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

[alsa-devel] [PATCH 1/4] ASoC: mc13783: Delete unneeded test before of_node_put
by Fabio Estevam 07 Oct '14

07 Oct '14

of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/mc13783.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2c1f1b0..388f90a 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -784,8 +784,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); out: - if (np) - of_node_put(np); + of_node_put(np); return ret; } -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put
by Fabio Estevam 07 Oct '14

07 Oct '14

of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..48179ff 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) clk_fail: clk_disable_unprepare(data->codec_clk); fail: - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
by Daniel Mack 07 Oct '14

07 Oct '14

In the SNDRV_PCM_STREAM_CAPTURE branch in soc_pcm_apply_msb(), look at sig_bits of the capture stream, not the playback one. Spotted by coverity. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 642c862..002311a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -352,7 +352,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) } else { for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->playback.sig_bits == 0) { + if (codec_dai->driver->capture.sig_bits == 0) { bits = 0; break; } -- 2.1.0

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[alsa-devel] [GIT PULL] ASoC updates for v3.18
by Mark Brown 07 Oct '14

07 Oct '14

The following changes since commit bfe01a5ba2490f299e1d2d5508cbbbadd897bbe9: Linux 3.17 (2014-10-05 12:23:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18 for you to fetch changes up to a5448c88b812390a3622e76d774e10c0da1fb970: Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next (2014-10-06 12:49:09 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.18 - More componentisation work from Lars-Peter, this time mainly cleaning up the suspend and bias level transition callbacks. - Real system support for the Intel drivers and a bunch of fixes and enhancements for the associated CODEC drivers, this is going to need a lot quirks over time due to the lack of any firmware description of the boards. - Jack detect support for simple card from Dylan Reid. - A bunch of small fixes and enhancements for the Freescale drivers. - New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX processors. ---------------------------------------------------------------- Anatol Pomozov (6): ASoC: rt5677: Add a configuration option for LDO2_POW pin ASoC: trace: Remove trailing new-lines in trace messages ASoC: rt5677: Add gpio-controller DTS documentation ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier ASoC: rt5677: Add dts properties for input/output differential configuration ASoC: ssm4567: Remove duplicated else-if branch Arnd Bergmann (2): ASoC: simple-card: fix regression in clock rate lookup ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is Axel Lin (4): ASoC: cs35l32: Remove unneeded regulator_bulk_free call in cs35l32_i2c_remove ASoC: cs42l56: Remove unneeded regulator_bulk_free call in cs42l56_remove ASoC: cs35l32: Simplify implementation of cs35l32_codec_set_sysclk ASoC: Remove return value checking for gpiochip_remove() Bard Liao (2): ASoC: rt5677: Add DMIC2 clock selection ASoC: rt286: Add depends on I2C Brian Austin (6): ASoC: cs35l32: Add support for CS35L32 Boosted Amplifier ASoC: cs35l32: Add bindings for CS35L32 ASoC: cs35l32: fix compile warning for i2c_probe ASoC: cs35l32: use true/false returns for bool functions ASoC: cs42l52: use true/false returns for bool functions ASoC: cs42l56: use true/false returns for bool functions Dan Murphy (1): ASoC: tas2552: Add DAPM calls for amp and PLL Dmitry Lavnikevich (1): ASoC: tlv320aic3x: fix PLL D configuration Dylan Reid (3): ASoC: simple-card: Add mic and hp detect gpios. ASoC: tegra: add mic detect gpio to tegra_max98090 ASoC: simple-card: Fix detect gpio documentation. Fabian Frederick (1): ASoC: cs35l32: remove second linux/slab.h inclusion Fabio Estevam (6): ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property ASoC: fsl ssi doc: Remove unused properties ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD ASoC: sgtl5000: Improve the error message on slave mode setting ASoC: sgtl5000: Do a sanity check on SYS_MCLK ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string Fabio Falzoi (1): ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS Fengguang Wu (1): ASoC: Intel: byt-rt5640: fix coccinelle warnings Geert Uytterhoeven (2): regmap: Split regmap_get_endian() in two functions ASoC: Allow SND_SOC_WM8978 to be selected manually Howard Mitchell (1): ASoC: core: Fix volsw_range funcs so SOC_DOUBLE_R_RANGE_TLV works. Jarkko Nikula (16): ASoC: Remove unused cache_only from struct snd_soc_codec ASoC: max98090: Add recovery for PLL lock failure ASoC: max98090: Move interrupt request from codec probe to i2c probe ASoC: max98090: Remove structure member irq from private data ASoC: max98090: Remove structure member extmic_mux from private data ASoC: max98090: Remove unused version define ASoC: max98090: Remove unused byte access macros ASoC: max98090: Enable both DMIC channels also when using mono configuration ASoC: Intel: byt-max98090: Set card as fully routed ASoC: Intel: byt-rt5640: Remove IN2N pin from DAPM route table ASoC: Intel: byt-rt5640: Add quirk for Asus T100 ASoC: rt5640: Add function for enabling DMIC from ACPI probed machine ASoC: Intel: byt-rt5640: Enable DMIC interface for default DAPM route ASoC: Intel: byt-rt5640: Add quirk for Dell Venue 8 Pro tablet ASoC: Intel: byt-rt5640: Set card as fully routed ASoC: dapm: Fix NULL pointer dereference when registering card with widgets Javier Martinez Canillas (1): regmap: Fix DT endianess parsing logic Jianqun (3): ASoC: rockchip-i2s: enable "hclk" for rockchip I2S controller ASoC: rockchip-i2s: fix rockchip i2s defination more reasonable ASoC: rockchip-i2s: add dma data to snd_soc_dai Julia Lawall (1): ASoC: rsnd: delete unneeded test before of_node_put Jyri Sarha (1): ASoC: tlv320aic31xx: Choose PLL p divider automatically Kuninori Morimoto (7): ASoC: simple-card: use asoc_simple_xxx prefix ASoC: simple-card: remove dai_link->cpu_dai_name when DT ASoC: simple-card: dai_link->init should be cared when multi DAI ASoC: simple-card: use common for_each_child_of_node() for loop ASoC: simple-card: fixup cpu_dai_name clear case ASoC: simple-card: tidyup use priv in parameter ASoC: simple-card: tidyup get dai_link/dai_props from priv Lars-Peter Clausen (59): ASoC: dmic: Add to SND_SOC_ALL_CODECS ASoC: edma-pcm: Include edma-pcm.h ASoC: odrodix2_max98090: Make non exported symbols static ASoC: sh: Fix dma direction type ASoC: samsung idma: Add proper annotation for casting iomem pointers ASoC: ab8500-codec: Drop bank prefix from AB8500_GPIO_DIR4_REG register define ASoC: Move debugfs registration to the component level ASoC: Consolidate platform and CODEC probe/remove ASoC: Make rtd->codec optional ASoC: Add component level probe/remove support ASoC: Move AUX dev support to the component level ASoC: Pass component instead of DAPM context to AUX dev init callback ASoC: Move component->probed check into soc_{remove,probe}_component() ASoC: Cleanup DAI module reference counting ASoC: Consolidate CPU and CODEC DAI removal ASoC: Consolidate CPU and CODEC DAI lookup ASoC: Automatically initialize regmap for all components ASoC: Remove support for legacy snd_soc_platform IO ASoC: Replace list_empty(&card->codec_dev_list) with !card->instantiated ASoC: sh: Don't opencode DMAengine API calls ASoC: txx9: Don't opencode DMAengine API calls ASoC: ac97-codec: Remove ASoC level IO support ASoC: Restore idle_bias_off initialization ASoC: ab8500-codec: Revert back to regmap ASoC: adau1373: Remove unnecessary suspend/resume bias level changes ASoC: lm49453: Remove unnecessary suspend/resume bias level changes ASoC: tlv320aic3x: Remove unnecessary suspend/resume bias level changes ASoC: wm8804: Remove unnecessary suspend/resume bias level changes ASoC: wm8995: Remove unnecessary suspend/resume bias level changes ASoC: Set card->instantiated to false when removing the card ASoC: Shutdown DAPM contexts when removing a card ASoC: Add support for automatically going to BIAS_OFF on suspend ASoC: Always run default suspend/resume code ASoC: adau1373: Cleanup manual bias level transitions ASoC: adau17x1: Cleanup manual bias level transitions ASoC: adav80x: Cleanup manual bias level transitions ASoC: ssm2518: Cleanup manual bias level transitions ASoC: ssm2602: Cleanup manual bias level transitions ASoC: 88pm860x-codec: Cleanup manual bias level transitions ASoC: jz4740: Cleanup manual bias level transitions ASoC: ml26124: Cleanup manual bias level transitions ASoC: sgtl5000: Cleanup bias level transitions ASoC: es8328: Cleanup manual bias level transitions ASoC: Add snd_soc_component_{get,set}_drvdata() ASoC: sst-haswell-pcm: Alloc state struct in driver probe() ASoC: sst-haswell-pcm: Move controls and DAPM elements to component ASoC: Remove table based DAPM/control setup support from snd_soc_platform_driver ASoC: cs42l52: Cleanup manual bias level transitions ASoC: cs42l56: Cleanup manual bias level transitions ASoC: cs42l73: Cleanup manual bias level transitions ASoC: da732x: Remove unnecessary idle_bias_off initialization ASoC: da732x: Remove unused codec field form da732x_priv struct ASoC: da732x: Cleanup manual bias level transitions ASoC: Remove locking in snd_soc_{new,free}_ac97_codec() ASoC: Fix snd_soc_{new,free}_ac97_codec() locking removal ASoC: wm8741: Remove unused wm8741_suspend define ASoC: wm8{350,753,971}: Use snd_soc_dapm_to_codec() instead of dapm->codec ASoC: rx51: Use snd_soc_dapm_to_codec() instead of dapm->codec ASoC: Remove CODEC pointer from snd_soc_dapm_context Mark Brown (31): Merge branch 'topic/es8328' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge branch 'fix/fsl-esai' of git://git.kernel.org/.../broonie/sound into asoc-fsl-esai Merge tag 'dt-endian' of git://git.kernel.org/.../broonie/regmap into asoc-fsl Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-esai Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-sai Merge branch 'fix/tlv320aic31xx' of git://git.kernel.org/.../broonie/sound into asoc-tlv320aic31xx Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-suspend Merge branch 'topic/suspend' of git://git.kernel.org/.../broonie/sound into asoc-es8328 Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/cs42l56' of git://git.kernel.org/.../broonie/sound into asoc-suspend Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/max98090', 'asoc/fix/samsung' and 'asoc/fix/tlv320aic31xx' into asoc-linus Merge branch 'fix/max98090' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/rt5640' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge tag 'v3.17-rc4' into asoc-simple Merge tag 'asoc-v3.17-rc4' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/ab8500', 'asoc/topic/ac97' and 'asoc/topic/cs35l32' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs42l52', 'asoc/topic/cs42l56' and 'asoc/topic/da732x' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/dmic', 'asoc/topic/drivers', 'asoc/topic/es8328' and 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-easi', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-ssi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/rockchip', 'asoc/topic/rsnd' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5677', 'asoc/topic/samsung', 'asoc/topic/sgtl5000' and 'asoc/topic/sh' into asoc-next Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/spdif', 'asoc/topic/ssm2602' and 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/suspend', 'asoc/topic/tas2552', 'asoc/topic/tegra', 'asoc/topic/tlv320aic31xx' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next Masanari Iida (1): ASoC: da732x: Remove unnecessary KERN_ERR in pr_err() Nicolin Chen (6): ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support ASoC: fsl_sai: Initialize with software reset ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode ASoC: fsl_sai: Add asynchronous mode support ASoC: fsl_sai: Make Synchronous and Asynchronous modes exclusive ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode Nikesh Oswal (1): ASoC: wm8994: Demux the microphone detection IRQ Oder Chiou (5): ASoC: rt5677: Revise the wrong name in the header file ASoC: rt5677: Add the GPIO function ASoC: rt5677: Add the TDM function ASoC: rt5677: Add sidetone function ASoC: rt5645: Add headset detect function Paul Handrigan (1): ASoC: cs4265: Add CHIP_ID as a readable register Peter Ujfalusi (4): ASoC: tlv320aic31xx: Correct interface register 2 variable name ASoC: tlv320aic31xx: Enable support for S24_LE format ASoC: davinci-mcasp: Convert the context save/restore to use array ASoC: davinvi-mcasp: Proper suspend/resume support while audio is active Rongjun Ying (1): ASoC: sirf: usp: Add bitclock inversion support Sean Cross (4): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver ASoC: fsl: Fix building of imx-es8328 on PPC Shengjiu Wang (6): ASoC: fsl_esai: refine esai for TDM support ASoC: fsl_esai: refine esai for TDM support ASoC: fsl-asoc-card: Fix build warning for maybe-uninitialized ASoC: fsl: fsl-asoc-card: Select SND_SOC_IMX_AUDMUX ASoC: fsl-asoc-card: move 'config SND_SOC_FSL_ASOC_CARD' to 'if SND_IMX_SOC' ASoC: fsl_ssi: refine ipg clock usage in this module Stefan Kristiansson (2): ASoC: ssm2602: add device tree bindings ASoC: ssm2602: add support for 11.025kHz and 22.5kHz sample rates Stephen Warren (1): regmap: of_regmap_get_endian() cleanup Subhransu S. Prusty (6): ASoC: Intel: mfld-pcm: Use function instead of ioctl ASoC: Intel: mfld-pcm: Change sst_ops prototypes to take dev parameter ASoC: mfld-compress: Use dedicated function instead of ioctl ASoC: mfld: pcm: Replace pr_ with dev_ ASoC: Intel: mrfld: Use snd_soc_dai_get_drvdata to derive drv data ASoC: Export dapm_kcontrol_get_value Sylwester Nawrocki (1): ASoC: Drop const from struct snd_soc_dai_link *of_node members Thierry Reding (2): ASoC: tas2552: Fix compilation warning for !PM_RUNTIME ASoC: rt286: Fix compilation warning for !PM Tomeu Vizoso (1): ASoC: fsl_spdif: Remove unused includes of linux/clk-private.h Vinod Koul (5): ASoC: Intel: Fix to use byte control interface ASoC: Intel: add mrfld DSP defines ASoC: Intel: mfld-pcm: don't call trigger ops to DSP for internal streams ASoC: Intel: mrfld: add bytes control for modules ASoC: Intel: mfld-pcm: add control for powering up/down dsp Wei Yongjun (1): ASoC: es8328: fix error return code in es8328_codec_probe() Xiubo Li (10): regmap: add DT endianness binding support. regmap: Add the DT binding documentation for endianness ASoC: simple-card: Fix the compile warning. ASoC: fsl-asrc: Convert to use regmap framework's endianness method. ASoC: fsl-esai: Convert to use regmap framework's endianness method. ASoC: fsl-spdif: Convert to use regmap framework's endianness method. ASoC: fsl-sai: Convert to use regmap framework's endianness method. ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'. ASoC: simple-card: Merge single and muti DAI link(s) code. ASoC: simple-card: Adjust the comments of simple card. .../devicetree/bindings/regmap/regmap.txt | 47 ++ .../devicetree/bindings/sound/adi,ssm2602.txt | 19 + .../devicetree/bindings/sound/cs35l32.txt | 62 ++ Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/fsl,esai.txt | 3 +- .../devicetree/bindings/sound/fsl,ssi.txt | 8 +- .../devicetree/bindings/sound/fsl-asoc-card.txt | 82 +++ .../devicetree/bindings/sound/fsl-sai.txt | 30 +- .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + Documentation/devicetree/bindings/sound/rt5677.txt | 59 ++ .../devicetree/bindings/sound/simple-card.txt | 4 + .../devicetree/bindings/sound/ssm4567.txt | 15 + .../devicetree/bindings/vendor-prefixes.txt | 1 + drivers/base/regmap/regmap-i2c.c | 2 + drivers/base/regmap/regmap-spi.c | 2 + drivers/base/regmap/regmap.c | 74 +- include/dt-bindings/sound/cs35l32.h | 26 + include/sound/rt5645.h | 3 + include/sound/rt5677.h | 13 +- include/sound/soc-dapm.h | 5 +- include/sound/soc.h | 101 +-- include/trace/events/asoc.h | 6 +- sound/soc/codecs/88pm860x-codec.c | 3 - sound/soc/codecs/Kconfig | 36 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 73 +- sound/soc/codecs/ac97.c | 15 - sound/soc/codecs/adau1373.c | 21 - sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 8 - sound/soc/codecs/adau17x1.h | 1 - sound/soc/codecs/adav80x.c | 23 +- sound/soc/codecs/cs35l32.c | 631 +++++++++++++++++ sound/soc/codecs/cs35l32.h | 93 +++ sound/soc/codecs/cs4265.c | 1 + sound/soc/codecs/cs42l52.c | 24 +- sound/soc/codecs/cs42l56.c | 27 +- sound/soc/codecs/cs42l73.c | 25 +- sound/soc/codecs/da732x.c | 27 +- sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 756 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/codecs/jz4740.c | 30 +- sound/soc/codecs/lm49453.c | 14 - sound/soc/codecs/max98090.c | 146 +++- sound/soc/codecs/max98090.h | 13 +- sound/soc/codecs/ml26124.c | 24 +- sound/soc/codecs/rt286.c | 2 + sound/soc/codecs/rt5640.c | 49 +- sound/soc/codecs/rt5640.h | 3 + sound/soc/codecs/rt5645.c | 99 +++ sound/soc/codecs/rt5645.h | 5 + sound/soc/codecs/rt5677.c | 327 ++++++++- sound/soc/codecs/rt5677.h | 171 ++++- sound/soc/codecs/sgtl5000.c | 42 +- sound/soc/codecs/ssm2518.c | 13 - sound/soc/codecs/ssm2602-i2c.c | 9 + sound/soc/codecs/ssm2602-spi.c | 7 + sound/soc/codecs/ssm2602.c | 39 +- sound/soc/codecs/ssm4567.c | 343 ++++++++++ sound/soc/codecs/tas2552.c | 70 +- sound/soc/codecs/tlv320aic31xx.c | 107 ++- sound/soc/codecs/tlv320aic31xx.h | 3 +- sound/soc/codecs/tlv320aic3x.c | 29 +- sound/soc/codecs/wm5100.c | 5 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8804.c | 19 - sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8962.c | 5 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8994.c | 18 +- sound/soc/codecs/wm8995.c | 19 - sound/soc/codecs/wm8996.c | 6 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-mcasp.c | 79 ++- sound/soc/davinci/edma-pcm.c | 2 + sound/soc/fsl/Kconfig | 26 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl-asoc-card.c | 574 ++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 6 +- sound/soc/fsl/fsl_esai.c | 19 +- sound/soc/fsl/fsl_esai.h | 8 +- sound/soc/fsl/fsl_sai.c | 58 +- sound/soc/fsl/fsl_sai.h | 8 +- sound/soc/fsl/fsl_spdif.c | 6 +- sound/soc/fsl/fsl_ssi.c | 89 ++- sound/soc/fsl/imx-es8328.c | 232 +++++++ sound/soc/generic/simple-card.c | 226 ++++-- sound/soc/intel/Makefile | 3 +- sound/soc/intel/byt-max98090.c | 1 + sound/soc/intel/byt-rt5640.c | 83 ++- sound/soc/intel/sst-atom-controls.c | 218 ++++++ sound/soc/intel/sst-atom-controls.h | 416 +++++++++++- sound/soc/intel/sst-haswell-pcm.c | 56 +- sound/soc/intel/sst-mfld-platform-compress.c | 38 +- sound/soc/intel/sst-mfld-platform-pcm.c | 106 +-- sound/soc/intel/sst-mfld-platform.h | 58 +- sound/soc/omap/rx51.c | 2 +- sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/Makefile | 2 +- sound/soc/rockchip/rockchip_i2s.c | 28 +- sound/soc/samsung/idma.c | 4 +- sound/soc/samsung/odroidx2_max98090.c | 4 +- sound/soc/samsung/speyside.c | 6 +- sound/soc/sh/fsi.c | 7 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/sh/siu_pcm.c | 4 +- sound/soc/sirf/sirf-usp.c | 24 +- sound/soc/soc-core.c | 669 ++++++++---------- sound/soc/soc-dapm.c | 26 +- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- sound/soc/soc-io.c | 28 - sound/soc/tegra/tegra_max98090.c | 40 ++ sound/soc/txx9/txx9aclc.c | 14 +- 119 files changed, 6323 insertions(+), 1359 deletions(-) create mode 100644 Documentation/devicetree/bindings/regmap/regmap.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2602.txt create mode 100644 Documentation/devicetree/bindings/sound/cs35l32.txt create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5677.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm4567.txt create mode 100644 include/dt-bindings/sound/cs35l32.h create mode 100644 sound/soc/codecs/cs35l32.c create mode 100644 sound/soc/codecs/cs35l32.h create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/codecs/ssm4567.c create mode 100644 sound/soc/fsl/fsl-asoc-card.c create mode 100644 sound/soc/fsl/imx-es8328.c create mode 100644 sound/soc/intel/sst-atom-controls.c

4 12

[alsa-devel] [PATCH v3 0/3] ASoC/ARM: tegra: apalis/colibri t30: sgtl5000 audio
by Marcel Ziswiler 07 Oct '14

07 Oct '14

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Changes in v3: - Revert to not using simple-audio-card being incompatible with tegra. - Rebased to for-next. Changes in v2: - Using simple-audio-card as suggested by Fabio. Marcel Ziswiler (3): ASoC: tegra: add tegra sgtl5000 machine driver ARM: tegra: apalis/colibri t30: integrate audio support ARM: tegra: enable sgtl5000 audio .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 45 +++++ arch/arm/boot/dts/tegra30-apalis.dtsi | 49 +++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 49 +++++ arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_sgtl5000.c | 216 +++++++++++++++++++++ 7 files changed, 372 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 sound/soc/tegra/tegra_sgtl5000.c -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: simple-card: Initialize headphone and mic GPIO numbers
by Geert Uytterhoeven 07 Oct '14

07 Oct '14

The uninitialized default of 0 for gpio_hp_det and gpio_mic_det doesn't play well with asm-generic's gpio_is_valid(): static inline bool gpio_is_valid(int number) { return number >= 0 && number < ARCH_NR_GPIOS; } Hence on r8a7740/armadillo-legacy: sh-mobile-hdmi sh-mobile-hdmi: SH Mobile HDMI Audio Codec sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Headphones sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Mic Jack After that the kernel log is spammed ca. 7 times per second with: sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Headphones Initialize the GPIO numbers with a negative number (-ENOENT) to fix this. Fixes: 3fe240326cc395c6 ("ASoC: simple-card: Add mic and hp detect gpios.") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- This is against sound-asoc/for-next sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fcb431fe20b4ae7b..d1b7293c133eaa45 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -501,6 +501,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; + priv->gpio_hp_det = -ENOENT; + priv->gpio_mic_det = -ENOENT; + /* Get room for the other properties */ priv->dai_props = devm_kzalloc(dev, sizeof(*priv->dai_props) * num_links, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Cleanup the comments
by Fabio Estevam 07 Oct '14

07 Oct '14

Fix grammar and typos. Besides that, also fix the comment about the range of SYS_MCLK, which is from 8 to 27 MHz. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..10160e7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -530,16 +530,16 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * set clock according to i2s frame clock, - * sgtl5000 provide 2 clock sources. - * 1. sys_mclk. sample freq can only configure to + * sgtl5000 provides 2 clock sources: + * 1. sys_mclk: sample freq can only be configured to * 1/256, 1/384, 1/512 of sys_mclk. - * 2. pll. can derive any audio clocks. + * 2. pll: can derive any audio clocks. * * clock setting rules: - * 1. in slave mode, only sys_mclk can use. - * 2. as constraint by sys_mclk, sample freq should - * set to 32k, 44.1k and above. - * 3. using sys_mclk prefer to pll to save power. + * 1. in slave mode, only sys_mclk can be used + * 2. as constraint by sys_mclk, sample freq should be set to 32 kHz, 44.1 kHz + * and above. + * 3. usage of sys_mclk is preferred over pll to save power. */ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) { @@ -549,8 +549,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * sample freq should be divided by frame clock, - * if frame clock lower than 44.1khz, sample feq should set to - * 32khz or 44.1khz. + * if frame clock is lower than 44.1 kHz, sample freq should be set to + * 32 kHz or 44.1 kHz. */ switch (frame_rate) { case 8000: @@ -603,7 +603,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * calculate the divider of mclk/sample_freq, - * factor of freq =96k can only be 256, since mclk in range (12m,27m) + * factor of freq = 96 kHz can only be 256, since mclk is in the range + * of 8 MHz - 27 MHz */ switch (sgtl5000->sysclk / sys_fs) { case 256: @@ -619,7 +620,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) SGTL5000_MCLK_FREQ_SHIFT; break; default: - /* if mclk not satisify the divider, use pll */ + /* if mclk does not satisfy the divider, use pll */ if (sgtl5000->master) { clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; @@ -795,7 +796,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) SGTL5000_LINEREG_D_POWERUP, SGTL5000_LINEREG_D_POWERUP); - /* when internal ldo enabled, simple digital power can be disabled */ + /* when internal ldo is enabled, simple digital power can be disabled */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINREG_SIMPLE_POWERUP, 0); @@ -1079,7 +1080,7 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) /* * sgtl5000 has 3 internal power supplies: * 1. VAG, normally set to vdda/2 - * 2. chargepump, set to different value + * 2. charge pump, set to different value * according to voltage of vdda and vddio * 3. line out VAG, normally set to vddio/2 * -- 1.9.1

2 1

[alsa-devel] [PATCH/alsa-lib] conf/ucm: PAZ00: add Toshiba AC100/Dynabook AZ new config
by zombah 07 Oct '14

07 Oct '14

Recent mainline kernels threat Toshiba AC100 audio hardware as hw:PAZ00 vs old hw:tegraalc5632. This patch adds config files for new hw name and include them to makefiles. Signed-off-by: zombah <zombah(a)gmail.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/PAZ00/HiFi.conf | 45 +++++++++++++++++++++++++++++ src/conf/ucm/PAZ00/Makefile.am | 4 +++ src/conf/ucm/PAZ00/PAZ00.conf | 39 +++++++++++++++++++++++++ src/conf/ucm/PAZ00/Record.conf | 65 ++++++++++++++++++++++++++++++++++++++++++ 6 files changed, 155 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/PAZ00/HiFi.conf create mode 100644 src/conf/ucm/PAZ00/Makefile.am create mode 100644 src/conf/ucm/PAZ00/PAZ00.conf create mode 100644 src/conf/ucm/PAZ00/Record.conf diff --git a/configure.ac b/configure.ac index b8353a0..734163c 100644 --- a/configure.ac +++ b/configure.ac @@ -653,6 +653,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ + src/conf/ucm/PAZ00/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index e089dde..bde89df 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 diff --git a/src/conf/ucm/PAZ00/HiFi.conf b/src/conf/ucm/PAZ00/HiFi.conf new file mode 100644 index 0000000..275516d --- /dev/null +++ b/src/conf/ucm/PAZ00/HiFi.conf @@ -0,0 +1,45 @@ +SectionVerb { + EnableSequence [ + cdev "hw:PAZ00" + + cset "name='Speaker Playback Switch' on" + cset "name='Int Spk Switch' on" + cset "name='Speaker Mix DAC2SPK Playback Switch' on" + ] + DisableSequence [ + cset "name='Speaker Playback Switch' off" + cset "name='Int Spk Switch' off" + cset "name='Speaker Mix DAC2SPK Playback Switch' off" + ] +} + +SectionDevice."Headset".0 { + Comment "3.5mm Headset" + + Value { + JackName "Paz00 Headset Jack" + } + + EnableSequence [ + cdev "hw:PAZ00" + + # Internal speaker + cset "name='Int Spk Switch' off" + + # Headphones + cset "name='Headphone Playback Switch' on" + cset "name='HPL Mix DACL2HP Playback Switch' on" + cset "name='HPR Mix DACR2HP Playback Switch' on" + ] + DisableSequence [ + cdev "hw:PAZ00" + + # Headphones + cset "name='Headphone Playback Switch' off" + cset "name='HPL Mix DACL2HP Playback Switch' off" + cset "name='HPR Mix DACR2HP Playback Switch' off" + + # Internal speaker + cset "name='Int Spk Switch' on" + ] +} diff --git a/src/conf/ucm/PAZ00/Makefile.am b/src/conf/ucm/PAZ00/Makefile.am new file mode 100644 index 0000000..1bf98a6 --- /dev/null +++ b/src/conf/ucm/PAZ00/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/PAZ00 +ucm_DATA = PAZ00.conf HiFi.conf Record.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/PAZ00/PAZ00.conf b/src/conf/ucm/PAZ00/PAZ00.conf new file mode 100644 index 0000000..8b9d904 --- /dev/null +++ b/src/conf/ucm/PAZ00/PAZ00.conf @@ -0,0 +1,39 @@ +Comment "PAZ00 alc5632" + +ValueDefaults { + PlaybackCTL "hw:0" + CaptureCTL "hw:0" +} + +SectionDefaults [ + cdev "hw:PAZ00" + + cset "name='Master Playback Switch' on" + cset "name='Master Playback Volume' 44,44" + cset "name='Headphone Playback Volume' 21,21" + cset "name='Speaker Playback Volume' 21,21" + cset "name='SpeakerOut Mux' 2" + cset "name='SpeakerOut N Mux' LN/-R" + cset "name='AB-D Amp Mux' 0" + cset "name='Left Headphone Mux' 1" + cset "name='Right Headphone Mux' 1" + cset "name='DMIC Boost Capture Volume' 4" + cset "name='Rec Capture Volume' 24" + + # Fix external mic record volumes + cset "name='Mic 1 Boost Volume' 0" + cset "name='Mic 2 Boost Volume' 0" + cset "name='Mic1 Playback Volume' 0" + cset "name='Mic2 Playback Volume' 0" + cset "name='Rec Capture Volume' 0" +] + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Music playback" +} + +SectionUseCase."Record" { + File "Record.conf" + Comment "Playback and capture" +} diff --git a/src/conf/ucm/PAZ00/Record.conf b/src/conf/ucm/PAZ00/Record.conf new file mode 100644 index 0000000..568e2fb --- /dev/null +++ b/src/conf/ucm/PAZ00/Record.conf @@ -0,0 +1,65 @@ +SectionVerb { + EnableSequence [ + cdev "hw:PAZ00" + + cset "name='Speaker Playback Switch' on" + cset "name='Int Spk Switch' on" + cset "name='Speaker Mix DAC2SPK Playback Switch' on" + + cset "name='DMICL Mix DMICL2ADC Capture Switch' on" + cset "name='DMIC En Capture Switch' on" + cset "name='DMIC PreFilter Capture Switch' on" + ] + DisableSequence [ + cset "name='Speaker Playback Switch' off" + cset "name='Int Spk Switch' off" + cset "name='Speaker Mix DAC2SPK Playback Switch' off" + + cset "name='DMICL Mix DMICL2ADC Capture Switch' off" + cset "name='DMIC En Capture Switch' off" + cset "name='DMIC PreFilter Capture Switch' off" + ] +} + +SectionDevice."Headset".0 { + Comment "3.5mm Headset" + + Value { + JackName "Paz00 Headset Jack" + } + + EnableSequence [ + cdev "hw:PAZ00" + + # Internal speaker + cset "name='Int Spk Switch' off" + + # Internal mic + cset "name='DMIC En Capture Switch' off" + + # Headphones + cset "name='Headphone Playback Switch' on" + cset "name='HPL Mix DACL2HP Playback Switch' on" + cset "name='HPR Mix DACR2HP Playback Switch' on" + + # External min + cset "name='Left Capture Mix MIC12REC_L Capture Switch' on" + ] + DisableSequence [ + cdev "hw:PAZ00" + + # Headphones + cset "name='Headphone Playback Switch' off" + cset "name='HPL Mix DACL2HP Playback Switch' off" + cset "name='HPR Mix DACR2HP Playback Switch' off" + + # External mic + cset "name='Left Capture Mix MIC12REC_L Capture Switch' on" + + # Internal speaker + cset "name='Int Spk Switch' on" + + # Internal mic + cset "name='DMIC En Capture Switch' on" + ] +} -- 2.0.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
by Takashi Iwai 07 Oct '14

07 Oct '14

There is another Thinkpad T440 with SSID 17aa:2212 that has a dock port. Reported-by: Siwei Luo <sluo(a)smartbeans.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69d1236365e0..2cc2568af016 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4799,7 +4799,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.1.2

1 0

[alsa-devel] [PATCH] ALSA: usb: caiaq: check for cdev->n_streams > 1
by Daniel Mack 07 Oct '14

07 Oct '14

Coverity spotted a possible DIV0 condition when cdev->n_streams is 0. Fix this by making sure the value is > 1 in snd_usb_caiaq_audio_init(). Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/caiaq/audio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 7103b09..2728447 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,6 +816,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } + if (cdev->n_streams < 2) { + dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); + return -EINVAL; + } + ret = snd_pcm_new(cdev->chip.card, cdev->product_name, 0, cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); -- 2.1.0

2 1