January 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [GIT PULL] ASoC updates for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8-2 for you to fetch changes up to c6affc0dba45bb2074ff56b21386d22eeaf682f8: Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-16 14:02:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for v3.14
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit cdef2e5f358db5fd27eec4c72eb295c232287813: sound: oss: remove last sleep_on users (2014-01-14 16:12:07 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-2 for you to fetch changes up to 701caa51a2ce74182d39380ca11defeb163d98c1: Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai (2014-01-16 12:44:01 +0000) ---------------------------------------------------------------- ASoC: More updates for v3.14 A few more updates for v3.14 since the last set, highlights include: - Lots of DMA updates from Lars-Peter - Improvements to the constraints handling code from Lars-Peter - A very helpful conversion of the TWL4030 driver to regmap from Peter - A new driver for the Freescale ESAI controller from Nicolin Chen - Conversion of some of the drivers to use params_width() ---------------------------------------------------------------- Alexander Shiyan (5): ASoC: fsl_ssi: Fix printing return code on clk error ASoC: mc13783: Use module_platform_driver_probe() ASoC: mc13783: Use core error messages if registration fails ASoC: mc13783: Drop fixed ADC & DAC ports usage ASoC: mc13783: trivial: Cleanup module Andreas Pretzsch (1): ASoC: ssm2602: add 16kHz sampling rate support Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read Charles Keepax (5): ASoC: wm_adsp: Factor out ADSP2 boot proceedure ASoC: wm_adsp: Start DSP booting earlier in the DAPM process ASoC: wm_adsp: Mark wm_adsp2_boot_work as static mfd: wm5110: Add registers for headphone short circuit control ASoC: wm5110: Add controls for headphone short circuit protection Fabio Estevam (1): ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int' Jean-Francois Moine (1): ASoC: pcm: Fix lack of platform bespoke_trigger() call Lars-Peter Clausen (24): ASoC: atmel: Don't set unused struct snd_pcm_hardware fields ASoC: dpcm: Add helper function for initializing runtime pcm ASoC: dpcm: Allow PCMs to omit the set of supported formats ASoC: fsl: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant ASoC: intel: Don't set unused struct snd_pcm_hardware fields ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields ASoC: sh: Don't set unused struct snd_pcm_hardware fields ASoC: ux500: Don't set unused struct snd_pcm_hardware fields ASoC: Allow PCMs to restrict the supported formats ASoC: pcm: Properly initialize hw->rate_max ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ALSA: Add helper function for intersecting two rate masks ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag Lee Jones (10): ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion ASoC: ux500: Provide better checking for Device Tree and/or Platform Data ASoC: Ux500: Match platform by device node when booting Device Tree ASoC: ux500_pcm: Stop pretending that we support varying address widths ASoC: ux500_pcm: Expect different saved DMA data when obtaining from DAI store ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case ASoC: ux500_pcm: Take out pointless dev_dbg() call ASoC: ux500_pcm: Differentiate between pdata and DT initialisation ASoC: ux500: Dramatically reduce the size of the DAI driver data struct Liam Girdwood (4): ASoC: docs: Update the Overview document ASoC: dpcm: Explicitly set BE DAI link supported stream directions ASoC: utils: Add internal call to determine if DAI is dummy. ASoC: sapm: Automatically connect DAI link widgets in DAPM graph. Mark Brown (30): Merge tag 'ib-asoc-3.14.2' of git://git.linaro.org/people/ljones/mfd into asoc-tlw4030 ASoC: ad1836: Use params_width() rather than explicit memory format ASoC: ad193x: Use params_width() rather than memory format ASoC: adau1373: Use params_width() rather than memory format ASoC: adau1701: Use params_width() rather than memory format ASoC: adav80x: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-arizona ASoC: alc5623: Use params_width() rather than memory format ASoC: alc5632: Use params_width() rather than memory format ASoC: cs42l51: Use params_width() rather than memory format ASoC: da7210: Use params_width() rather than memory format ASoC: da7213: Use params_width() rather than memory format ASoC: da732x: Use params_width() rather than memory format ASoC: da9055: Use params_width() rather than memory format ASoC: isabelle: Use params_width() rather than memory format ASoC: max98088: Use params_width() rather than memory format ASoC: max98090: Use params_width() rather than memory format ASoC: max98095: Use params_width() rather than memory format ASoC: max9850: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-pcm Merge branch 'topic/axi' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge branch 'topic/samsung' of git://git.kernel.org/.../broonie/sound into asoc-dma ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into for-tiwai Merge remote-tracking branch 'asoc/topic/core' into for-tiwai Merge remote-tracking branch 'asoc/topic/dapm' into for-tiwai Merge remote-tracking branch 'asoc/topic/dma' into for-tiwai Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai Merge remote-tracking branch 'asoc/topic/arizona' into for-tiwai Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai Markus Pargmann (13): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ASoC: fsl-ssi doc: Add list of supported compatibles ASoC: fsl-ssi: Fix stats compile warning ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic3x: Add tlv320aic32x4 as compatible ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (2): ASoC: fsl_ssi: Set default slot number for common cases ASoC: fsl_esai: Add ESAI CPU DAI driver Peter Ujfalusi (11): mfd: twl-core: Simplify IO wrapper functions by moving common code out mfd: twl-core: API to set the regcache bypass for a given regmap in twl mfd: twl-core: Enable regcache for audio registers ASoC: twl4030: Separate write condition checking from I/O function ASoC: twl4030: Remove check defaults functionality ASoC: twl4030: Remove reset registers functionality ASoC: twl4030: Introduce local ctl register cache ASoC: twl4030: Remove local reg cache ASoC: twl4030: Parameter alignment fixes (for code consistency) ASoC: twl4030: Move the ctl cache update local to twl4030_write() function ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() Takashi Iwai (1): ASoC: bcm: Remove obsoleted Kconfig dependency Wei Yongjun (2): ASoC: ux500: Fix sparse non static symbol warning ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe() Xiubo Li (8): ASoC: simple-card: Fix the sysclk selection. ASoC: simple-card: fix the DAPM routes map parsing ASoC: simple-card: keep the property's name the same pattern ASoC: fsl-sai: Clean up the code ASoC: simple-card: fix a bug where cinfo will be NULL before using it ASoC: simple-card: fix the cinfo error check ASoC: simple-card: use snd_soc_card_set/get_drvdata ASoC: simple-card: fix one bug to writing to the platform data .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ .../devicetree/bindings/sound/fsl,ssi.txt | 7 +- .../devicetree/bindings/sound/simple-card.txt | 2 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 1 + Documentation/sound/alsa/soc/overview.txt | 27 +- arch/arm/mach-ux500/board-mop500-audio.c | 8 +- drivers/dma/pl330.c | 1 + drivers/mfd/twl-core.c | 190 ++++- drivers/mfd/wm5110-tables.c | 6 + include/linux/dmaengine.h | 28 + include/linux/i2c/twl.h | 5 +- include/linux/mfd/arizona/registers.h | 27 + include/linux/platform_data/asoc-ux500-msp.h | 9 +- include/sound/pcm.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 4 + sound/core/pcm_misc.c | 39 + sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - sound/soc/bcm/Kconfig | 1 - sound/soc/codecs/ad1836.c | 10 +- sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 10 +- sound/soc/codecs/adau1701.c | 36 +- sound/soc/codecs/adav80x.c | 30 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 8 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs42l51.c | 14 +- sound/soc/codecs/da7210.c | 10 +- sound/soc/codecs/da7213.c | 10 +- sound/soc/codecs/da732x.c | 10 +- sound/soc/codecs/da9055.c | 10 +- sound/soc/codecs/isabelle.c | 6 +- sound/soc/codecs/max98088.c | 6 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/mc13783.c | 34 +- sound/soc/codecs/ssm2602.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/twl4030.c | 376 ++++------ sound/soc/codecs/wm5110.c | 7 + sound/soc/codecs/wm_adsp.c | 199 +++-- sound/soc/codecs/wm_adsp.h | 12 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dma.c | 7 - sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_ssi.c | 586 ++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 3 - sound/soc/fsl/imx-pcm-fiq.c | 3 - sound/soc/fsl/mpc5200_dma.c | 4 - sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/generic/simple-card.c | 70 +- sound/soc/intel/sst_platform.c | 10 - sound/soc/intel/sst_platform.h | 4 - sound/soc/kirkwood/kirkwood-dma.c | 16 - sound/soc/mxs/mxs-pcm.c | 6 - sound/soc/nuc900/nuc900-pcm.c | 3 - sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/sh/dma-sh7760.c | 17 - sound/soc/sh/fsi.c | 6 - sound/soc/sh/rcar/core.c | 6 - sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 75 +- sound/soc/soc-generic-dmaengine-pcm.c | 66 +- sound/soc/soc-pcm.c | 58 +- sound/soc/soc-utils.c | 7 + sound/soc/ux500/mop500.c | 2 + sound/soc/ux500/ux500_msp_dai.c | 146 ++-- sound/soc/ux500/ux500_msp_i2s.c | 56 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- sound/soc/ux500/ux500_pcm.c | 65 +- 80 files changed, 2656 insertions(+), 1034 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h

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[alsa-devel] different sample rate in alsa.conf file, ALSA DMIX mode
by Pavel Botev 16 Jan '14

16 Jan '14

Hi ALSA support team, Can we set more sample rate for DMIX mode in the alsa.conf file than the default one (defaults.pcm.dmix.rate 48000)? Can we use one alsa.conf file, with many sample rates (8KHz, 11.025KHz, 16KHz, 22.05KHz, 32KHz, 44.1KHz, 48KHz, 64KHz, 88.2KHz and 96KHz), not with just one default sample rate? The problem is that when we use one sample rate in alsa.conf and other in the ALSA application, we have resampling executed which is very CPU intensive. Thus we search a way to use alsa.conf file with many sample rates, thus avoiding resampling. If not possible, can we change the algorithm of resampling, thus have low CPU usage when resampling is activated? Thanks for your time. Best regards, Pavel

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for some Dell machines
by Hui Wang 16 Jan '14

16 Jan '14

When we plug a 3-ring headset on some Dell machines, the headset mic can't be detected, after apply this patch, the headset mic can work well on all those machines. On the machine with the Subsytem ID 0x10280610, if we use ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be detected and work well, but the sound can't be outputed via headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE can fix this problem. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: David Chen <david.chen(a)canonical.com> Tested-by: Cyrus Lien <cyrus.lien(a)canonical.com> Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1c16457..f9b22fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4253,7 +4253,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060f, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), @@ -4264,6 +4265,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5065,6 +5071,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

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[alsa-devel] [3.13-rc8] AM335x BeagleBone Buffer XRUN
by Jack Mitchell 16 Jan '14

16 Jan '14

I am currently trying to debug a buffer overrun when capturing with the BeagleBone Black and Audio Cape attached on 3.13-rc8, this is pretty much the same setup as the am335x-evm. The following is the traceback captured using the XRUN_BUFFER_DEBUG kernel config option. I think an interrupt is taking too long to return. Example Traceback 1 (Writing to uSD card) > [ 802.578728] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 802.593139] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 802.602174] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 802.610659] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > [ 802.618596] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 802.627531] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 802.638281] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 802.649131] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 802.659345] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 802.669735] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 802.680035] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 802.689697] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 802.699542] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 802.708750] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 802.718237] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 802.727618] Exception stack(0xc088df58 to 0xc088dfa0) > [ 802.732912] df40: 00000001 00000001 > [ 802.741479] df60: 00000000 c0897cb0 c088c000 c088c000 00000000 c0933bd4 c08949a0 c0894880 > [ 802.750047] df80: c0894930 c059f4a8 c088c020 c088dfa0 c0082ef8 c000fd00 20070013 ffffffff > [ 802.758620] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c000fd00>] (arch_cpu_idle+0x20/0x3c) > [ 802.767283] [<c000fd00>] (arch_cpu_idle+0x20/0x3c) from [<c008db84>] (cpu_startup_entry+0x120/0x1f0) > [ 802.776858] [<c008db84>] (cpu_startup_entry+0x120/0x1f0) from [<c0826a9c>] (start_kernel+0x304/0x364) Example Traceback 2 (Writing to tmpfs) > [ 1310.025256] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > [ 1310.031046] CPU: 0 PID: 998 Comm: avconv Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 1310.039580] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 1310.048615] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 1310.057102] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > 1310.065039] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 1310.073974] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 1310.084724] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 1310.095574] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 1310.105788] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 1310.116178] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 1310.126477] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 1310.136138] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 1310.145982] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 1310.155190] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.164677] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 1310.174056] Exception stack(0xde0f7ee0 to 0xde0f7f28) > [ 1310.179354] 7ee0: 00000001 00000000 00000000 de3f8a80 00000000 00000002 c0047968 00000054 > [ 1310.187920] 7f00: 00000000 b5440c64 c0933670 beb92fb8 de0f6028 de0f7f28 c0082eb0 c0047348 > [ 1310.196480] 7f20: 60000113 ffffffff > [ 1310.200153] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c0047348>] (__do_softirq+0xa8/0x2f4) > [ 1310.208817] [<c0047348>] (__do_softirq+0xa8/0x2f4) from [<c0047968>] (irq_exit+0xac/0x104) > [ 1310.217479] [<c0047968>] (irq_exit+0xac/0x104) from [<c000f970>] (handle_IRQ+0x54/0xb4) > [ 1310.225868] [<c000f970>] (handle_IRQ+0x54/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.235347] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592c3c>] (__irq_usr+0x3c/0x60) > [ 1310.244724] Exception stack(0xde0f7fb0 to 0xde0f7ff8) > [ 1310.250019] 7fa0: beb94588 beb92ab8 beb9301c beb93038 > [ 1310.258586] 7fc0: 00088ca0 00000140 00000160 b5440f18 00000080 b5440c64 beb92ab8 beb92fb8 > [ 1310.267150] 7fe0: 4b000001 beb9267c b5f4d6a0 b5f4c8c8 80000010 ffffffff I have done some tests and capturing to a uSD card is the easiest way to trigger XRUNS, where they occur often. Capturing to the eMMC is a little bit better. Capturing directly to tmpfs is nearly perfect, apart from very rare XRUN occurrences (10 minutes average maybe). This makes sense as EDMA is used to perform writes to the uSD and eMMC devices, so if they're being hogged with writes, then there is no time to transfer out the sound buffer and we get an overrun. So, does anybody have ideas where to start debugging this further? I have tried to increase the buffer size from 32K up to the supposed max of 128K but I can't find where to do it... if anyone could shed any light I would most grateful, do you think increasing the buffer would solve this, as it would give more time to trigger the irq and get the data out before overrun? I have tried multiple userspace programs to do the capture and the best so far is arecord initialized like this: nice -n -20 arecord -D hw:0,0 -f cd -v jack-test.wav > root@beaglebone:~# nice -n -20 arecord -D hw:0,0 -f dat -v jack-test.wav > Recording WAVE 'jack-test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > DEBUG: buffer_time = 682667 > DEBUG: max_buffer_size = 32768 > Hardware PCM card 0 'AM335x-EVM' device 0 subdevice 0 > Its setup is: > stream : CAPTURE > access : RW_INTERLEAVED > format : S16_LE > subformat : STD > channels : 2 > rate : 48000 > exact rate : 48000 (48000/1) > msbits : 16 > buffer_size : 32768 > period_size : 2048 > period_time : 42666 > tstamp_mode : NONE > period_step : 1 > avail_min : 2048 > period_event : 0 > start_threshold : 1 > stop_threshold : 32768 > silence_threshold: 0 > silence_size : 0 > boundary : 1073741824 > appl_ptr : 0 > hw_ptr : 0 > overrun!!! (at least 221.598 ms long) > Status: > state : XRUN > trigger_time: 1389534120.970142840 > tstamp : 1389534121.191651631 > delay : 0 > avail : 32768 > avail_max : 32768 If you need any more information or debugging output please let me know. Regards, -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer http://www.embed.me.uk --

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Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 16 Jan '14

16 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 09:03:25 schrieb Raymond Yau: > > Post the output of > > > > pactl list > > > > pulseaudio verbose log > > I jump in here as well, same problem on a ThinkPad T520, sound volume is > way > too low - watching movie via loudspeakers is fairly impossible. > > Any hint you could give me? > post the output of alsa-info.sh , there is a bug in alsa-driver which treat dock headphone and headphone to support surround 4.0 Profile: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 60, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6000, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6060, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 700, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 760, available: yes)

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[alsa-devel] [PATCH] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card | 0 sound/soc/generic/simple-card. | 0 sound/soc/generic/simple-card.c | 7 ++++++- 3 files changed, 6 insertions(+), 1 deletion(-) create mode 100644 sound/soc/generic/simple-card create mode 100644 sound/soc/generic/simple-card. diff --git a/sound/soc/generic/simple-card b/sound/soc/generic/simple-card new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card. b/sound/soc/generic/simple-card. new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0e2fbdd..b40b57c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -143,6 +143,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -257,7 +261,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

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[alsa-devel] [PATCHv2] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d437c80..711d6a1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -152,6 +152,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -266,7 +270,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

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[alsa-devel] Problems with simple-card
by Jean-Francois Moine 16 Jan '14

16 Jan '14

Xiubo, I did a mistake in the v1 of my 'ASoC: simple-card: simplify code': I did not initialize the pointer to the asoc_simple_card_dai_init() function when DT. Then, I fixed that, and the simple card does not work for me. First, without any 'format' in the DT, I get a fmt for each CPU / CODEC DAI: SND_SOC_DAIFMT_CBS_CFS is always set. Well, some code is executed for nothing, but this is not critical. The main problem is about sysclk: when there is no clock in the DT, you get the clock of the CPU or CODEC DAI. In my system, two clocks are declared in the controller CPU DAI, but there is no .set_sysclk pointer. So, snd_soc_dai_set_sysclk() returns -EINVAL and the card is not created. As I don't know why you need these fmt and sysclk, may you have a look at these problems? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: simple-card: simplify the daifmt code
by Xiubo Li 16 Jan '14

16 Jan '14

In the asoc_simple_card_parse_of() will parse the device node's CPU/CODEC DAI commone fmts, and then in asoc_simple_card_sub_parse_of() will parse the CPU/CODEC DAI's sub-node fmts, so we can combine the info->daifmt and info->set.fmt in asoc_simple_card_sub_parse_of() not while just before set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b6014fe..0e2fbdd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,15 +16,12 @@ #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, - struct asoc_simple_dai *set, - unsigned int daifmt) + struct asoc_simple_dai *set) { int ret; - daifmt |= set->fmt; - - if (daifmt) { - ret = snd_soc_dai_set_fmt(dai, daifmt); + if (set->fmt) { + ret = snd_soc_dai_set_fmt(dai, set->fmt); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_fmt error\n"); return ret; @@ -48,14 +45,13 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai); if (ret < 0) return ret; @@ -64,6 +60,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, + unsigned int daifmt, struct asoc_simple_dai *dai, struct device_node **node) { @@ -89,6 +86,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * and specific "format" if it has */ dai->fmt = snd_soc_of_parse_daifmt(np, NULL); + dai->fmt |= daifmt; /* * dai->sysclk come from @@ -149,7 +147,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->cpu_dai, of_cpu); if (ret < 0) @@ -159,7 +157,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->codec_dai, of_codec); if (ret < 0) -- 1.8.4

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