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[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by natunobilis＠gentoobr.org 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

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[alsa-devel] Surround problems on ALC883
by Pavel Patz 18 Apr '07

18 Apr '07

Hi all, I have problems with chip Realtek ALC883 (lspci: 00:06.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)) and surround sound. Stereo sound is working (games, music) fine - all 6 speaker are playing. But when I tried to run "aplay chan-id.wav", i've heard only "front left" (front & rear left + center were playing) and "front right" (right front & rear were playing), then was silence until the end of the file. Running "aplay -Dsurround51 chan-id.wav" causes that aplay hangs-up without any output. Running it with -v parameter produces this output: Playing WAVE 'Desktop/Surround-SDL-testfiles/chan-id.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 6 Soft volume PCM Control: PCM Playback Volume min_dB: -51 resolution: 256 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 and then hang up until i press ctrl+c. Same results i get with speaker-test. I'm using alsa-lib-1.0.14-0.4.rc3.fc7 I've found similar problem in archive (about half a year old) but it still seems it isn't working. Is there any possibility, that this will be fixed in future? Thanks for all your help (and please leave me in CC, i'm not on mailing-list). Following lsmod & /proc/asound/card0/codec#0 [cdome@andromeda ~]$ /sbin/lsmod |grep snd snd_hda_intel 27936 2 snd_hda_codec 262400 1 snd_hda_intel snd_seq_dummy 11908 0 snd_seq_oss 40448 0 snd_seq_midi_event 16128 1 snd_seq_oss snd_seq 60704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 16404 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 50944 0 snd_mixer_oss 24320 2 snd_pcm_oss snd_pcm 90888 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 30088 2 snd_seq,snd_pcm snd 67624 12 snd_hda_intel,snd_hda_codec,snd_seq_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 17505 2 snd snd_page_alloc 16912 2 snd_hda_intel,snd_pcm [cdome@andromeda ~]$ cat /proc/asound/card0/codec#0 Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x1458e601 Revision Id: 0x100002 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x17 0x17] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x04] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01a19c40: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01813c41: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9933014f: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x01cb7160: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

3 8

[alsa-devel] snd_pcm_hardware_t.info question
by Ciaccia 18 Apr '07

18 Apr '07

Hi all, I am trying to fix (improve?) a driver for an embedded ARM device I bought months ago, and for some reasons some ALSA applications work fine while other ones don't... Looking at the driver I got, I noticed that the info field in the snd_pcm_hardware_t struct just defines: SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE while the info in the tutorial "writing an alsa driver" defines: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID Furthermore, the driver defines a copy callback (actually 2, one for playback and one for capture), where it copies the data from the use space to the dma buffer. Since the dma buffer was preallocated with snd_pcm_lib_preallocate_pages_for_all and is accessible from the "outside", I wonder why the original author did this... I have some questions for you gurus: - what is the difference between SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID? What is SNDRV_PCM_INFO_BLOCK_TRANSFER? - The code calls snd_pcm_lib_preallocate_pages_for_all function as following: /* allocate the pcm(DMA) memory */ ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,0, 4*128*1024, 4*128*1024); is the DMA_TYPE right for an ARM device? Is NULL a correct value for data in this case? - In case I get rid of the copy callback, how do I specify the format of the stream? How does Where can I find an example for that? Thanks a lot Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 5

[alsa-devel] [PATCH] sound: convert "sound" subdirectory to UTF-8
by John Anthony Kazos Jr. 17 Apr '07

17 Apr '07

From: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> Convert the "sound" subdirectory to UTF-8. The following files are changed. sound/oss/{trident.c,es1371.c,pas2_pcm.c} sound/oss/emu10k1/{passthrough.h,passthrough.c} sound/pci/mixart/mixart.c Signed-off-by: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> --- I can't get my mail client to send in ISO-8859-1 explicitly instead of whatever it decides I want, so the actual patch is attached in octet-stream format. The below patch is just for reference and will very likely NOT work, unless this time it picks ISO-8859-1. Also, since the patch includes both encodings, whichever encoding you select while viewing it will display one correct and one garbled line. However, the bytecodes -are- correct. diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.c -- Emu10k1 digital passthrough - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.h -- Emu10k1 digital passthrough header file - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/es1371.c linux-2.6.21-rc7.mod/sound/oss/es1371.c --- linux-2.6.21-rc7.orig/sound/oss/es1371.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/es1371.c 2007-04-17 13:55:45.000000000 -0400 @@ -100,7 +100,7 @@ * Tjeerd Mulder <tjeerd.mulder(a)fujitsu-siemens.com> * 05.01.2001 0.29 Hopefully updates will not be required anymore when Creative bumps * the CT5880 revision. - * suggested by Stephan Müller <smueller(a)chronox.de> + * suggested by Stephan MÃ¼ller <smueller(a)chronox.de> * 31.01.2001 0.30 Register/Unregister gameport * Fix SETTRIGGER non OSS API conformity * 14.07.2001 0.31 Add list of laptops needing amplifier control diff -uprN linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c --- linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c 2007-04-17 13:55:45.000000000 -0400 @@ -85,7 +85,7 @@ static int pcm_set_speed(int arg) * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. * * I cleared bit 5 of these values, since that bit controls the master - * mute flag. (Olav Wölfelschneider) + * mute flag. (Olav WÃ¶lfelschneider) * */ #if !defined NO_AUTO_FILTER_SET diff -uprN linux-2.6.21-rc7.orig/sound/oss/trident.c linux-2.6.21-rc7.mod/sound/oss/trident.c --- linux-2.6.21-rc7.orig/sound/oss/trident.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/trident.c 2007-04-17 13:55:45.000000000 -0400 @@ -18,7 +18,7 @@ * Ollie Lho <ollie(a)sis.com.tw> SiS 7018 Audio Core Support * Ching-Ling Lee <cling-li(a)ali.com.tw> ALi 5451 Audio Core Support * Matt Wu <mattwu(a)acersoftech.com.cn> ALi 5451 Audio Core Support - * Peter Wächtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support + * Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support * Muli Ben-Yehuda <mulix(a)mulix.org> * * @@ -89,7 +89,7 @@ * use set_current_state, properly release resources on failure in * trident_probe, get rid of check_region * v0.14.9c - * August 10 2001 Peter Wächtler <pwaechtler(a)loewe-komp.de> + * August 10 2001 Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> * added support for Tvia (formerly Integraphics/IGST) CyberPro5050 * this chip is often found in settop boxes (combined video+audio) * v0.14.9b diff -uprN linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c --- linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c 2007-04-17 13:55:45.000000000 -0400 @@ -472,7 +472,7 @@ static int snd_mixart_prepare(struct snd struct snd_mixart *chip = snd_pcm_substream_chip(subs); struct mixart_stream *stream = subs->runtime->private_data; - /* TODO de façon non bloquante, réappliquer les hw_params (rate, bits, codec) */ + /* TODO de faÃ§on non bloquante, rÃ©appliquer les hw_params (rate, bits, codec) */ snd_printdd("snd_mixart_prepare\n");

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[alsa-devel] Request for CS4280 Manual
by Amar singh 17 Apr '07

17 Apr '07

Hello Sir, I'm a newbie in Linux Deivce Drivers and Kernel Development. We're writing our own driver for sound card - Cirrus Logic, Sound Fusion CS4280. in Fedora Core 6 Linux flavour. For that I've been looking for the programming manual of the above said sound card, for a long time now, but didn't succeed in getting one. Since nothing being fruitful for me in this case, we're stuck up with our project, at the start. Therefore I'd kindly like to request you to plz, send me, if you can, the programming manual in the pdf/doc format or a link to the place from where we can get that document. We'd really appreciate your help for the matter. Looking ahead for positive reply from you. Thanks and Regards, Amar singh Emertxe Information Technologies (P) Ltd., Bangalore - 56. India.

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[alsa-devel] [PATCH] ASoC Kconfig description.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch makes the ASoC Kconfig descriptions a little more meaningful. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] Why do I not see patches in the emails?
by Timur Tabi 17 Apr '07

17 Apr '07

Can someone tell me why I can't see that actual patches in the posts that include them? I'm guessing it's some odd configuration option, but I went to http://mailman.alsa-project.org/mailman/options/alsa-devel and I didn't see anything applicable. Here's an example. As you can see, there is a text/plain attachment where the patch would be, but it's empty. Sender: alsa-devel-bounces(a)alsa-project.org Errors-To: alsa-devel-bounces(a)alsa-project.org Return-Path: alsa-devel-bounces(a)alsa-project.org X-OriginalArrivalTime: 16 Apr 2007 13:52:51.0309 (UTC) FILETIME=[864861D0:01C7802E] --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain Content-Transfer-Encoding: 7bit This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel --=-Uhr6lvaXHMfZKXRJXsBj-- -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] FW: Individual PCM stream control
by Nagalla, Hari 17 Apr '07

17 Apr '07

Thanks for the reply.. We followed your suggestions and we see the PCM stream controls from the user space after we have created the controls at the driver using IFACE_PCM tags.. But when I try to set the 'index' in 'snd_ctl_elem_id' to reflect the PCM substream number, it seems to always get reset to '0'. Can you please suggest, how to set the index. --- snapshot of the code -------- snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_numid(id, ctl_num); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_PCM); fprintf(stdout,"setting the mute/unmute for stream index 0x%x \n", (unsigned int) stream_id); snd_ctl_elem_id_set_index(id,(unsigned int) stream_id); if (handle == NULL && (err = snd_ctl_open(&handle, card, 0)) < 0) { fprintf(stderr,"Control %s open error: %s\n", card, snd_strerror(err)); return err; } snd_ctl_elem_info_set_id(info, id); if ((err = snd_ctl_elem_info(handle, info)) < 0) { fprintf(stderr,"Cannot find the given element from control %s\n", card); return err; } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(control, id); show_control_id(id); printf("\n"); fprintf(stdout, "Retrieving the stream index 0x%x \n",snd_ctl_elem_id_get_index(id)); ---------------------------------------------------------------- Thanks a bunch !! Hari > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, April 04, 2007 11:21 AM > To: Nagalla, Hari > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Individual PCM stream control > > At Fri, 30 Mar 2007 05:47:11 -0500, > Nagalla, Hari wrote: > > > > Thanks for the advice.. Couple of inline questions below.. > > > > > > > -----Original Message----- > > > > I am new to ALSA, so some help from experienced folks would help.. > > > > > > > > We are trying to develop ALSA support for our DSP based mixer. In > > our > > > > architecture DSP performs the mixing, sample rate conversion etc. My > > > > question is how, can does the user space applications exercise > > > > controlling individual PCM stream to the mixer? Ex: individual > > stream > > > > gain control, individual stream mute, etc.. Any user space example > > code > > > > would be useful too.. > > > > > > Most drivers create PCM volume controls with IFACE_PCM tag, and set up > > > the volume appropriately at each PCM open, or let alsa-lib configs to > > > set up the initial volumes. Usually, the control element index > > > corresponds to the PCM substream number. > > > > Which alsa library API lets us to set up the PCM stream volume? And can > > it be used dynamically when the stream is active? > > It is controlled via the normal control API. Open the snd_ctl_t > handle for the same card (e.g. "hw:0" or "default"), and change the > value of the element. However, there is no standard name for the > individual PCM volume controls, and currrently depending on the driver > implementation... > > > I could n't figure out 'index' of the control element. > > It's the index in snd_ctl_elem_id. You can access via > snd_ctl_elem_id_{get|set}_index() in alsa-lib. > > > Takashi

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Re: [alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 17 Apr '07

17 Apr '07

I applied the first hint from Jaroslav (at the end of this post the new asound.conf), but the problem seems not gone: one time each 256 seconds I get a very strong cpu load spike, that is maybe related (my guess) to the usage of plugin (maybe periodic cleaning each 256 secs?). When there is no spike, the cpu load is 0.0% with occasional 0.1% any 60-90 secs. I ran the latency test code with plughw:0 (I can't use latency.c with my virtual cards because they can't link the in and out togheter) using the blocking mode, real time, no poll. Any 256 secs, spike! (a cpu load for latency.c of 45.00%, for roughly a sec, with the cumulated cpu time of the latency.c app reported from top -d 1.0 passing from: 0:00.01 to: 0:00.64 to: 0:01.26 etc, any 256 seconds. latency.c output: ./latency -P plughw:0 -C plughw:0 -f S16_LE -c 1 -r 8000 -m 320 -M 960 -s 600 -b Playback device is plughw:0 Capture device is plughw:0 Parameters are 8000Hz, S16_LE, 1 channels, blocking mode Wanted tick time: 0us, poll mode: no Loop limit is 4800000 frames, minimum latency = 320, maximum latency = 960 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 2 <- 0 3 <- 0 4 <- 0 5 <- 0 6 <- 0 7 <- 0 8 <- 0 9 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 10 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 12 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Trying latency 320 frames, 40000.000us, 40.000000ms (25.0000Hz) Success Playback: *** frames = 4800320 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482713139 delay : 317 avail : 3 avail_max : 192 Capture: *** frames = 4800000 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482757926 delay : 1 avail : 1 avail_max : 189 Maximum read: 160 frames Maximum read latency: 20000.000us, 20.000000ms (50.0000Hz) Hardware sync Playback time = 1176471431.639255, Record time = 1176471431.639255, diff = 0 new asound.conf: pcm.c3 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 2 } } pcm.capt3 { type plug slave.pcm c3 } pcm.c4 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 3 } } pcm.capt4 { type plug slave.pcm c4 } pcm.c5 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 4 } } pcm.capt5 { type plug slave.pcm c5 } pcm.c6 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 5 } } pcm.capt6 { type plug slave.pcm c6 } pcm.c7 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 6 } } pcm.capt7 { type plug slave.pcm c7 } pcm.c8 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 7 } } pcm.capt8 { type plug slave.pcm c8 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 } On 4/13/07, Giovanni Maruzzelli <gmaruzz.lists(a)gmail.com> wrote: > Thanks a lot Jaroslav, I'll apply your suggestions and I'll report here > > On 4/13/07, Jaroslav Kysela <perex(a)suse.cz> wrote: > > On Fri, 13 Apr 2007, Giovanni Maruzzelli wrote: > > > > > Dear Alsa developers, > > > > > > I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. > > > > > > I want to use the /etc/asound.conf (.asoundrc) to split the card (that > > > has 8 in and 8 out - but I use just 6 in/out), with minimal latency. > > > > > > I mean, I want to record and play (duplex) from/to each virtual card > > > concurrently and from separate application instances. > > > > > > My app works with S16_LE, 8000, mono. > > > > > > Is the following asound.conf (.asoundrc) correct and efficient for this? > > > > Use dsnoop bindings rather than routing (ttable) for capture too, > > otherwise alsa-lib works with all channels internally. > > > > > I got spikes of cpu load (at 5-6%) on each virtual card any roughly > > > 256 seconds when using a(play|record) to record or play from/to any > > > > aplay|arecord is not very good test. Use latency test code - available in > > alsa-lib/test directory. > > > > Jaroslav > > > > ----- > > Jaroslav Kysela <perex(a)suse.cz> > > Linux Kernel Sound Maintainer > > ALSA Project, SUSE Labs > > >

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[alsa-devel] [PATCH 1/2] ASoC Samsung S3c24xx updates - audio DMA cleanup.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch cleans up the audio DMA for the Samsung S3C24xx platform. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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