to platform_get_resource and the call to devm_ioremap_resource more clear, the former is also moved down to be adjacent to the latter. In many cases, this patch changes the specific error value that is returned on failure of platform_get_resource. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ ( res = platform_get_resource(pdev, IORESOURCE_MEM, n); - if (res == NULL) { ... $goto l;\|return ret;$ } e = devm_ioremap_resource(e1, res); | - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); ) // </smpl>

2 3

[alsa-devel] HDMI audio is not working on my new desktop
by Alexander E. Patrakov 14 Aug '13

14 Aug '13

Hello. I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. It has one DVI and two HDMI outputs. However, I cannot get any sound over HDMI. The same OS image works perfectly on my laptop that also has an HDMI output. This is not a pulseaudio problem, as I don't get any sound even if I test with this command: pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 Alsa-info.sh has uploaded its output to http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 Please note this error in dmesg, it may be relevant: [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 -- Alexander E. Patrakov

1 0

Re: [alsa-devel] Query on Audio DMA using DMAEngine
by Lars-Peter Clausen 14 Aug '13

14 Aug '13

Added alsa-devel to Cc. On 06/28/2013 05:27 AM, Fernandes, Joel wrote: > Hi Lars, > > Hope you are doing well. > > I am implementing Cyclic DMA support in the EDMA driver that is used by Davinci and now newer TI SoCs. > I am thinking once I am done I can plug it into the snd_dmaengine framework. > > Currently however, the davinci-pcm code directly programs the EDMA. That is what I am working to replace with a single driver and adapt to the snd dmaengine framework. However, once the current code in davinci-pcm uses internal RAM as an intermediate step in the whole DMA process (First data is TX to IRAM from DRAM and then from DRAM to the audio device). > > Do you have any ideas on how we can adapt to the framework, such that we can till use the IRAM? Are there any existing implementations out there that do something similar? Hm, I guess using the snd_dmaengine_pcm helper functions here shouldn't be too hard. Using the generic snd_dmaengine_pcm driver will require some extensions to it though. The mmp platform (pxa/mmp-pcm.c) is also using some kind of on-chip memory, so having support for this in the generic driver certainly makes sense. For the chaining you'd probably have to extend the dmaengine framework, since this kind of interleaved mem-to-mem and mem-to-dev cyclic transfer is currently not possible. I'm wondering though why do you need to copy the data to RAM first, is it not possible to map the IRAM to userspace? - Lars

6 28

[alsa-devel] [PATCH v5 0/1] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 14 Aug '13

14 Aug '13

This patch series adds DT support to the kirkwood/mvebu audio subsystem. History v5 - change the DT name to "marvell,mvebu-audio" v4 - change the DT name to "mrvl,mvebu-audio" - remove previous patches 1..3 which have been applied by Russell King v3 - change the DT name from "kirkwood-i2s" to "mvebu-pcm-audio" - apply to linux-next - let NULL as the internal i2s clock for compatibility - change i2s external clock name to "extclk" instead of "external" - cancel previous patch 1/4. Russell's patch should be applied instead. v2 - fix compilation errors -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH] ASoC: pcm1681: Add DAPM support
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 51b1866..651ce09 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -206,6 +206,28 @@ static const struct snd_soc_dai_ops pcm1681_dai_ops = { .digital_mute = pcm1681_digital_mute, }; +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); static const struct snd_kcontrol_new pcm1681_controls[] = { @@ -258,6 +280,10 @@ static const struct regmap_config pcm1681_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), }; static const struct i2c_device_id pcm1681_i2c_id[] = { -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH 2/3 v2] ASoC: wm8997: Initial CODEC driver
by Charles Keepax 13 Aug '13

13 Aug '13

The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus interfacing, for smartphones, tablets and other portable audio devices based on the Arizona platform. This patch adds the wm8997 CODEC driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1187 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 5 files changed, 1228 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..14d2888 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -143,8 +144,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -492,6 +495,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..2d327ad 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -235,6 +236,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index de62581..c460fca 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -199,9 +199,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 0000000..3ba0a98 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1187 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, + { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, + + { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, + { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, + + { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, + { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, + + { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, + { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 0000000..5e91c6a --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(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 _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/3] ASoC: max9877: Convert to use regmap API
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9877.c | 187 ++++++++++++++++++++++++++++----------------- 1 file changed, 118 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74c..7e2fe50 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,27 +14,21 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <sound/tlv.h> #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static int max9877_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -45,8 +39,14 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int mask = mc->max; unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = @@ -65,18 +65,21 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int invert = mc->invert; unsigned int val = (ucontrol->value.integer.value[0] & mask); + bool change; + int ret; if (invert) val = mask - val; - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change); + if (ret != 0) + return ret; - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_2reg(struct snd_kcontrol *kcontrol, @@ -88,9 +91,18 @@ static int max9877_get_2reg(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int mask = mc->max; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[0] = (val >> shift) & mask; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + ret = regmap_read(regmap, reg2, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[1] = (val >> shift) & mask; return 0; } @@ -106,77 +118,99 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; + bool change1, change2; + int ret; + + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change1); + if (ret != 0) + return ret; + + ret = regmap_update_bits_check(regmap, reg2, mask << shift, + val2 << shift, &change2); + if (ret != 0) + return ret; + + if (change1 || change2) + return 1; + else + return 0; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OUTMODE_MASK; + if (val) + val--; - if (value) - value -= 1; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; + unsigned int val; + bool change; + int ret; - value += 1; + val = ucontrol->value.integer.value[0] + 1; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OUTMODE_MASK, val, &change); + if (ret != 0) + return ret; - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OSC_MASK; + val >>= MAX9877_OSC_OFFSET; - value = value >> MAX9877_OSC_OFFSET; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + unsigned int val; + bool change; + int ret; + + val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OSC_MASK, val, &change); + if (ret != 0) + return ret; + + if (change) + return 1; + else return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; } static const unsigned int max9877_pgain_tlv[] = { @@ -258,19 +292,34 @@ int max9877_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; + static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; + + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - max9877_write_regs(); + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); return 0; } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + regmap = NULL; return 0; } -- 1.8.4.rc2

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[alsa-devel] [PATCH v2] extcon: arizona: Clear trig_sts bits on all paths
by Charles Keepax 13 Aug '13

13 Aug '13

We want the trig_sts bits to be cleared in all cases where we consider the jack detection interrupt to have been handled. Specifically, if a duplicate detection event was suppressed these bits were not cleared causing the CODEC to not enter a low power state. This patch clears the bits on the duplicate detection code path. Reported-by: Ryo Tsutsui <ryo.tsutsui(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/extcon/extcon-arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 7a1b4a7..4df68de 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -1034,6 +1034,7 @@ static irqreturn_t arizona_jackdet(int irq, void *data) else info->micd_timeout = DEFAULT_MICD_TIMEOUT; +out: /* Clear trig_sts to make sure DCVDD is not forced up */ regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG, ARIZONA_MICD_CLAMP_FALL_TRIG_STS | @@ -1041,7 +1042,6 @@ static irqreturn_t arizona_jackdet(int irq, void *data) ARIZONA_JD1_FALL_TRIG_STS | ARIZONA_JD1_RISE_TRIG_STS); -out: mutex_unlock(&info->lock); pm_runtime_mark_last_busy(info->dev); -- 1.7.2.5

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[alsa-devel] [PATCH] Add PM support to the rme96 driver
by Knut Petersen 13 Aug '13

13 Aug '13

Hi everybody! The attached patch adds PM support to the rme96 driver, it should be applied after v2 of the patch adding pcm stream synchronization support posted on 2013-08-08. It does work perfectly here on an RME Digi96/8 PAD: After connecting analog output and analog input executing #! /bin/sh # arecord -D hw:0,0 -d 62 testout.wav & aplay -D hw:0,0 -d 60 testin.wav produces an almost perfect recording even when the test machine is suspended / hibernated and resumed several times. cu, Knut

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[alsa-devel] [PATCH] ASoC: dapm: Ensure kcontrol list is initialised
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure that the recently added path kcontrol list is initialised otherwise we may crash trying to delete routes that don't have kcontrols. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b78eaa1..1e0e4ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2357,6 +2357,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, path->sink = wsink; path->connected = connected; INIT_LIST_HEAD(&path->list); + INIT_LIST_HEAD(&path->list_kcontrol); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); -- 1.8.4.rc1

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[alsa-devel] [PATCH] ASoC: dapm: Don't create routes when creating kcontrols
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Attempting to create the route as part of adding a mux control causes us to attempt to add the same route twice since we loop over all sources for the mux after creating the control. Instead do the addition in the callers. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 662a904..b78eaa1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -665,7 +665,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * create it. Either way, add the widget into the control's widget list */ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, - int kci, struct snd_soc_dapm_path *path) + int kci) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; @@ -766,7 +766,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, return ret; w->kcontrols[kci] = kcontrol; - dapm_kcontrol_add_path(kcontrol, path); return 0; } @@ -790,9 +789,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; } - ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, i); if (ret < 0) return ret; + + dapm_kcontrol_add_path(w->kcontrols[i], path); } } @@ -821,7 +822,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) path = list_first_entry(&w->sources, struct snd_soc_dapm_path, list_sink); - ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0); if (ret < 0) return ret; -- 1.8.4.rc1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Allow auto_mute_via_amp on bind mute controls
by Takashi Iwai 13 Aug '13

13 Aug '13

The auto-mute using the amp currently works only for a single amp on a pin. Make it working also with HDA_CTL_BIND_MUTE type, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f6c0344..6ed2209 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -816,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); enum { HDA_CTL_WIDGET_VOL, @@ -833,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = { .put = hda_gen_mixer_mute_put, /* replaced */ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), }, - HDA_BIND_MUTE(NULL, 0, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_bind_switch_get, + .put = hda_gen_bind_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, }; /* add dynamic controls from template */ @@ -940,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, } /* playback mute control with the software mute bit check */ -static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; @@ -952,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] &= enabled; ucontrol->value.integer.value[1] &= enabled; } +} +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { -- 1.8.3.1

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Add inputs for noise and mic mixers
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f38c52d..8bbddc1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1423,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1552,6 +1549,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- 1.8.4.rc1

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[alsa-devel] [PATCH v2 0/2] ALSA: Implement core jack support for kcontrol
by Felipe F. Tonello 12 Aug '13

12 Aug '13

From: "Felipe F. Tonello" <eu(a)felipetonello.com> On this second version I made sure that the patch series can be bisected. Also, since HDA codec has it's own jack infrastructure, I just disabled the support of the core ALSA jack in it. Why? Because it's causing duplication of controls, although the input events works fine. A HDA codec developer might give us more information about it. Patch 1) Is the actual implementation and fixes for users that uses the API. Patch 2) Name fixes. Just to make sure no jack has " Jack" duplicated at the end of its name. Felipe F. Tonello (2): ALSA: Added jack detection KControl support ALSA: SoC: Fix jack names according new API include/sound/control.h | 2 +- include/sound/jack.h | 8 ++-- include/sound/soc.h | 2 +- sound/core/Kconfig | 1 + sound/core/ctljack.c | 15 ++++++- sound/core/jack.c | 88 +++++++++++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 8 ---- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 4 +- sound/soc/mid-x86/mfld_machine.c | 6 +-- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 +- sound/soc/omap/omap-twl4030.c | 4 +- sound/soc/omap/rx51.c | 6 +-- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/ttc-dkb.c | 10 ++--- sound/soc/pxa/z2.c | 4 +- sound/soc/samsung/goni_wm8994.c | 7 +-- sound/soc/samsung/h1940_uda1380.c | 4 +- sound/soc/samsung/littlemill.c | 10 ++--- sound/soc/samsung/lowland.c | 6 +-- sound/soc/samsung/rx1950_uda1380.c | 4 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/speyside.c | 6 +-- sound/soc/samsung/tobermory.c | 4 +- sound/soc/soc-jack.c | 5 ++- sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_rt5640.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 8 ++-- 30 files changed, 163 insertions(+), 77 deletions(-) -- 1.8.3.1

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[alsa-devel] [RFC PATCH] ASoC: omap-mcbsp: Add support for highter rate up to 384000
by Michael Trimarchi 12 Aug '13

12 Aug '13

Add support for highter rate up to 384000Khz. There are codecs in the market that can play up to this frequency. Test was done just with: #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) playing wav file 32bit/192000Khz stereo. Some glitch has observed but this should depend on the selected min buffer bytes that was left untouched in aplay and in the omap-pcm driver. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..1e95f6a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,9 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS | \ + SNDRV_PCM_RATE_KNOT) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ @@ -574,11 +576,15 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = { .channels_max = 16, .rates = OMAP_MCBSP_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 384000, }, .capture = { .channels_min = 1, .channels_max = 16, .rates = OMAP_MCBSP_RATES, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &mcbsp_dai_ops, -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: dt: Move ALC5632 pin list from Tegra board binding to CODEC binding
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../devicetree/bindings/sound/alc5632.txt | 20 ++++++++++++++++++ .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 ++-------------------- 2 files changed, 22 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt index 8608f74..b913b74 100644 --- a/Documentation/devicetree/bindings/sound/alc5632.txt +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -13,6 +13,26 @@ Required properties: - #gpio-cells : Should be two. The first cell is the pin number and the second cell is used to specify optional parameters (currently unused). +Pins on the device (for linking into audio routes): + + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + + Example: alc5632: alc5632@1e { diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt index 05ffecb..8b8903e 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,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 ALC5632's pins: - - ALC5632 pins: - - * SPK_OUTP - * SPK_OUTN - * HP_OUT_L - * HP_OUT_R - * AUX_OUT_P - * AUX_OUT_N - * LINE_IN_L - * LINE_IN_R - * PHONE_P - * PHONE_N - * MIC1_P - * MIC1_N - * MIC2_P - * MIC2_N - * MICBIAS1 - * DMICDAT - - Board connectors: + sinks are the ALC5632's pins as documented in the binding for the device + and: * Headset Stereophone * Int Spk -- 1.8.4.rc1

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? Let me clarify that again: I read the above as the two options are a) DSD material is augmented by DOP markers and send as such over a regular PCM interface, or b) convert DSD to PCM in software and then send PCM samples over a regular PCM interface. What applications need to use for my new approach is another is: c) take DSD samples, bit reverse them if necessary, and then send them over the generic DSD data pipe to the Linux kernel. Device-specific details are taken care of by the kernel layer. I'm not aware of any software that uses that new DSD format type directly except for my example program. Daniel

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

[please never drop people from the Cc: list but always use 'reply to all'!] On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Given than none of devices have a chance to report their capabilities to > the host, finding out which modes to use is a trial-and-error game, > unless you have access to the firmware sources. > > > Hence, please just look up the USB IDs and alternate interface settings > of your connected device with lsusb, and then augment the code in > snd_usb_interface_dsd_format_quirks() (file sound/usb/quirks.c) and play > around with different possibilities (bitreverse on/off, DSD_U8 vs > DSD_U16 etc). Follow the example that I provided in that function. > > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? It's more complicated than that. Applications were always free to use a usual PCM stream and stuff it with DOP/DSD samples. This is the whole idea of DOP: allow applications to transport DSD samples via regular audio streams. The downside of this is that applications are in charge of adding the DOP markers manually, and information about bit-reversing and marker format (8 bit vs. 16 bit) is up to the application, which is not the right place for it. The patch series I brought in addressed this in a way that all that information about device-specific details is now known to the kernel, which can hence offer a standard compliant interface to userspace. [There's yet one more level of confusion as some file formats store the samples bit-reversed, but there's nothing we can do about that from kernel space except for defining a stable interface application have to obey to in order to be compatible. That is, samples are expected to always carry the oldest bit in the most significant bit of a byte.] So my question is: are you 100% sure that MPD uses the newly introduces ALSA formats, or are they doing the DOP conversion in manually in their own software layers? > You have to recompile and rmmod/insmod the snd-usb driver every time you > make a change, and then you need some sort of userspace tool to test it. > I published one here: > > This may be more than I can do. Can you add the options (bitreverse > on/off, DSD_U8 vs > DSD_U16 etc) to the alas-dsd-player? This would allow me to quickly test > the devices. There are #if0'ed blocks that you can opt-in for bit reversal, but as the marker format is different for 8 and 16 bits, and up to the kernel space now, the dsd player does not know about them. That was the entire intention of bringing that logic to kernel space in the first place. > Take a look at the DSD database so you can see what we are up against. I > could even add the tests results to the database. > https://docs.google.com/spreadsheet/ccc?key=0AgVhKcl_3lHfdFVyenBBNjNpQ2lieG… Nice list. Only devics with USB interface are interesting at this point, but that's a good start. As I said: you need to add the quirks for the devices to the quirks list. Other than that, there's no chance to test them. Copy'n pasting code inside that function and recompiling the kernel is really not as hard as you might think it is. Just search the internet for "compile kernel git <your-distribution-here>" and follow the links. Thanks, Daniel

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[alsa-devel] [PATCH V4 0/4] Add i2s support on smdk5420
by Padmavathi Venna 12 Aug '13

12 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420. Changes since V3: - Addressed review comments by Tomasz Figa related to const qualifier for samsung_i2s_dai_data - Removed passing quirks as driver data for non-dt platforms. - Separated out adding i2s nodes and enabling audio support on 5420 into different patch set as they are dependent on some of already posted but not yet merged i2c, dwmmc, dma and audss clock controller patches. Changes since V2: - Separated out driver side changes and dts changes in two patch sets. - Replaced samsung,s3c6410-i2s-v4 with samsung,s3c6410-i2s-multi for more clarity as suggested by Tomasz Figa. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is made based on Mark Brown for-next branch on sound.git. Padmavathi Venna (4): ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: exynos5250: move common i2s properties to exynos5 dtsi ARM: dts: Change i2s compatible string on exynos5250 .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++-- arch/arm/boot/dts/exynos5.dtsi | 21 +++ arch/arm/boot/dts/exynos5250.dtsi | 17 +--- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++ sound/soc/samsung/i2s.c | 143 +++++++++++++++----- 6 files changed, 157 insertions(+), 62 deletions(-) -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: arizona: Add widget<->mux route into mux route macro
by Charles Keepax 12 Aug '13

12 Aug '13

The routes linking the widget and the input mux were being added manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds the routes to the macro. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/wm5102.c | 41 ++++++++++++----------------------------- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 18 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08c..bf67d65 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd23..9e9a0c1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1499,23 +1499,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1567,22 +1550,22 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e7cb4b..0cfc5e6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -979,10 +979,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- 1.7.2.5

2 1

[alsa-devel] Another possible race in deferred probing...
by Russell King - ARM Linux 12 Aug '13

12 Aug '13

There is no better way to provoke bugs than to show a device to someone who has never seen it before. Obviously, I didn't get much chance to investigate what happened, so all I have are the kernel messages: kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CPU DAI mvebu-audio.1 not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral mvebu-audio mvebu-audio.1: found external clock kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral and then when manually removing the kirkwood-spdif module and re-inserting it: kirkwood-spdif-audio kirkwood-spdif-audio.1: dit-hifi <-> mvebu-audio.1 mapping ok First, remember that Ubuntu loads kernel modules from separate threads, so module loading here is multithreaded. What I suspect happened was the spdif-dit module was inserted and initialized between kirkwood-spdif starting to be re-probed, and returning from its probe function with -EPROBE_DEFER, and I suspect that spdif-dit was the last module. As there was no further module loading or driver activity, there was no attempt to re-probe the kirkwood-spdif module. This seems to be a very rare event (it's the first time I've seen it since the last fix to the deferred probing, and that's many hundreds of reboots.) Nevertheless, the fact that it has happened indicates that deferred probing still has holes that can leave modular drivers unbound after boot. Let's consider this scenario, which from a quick scan of the code looks like it's possible to happen: Thread 0 Thread 1 deferred probe triggered driver A's probe function called (kirkwood_spdif_probe in my case) takes lock L to lookup resource A checks for resource A (the spdif_dit codec) releases lock L resource A is not found ----- context switch ------ module init function called (to register spdif_dit_driver) driver B's probe function called (to register the spdif_dit codec via snd_soc_register_codec) takes lock L for adding resource A registers resource A releases lock L driver B's probe function completes deferred probe triggered, deferred probe list empty module init completes ----- context switch ------ probe returns -EPROBE_DEFER deferred probe mutex taken added to deferred probe list deferred probe mutex released This can happen because there's no locking between looking up the resource and having the driver added to the list of devices waiting to be probed. Given that deferred probing is becoming the "normal" thing for device drivers, these races are only going to get worse. Consider that as we're working on ARM towards having a single zImage bootable on many boards, most drivers will be modules, and if they're inserted into the kernel in a multi-threaded way, the above race is going to continually come back and bite us. I don't see an easy solution to this; we can't serialise device probes, because we have some places where device driver probe functions register devices themselves, which then go on to have their drivers probed (nested probes). Maybe this reflects back into Greg's whole argument against platform devices: maybe we need a struct device which lists _all_ the resources which a device requires, have the bus layer safely check for the presence of all those resources before calling the probe function, and move the deferred probe out of the core into the bus layer(s). Another solution would be to have a way to attach to the device a list of resources which failed, and have the driver core scan that list after the probe returns rechecking to see if all the resources are now present. That to me sounds particularly horrid, invasive and quite disgusting to implement safely.

1 0

[alsa-devel] [PATCH] ALSA: compress: update the MAINTAINER entry
by Vinod Koul 12 Aug '13

12 Aug '13

add missing uapi/ headers and Documentation files Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- MAINTAINERS | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 38a870e..a40e393 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7667,7 +7667,9 @@ M: Vinod Koul <vinod.koul(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported +F: Documentation/sound/alsa/compress_offload.txt F: include/sound/compress_driver.h +F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c -- 1.7.0.4

2 1

[alsa-devel] [PATCH 2/2] 6fire: make buffers DMA-able (midi)
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch makes midi output buffer DMA-able by allocating it separately. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/6fire/midi.c b/sound/6fire/midi.c --- a/sound/6fire/midi.c 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.c 2013-08-11 10:51:05.682482388 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/6fire/midi.h b/sound/6fire/midi.h --- a/sound/6fire/midi.h 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.h 2013-08-11 10:51:05.682482388 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

2 1

[alsa-devel] [PATCH] 6fire: fix URB transfer buffer for midi output
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch fixes URB transfer buffer allocation for midi output to be DMA-able. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.c 2013-08-07 16:32:31.363378104 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.h 2013-08-07 16:32:31.363378104 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

4 7

[alsa-devel] LG LW25 pin fix
by Joonas Saarinen 12 Aug '13

12 Aug '13

Hello, I discovered that the LG LW25 laptop needs a small fix to its pin mapping. The functions of blue and green jacks simply have to be interchanged. The rest of the configuration is correct. Can this fix be made permanent in the driver? The fixed mappings are: 0x1a 0x0181344f 0x1b 0x0321403f Here's the full correct map: [codec] 0x10ec0880 0x00001854 0 [pincfg] 0x14 0xb7031110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x411111f0 0x18 0x01a19c22 0x19 0xb7a31120 0x1a 0x0181344f 0x1b 0x0321403f 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x0145311e 0x1f 0x411111f0 Thanks, Joonas

2 5

Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Jesus R 12 Aug '13

12 Aug '13

Daniel/team, I have access to a few DSD devices and or USB based devices. I can provide any information you need to help with allowing ALSA to identify the device's capabilities. Just let me know how to get the information you need from command line:) I also have a list of known DSD/DoP devices in Google Drive that may be helpful. Let me know if it's okay to post the link. Jesus R sonore.us

2 2

[alsa-devel] [PATCH] ASoC: adau1701: Use gpio_set_value_cansleep()
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The GPIO manipulation done by this driver is never in atomic context so we can use gpio_set_value_cansleep() and support GPIOs that can't be set from atomic context. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/adau1701.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2c10252..ebff112 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 256: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; case 384: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 0: /* fallback */ case 512: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 12 Aug '13

12 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Please ignore the previous patch. I have tested speaker and headphone playback on this patch. --- sound/soc/codecs/rt5640.c | 389 +++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5640.h | 15 ++ 2 files changed, 352 insertions(+), 52 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..cae3f46 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -378,21 +376,119 @@ static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; +static const char * const rt5640_mute_mode[] = {"mute", "unmute"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_hp_enum, 0, 0, rt5640_mute_mode); +static const SOC_ENUM_SINGLE_DECL(rt5640_spo_enum, 0, 0, rt5640_mute_mode); + static const SOC_ENUM_SINGLE_DECL( rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static int rt5640_spo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->spo_l_mute; + ucontrol->value.integer.value[1] = !rt5640->spo_r_mute; + + return 0; +} + +static int rt5640_spo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_DIG1) & RT5640_PWR_CLS_D; + + if (ucontrol->value.integer.value[0]) { + rt5640->spo_l_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->spo_l_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, + RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->spo_r_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->spo_r_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + } + + return 0; +} + +static int rt5640_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->hp_l_mute; + ucontrol->value.integer.value[1] = !rt5640->hp_r_mute; + + return 0; +} + +static int rt5640_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + + if (ucontrol->value.integer.value[0]) { + rt5640->hp_l_mute = false; + if (pow & RT5640_PWR_HP_L) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->hp_l_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->hp_r_mute = false; + if (pow & RT5640_PWR_HP_R) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->hp_r_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, RT5640_R_MUTE); + } + + + return 0; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("Speaker Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_spo_get, rt5640_spo_put), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("HP Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_hp_get, rt5640_hp_put), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -868,33 +964,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +1012,216 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_spo_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_r_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_spo_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_l_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, + RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | + RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(20000, 25000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CO_MASK | + RT5640_HP_SG_MASK, + RT5640_HP_CO_EN | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, pow; + + regmap_read(rt5640->regmap, RT5640_PWR_ANLG1, &pow); + if (rt5640->hp_l_mute || !(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (rt5640->hp_r_mute || !(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + /* headphone unmute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK, RT5640_SMT_TRIG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK, RT5640_RSTN_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_RSTN_DIS | RT5640_HP_L_SMT_EN | + RT5640_HP_R_SMT_EN); + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(40000, 45000); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | RT5640_HP_R_SMT_DIS); + +} + +static void rt5640_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK, RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK, RT5640_RSTP_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_RSTP_DIS | + RT5640_HP_L_SMT_EN | RT5640_HP_R_SMT_EN); + + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + usleep_range(40000, 45000); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | + RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | + RT5640_HP_R_SMT_DIS); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK | + RT5640_HP_CD_PD_MASK | + RT5640_HP_CO_MASK | + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_SMT_TRIG_DIS | + RT5640_HP_CD_PD_EN | + RT5640_HP_CO_DIS | RT5640_HP_CP_PD | + RT5640_HP_SG_EN | RT5640_HP_CB_PD); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1411,23 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO L Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_l_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPO R Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_r_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1396,13 +1683,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"SPO L Amp", NULL, "SPOL MIX"}, + {"SPO R Amp", NULL, "SPOR MIX"}, + {"SPOLP", NULL, "SPO L Amp"}, + {"SPOLN", NULL, "SPO L Amp"}, + {"SPORP", NULL, "SPO R Amp"}, + {"SPORN", NULL, "SPO R Amp"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,7 +1701,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, + {"HPOL", NULL, "HP Amp"}, {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP Amp"}, {"HPOR", NULL, "HP R Amp"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, @@ -1792,17 +2083,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +2133,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..cb6bce9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,10 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool spo_l_mute; + bool spo_r_mute; + bool hp_l_mute; + bool hp_r_mute; }; #endif -- 1.8.1.1.439.g50a6b54

5 26

[alsa-devel] [PATCH] ASoC: compress: Use power efficient workqueue
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> There is no need for the power down work to be done on a per CPU workqueue especially considering the fairly long delay before powerdown. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d220150..53c9ecd 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { 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 */ -- 1.8.4.rc1

2 1

[alsa-devel] BeagleBoard driver development (PCM3168)
by Michael Jordan 12 Aug '13

12 Aug '13

Okay, before I went too far I thought to download archives and found that Klimann has been working on this, but last post I see is April 2013. I picked through and got what I think are the last discussed of the 3 files omap-pcm3168.c pcm3168.c and pcm3168.h. I did not mean to cross paths with anybody. I will compare these to what I have worked up and see what I can contribute. I am a bit confused about one thing though. When I began this (before I knew I might be stepping on somebody else's work) I downloaded 3.10 sources from kernel.org. I have previously and again grepped through the sources and I find no mention of pcm3168. Has this project just not made it that far? And do I have the latest versions (plucked from the text archives, or are they in a git somewhere that I don't know about? Michael

1 0

[alsa-devel] Hello
by Michael Jordan 11 Aug '13

11 Aug '13

Hello all, I am involved in a university project using a TI OMAPL138 EXPKIT (Logic PD Zoom Board) to create a sound processing box. I am tasked with developing the code to replace the TLV320AIC3106 with a PCM3168A due to the greater ADC/DAC count. We have already accomplished the hardware end, and I have been exercising my linux skills at trying to put this together. I found this list and subscribed, and I am glad I sat and listened for a bit before I said anything. 457 patches in the last 10 days! My fingers would fall off! I will get together what I have been working on (2 files pcm3168a.c and pcm3168a.h) and post them. I have some questions, obviously, but I will follow up with them in a bit. If someone could help me along it would be greatly appreciated. Or if I can just provide some relevant information, I would be happy to do that as well. Thanks, Michael

1 0

[alsa-devel] [PATCH] ASoC: doc: Add documentation to MAINTAINERS patterns
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 427a930..07e2bc8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7710,6 +7710,7 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported +F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH 1/16] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 11 Aug '13

11 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. sound/soc/pxa/mioa701_wm9713.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..20fdce6 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -133,10 +133,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

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[alsa-devel] [PATCH] ASoC: cs4271: fix dapm routing table
by Daniel Mack 11 Aug '13

11 Aug '13

The widgets are names "AINA" and "AINB", which is also what the datasheets calls the terminals. Fix the routing table to match these names. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 35a6a67..a20f1bb 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -184,8 +184,8 @@ SND_SOC_DAPM_OUTPUT("AOUTB-"), }; static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { - { "Capture", NULL, "AINL" }, - { "Capture", NULL, "AINR" }, + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, { "AOUTA+", NULL, "Playback" }, { "AOUTA-", NULL, "Playback" }, -- 1.8.3.1

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[alsa-devel] [PATCH] usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
by Maksim A. Boyko 11 Aug '13

11 Aug '13

Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 (ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310) for the Logitech HD Webcam C310. Reported-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Tested-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 3.10.5+ Signed-off-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> -- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ static void volume_control_quirks(struct case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- WBR, Maksim Boyko

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