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[alsa-devel] [PATCH 7/9] ARM: DaVinci: ASoC: Adds ALSA SoC DIT/DIR driver support for DM646X
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds ALSA SoC DIT/DIR driver support for DM646X Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.c | 63 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 63 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.c diff --git a/sound/soc/codecs/codec_stubs.c b/sound/soc/codecs/codec_stubs.c new file mode 100644 index 0000000..60d3446 --- /dev/null +++ b/sound/soc/codecs/codec_stubs.c @@ -0,0 +1,63 @@ +/* + * ALSA SoC DaVinci DIT/DIR driver + * + * TI DaVinci audio controller can operate in DIT/DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai dit_stub_dai[] = { + { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +struct snd_soc_dai dir_stub_dai[] = { + { + .name = "DIR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +static int __init dit_modinit(void) +{ + return snd_soc_register_dai(dit_stub_dai); +} +module_init(dit_modinit); + +static void __exit dit_exit(void) +{ + snd_soc_unregister_dai(dit_stub_dai); +} +module_exit(dit_exit); + -- 1.5.6

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Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 18 Mar '09

18 Mar '09

Hi Giuliano, At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. > > > > - run checkpatch.pl against the patch and fix issues reported there > > Ok ! Any progress on this? I'm asking it because 2.6.29 kernel will likely go out soon, and all patches should be merged before the merge-window is opened. If you have no time, either we skip it or I'll do a quick fix. thanks, Takashi

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[alsa-devel] [PATCH 3/9] ARM: DaVinci: ASoC: Adds header file for McASP Audio Layer for TI DM646X processor
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds header file for McASP Audio Layer for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-i2s-mcasp.h | 312 +++++++++++++++++++++++++++++++++ 1 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..7d2f7bb --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,312 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#ifndef DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include "davinci-pcm.h" + + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define FREE BIT(0) +#define SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ + /* and Transmit Bad Clock Detect /32 Counter + Reset */ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ -- 1.5.6

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[alsa-devel] Correct macro for Headset widgets
by Lopez Cruz, Misael 18 Mar '09

18 Mar '09

What is the correct way to declare a Headset widget in a machine driver? I originally thought that by using SND_SOC_DAPM_HP macro was enough, but it's not since the headset is both headphones and microphone. In my machine driver I have the following widget and map for headset: SND_SOC_DAPM_HP("Headset Jack", NULL) ... /* Headset: HSMIC (with bias), HSOL, HSOR */ {"Headset Jack", NULL, "HSOL"}, {"Headset Jack", NULL, "HSOR"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Jack"}, but the "Headset Mic Bias" widget never gets powered when the "Headset Jack" pin is enabled. When the audio map is scanned, only valid routes are powered: DAC-Output, Input-DAC, Input-Output, DAC-ADC. And "Headset Jack" (declared as SND_SOC_DAPM_HP) is not a valid input endpoint, then the capture route becomes invalid. Can we have a macro specifically for Headset widgets? Thanks, -Misa

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[alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Jason Holmes 18 Mar '09

18 Mar '09

Hi all, I have a Dell SP2309W monitor which contains an integrated webcam and microphones. I have been unable to get the integrated microphones to record any sound. I ran utils_alsa-info.sh and the link with the results in posted below: http://www.alsa-project.org/db/?f=590e122a466ce09cd1f911ba7146a87df2b5cbeb The integrated microphone seems to be made by OmniVision Technologies. The results of running arecord -l are: **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Webcam [Monitor Webcam], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: default [USB Sound Device ], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 and the results of arecord --device=default:CARD=Webcam test are: ALSA lib pcm_direct.c:866:(snd1_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dsnoop.c:604:(snd_pcm_dsnoop_open) unable to initialize slave arecord: main:590: audio open error: Invalid argument Thanks for your help. -- Jason Holmes

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[alsa-devel] Cleanup patches for snd-usb-caiaq
by Daniel Mack 18 Mar '09

18 Mar '09

Following up to this mail are 3 patches to clean up the snd-usb-caiaq driver. [PATCH 1/3] snd-usb-caiaq: only warn once on streaming errors [PATCH 2/3] snd-usb-caiaq: drop bogus iso packets [PATCH 3/3] snd-usb-caiaq: bump version number

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Re: [alsa-devel] Bad page state in process 'rmmod' cs4281
by "Timo T.Hyrsylä" 18 Mar '09

18 Mar '09

On 13.3.2009 13:04, Takashi Iwai wrote: > At Mon, 09 Mar 2009 12:24:51 +0100, > wrote: >> At Mon, 09 Mar 2009 10:32:02 +0200, >> Timo T.Hyrsylä wrote: >>> Trying to make Alsa work on our embedded arm system with ixdp425 and CS4281. >>> >>> kernel is 2.6.20.3 and the Alsas (driver+lib+utils) latest 1.0.19 >>> >>> Other than the two below errors, the system seems to work ok. >>> >>> When removing the snd_cs4281 module, dmesg reports multiple errors about >>> bad page >>> >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007300 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007320 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> >>> I've managed to track down the error upto dma_free_coherent. >>> >>> snd_pcm_free > snd_pcm_lib_preallocate_free_for_all > >>> snd_pcm_lib_preallocate_free > snd_pcm_lib_preallocate_dma_free > >>> snd_dma_free_pages > snd_free_dev_pages > dma_free_coherent >>> >>> >>> Another issue, possibly related, is that when running aplay with --mmap >>> it will crash with: >>> >>> Internal error: Oops: f3 [#1] >>> PC is at snd_pcm_mmap_data_nopage+0x10c/0x148 [snd_pcm] >>> LR is at __handle_mm_fault+0x16c/0x8ac >> This is likely the problem with dma_alloc_coherent() and mmap on >> a non-coherent architecture. I have patches for these, but only for >> recent kernels. The patches are found in test/dma-fix branch of sound >> git tree: >> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/dma-fix > > Alternatively, an easier "fix" would be simply disable mmap support. > Remove SNDRV_PCM_INFO_MMAP* from snd_pcm_hardware definitions. > > This may disable the conversion plugin usage in the earlier alsa-lib > versions, though. > > > Takashi Tried to apply the patch changes into my version, but it did not affect the unloading 'bad page' problem at all. Instead it did make all play attempts (even non MMAP) to crash. (no crash details available since I desided to forget about it for now, working on unrelated issues for awhile) I did disable the MMAP so that player wont crash when requesting MMAP. System still dumps the bad pages at unload, play and capture seem to work OK, not tested extensively thou. -- Timo

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 17 Mar '09

17 Mar '09

The following changes since commit 852fd9e50f62b4ea7afe26eee0710464de4801b8: Mark Brown (1): ASoC: Each PXA AC97 DAI needs a separate ops are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Atsushi Nemoto (1): ASoC: Only deregister AC97 dev if it's name was not "AC97" Daniel Mack (1): ASoC: switch PXA SSP driver from network mode to PSP Mark Brown (3): ASoC: Fix non-networked I2S mode for PXA SSP ASoC: Fix Zylonite for non-networked SSP mode Merge branch 'pxa-ssp' into for-2.6.30 sound/soc/pxa/pxa-ssp.c | 55 ++++++++++++++++++++++++++++++++++++---------- sound/soc/pxa/zylonite.c | 55 +++++++++++++++++++++------------------------- sound/soc/soc-core.c | 3 +- 3 files changed, 70 insertions(+), 43 deletions(-)

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[alsa-devel] Tenx USB audio adapter
by PETER EASTHOPE 17 Mar '09

17 Mar '09

Folk, I was advised to ask about the Tenx USB audio adapter here. lshal claims it is a standard HID device and a standard USB device. Ref. http://carnot.yi.org/lshal1.out Nevertheless, sound in and sound out both fail in my system. Some information is at the bottom of this page. "http://carnot.yi.org/DebianPage.html#USBAudio" Any suggestions? Thanks, ... Peter E. -- http://members.shaw.ca/peasthope/ http://carnot.yi.org/ = http://carnot.pathology.ubc.ca/

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[alsa-devel] Fedora 10 x86_64 alsa problem: Realtek ALC889A Audio Codec(ATI azalia) not detected on alsa
by Emmanuel Chanel 17 Mar '09

17 Mar '09

I wrote this on https://bugzilla.redhat.com/show_bug.cgi?id=489828 I've used the driver of kernel package. And Mr. Jaroslav Kysela replied me that I could try snapshot source. (Thx, Mr. Jaroslav Kysela. And on #alsa on Freenode, Thingol and gnubien advised me much. Thank you!) But I couldn't get sound back none of both: stable(alsa-driver-1.0.19) and snapshot (alsa-driver-1.0.19.34.g70f78.363.g0ec3b). My alsa info is on http://www.alsa-project.org/db/?f=af6a6e787d4bd5212bf766bf5d09fda1ab5e2300 I use desktop. Its mother board is GA-MA69H-S3H . The soundcard is onboard. Realtek ALC889A Audio Codec. # lspci |grep -i "multi\|audio\|040[0-3]" # Audio device: ATI Technologies Inc SB600 Azalia (Intel HDA) When I installed stable release of alsa-driver --with-debug=verbose , dmesg says HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:2218: chipset global capabilities = 0x4401 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:786: codec_mask = 0x8 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1214: hda_intel: codec #3 probed OK ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/../../alsa-kernel/pci/hda/hda_codec.c:788: hda_codec: no AFG or MFG node found ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1298: hda-intel: no codecs initialized HDA Intel 0000:00:14.2: PCI INT A disabled -- Emmanuel Chanel <emmanuelchanel(a)gmail.com>

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