November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Put WM8958 and WM1811 MICBIAS into bypass mode when no audio
by Mark Brown 29 Nov '11

29 Nov '11

When we don't have any active audio we can put the microphone biases into bypass mode to save power at the expense of performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 files changed, 39 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2cb2518..51fefb8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1980,6 +1980,18 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: + /* MICBIAS into regulating mode */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, 0); + break; + default: + break; + } break; case SND_SOC_BIAS_STANDBY: @@ -2032,7 +2044,20 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } - + /* MICBIAS into bypass mode on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, + WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, + WM8958_MICB2_MODE); + break; + default: + break; + } break; case SND_SOC_BIAS_OFF: @@ -3354,6 +3379,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + /* Put MICBIAS into bypass mode by default on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, WM8958_MICB2_MODE); + break; + default: + break; + } + wm8994_update_class_w(codec); wm8994_handle_pdata(wm8994); -- 1.7.7.3

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Re: [alsa-devel] Alsa asynchonus playnback
by Clemens Ladisch 29 Nov '11

29 Nov '11

Martin schmidt wrote: > my problem is, the sound has an Echo (in german Hall-Effekt). > > int write_to_audiostream(short*buf) > { > memcpy(audiobuffer[audiobuffercounter],buf,audio_buffersize*sizeof(short)); > int w= snd_pcm_writei (pcm_handle, audiobuffer[audiobuffercounter], 2* audio_buffersize); This looks correct (as long as the buffer is correctly filled). Do you ever check w? > audiobuffercounter = ++audiobuffercounter % 10; It is a bad idea to disable compiler warnings, or to not heed them. > in a Timer-Event Why a timer? How is the timer synchronized to the sample clock? Regards, Clemens

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[alsa-devel] [PATCH] ASoC: Ensure SYSCLK is enabled for WM8958 accessory detection
by Mark Brown 29 Nov '11

29 Nov '11

Ensure SYSCLK is enabled while running accessory detection on WM8958. It is always required so there is no sense in requiring machine drivers to individually do this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2e577e6..94c0efd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2919,6 +2919,8 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, cb_data = codec; } + snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS"); + wm8994->micdet[0].jack = jack; wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; @@ -2928,6 +2930,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } else { snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); + snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS"); } return 0; -- 1.7.7.3

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[alsa-devel] Alsa asynchonus playnback
by Martin schmidt 29 Nov '11

29 Nov '11

Hello, I want to play audiodata using alsa and do as it is explained in http://alsa.opensrc.org/Asynchronous_Playback_%28Howto%29 http://equalarea.com/paul/alsa-audio.html#interruptex 1. open device 2 define a callback void MyCallback(snd_async_handler_t *pcm_callback) { ... } err = snd_async_add_pcm_handler(&pcm_callback, pcm_handle, MyCallback, NULL); if(err<0){ fprintf (stderr, "err = %d --> snd_async_add_pcm_handler: (%s)\n",err,snd_strerror (err)); } return -38 ( function is not implemented )??? i a loop I generate the audiodata void produce_audiodata(short* left, short *right) { for(int k = 0; k< max;k++){ audio[2*k]= left[k]; audio[2*k+1]= right[k]; } Mycallback(...) is it the right place to call it ? } thanks

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 29 Nov '11

29 Nov '11

The following changes since commit 12a7a709a09aac117b630264cdd526e20d4d0ce2: ASoC: Remove conditional I2C usage from tlv320aic3x driver (2011-11-22 11:59:50 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 A very big update but mostly consisting of lots of small changes that clean up many different drivers. Axel Lin (23): ASoC: cs42l73: Unify the way to define bits of register ASoC: cs42l73: Make inv and format to be unsigned int ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET ASoC: max9877: Update register if either val or val2 is changed ASoC: cs42l51: Fix off-by-one for reg_cache_size ASoC: Convert ep93xx directory to module_platform_driver ASoC: Convert s6000 directory to module_platform_driver ASoC: Convert mxs directory to module_platform_driver ASoC: Convert blackfin directory to module_platform_driver ASoC: Convert imx directory to module_platform_driver ASoC: Convert jz4740 directory to module_platform_driver ASoC: Convert nuc900 directory to module_platform_driver ASoC: Convert kirkwood directory to module_platform_driver ASoC: Convert mid-x86 directory to module_platform_driver ASoC: Convert atmel directory to module_platform_driver ASoC: Convert au1x directory to module_platform_driver ASoC: Convert pxa directory to module_platform_driver ASoC: Convert sh directory to module_platform_driver ASoC: Convert txx9 directory to module_platform_driver ASoC: Convert davinci directory to module_platform_driver ASoC: Convert omap directory to module_platform_driver ASoC: Convert fsl directory to module_platform_driver ASoC: Convert tegra_spdif to use module_platform_driver() Boojin Kim (1): ASoC: SAMSUNG: Fix build error Brian Austin (1): ASoC: Convert CS42L73 to devm_kzalloc() Daniel Mack (1): ASoC: cs4720: use snd_soc_cache_sync() Eric Miao (1): ASoC: skip resume of soc-audio devices without codecs Kuninori Morimoto (1): ASoC: fsi-ak4642: modify specification method of FSI / ak464x Lars-Peter Clausen (9): ASoC: Constify snd_soc_dai_ops structs ASoC: Cleanup duplicated const ASoC: ad193x: Use table based DAPM and controls setup ASoC: ad193x: Provide dB ranges for the volume controls ASoC: ad193x: Make enum items const char * const ASoC: ad193x: Remove non-functional DAPM route controls ASoC: ad193x: Add sysclk DAPM supply ASoC: ad193x: Use snd_soc_update_bits where appropriate ASoC: ad193x: Convert to direct regmap API usage Mark Brown (30): Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 ASoC: Staticise and constify cs42l73_reg_defaults ASoC: Convert wm8776 to table based control and DAPM init Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 ASoC: Remove unused variable in wm8776 driver ASoC: Add fully_routed flag to Speyside machines ASoC: Convert Samsung directory to module_platform_driver ASoC: Staticise non-exported symbols in sta32x ASoC: Use devm_kzalloc() in wm5100 ASoC: Remove driver versioning from ak4642 ASoC: Remove unneeded platform_device.h inclusions from CODECs ASoC: Correct name of Speyside Main Speaker widget ASoC: Convert wm1250-ev1 driver to use devm_kzalloc() ASoC: Convert wm9081 driver to use devm_kzalloc() ASoC: Convert wm8996 to use devm_kzalloc() Merge branch 'for-3.2' into for-3.3 ASoC: Fix CODEC enumeration for auto_nc_codec_pins ASoC: Log automatic pin disconnection per CODEC rather than per card ASoC: Convert WM8962 to devm_kzalloc() ASoC: Convert CODEC drivers to module_platform_driver ASoC: Convert WM8903 MICBIAS to a supply widget ASoC: Error out if we can't generate a LRCLK at all for WM8994 ASoC: Add basic 1277-EV1 Littlemill audio driver Merge branch 'for-3.2' into for-3.3 ASoC: Supply dcs_codes for newer WM1811 revisions Merge branch 'for-3.2' into for-3.3 Paul Bolle (1): ASoC: drop support for PlayPaq with WM8510 Peter Ujfalusi (2): ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC ASoC: sdp4430: Add support for digital microphones Stephen Warren (11): ASoC: Tegra PCM: Use module_platform_driver ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver ASoC: Tegra: Move DAS configuration into machine drivers ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver ASoC: Tegra DAS: Add device tree binding ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver ASoC: Tegra TrimSlice machine: Use devm_ APIs and module_platform_driver ASoC: Tegra I2S: Remove dependency on pdev->id ASoC: Implement fully_routed card property ASoC: Tegra+WM903 machine: Set the new fully_routed flag ASoC: TrimSlice machine: Set the new fully_routed flag Timur Tabi (1): ASoC: mpc8610: tell the CS4270 codec that it's the master .../devicetree/bindings/sound/tegra20-das.txt | 12 + arch/arm/mach-shmobile/board-ap4evb.c | 15 +- arch/arm/mach-shmobile/board-mackerel.c | 14 +- arch/sh/boards/mach-se/7724/setup.c | 14 +- include/sound/sh_fsi.h | 12 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/soc/atmel/Kconfig | 21 +- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/atmel-pcm.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 14 +- sound/soc/atmel/playpaq_wm8510.c | 473 ----------------- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/db1000.c | 13 +- sound/soc/au1x/db1200.c | 13 +- sound/soc/au1x/dbdma2.c | 13 +- sound/soc/au1x/dma.c | 13 +- sound/soc/au1x/i2sc.c | 13 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 15 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +- sound/soc/blackfin/bf5xx-ac97.c | 13 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +- sound/soc/blackfin/bf5xx-i2s.c | 15 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +- sound/soc/blackfin/bf5xx-tdm.c | 14 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 16 +- sound/soc/codecs/ac97.c | 14 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad1836.h | 2 +- sound/soc/codecs/ad193x.c | 190 ++++---- sound/soc/codecs/ad193x.h | 17 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/ad73311.c | 12 +- sound/soc/codecs/ads117x.c | 12 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 3 +- sound/soc/codecs/ak4641.c | 5 +- sound/soc/codecs/ak4642.c | 7 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 3 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 14 +- sound/soc/codecs/cs4270.c | 13 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l51.c | 5 +- sound/soc/codecs/cs42l73.c | 16 +- sound/soc/codecs/cs42l73.h | 18 +- sound/soc/codecs/cx20442.c | 12 +- sound/soc/codecs/da7210.c | 3 +- sound/soc/codecs/dfbmcs320.c | 12 +- sound/soc/codecs/dmic.c | 12 +- sound/soc/codecs/jz4740.c | 14 +- sound/soc/codecs/max98088.c | 5 +- sound/soc/codecs/max98095.c | 7 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9877.c | 10 +- sound/soc/codecs/pcm3008.c | 12 +- sound/soc/codecs/rt5631.c | 3 +- sound/soc/codecs/sgtl5000.c | 3 +- sound/soc/codecs/sn95031.c | 22 +- sound/soc/codecs/spdif_transciever.c | 13 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/sta32x.c | 7 +- sound/soc/codecs/stac9766.c | 16 +- sound/soc/codecs/tlv320aic23.c | 3 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/tlv320dac33.c | 3 +- sound/soc/codecs/twl4030.c | 16 +- sound/soc/codecs/twl6040.c | 14 +- sound/soc/codecs/uda134x.c | 14 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wl1273.c | 14 +- sound/soc/codecs/wm1250-ev1.c | 10 +- sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm5100.c | 12 +- sound/soc/codecs/wm8350.c | 14 +- sound/soc/codecs/wm8400.c | 14 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 3 +- sound/soc/codecs/wm8580.c | 5 +- sound/soc/codecs/wm8711.c | 3 +- sound/soc/codecs/wm8727.c | 12 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 3 +- sound/soc/codecs/wm8737.c | 3 +- sound/soc/codecs/wm8741.c | 3 +- sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8770.c | 3 +- sound/soc/codecs/wm8776.c | 19 +- sound/soc/codecs/wm8782.c | 12 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 3 +- sound/soc/codecs/wm8903.c | 7 +- sound/soc/codecs/wm8904.c | 3 +- sound/soc/codecs/wm8940.c | 3 +- sound/soc/codecs/wm8955.c | 3 +- sound/soc/codecs/wm8960.c | 3 +- sound/soc/codecs/wm8961.c | 3 +- sound/soc/codecs/wm8962.c | 12 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8974.c | 3 +- sound/soc/codecs/wm8978.c | 3 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 3 +- sound/soc/codecs/wm8991.c | 3 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 26 +- sound/soc/codecs/wm8995.c | 6 +- sound/soc/codecs/wm8996.c | 7 +- sound/soc/codecs/wm9081.c | 8 +- sound/soc/codecs/wm9705.c | 14 +- sound/soc/codecs/wm9712.c | 16 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_hubs.c | 1 - sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/davinci/davinci-mcasp.c | 14 +- sound/soc/davinci/davinci-pcm.c | 12 +- sound/soc/davinci/davinci-vcif.c | 14 +- sound/soc/ep93xx/edb93xx.c | 12 +- sound/soc/ep93xx/ep93xx-ac97.c | 14 +- sound/soc/ep93xx/ep93xx-i2s.c | 15 +- sound/soc/ep93xx/ep93xx-pcm.c | 13 +- sound/soc/ep93xx/simone.c | 12 +- sound/soc/ep93xx/snappercl15.c | 13 +- sound/soc/fsl/fsl_dma.c | 15 +- sound/soc/fsl/fsl_ssi.c | 17 +- sound/soc/fsl/mpc5200_dma.c | 12 +- sound/soc/fsl/mpc5200_psc_ac97.c | 20 +- sound/soc/fsl/mpc5200_psc_i2s.c | 18 +- sound/soc/fsl/mpc8610_hpcd.c | 24 +- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +- sound/soc/imx/imx-pcm-fiq.c | 12 +- sound/soc/imx/imx-ssi.c | 15 +- sound/soc/jz4740/jz4740-i2s.c | 14 +- sound/soc/jz4740/jz4740-pcm.c | 12 +- sound/soc/kirkwood/kirkwood-dma.c | 12 +- sound/soc/kirkwood/kirkwood-i2s.c | 14 +- sound/soc/mid-x86/mfld_machine.c | 14 +- sound/soc/mid-x86/sst_platform.c | 14 +- sound/soc/mxs/mxs-pcm.c | 12 +- sound/soc/mxs/mxs-saif.c | 14 +- sound/soc/mxs/mxs-sgtl5000.c | 12 +- sound/soc/nuc900/nuc900-ac97.c | 15 +- sound/soc/nuc900/nuc900-pcm.c | 12 +- sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-dmic.c | 549 ++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 +++ sound/soc/omap/omap-hdmi.c | 14 +- sound/soc/omap/omap-mcbsp.c | 14 +- sound/soc/omap/omap-mcpdm.c | 14 +- sound/soc/omap/omap-pcm.c | 12 +- sound/soc/omap/omap4-hdmi-card.c | 12 +- sound/soc/omap/sdp4430.c | 85 +++- sound/soc/pxa/hx4700.c | 13 +- sound/soc/pxa/mioa701_wm9713.c | 13 +- sound/soc/pxa/palm27x.c | 13 +- sound/soc/pxa/pxa-ssp.c | 14 +- sound/soc/pxa/pxa2xx-ac97.c | 18 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 12 +- sound/soc/s6000/s6000-i2s.c | 14 +- sound/soc/s6000/s6000-pcm.c | 12 +- sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/ac97.c | 16 +- sound/soc/samsung/dma.c | 12 +- sound/soc/samsung/i2s.c | 14 +- sound/soc/samsung/idma.c | 12 +- sound/soc/samsung/littlemill.c | 227 ++++++++ sound/soc/samsung/lowland.c | 12 +- sound/soc/samsung/pcm.c | 14 +- sound/soc/samsung/s3c2412-i2s.c | 14 +- sound/soc/samsung/s3c24xx-i2s.c | 14 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 16 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 16 +- sound/soc/samsung/s3c24xx_uda134x.c | 14 +- sound/soc/samsung/smdk_wm8580pcm.c | 14 +- sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/spdif.c | 14 +- sound/soc/samsung/speyside.c | 17 +- sound/soc/samsung/speyside_wm8962.c | 13 +- sound/soc/sh/dma-sh7760.c | 12 +- sound/soc/sh/fsi-ak4642.c | 127 +---- sound/soc/sh/fsi-hdmi.c | 13 +- sound/soc/sh/fsi.c | 15 +- sound/soc/sh/hac.c | 14 +- sound/soc/sh/siu_dai.c | 15 +- sound/soc/sh/ssi.c | 14 +- sound/soc/soc-core.c | 12 +- sound/soc/soc-dapm.c | 73 +++ sound/soc/tegra/tegra_das.c | 53 +-- sound/soc/tegra/tegra_i2s.c | 137 ++---- sound/soc/tegra/tegra_i2s.h | 1 + sound/soc/tegra/tegra_pcm.c | 13 +- sound/soc/tegra/tegra_spdif.c | 14 +- sound/soc/tegra/tegra_wm8903.c | 79 ++-- sound/soc/tegra/trimslice.c | 45 +- sound/soc/txx9/txx9aclc-ac97.c | 13 +- sound/soc/txx9/txx9aclc.c | 12 +- 210 files changed, 1629 insertions(+), 2476 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-das.txt delete mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h create mode 100644 sound/soc/samsung/littlemill.c

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 29 Nov '11

29 Nov '11

The following changes since commit ed3e80c4c991a52f9fce3421536a78e331ae0949: ASoC: Ensure WM8731 register cache is synced when resuming from disabled (2011-11-21 16:58:46 +0000) The usual series of small updates, the one core update (from Eric) is small and well targetted. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 Axel Lin (3): ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET ASoC: max9877: Update register if either val or val2 is changed ASoC: cs42l51: Fix off-by-one for reg_cache_size Boojin Kim (1): ASoC: SAMSUNG: Fix build error Daniel Mack (1): ASoC: cs4720: use snd_soc_cache_sync() Eric Miao (1): ASoC: skip resume of soc-audio devices without codecs Mark Brown (3): ASoC: Correct name of Speyside Main Speaker widget ASoC: Error out if we can't generate a LRCLK at all for WM8994 ASoC: Supply dcs_codes for newer WM1811 revisions Paul Bolle (1): ASoC: drop support for PlayPaq with WM8510 Timur Tabi (1): ASoC: mpc8610: tell the CS4270 codec that it's the master sound/soc/atmel/Kconfig | 21 +-- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/playpaq_wm8510.c | 473 -------------------------------------- sound/soc/codecs/ad1836.h | 2 +- sound/soc/codecs/cs4270.c | 10 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/max9877.c | 10 +- sound/soc/codecs/wm8994.c | 7 + sound/soc/fsl/mpc8610_hpcd.c | 24 ++- sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 6 + 12 files changed, 40 insertions(+), 522 deletions(-) delete mode 100644 sound/soc/atmel/playpaq_wm8510.c

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[alsa-devel] ALSA: hda_intel - revert a quirk that affect VIA chipsets
by Marc Vertes 29 Nov '11

29 Nov '11

This quirk sould be reverted. It has the following probems: 1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the ID used matches a much broader range, potentially all boards containing a VIA chipset model in the family of vendor VIA 0x1106 and audio device ID 0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets. 2) VIA chipsets rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.h… Signed-of-by: Marc Vertes <mvertes(a)sigfox.com> --- linux-3.2-rc3/sound/pci/hda/hda_intel.c.orig 2011-11-24 05:20:28.000000000 +0100 +++ linux-3.2-rc3/sound/pci/hda/hda_intel.c 2011-11-28 17:29:01.044095455 +0100 @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),

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[alsa-devel] [PATCH v2 1/7] firmware: Sigma: Prevent out of bounds memory access
by Lars-Peter Clausen 29 Nov '11

29 Nov '11

The SigmaDSP firmware loader currently does not perform enough boundary size checks when processing the firmware. As a result it is possible that a malformed firmware can cause an out of bounds memory access. This patch adds checks which ensure that both the action header and the payload are completely inside the firmware data boundaries before processing them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- Mark, in case you are ok with these patches could you take them through your ASoC tree? The firmware subsystem is orphaned. Changes since v1: * Minor code style issues fixes * Add comment exlaining the return value of process_sigma_action Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/firmware/sigma.c | 76 +++++++++++++++++++++++++++++++++------------- include/linux/sigma.h | 5 --- 2 files changed, 55 insertions(+), 26 deletions(-) diff --git a/drivers/firmware/sigma.c b/drivers/firmware/sigma.c index f10fc52..c780baa 100644 --- a/drivers/firmware/sigma.c +++ b/drivers/firmware/sigma.c @@ -14,13 +14,34 @@ #include <linux/module.h> #include <linux/sigma.h> -/* Return: 0==OK, <0==error, =1 ==no more actions */ +static size_t sigma_action_size(struct sigma_action *sa) +{ + size_t payload = 0; + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + payload = sigma_action_len(sa); + break; + default: + break; + } + + payload = ALIGN(payload, 2); + + return payload + sizeof(struct sigma_action); +} + +/* + * Returns a negative error value in case of an error, 0 if processing of + * the firmware should be stopped after this action, 1 otherwise. + */ static int -process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) +process_sigma_action(struct i2c_client *client, struct sigma_action *sa) { - struct sigma_action *sa = (void *)(ssfw->fw->data + ssfw->pos); size_t len = sigma_action_len(sa); - int ret = 0; + int ret; pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, sa->instr, sa->addr, len); @@ -29,44 +50,50 @@ process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - if (ssfw->fw->size < ssfw->pos + len) - return -EINVAL; ret = i2c_master_send(client, (void *)&sa->addr, len); if (ret < 0) return -EINVAL; break; - case SIGMA_ACTION_DELAY: - ret = 0; udelay(len); len = 0; break; - case SIGMA_ACTION_END: - return 1; - + return 0; default: return -EINVAL; } - /* when arrive here ret=0 or sent data */ - ssfw->pos += sigma_action_size(sa, len); - return ssfw->pos == ssfw->fw->size; + return 1; } static int process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) { - pr_debug("%s: processing %p\n", __func__, ssfw); + struct sigma_action *sa; + size_t size; + int ret; + + while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { + sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + + size = sigma_action_size(sa); + ssfw->pos += size; + if (ssfw->pos > ssfw->fw->size || size == 0) + break; + + ret = process_sigma_action(client, sa); - while (1) { - int ret = process_sigma_action(client, ssfw); pr_debug("%s: action returned %i\n", __func__, ret); - if (ret == 1) - return 0; - else if (ret) + + if (ret <= 0) return ret; } + + if (ssfw->pos != ssfw->fw->size) + return -EINVAL; + + return 0; } int process_sigma_firmware(struct i2c_client *client, const char *name) @@ -89,7 +116,14 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) /* then verify the header */ ret = -EINVAL; - if (fw->size < sizeof(*ssfw_head)) + + /* + * Reject too small or unreasonable large files. The upper limit has been + * chosen a bit arbitrarily, but it should be enough for all practical + * purposes and having the limit makes it easier to avoid integer + * overflows later in the loading process. + */ + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) goto done; ssfw_head = (void *)fw->data; diff --git a/include/linux/sigma.h b/include/linux/sigma.h index e2accb3..9a138c2 100644 --- a/include/linux/sigma.h +++ b/include/linux/sigma.h @@ -50,11 +50,6 @@ static inline u32 sigma_action_len(struct sigma_action *sa) return (sa->len_hi << 16) | sa->len; } -static inline size_t sigma_action_size(struct sigma_action *sa, u32 payload_len) -{ - return sizeof(*sa) + payload_len + (payload_len % 2); -} - extern int process_sigma_firmware(struct i2c_client *client, const char *name); #endif -- 1.7.7.3

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[alsa-devel] [PATCH RESEND] ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt
by Axel Lin 29 Nov '11

29 Nov '11

What we want is to clear BIT[5:4](PCM_MODE_MASK) and BIT[3](PCM_BIT_ORDER) bits, but current code clears BIT[2:0]. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- This patch was sent on http://comments.gmane.org/gmane.linux.alsa.devel/91501 and Brian replied "tested and acked" for both patches. [PATCH 1/2] ASoC: cs42l73: Unify the way to define bits of register is already merged, but [PATCH 2/2] ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt is missed. sound/soc/codecs/cs42l73.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 0306e72..a2af595 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1024,7 +1024,8 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } if (spc & SPDIF_PCM) { - spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ + spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ASoC: Convert smdk_wm8994pcm to use module_platform_driver()
by Axel Lin 29 Nov '11

29 Nov '11

Use the module_platform_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/smdk_wm8994pcm.c | 14 +------------- 1 files changed, 1 insertions(+), 13 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index da9c2a2..23c7fb7 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -158,19 +158,7 @@ static struct platform_driver snd_smdk_driver = { .remove = __devexit_p(snd_smdk_remove), }; -static int __init smdk_audio_init(void) -{ - return platform_driver_register(&snd_smdk_driver); -} - -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_driver_unregister(&snd_smdk_driver); -} - -module_exit(smdk_audio_exit); +module_platform_driver(snd_smdk_driver); MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); -- 1.7.5.4

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