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

[alsa-devel] ASoC updates for v3.10
by Mark Brown 15 Apr '13

15 Apr '13

The following changes since commit b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10 for you to fetch changes up to 5cbad7d39ad229c68a724e5e139fd845b93766b2: Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next (2013-04-12 13:57:31 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A bunch of changes here, the most interesting one subsystem wise being Morimoto-san's work to create snd_soc_component which doesn't do much for now but will be pretty important going forwards: - Add a new component object type which will form the basis of moving to a more generic handling of SoC and off-SoC components, contributed by Kuninori Morimoto. - A fairly large set of cleanups for the dmaengine integration from Lars-Peter Clausen, starting to move towards being able to have a generic driver based on the library. - Performance optimisations to DAPM from Ryo Tsutsui. - Support for mixer control sharing in DAPM from Stephen Warren. - Multiplatform ARM cleanups from Arnd Bergmann. - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack. ---------------------------------------------------------------- Alban Bedel (1): ASoC: wm8903: Add the DAC boost control Alexandru Gheorghiu (1): ASoC: core: Use PTR_RET function Andrey Smirnov (2): ASoC: si476x: Convert SI476X codec to use regmap ASoC: si476x: Cosmetic changes to SI476X codec driver Arnd Bergmann (4): ASoC: samsung: use irq resource for idma ASoC: samsung: fix module_device_table ASoC: samsung: export idma_reg_addr_init ASoC: samsung: move plat/ headers to local directory Charles Keepax (12): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK ASoC: arizona: Don't enable FLL on REFCLK configuration ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls ASoC: wm0010: Constify usage of firmware filenames ASoC: wm0010: Report filename when we fail to load firmware Dan Carpenter (1): ASoC: tas5086: signedness bug in tas5086_hw_params() Daniel Mack (7): ASoC: ak4104: convert to direct regmap API usage ASoC: ak4104: correct tranceiver enable handling ASoC: Add codec driver for AK5386 ALSA: ASoC: add codec driver for TI TAS5086 ASoC: davinci-mcasp: clean up davinci_hw_common_param() ASoC: cs4271: convert to direct regmap API usage ASoC: cs4271: switch to mute_stream Dylan Reid (1): ASoC: max98088: Add TLV data for volume controls. Fabio Estevam (2): ASoC: fsl: imx-audmux: Use devm_clk_get() ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period Heiko Stübner (1): ASoC: samsung: fix neo1973-wm8753 compilation Jeeja KP (1): ASoC: compress - add support for metadata apis Kuninori Morimoto (61): ASoC: core: tidyup snd_soc_register_codec() fail case ASoC: core: use snd_soc_register_dais() on codec ASoC: core: remove codec from list if registration failed ASoC: fsi: remove unused irq ASoC: add snd_soc_register_component() ASoC: add .name for snd_soc_component_driver ASoC: fsi: use snd_soc_register_component() instead of snd_soc_register_dais() ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: snd_soc_register_component() uses properly snd_soc_register_dai[s]() ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function ASoC: switch over to use snd_soc_register_component() on sh4 siu Lars-Peter Clausen (28): ASoC: core: Drop unused "dapm" field form soc_enum struct ASoC: core: Remove unused "n_widgets" field from snd_soc_dapm struct ASoC: spear_pcm: Staticize non-exported structs ASoC: omap: Call omap_mcbsp_set_threshold() from mcbsp hw_params ASoC: ux500_pcm: Remove duplicated SNDRV_PCM_HW_PARAM_PERIODS constraint ASoC: spear_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: omap-pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: tegra_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: atmel-pcm-dma: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: speaer_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: atmel-pcm-dma: Do not use snd_dmaengine_pcm_{set,get}_data() ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params ASoC: mmp-pcm: Allocate dma filter parameters on the stack ASoC: imx-pcm: Embed the imx_dma_data struct in the dma_params struct ASoC: mxs: Embed the mxs_dma_data struct in the mxs_pcm_dma_params struct ASoC: dmaengine-pcm: Remove snd_dmaengine_pcm_{set,get}_data ASoC: Constify the 'driver' field of snd_soc_platform ASoC: Constify the 'ops' field of snd_soc_platform_driver ASoC: Constify the 'compr_ops' field of snd_soc_platform_driver ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: omap: Use common DAI DMA data ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: ep93xx_pcm: Fix compile error ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture Mark Brown (72): ASoC: arizona: Only allow input volume updates when inputs are enabled ASoC: wm8960: Add input boost volume control ASoC: omap: Check regulator enable for DAC on Pandora ASoC: arizona: If we only have a clock to synchronise with make it REFCLK mfd: arizona: Define additional FLL control registers mfd: wm5102: Map in additional FLL control registers ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies ASoC: arizona: Optimise FLL loop gains ASoC: arizona: Ensure synchroniser is disabled when not needed ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 Merge tag 'v3.9-rc2' into asoc-core ASoC: arizona: Provide defines for the clock rates ASoC: wm_adsp: Handle old .bin files ASoC: arizona: Ensure we clock two channels for I2S mode ASoC: wm8994: Support constraining the maximum number of channels clocked mfd: wm5102: Add additional speaker control registers ASoC: arizona: Factor out speaker widgets from CODEC drivers ASoC: arizona: Log thermal events ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged ASoC: arizona: Basic support for ISRC rate selection ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates extcon: arizona: Factor out magic application ASoC: arizona: Fix interaction between headphone outputs and identification extcon: arizona: Fix interaction between headphone outputs and identification Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: arizona: Add delay after powering up line level outputs Merge branch 'topic/pxa' of git://git.kernel.org/.../broonie/sound into asoc-component ASoC: wm5102: Implement OSR support ASoC: wm2000: Expose some more registers for diagnostics ASoC: wm_adsp: Provide defines for firmwares ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Merge branch 'topic/omap' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung ASoC: core: Merge Samsung fixes to avoid trivial context conflict ASoC: wm8994: Remove duplicate revision cache ASoC: wm_adsp: Add support for firmware wide coefficient blocks Merge remote-tracking branch 'asoc/fix/compress' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tegra' into asoc-next Merge remote-tracking branch 'asoc/fix/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' 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/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/component' 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/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm-hubs' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Michal Bachraty (1): ASoC: davinci-mcasp: Add support for multichannel playback Paul Bolle (3): ASoC: samsung: remove last traces of neo1973-gta01 ASoC: codecs: remove hidden prompt ASoC: Ux500: remove test for undefined Kconfig macro Paul Handrigan (1): ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Peter Ujfalusi (4): ASoC: omap3pandora: Fix compilation error ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address Ryo Tsutsui (2): ASoC: dapm: Remove redundant clear_walk() for supply widgets ASoC: dapm: Only clear paths we've walked Sachin Kamat (5): ASoC: max98090: Remove unneeded version.h header include ASoC: max98090: Convert to devm_regmap_init_i2c() ASoC: max98090: Make struct dev_pm_ops const ASoC: max98090: Fix checkpatch errors related to spacing ASoC: max98090: Remove unnecessary braces Sascha Hauer (1): ASoC: imx-ssi: Fix AC97 rates Sebastien Guiriec (2): ASoC: omap-mcpdm: Clean up with devm_* function ASoC: omap-dmic: Clean up with devm_request_and_ioremap Shawn Guo (1): ASoC: imx-sgtl5000: use of_node to match cpu dai Silviu-Mihai Popescu (1): ASoC: omap: convert to devm_ioremap_resource() Stephen Warren (10): ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init ASoC: tegra_wm8753: minor cleanup ASoC: tegra_wm9712: assume CONFIG_OF ASoC: tegra_alc5632: assume CONFIG_OF, and other cleanup ASoC: tegra trimslice: assume CONFIG_OF, and other cleanup ASoC: tegra_wm8903: assume CONFIG_OF, remove platform data ASoC: tegra: add Tegra114 support to the AHUB driver ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c ASoC: export snd_soc_register_component ASoC: dapm: Implement mixer control sharing Tim Gardner (2): ASoC: adau1373: adau1373_hw_params: Silence overflow warning ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Vinod Koul (1): ASoC: mid-x86 - add support for meaadata apis Wei Yongjun (2): ASoC: tas5086: use module_i2c_driver to simplify the code ASoC: arizona: remove duplicated include from arizona.c Yegor Yefremov (2): ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode ASoC: davinci-mcasp: don't overwrite DIT settings Zoltan Puskas (1): ASoC: atmel: Add slave mode support to SSC in DSP Mode A Documentation/devicetree/bindings/sound/ak5386.txt | 19 + .../bindings/sound/nvidia,tegra30-ahub.txt | 26 +- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 - arch/arm/plat-samsung/devs.c | 6 + drivers/extcon/extcon-arizona.c | 107 ++-- drivers/mfd/wm5102-tables.c | 10 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/registers.h | 42 ++ include/linux/mfd/wm8994/pdata.h | 8 + include/sound/dmaengine_pcm.h | 27 +- include/sound/soc-dai.h | 8 - include/sound/soc-dapm.h | 1 - include/sound/soc.h | 29 +- include/sound/tas5086.h | 7 + include/sound/tegra_wm8903.h | 26 - sound/soc/atmel/atmel-pcm-dma.c | 25 +- sound/soc/atmel/atmel_ssc_dai.c | 54 +- sound/soc/au1x/ac97c.c | 9 +- sound/soc/au1x/i2sc.c | 9 +- sound/soc/au1x/psc-ac97.c | 9 +- sound/soc/au1x/psc-i2s.c | 9 +- sound/soc/blackfin/bf5xx-ac97.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 9 +- sound/soc/blackfin/bf6xx-i2s.c | 9 +- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 18 +- sound/soc/cirrus/ep93xx-i2s.c | 25 +- sound/soc/cirrus/ep93xx-pcm.c | 37 +- sound/soc/cirrus/ep93xx-pcm.h | 20 - sound/soc/cirrus/simone.c | 2 - sound/soc/cirrus/snappercl15.c | 1 - sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/ak4104.c | 55 +- sound/soc/codecs/ak5386.c | 152 +++++ sound/soc/codecs/arizona.c | 530 +++++++++++++++--- sound/soc/codecs/arizona.h | 34 +- sound/soc/codecs/cs4271.c | 166 +++--- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/max98088.c | 30 +- sound/soc/codecs/max98090.c | 45 +- sound/soc/codecs/si476x.c | 47 +- sound/soc/codecs/tas5086.c | 591 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2000.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 131 ++--- sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 54 +- sound/soc/codecs/wm5110.h | 6 +- sound/soc/codecs/wm8903.c | 6 + sound/soc/codecs/wm8960.c | 10 + sound/soc/codecs/wm8994.c | 68 ++- sound/soc/codecs/wm8994.h | 3 +- sound/soc/codecs/wm_adsp.c | 124 +++- sound/soc/codecs/wm_adsp.h | 5 +- sound/soc/codecs/wm_hubs.c | 9 +- sound/soc/davinci/davinci-i2s.c | 15 +- sound/soc/davinci/davinci-mcasp.c | 97 +++- sound/soc/davinci/davinci-pcm.c | 16 +- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/davinci-vcif.c | 11 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/fsl/fsl_ssi.c | 45 +- sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 58 +- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-pcm.h | 19 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 42 +- sound/soc/fsl/imx-ssi.h | 7 +- sound/soc/fsl/mpc5200_psc_ac97.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mid-x86/sst_platform.c | 20 +- sound/soc/mid-x86/sst_platform.h | 2 + sound/soc/mxs/mxs-pcm.c | 43 +- sound/soc/mxs/mxs-pcm.h | 4 +- sound/soc/mxs/mxs-saif.c | 17 +- sound/soc/nuc900/nuc900-ac97.c | 9 +- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 +- sound/soc/omap/mcbsp.h | 7 +- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 47 +- sound/soc/omap/omap-hdmi.c | 33 +- sound/soc/omap/omap-mcbsp.c | 27 +- sound/soc/omap/omap-mcpdm.c | 118 ++-- sound/soc/omap/omap-pcm.c | 83 +-- sound/soc/omap/omap-pcm.h | 40 -- sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 9 +- sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/pxa/mmp-pcm.c | 33 +- sound/soc/pxa/mmp-sspa.c | 9 +- sound/soc/pxa/pxa-ssp.c | 9 +- sound/soc/pxa/pxa2xx-ac97.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/s6000/s6000-i2s.c | 9 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/ac97.c | 14 +- sound/soc/samsung/goni_wm8994.c | 11 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 33 +- sound/soc/samsung/idma.c | 11 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/pcm.c | 11 +- .../include/plat => sound/soc/samsung}/regs-ac97.h | 0 .../include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 9 +- sound/soc/samsung/s3c-i2s-v2.h | 7 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 13 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/spdif.c | 11 +- sound/soc/sh/fsi.c | 14 +- sound/soc/sh/hac.c | 10 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/siu_dai.c | 11 +- sound/soc/sh/ssi.c | 10 +- sound/soc/soc-compress.c | 41 +- sound/soc/soc-core.c | 130 ++++- sound/soc/soc-dapm.c | 320 ++++++----- sound/soc/soc-dmaengine-pcm.c | 68 ++- sound/soc/soc-io.c | 5 +- sound/soc/spear/spdif_in.c | 9 +- sound/soc/spear/spdif_out.c | 9 +- sound/soc/spear/spear_pcm.c | 22 +- sound/soc/tegra/tegra20_ac97.c | 28 +- sound/soc/tegra/tegra20_ac97.h | 4 +- sound/soc/tegra/tegra20_i2s.c | 28 +- sound/soc/tegra/tegra20_i2s.h | 4 +- sound/soc/tegra/tegra20_spdif.c | 22 +- sound/soc/tegra/tegra20_spdif.h | 4 +- sound/soc/tegra/tegra30_ahub.c | 72 ++- sound/soc/tegra/tegra30_ahub.h | 24 +- sound/soc/tegra/tegra30_i2s.c | 28 +- sound/soc/tegra/tegra30_i2s.h | 4 +- sound/soc/tegra/tegra_alc5632.c | 15 +- sound/soc/tegra/tegra_asoc_utils.c | 34 +- sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 48 +- sound/soc/tegra/tegra_pcm.h | 7 - sound/soc/tegra/tegra_wm8753.c | 15 +- sound/soc/tegra/tegra_wm8903.c | 179 +++--- sound/soc/tegra/tegra_wm9712.c | 7 +- sound/soc/tegra/trimslice.c | 56 +- sound/soc/txx9/txx9aclc-ac97.c | 9 +- sound/soc/ux500/ux500_msp_dai.c | 13 +- sound/soc/ux500/ux500_msp_dai.h | 5 - sound/soc/ux500/ux500_pcm.c | 76 +-- sound/soc/ux500/ux500_pcm.h | 14 - 164 files changed, 3437 insertions(+), 1710 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 include/sound/tas5086.h delete mode 100644 include/sound/tegra_wm8903.h delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/codecs/ak5386.c create mode 100644 sound/soc/codecs/tas5086.c delete mode 100644 sound/soc/omap/omap-pcm.h rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%)

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic support for Asus X101CH
by David Henningsson 15 Apr '13

15 Apr '13

With this patch, a TRRS headset mic cannot be successfully detected on the Asus X101CH, and we can also distinguish between headphone and headset automatically. Buglink: https://bugs.launchpad.net/bugs/1169138 Co-authored-by: Kailang <kailang(a)realtek.com> Tested-by: Luis Henriques <luis.henriques(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Now that we have better headset mic support infrastructure, I remembered I forgot to upstream this patch. It was originally provided by Realtek (therefore, I cannot answer for the msleeps and exactly why this is needed). I just rewrote it for the current kernel version. 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 6cf27f5..1b7aed5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2680,6 +2680,34 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->gen.automute_hook = alc269_quanta_automute; } +static void alc269_x101_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + msleep(100); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); + msleep(500); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc269_x101_hp_automute_hook; + } +} + + /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -2837,6 +2865,9 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_ASUS_X101_FUNC, + ALC269_FIXUP_ASUS_X101_VERB, + ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3027,30 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_ASUS_X101_FUNC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_x101_headset_mic, + }, + [ALC269_FIXUP_ASUS_X101_VERB] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0310}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_FUNC + }, + [ALC269_FIXUP_ASUS_X101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x04a1182c }, /* Headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_VERB + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3044,6 +3099,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- 1.7.9.5

2 2

[alsa-devel] [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 15 Apr '13

15 Apr '13

Hi, This series adds DT support for phycore-ac97 using the fsl-ssi driver. In version 2 I discarded most of the imx-ssi work and integrated the ac97-slave support in fsl-ssi, including support for imx-pcm-fiq as alternative to dma. There are some other changes to get phycore-ac97 working. The first two patches have notes about the detailed changes since version 1. The rest of the patches is new and necessary for the ac97-slave support. Regards, Markus Markus Pargmann (11): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: Add support for DMA init by ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add SACNT definitions ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Setup generic imx dma params ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/powerpc/fsl/ssi.txt | 73 - Documentation/devicetree/bindings/sound/fsl,ssi.txt | 13 b/Documentation/devicetree/bindings/sound/fsl,ssi.txt | 73 + b/Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 12 b/arch/arm/mach-imx/mach-pca100.c | 7 b/arch/arm/mach-imx/mach-pcm043.c | 7 b/sound/soc/fsl/Kconfig | 2 b/sound/soc/fsl/fsl_ssi.c | 70 + b/sound/soc/fsl/fsl_ssi.h | 8 b/sound/soc/fsl/imx-pcm-dma.c | 31 b/sound/soc/fsl/imx-pcm-fiq.c | 14 b/sound/soc/fsl/imx-pcm.h | 6 b/sound/soc/fsl/imx-ssi.c | 7 b/sound/soc/fsl/imx-ssi.h | 1 b/sound/soc/fsl/phycore-ac97.c | 185 ++++ sound/soc/fsl/fsl_ssi.c | 379 +++++++--- sound/soc/fsl/imx-pcm.h | 9 17 files changed, 664 insertions(+), 233 deletions(-)

4 34

[alsa-devel] Stupid question #1 of many -- unit test harness?
by TJ Olaes 15 Apr '13

15 Apr '13

Hi. So, I'm attempting to write the driver for the eClaro card for a couple of reasons, one of which is that I currently don't have sound on mine. So... is there some kind of test harness framework for new drivers? I have no prior idea how to move forward to build this thing, so I'm snooping around in the oxygen source files and trying to copy things that look like they make sense. For instance, I noted that there's mention of an "AK4396" related to output. I assume this is some kind of chip, and I saw in the Claro code that something called "ak4396_init(...)" is called, so I just copied that line and pasted it into the eclaro_init(...) function I'm writing. I have no idea how I'm going to test this short of just compiling the kernel and booting into it. I feel like that's the brute-force, non-smart way to do it, and that you guys who've been doing this for a while might have ideas and recommendations on how to better test my code so I'm not rebooting my machine each time I change a line and recompile. I'd appreciate your guidance, community. Thanks. -TJ

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: disable autopm for MIDI devices
by Clemens Ladisch 15 Apr '13

15 Apr '13

Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) introduced autopm for all USB audio/MIDI devices. However, many MIDI devices, such as synthesizers, do not merely transmit MIDI messages but use their MIDI inputs to control other functions. With autopm, these devices would get powered down as soon as the last MIDI port device is closed on the host. Even some plain MIDI interfaces could get broken: they automatically send Active Sensing messages while powered up, but as soon as these messages cease, the receiving device would interpret this as an accidental disconnection. Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input) introduced another regression: some devices (e.g. the Roland GAIA SH-01) are self-powered but do a reset whenever the USB interface's power state changes. To work around all this, just disable autopm for all USB MIDI devices. Reported-by: Laurens Holst Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/midi.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 34b9bb7..e5fee18 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -126,7 +126,6 @@ struct snd_usb_midi { struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; - bool autopm_reference; unsigned int opened[2]; unsigned char disconnected; unsigned char input_running; @@ -1040,7 +1039,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, { struct snd_usb_midi* umidi = substream->rmidi->private_data; struct snd_kcontrol *ctl; - int err; down_read(&umidi->disc_rwsem); if (umidi->disconnected) { @@ -1051,13 +1049,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, mutex_lock(&umidi->mutex); if (open) { if (!umidi->opened[0] && !umidi->opened[1]) { - err = usb_autopm_get_interface(umidi->iface); - umidi->autopm_reference = err >= 0; - if (err < 0 && err != -EACCES) { - mutex_unlock(&umidi->mutex); - up_read(&umidi->disc_rwsem); - return -EIO; - } if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1080,8 +1071,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } - if (umidi->autopm_reference) - usb_autopm_put_interface(umidi->iface); } } mutex_unlock(&umidi->mutex); @@ -2256,6 +2245,8 @@ int snd_usbmidi_create(struct snd_card *card, return err; } + usb_autopm_get_interface_no_resume(umidi->iface); + list_add_tail(&umidi->list, midi_list); return 0; }

2 1

[alsa-devel] [PATCH] ALSA: hda - Implement headset jack functionality for some Dell hw
by David Henningsson 15 Apr '13

15 Apr '13

On some machines, there is a headset jack that can support both headphone, headsets (of both CTIA and OMTP type) and mic-in. On other machines, the headset jack supports headphone, headsets (both CTIA and OMTP), but not mic-in. This patch implements that functionality as different capture sources. Buglink: https://bugs.launchpad.net/bugs/1169143 Tested-by: David Chen <david.chen(a)canonical.com> Co-authored-by: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 386 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 386 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6cf27f5..f3307eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -52,6 +52,20 @@ enum { ALC_INIT_GPIO3, }; +enum { + ALC_HEADSET_MODE_UNKNOWN, + ALC_HEADSET_MODE_UNPLUGGED, + ALC_HEADSET_MODE_HEADSET, + ALC_HEADSET_MODE_MIC, + ALC_HEADSET_MODE_HEADPHONE, +}; + +enum { + ALC_HEADSET_TYPE_UNKNOWN, + ALC_HEADSET_TYPE_CTIA, + ALC_HEADSET_TYPE_OMTP, +}; + struct alc_customize_define { unsigned int sku_cfg; unsigned char port_connectivity; @@ -87,6 +101,11 @@ struct alc_spec { unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ + hda_nid_t headset_mic_pin; + hda_nid_t headphone_mic_pin; + int current_headset_mode; + int current_headset_type; + /* hooks */ void (*init_hook)(struct hda_codec *codec); #ifdef CONFIG_PM @@ -2798,6 +2817,302 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc_headset_mode_unplugged(struct hda_codec *codec) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + alc_write_coef_idx(codec, 0x45, 0xc429); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val & 0xbfff); + alc_write_coef_idx(codec, 0x06, 0x2104); + alc_write_coef_idx(codec, 0x1a, 0x0001); + alc_write_coef_idx(codec, 0x26, 0x0004); + alc_write_coef_idx(codec, 0x32, 0x42a3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x18, 0x7308); + alc_write_coef_idx(codec, 0x6b, 0xc429); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to unplugged mode.\n"); +} + + +static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, + hda_nid_t mic_pin) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xc429); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val | 1<<14); + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x1a, 0x0021); + alc_write_coef_idx(codec, 0x26, 0x008c); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + case 0x10ec0292: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0x19, 0xa208); + alc_write_coef_idx(codec, 0x2e, 0xacf0); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0xb5, 0x1040); + val = alc_read_coef_idx(codec, 0xc3); + alc_write_coef_idx(codec, 0xc3, val | 1<<12); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + } + snd_printdd("Headset jack set to mic-in mode.\n"); +} + +static void alc_headset_mode_default(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x6b, 0xc429); + alc_write_coef_idx(codec, 0x18, 0x7308); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0041); + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to headphone (default) mode.\n"); +} + +/* Iphone type */ +static void alc_headset_mode_ctia(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to iPhone-style headset mode.\n"); +} + +/* Nokia type */ +static void alc_headset_mode_omtp(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xe429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xe429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d50); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to Nokia-style headset mode.\n"); +} + +static void alc_determine_headset_type(struct hda_codec *codec) +{ + int val; + bool is_ctia = false; + struct alc_spec *spec = codec->spec; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd029); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + msleep(300); + val = alc_read_coef_idx(codec, 0x6c); + is_ctia = (val & 0x001c) == 0x001c; + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0c00); + msleep(300); + val = alc_read_coef_idx(codec, 0xbe); + is_ctia = (val & 0x1c02) == 0x1c02; + break; + } + + snd_printdd("Headset jack detected iPhone-style headset: %s\n", + is_ctia ? "yes" : "no"); + spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; +} + +static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; + + if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} + +static void alc_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + alc_update_headset_mode(codec); +} + +static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + snd_hda_gen_hp_automute(codec, jack); +} + +static void alc_probe_headset_mode(struct hda_codec *codec) +{ + int i; + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + /* Find mic pins */ + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].is_headset_mic && !spec->headset_mic_pin) + spec->headset_mic_pin = cfg->inputs[i].pin; + if (cfg->inputs[i].is_headphone_mic && !spec->headphone_mic_pin) + spec->headphone_mic_pin = cfg->inputs[i].pin; + } + + spec->gen.cap_sync_hook = alc_update_headset_mode_hook; + spec->gen.automute_hook = alc_update_headset_mode; + spec->gen.hp_automute_hook = alc_update_headset_jack_cb; +} + +static void alc_fixup_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + alc_probe_headset_mode(codec); + break; + case HDA_FIXUP_ACT_INIT: + spec->current_headset_mode = 0; + alc_update_headset_mode(codec); + break; + } +} + +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + } + else + alc_fixup_headset_mode(codec, fix, action); +} + +static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + int val; + alc_write_coef_idx(codec, 0xc4, 0x8000); + val = alc_read_coef_idx(codec, 0xc2); + alc_write_coef_idx(codec, 0xc2, val & 0xfe); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } + alc_fixup_headset_mode(codec, fix, action); +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2837,6 +3152,10 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_HEADSET_MODE, + ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3315,35 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, + [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21014020 }, /* dock line out */ + { 0x19, 0x21a19030 }, /* dock mic */ + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC269_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode, + }, + [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_no_hp_mic, + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3030,6 +3378,26 @@ static const struct hda_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c7, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ec, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ed, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ee, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f3, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -3536,6 +3904,8 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC668_FIXUP_HEADSET_MODE, }; static const struct hda_fixup alc662_fixups[] = { @@ -3696,6 +4066,20 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE + }, + [ALC668_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc668, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -3704,6 +4088,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), -- 1.7.9.5

2 1

[alsa-devel] PROBLEM: iMX6 SSI - set up as a master of external audio codec
by Andrey Gelman 15 Apr '13

15 Apr '13

Hi, everyone ! I'm bringing up a wm8731 audio codec on an iMX6 based board. I use the slave mode, i.e. SSI outputs all the clocks towards the codec IC. I work with 3.0.35 Android kernel provided by freescale. While, generally, the playback is Ok, I face 2 issues: - cannot play back mono streams (too high rate, one speaker produces only noise) - cannot play back streams with sampling rate as low as 8kHz (the sound is accompanied by high distortion) In the mainline kernel (commits to v3.6), there is a workaround to the 1-st issue by simply removing mono support from imx-ssi.c. Similar approach would fix the 2-nd issue as well - SSI would report it does not support 8kHz sampling rate. In a later commit, this workaround has been reverted, enabling mono support back. My question is this: Is this a real limitation of SSI ? Does it support mono playback or not ? Does SSI support 8kHz sampling rate ? If yes, what could be possible reason to the distortions I get, while higher rates are played back well ? Is there a real fix to these issues, or only workarounds are available ? Thanks, Andrey Gelman

1 0

[alsa-devel] [PATCH 20/30] ASoC: samsung: convert to dmaengine API
by Arnd Bergmann 15 Apr '13

15 Apr '13

In order to build the exynos kernel with CONFIG_ARCH_MULTIPLATFORM, we must convert all users of the Samsung private DMA interface to the generic dmaengine API. This version of the patch adds the generic dmaengine API as an alternative to the existing samsung specific one. Once all the older platforms provide support for the common dmaengine interfaces, we can remove the old code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/dma.c | 219 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/dma.h | 15 +++- sound/soc/samsung/i2s.c | 2 - sound/soc/samsung/pcm.c | 1 - sound/soc/samsung/spdif.c | 1 - 5 files changed, 232 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b7926..9fd53df 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -22,8 +22,14 @@ #include <sound/pcm_params.h> #include <asm/dma.h> + +#ifdef CONFIG_SAMSUNG_DMADEV #include <mach/hardware.h> #include <mach/dma.h> +#else +#include <linux/dmaengine.h> +#include <linux/amba/pl330.h> +#endif #include "dma.h" @@ -62,11 +68,13 @@ struct runtime_data { static void audio_buffdone(void *data); +#ifdef CONFIG_SAMSUNG_DMADEV /* dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ + static void dma_enqueue(struct snd_pcm_substream *substream) { struct runtime_data *prtd = substream->runtime->private_data; @@ -265,6 +273,217 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } +#else +/* dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. + */ + +static void dma_enqueue(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned long period = prtd->dma_period; + unsigned int limit; + enum dma_transfer_direction direction; + struct dma_chan *chan = prtd->params->ch; + struct dma_async_tx_descriptor *desc; + + pr_debug("Entered %s\n", __func__); + + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); + + while (prtd->dma_loaded < limit) { + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + period) > prtd->dma_end) { + period = prtd->dma_end - pos; + pr_debug("%s: corrected dma len %ld\n", + __func__, period); + } + + desc = dmaengine_prep_dma_cyclic(chan, pos, + prtd->dma_period*limit, period, direction, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + + if (desc) { + desc->callback = audio_buffdone; + desc->callback_param = substream; + dmaengine_submit(desc); + } + + prtd->dma_loaded++; + pos += period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } + + prtd->dma_pos = pos; +} + +static void audio_buffdone(void *data) +{ + struct snd_pcm_substream *substream = data; + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (prtd->state & ST_RUNNING) { + prtd->dma_pos += prtd->dma_period; + if (prtd->dma_pos >= prtd->dma_end) + prtd->dma_pos = prtd->dma_start; + + if (substream) + snd_pcm_period_elapsed(substream); + } +} + +static int dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + struct dma_slave_config config; + dma_cap_mask_t mask; + + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, channel %d\n", prtd->params, + prtd->params->channel); + + dma_cap_zero(mask); + dma_cap_set(DMA_CYCLIC, mask); + + prtd->params->ch = dma_request_slave_channel_compat(mask, + pl330_filter, (void *)prtd->params->channel, + rtd->cpu_dai->dev, prtd->params->ch_name); + + memset(&config, 0, sizeof(struct dma_slave_config)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.direction = DMA_MEM_TO_DEV; + config.dst_addr = prtd->params->dma_addr; + config.dst_addr_width = prtd->params->dma_size; + config.dst_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } else { + config.direction = DMA_DEV_TO_MEM; + config.src_addr = prtd->params->dma_addr; + config.src_addr_width = prtd->params->dma_size; + config.src_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int dma_hw_free(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + dmaengine_terminate_all(prtd->params->ch); + dma_release_channel(prtd->params->ch); + prtd->params = NULL; + } + + return 0; +} + +static int dma_prepare(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* flush the DMA channel */ + dmaengine_terminate_all(prtd->params->ch); + + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + dma_enqueue(substream); + + return ret; +} + +static int dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->state |= ST_RUNNING; + dma_async_issue_pending(prtd->params->ch); + break; + + case SNDRV_PCM_TRIGGER_STOP: + prtd->state &= ~ST_RUNNING; + dmaengine_terminate_all(prtd->params->ch); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} +#endif static snd_pcm_uframes_t dma_pointer(struct snd_pcm_substream *substream) diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..6bd1857 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,13 +12,24 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H +#ifdef CONFIG_SAMSUNG_DMADEV +#include <mach/dma.h> +#endif + struct s3c_dma_params { +#ifdef CONFIG_SAMSUNG_DMADEV struct s3c2410_dma_client *client; /* stream identifier */ + unsigned ch; + struct samsung_dma_ops *ops; +#else + struct s3c2410_dma_client { + char *name; /* unused */ + } *client; + struct dma_chan *ch; +#endif int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ - unsigned ch; - struct samsung_dma_ops *ops; char *ch_name; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e3..61f2622 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/dma.h> - #include <linux/platform_data/asoc-s3c.h> #include "dma.h" diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 13bab79..b5f267c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5008e5b..ee792aa 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" -- 1.8.1.2

4 5

[alsa-devel] [PATCH v2] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 15 Apr '13

15 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Calculate nsec from frames, subtract delay for playback as well as capture. sound/pci/hda/hda_intel.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..3e6f2c6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,23 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + u64 codec_frames, codec_nsecs; + + if (!hinfo->ops.get_delay) + return nsec; + + codec_frames = hinfo->ops.get_delay(hinfo, apcm->codec, substream); + codec_nsecs = div_u64(codec_frames * 1000000000LL, + substream->runtime->rate); + + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1914,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_subtract_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 9

[alsa-devel] Can Realtek HD-A codec ALC282 support D3cold?
by Lin, Mengdong 15 Apr '13

15 Apr '13

Hi, We want to put an idle HD-A codec into D3cold for more power saving and preparation for being power gated. The Realtek ALC282 is used in our test. Could anyone help to confirm whether ALC282 can support D3cold or not? I'm told that ALC282 can support D3cold. But my test shows it cannot. At first, ALC282 HW does not report that it can support D3cold. Its parameter "Supported Power States" returns 0xc000000f, means it only supports: EPSS, CLKSTOP, D3hot, D2, D1 and D0. Is this a just mistake in reporting the bit, or D3cold is not supported all? Then when I explicitly set the audio function group to D3cold by verb 0x705, and then get its power state after 200ms by verb 0xf05, The codec response is 0x00000000: No PS-Error, both PS-Act and PS-set is D0. So it seems the command to D3cold is not processed by codec HW at all. Is there someone from Realtek or with experience on ALC282 here can share some information about this issue? Many thanks Mengdong

1 0