July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] snd-hda-intel acl269 toshiba r700 speakers work, headphones do not
by Thomas Hoffmann 21 Jul '10

21 Jul '10

Dear Alsa developers, Sound works from the built in laptop speakers. When headphones are plugged in, the speakers properly mute, but there is no sound coming from the headphones. I've tried some of the options to the alsa-base.conf file to no avail; generally they made it so sound came out of the speakers always, even when headphones were plugged in. I previously sent an e-mail to this list server: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5064 where they suggested I e-mail you. My system info is available from http://www.alsa-project.org/db/?f=f9aaf6e43b669cf2c4b2215781a414b12843e2a3 Best, Tom

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[alsa-devel] [PATCH 1/3] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] Audio dropouts with pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32).
by VDR User 20 Jul '10

20 Jul '10

Hi. I've been experiencing audio drops on dvb streams that use ac3 audio (h264+ac3 hdtv streams) and occasionally mp2 audio (sdtv channels) . When the ac3 audio drops out, it usually recovers after a few seconds though sometimes it never does. When mp2 drops out, it almost never recovers. Retuning the channel always fixes the drop out. When this happens there's usually no interruption in the video so I question whether stream errors are causing the problem. When a dropout occurs, I always see the following in my xine log: pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). I (and others apparently) have been experiencing this problem for quite some time. I have read about and tried adding notsc to no kernel boot line, which has had no affect. Here's a clip from the most recent drop outs: audio jump, diff=-90720 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21713 0-frames to fill a gap of 40723 pts video_out: throwing away image with pts 176336725 because it's too old (diff : 10842). video_out: throwing away image with pts 176339939 because it's too old (diff : 16618). video jump 200 frames delivered, 0 frames skipped, 2 frames discarded video jump audio jump, diff=-103680 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21233 0-frames to fill a gap of 39823 pts video_out: throwing away image with pts 183068800 because it's too old (diff : 6699). video jump video_out: throwing away image with pts 183071835 because it's too old (diff : 12728). pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). 200 frames delivered, 0 frames skipped, 2 frames discarded Some system specs are: debian testing kernel 2.6.34.1 alsa-base 1.0.23+dfsg-1 alsa-utils 1.0.23-2 alsa drivers compiled from today's snapshot (2010-07-20). in /etc/modprobe.d/alsa-base.conf: options snd-hda-intel enable_msi=1 probe_mask=0xffff,0xfff2 lsmod: snd_hda_codec_nvhdmi 9750 1 snd_hda_intel 14764 0 snd_hda_codec 48881 2 snd_hda_codec_nvhdmi,snd_hda_intel snd_hwdep 3566 1 snd_hda_codec snd_pcm 44172 2 snd_hda_intel,snd_hda_codec snd_timer 11991 1 snd_pcm snd 28664 6 snd_hda_codec_nvhdmi,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 548 1 snd snd_page_alloc 4669 2 snd_hda_intel,snd_pcm I'd greatly appreciate any help in getting this problem worked out and hopefully solved once and for all. Thanks in advance for _anything_. Best regards, Derek

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[alsa-devel] [PATCH] wm8960: fix compile error
by Axel Lin 20 Jul '10

20 Jul '10

This patch fixes compile error: sound/soc/codecs/wm8960.c: In function 'wm8960_hw_params': sound/soc/codecs/wm8960.c:511: error: 'codec' undeclared (first use in this function) sound/soc/codecs/wm8960.c:511: error: (Each undeclared identifier is reported only once sound/soc/codecs/wm8960.c:511: error: for each function it appears in.) make[3]: *** [sound/soc/codecs/wm8960.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against Liam's multi-component branch. Axel sound/soc/codecs/wm8960.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e6df6e3..599c559 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -508,8 +508,8 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct snd_soc_codec *codec = rtd->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; -- 1.5.4.3

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 0c74a939d84730818bc9bf1efa5ad7184bb2f248: ASoC: au1x: fix section mismatch in psc-i2s.c (2010-07-13 12:39:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (2): ASoC: Select wm_hubs automatically for WM8994 ASoC:Support Samsung SoC(S5P) in I2Sv2 Eric Bénard (1): ASoC: eukrea-tlv320: add support for cpuimx35sd Kulikov Vasiliy (3): ASoC: davinci: check kzalloc() result (typo) ASoC: imx: check kzalloc() result and fix memory leak ASoC: imx: check kzalloc() result and fix memory leak Kuninori Morimoto (1): ASoC: fsi: Add specified ID for soc-audio Mark Brown (8): Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Handle read failures in codec_reg ASoC: Remove duplicate AUX definition from WM8776 Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Explicitly disable DC servo on WM hubs headphone powerdown ASoC: Implement WM8994 AIF1ADC2 paths Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode Sekhar Nori (1): ASoC: davinci: let platform data define edma queue numbers arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 +- arch/arm/mach-davinci/include/mach/asp.h | 3 +- include/sound/sh_fsi.h | 3 ++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/wm8727.c | 2 + sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8994.c | 32 ++++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.c | 2 + sound/soc/davinci/davinci-i2s.c | 10 +++++++++ sound/soc/davinci/davinci-mcasp.c | 6 +++- sound/soc/davinci/davinci-pcm.c | 5 ++- sound/soc/davinci/davinci-pcm.h | 3 +- sound/soc/davinci/davinci-vcif.c | 2 +- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 +- sound/soc/imx/imx-pcm-dma-mx2.c | 6 ++++- sound/soc/imx/imx-pcm-fiq.c | 6 ++++- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/sh/fsi-ak4642.c | 4 +- sound/soc/sh/fsi-da7210.c | 4 +- sound/soc/soc-core.c | 22 +++++++++++++++---- 24 files changed, 103 insertions(+), 29 deletions(-)

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[alsa-devel] [PATCH RFC 0/1] ASoC: WL1273 FM Radio codes
by Matti J. Aaltonen 20 Jul '10

20 Jul '10

Hello. This is the ALSA part of our WL1273 FM Radio driver. I've sent the complete driver to the linux-media list for review. But as people on that list are not in general ALSA specialists I'm sending the codec part here. So all comments are welcome. Thank you, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 593 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 42 ++++ 2 files changed, 635 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

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[alsa-devel] ASoC update for 2.6.35
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 637727838a5e82bc9285ab078a793eaae590bacb: ASoC: fsi: fixup wrong value setting order of TDM (2010-07-13 12:26:26 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (1): ASoC: Select wm_hubs automatically for WM8994 Mark Brown (1): ASoC: Remove duplicate AUX definition from WM8776 Another series of small, driver-specific fixes. sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/wm8727.c | 2 ++ sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 17 +++++++++++++++-- 1 files changed, 15 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 0/12] sound/alsa/soc/codec: fix memory leak and resource relaim in error path
by Axel Lin 20 Jul '10

20 Jul '10

This serial of patches fixes memory leak and resource relaim in error path [PATCH 1/12] ad1836: fix a memory leak if another ad1836 is registered [PATCH 2/12] ak4642: fix a memory leak if failed to initialise AK4642 [PATCH 3/12] da7210: fix a memory leak if failed to initialise da7210 audio codec [PATCH 4/12] wm8523: fix resource reclaim in wm8523_register error path [PATCH 5/12] wm8711: fix a memory leak if another WM8711 is registered [PATCH 6/12] wm8904: fix resource reclaim in wm8904_register error path [PATCH 7/12] wm8940: fix resource reclaim in wm8940_register error path [PATCH 8/12] wm8955: fix resource reclaim in wm8955_register error path [PATCH 9/12] wm8961: fix resource reclaim in wm8961_register error path [PATCH 10/12] wm8974: fix a memory leak if another WM8974 is registered [PATCH 11/12] wm8978: fix a memory leak if another WM8978 is registered [PATCH 12/12] wm9081: fix resource reclaim in wm9081_register error path Regards, Axel

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Re: [alsa-devel] [PATCH v2] asoc: davinci: let platform data define edma queue numbers
by Mark Brown 20 Jul '10

20 Jul '10

On Mon, Jul 19, 2010 at 12:31:16PM +0530, Sekhar Nori wrote: > Currently the EDMA queue to be used by for servicing ASP through > internal RAM is fixed to EDMAQ_0 and that to service internal RAM > from external RAM is fixed to EDMAQ_1. Applied, thanks. It's "ASoC", BTW.

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