July 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] non-interleaved mode
by Steve Longerbeam 11 Jul '07

11 Jul '07

Hi, The driver for the sound engine I'm working on requires NONINTERLEAVED. Will ALSA core give my driver a full period of left-only samples, followed by a full period of right-only samples? Or are left/right samples split within a single period (ie. first half of period is left samples, second half is right samples)? Thanks, Steve

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Re: [alsa-devel] Errors in application building
by Paul Kavan 11 Jul '07

11 Jul '07

On 7/3/07, Lee Revell <rlrevell(a)joe-job.com> wrote: > > Use gdb, strace is not useful for this > > On 7/3/07, Paul Kavan <pkavan(a)gmail.com> wrote: > > Hello all: > > > > Just tried to run with the warnings: > > > > > > > > > > > > > grhuser@pcL7:$ arm-linux-gcc -lasound -Wall -o grh_audio grh_audio.c > > > grh_audio.c: In function 'grh_capture': > > > grh_audio.c:112: warning: pointer targets in passing argument 3 of > > 'snd_pcm_hw_params_set_rate_near' differ in signedness > > > grh_audio.c: In function 'grh_playback': > > > grh_audio.c:207: warning: pointer targets in passing argument 3 of > > 'snd_pcm_hw_params_set_rate_near' differ in signedness > > > > > > Which comes from the code: > > > > > > 112: if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, > > hw_params, &sample_rate, 0)) < 0) > > > > > > > > > 207: if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, > > hw_params, &sample_rate, 0)) < 0) > > > > However, I get a seg fault when I ran it. The strace of this is: > > > > /usr/sbin $ strace grh_audio > > execve("/usr/sbin/grh_audio", ["grh_audio"], [/* 7 vars */]) = 0 > > mmap2(NULL, 20, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) > = > > 0x40005000 > > stat("/etc/ld.so.cache", {st_mode=S_IFREG|0644, st_size=1150, ...}) = 0 > > open("/etc/ld.so.cache", O_RDONLY) = 3 > > mmap2(NULL, 1150, PROT_READ, MAP_SHARED, 3, 0) = 0x40006000 > > close(3) = 0 > > open("/usr/lib/libasound.so.2", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0755, st_size=762072, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0x\367\1\000"..., > > 4096) = 4096 > > mmap2(NULL, 794624, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x4000e000 > > mmap2(0x4000e000, 745564, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) > > = 0x4000e000 > > mmap2(0x400cc000, 15508, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0xb6) = 0x400cc000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libgcc_s.so.1", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=44167, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0X\25\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 65536, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x400d0000 > > mmap2(0x400d0000, 28800, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x400d0000 > > mmap2(0x400df000, 548, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0x7) > > = 0x400df000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0000\253\0"..., > > 4096) = 4096 > > mmap2(NULL, 360448, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x400e0000 > > mmap2(0x400e0000, 302380, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) > > = 0x400e0000 > > mmap2(0x40132000, 5172, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0x4a) = 0x40132000 > > mmap2(0x40134000, 16020, PROT_READ|PROT_WRITE, > > MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x40130 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libm.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=45812, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0t\22\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 81920, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x40138000 > > mmap2(0x40138000, 41428, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x40138000 > > mmap2(0x4014a000, 4100, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0xa) > > = 0x4014a000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libdl.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=8900, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0(\10\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 40960, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x4014c000 > > mmap2(0x4014c000, 5760, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x4014c000 > > mmap2(0x40155000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0x1) > > = 0x40155000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libpthread.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=63852, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0D<\0\0004"..., > > 4096) = 4096 > > mmap2(NULL, 106496, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x40156000 > > mmap2(0x40156000, 40952, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x40156000 > > mmap2(0x40168000, 21732, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0xa) = 0x40168000 > > mmap2(0x4016e000, 5788, PROT_READ|PROT_WRITE, > > MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x4016e0 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libgcc_s.so.1", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=44167, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > munmap(0x40006000, 1150) = 0 > > stat("/lib/ld-uClibc.so.0", {st_mode=S_IFREG|0755, st_size=21096, ...}) > = 0 > > mprotect(0x40132000, 4096, PROT_READ) = 0 > > mprotect(0x4014a000, 4096, PROT_READ) = 0 > > mprotect(0x40155000, 4096, PROT_READ) = 0 > > mprotect(0x40168000, 4096, PROT_READ) = 0 > > mprotect(0x4000c000, 4096, PROT_READ) = 0 > > ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, {B115200 opost isig icanon echo > > ...}) = 0 > > ioctl(1, SNDCTL_TMR_TIMEBASE or TCGETS, {B115200 opost isig icanon echo > > ...}) = 0 > > getpid() = 756 > > getrlimit(RLIMIT_STACK, {rlim_cur=8192*1024, rlim_max=RLIM_INFINITY}) = > 0 > > setrlimit(RLIMIT_STACK, {rlim_cur=2040*1024, rlim_max=RLIM_INFINITY}) = > 0 > > rt_sigaction(SIGRTMIN, {0x4015fd70, [], 0x4000000 /* SA_??? */}, NULL, > 8) = > > 0 > > rt_sigaction(SIGRT_1, {0x4015fc98, [RTMIN], 0x4000000 /* SA_??? */}, > NULL, > > 8) = 0 > > rt_sigaction(SIGRT_2, {0x4015f2d0, [], 0x4000000 /* SA_??? */}, NULL, 8) > = 0 > > rt_sigprocmask(SIG_BLOCK, [RTMIN], NULL, 8) = 0 > > rt_sigprocmask(SIG_UNBLOCK, [RT_1], NULL, 8) = 0 > > --- SIGSEGV (Segmentation fault) @ 0 (0) --- > > +++ killed by SIGSEGV +++ > > Process 756 detached > > > > > > I am still trying to locate why those warnings are popping up, but as > that > > part of the code is not called immediately, I am wondering why the seg > fault > > is occurring. Any clues? Lee: Have attempted to run this with gdb. However, I get: /usr/sbin $ gdb grh_audio dlopen failed on 'libthread_db.so.1' - File not found GDB will not be able to debug pthreads. GNU gdb 6.5 Copyright (C) 2006 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "arm-linux-uclibc"...(no debugging symbols found) (gdb) Trying to determine why...will let you know asap what I find. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] Query regarding sampling rates etc
by Pharaoh . 11 Jul '07

11 Jul '07

Hi, This is a very vague question, just thought of asking it after hours of debugging. I am testing my alsa driver, written for omap 730/850 based board. I am testing it by using aplay. Despite checking clocks, sampling rates etc many times and after doing many combinations thereof, I cant hear anything more than a noise which changes and becomes more like the original .wav file as I change various params like sampling rates/clocks etc. It becomes close to the actual thing, but not better. Any idea what are the params I should be tweaking? I know this is h/w dependent. -pharaoh

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Re: [alsa-devel] RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 11 Jul '07

11 Jul '07

> You'll find attached an equivalent fix that checks alt-settings > instead of device_setup param: Can you try it Hakan ? Ok, patch tested now. Seem to work. /Håkan

1 0

[alsa-devel] Alsa lib configure asoundlib.h
by Fausto Carvalho Marques SIlva 11 Jul '07

11 Jul '07

In salsa library(this library really saved my life, from am alsa-lib of 600K to 45K!!!), developed by Takashi Iwai, the creation of asoundlib.h is dynamic and based on the configure options ( --disable-mixer will remove mixer.h from asoundlib.h). Can alsa-lib do the same??I think it is a good "cleanup" way of doing client header files. Example from configure.ac from salsa-0.0.6: " echo "Creating asoundlib.h..." cp "$srcdir"/src/asoundlib-head.h src/asoundlib.h test "$mixer" = "yes" && echo "#include \"mixer.h\"" >> src/asoundlib.h test "$rawmidi" = "yes" && echo "#include \"rawmidi.h\"" >> src/asoundlib.h test "$hwdep" = "yes" && echo "#include \"hwdep.h\"" >> src/asoundlib.h test "$timer" = "yes" && echo "#include \"timer.h\"" >> src/asoundlib.h test "$sndconf" = "yes" && echo "#include \"conf.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seq_event.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seq.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seqmid.h\"" >> src/asoundlib.h cat "$srcdir"/src/asoundlib-tail.h >> src/asoundlib.h " Fausto Carvalho Marques Silva

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[alsa-devel] Mercurial repos
by Adrian McMenamin 11 Jul '07

11 Jul '07

Are the browsable repos broken - they don't seem to have been updated for days, though I've been told my patch has been applied.

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Re: [alsa-devel] Writing a new plugin for ALSA
by Nobin Mathew 11 Jul '07

11 Jul '07

On 7/10/07, Takashi Iwai <tiwai(a)suse.de> wrote: > At Tue, 10 Jul 2007 17:45:03 +0530, > Sudhanshu Saxena wrote: > > > > Hi Takashi, > > > > I am working on this issue along with Nobin, for multiple multiple streams (at > > different rates) playback based on a priority. > > > > I attempted suspend and resume returning -ESTRPIPE, but it has two issues > > 1) All applications may need a change in Suspend logic specific to wait for higher > > priority stream to run to completion. > > Basically every app is _required_ to handle of suspend/resume > properly. > > > 2) I have to still find out a mechanism to reset the hardware settings are reconfigure > > it for second stream. > > snd_pcm_hw_reset has an access to reset using IOCTL "SNDRV_PCM_IOCTL_RESET" , I hope > > that can help, I am not > > very sure on this. > > Well, I don't think the plugin itself requires the reset. > > > As mentioned by Nobin we want to skip any upsampling caused when we use the dmix > > (rate) plugin. > > I was trying to find the way out, with a plugin with following changes: > > 1) All streams will open pcm with a priority. (Pre decided for the system) > > 2) Plug-in will have shared memory to hold the priority list of all applications which > > are accessing sound. > > 3) At any instance only one stream will be active. > > 4) All applications will hold the context of HW params and SW params (accessible > > through apis plugin will support). > > 5) If an application with a higher priority calls snd_pcm_open shared memory with the > > priority list will be updated and > > when the currently active stream attempts to re-write the data it will check for > > priority list. This will cause saving of params, drain of stream and reset of hw as > > mentioned using above IOCTL and the active stream would be changed to > > currently highest in the priority list. > > 6) The higher priority stream can then start writing data, on completion of access > > similar sequence of drain, hw reset will be caused and the active stream will be > > changed to next in the priority list. > > 7) On resume of lower priority stream the hw params will be set again. > > The question is how the application decides which stream to use. > If it can open a PCM with a different name (to snd_pcm_open call), and > if you have three individual PCM devices, the whole switching can > be done inside the kernel driver. That is, when an SCO stream is > opened during audio stream(s) is running, the driver triggers SUSPEND > mode for the audio stream. Then the app will stop and restart the > stream by itself. This means you don't need any change in user-space > side but needs some additions in the driver side. We thought about implementing this in the kernel driver, but after some investigation, we found that we are serializing the driver so much by providing three interfaces (audio, sco and Nav). say pcmC0D0p/c, pcmC0D1p/c and pcmC0D2p/c. Consider the case of two apps audio and sco. We have two interfaces for this pcmC0D0p/c and pcmC0D1p/c priority is audio < sco so interface priority is pcmC0D0p/c < pcmC0D1p/c. Assume that audio is already opened. When we open sco interface audio interface can be in any state, it i can be in paused state or it is about to close etc. So we will have so many states of application in the kernel driver. we need to maintain all these states of application in the kernel driver. Say tomorrow one more application came in to picture, then this complexity will increase. Locking of states of streams is also required in this case. correct me if I am going wrong > > BTW, any reason to drop Cc to alsa-devel ML? Such an open discussion > is very important for opensource development. I think Sudhanshu missed it, I am copying to alsa-devel > > > Takashi > thanks Nobin mathew

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[alsa-devel] [PATCH] hda-codec - Add VIA HDA to si3054
by Claudio Matsuoka 10 Jul '07

10 Jul '07

The attached patch is needed to proper recognition of the si3054 modem in the Clevo m540 laptop. Thanks, Claudio

1 0

Re: [alsa-devel] default setting not fully working with new ALSA device
by Tobin Davis 10 Jul '07

10 Jul '07

Try this patch. It adds support for both ALC883 and HDMI ports on your system. Summary: HDA-Intel - Add support for MSI K9AGM2-FIH motherboard This patch adds support for the MSI K9AGM2-FIH on-board audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> On Thu, 2007-07-05 at 18:34 +0200, Peter Matthias wrote: > The Board is a MSI K9AGM2-FIH, Sockel AM2, ATI RS690G, mATX, PCIe > > The script gave following output: > "Your ALSA information is located at http://pastebin.ca/604529" > > Other drivers are statically linked in my kernel. > > Thank you for your work! In case of any questions, contact me. > > Peter > > > Am Mittwoch, 4. Juli 2007 schrieben Sie: > > Could you post the output from > > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh ? This > > script will give us more detailed information on what sound chip it is. > > We will also need to know the brand of board, for the driver file. > > > > Thanks, > > > > Tobin > > > > On Wed, 2007-07-04 at 16:48 +0200, Peter Matthias wrote: > > > Hello, > > > > > > I have got a 690G board with following lspic -nv output: > > > > > > 01:05.2 0403: 1002:7919 > > > Subsystem: 1002:7919 > > > Flags: bus master, fast devsel, latency 64, IRQ 19 > > > Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] > > > Capabilities: [50] Power Management version 2 > > > Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 > > > Enable- > > > > > > With the default setting, microphone does not work. With module > > > parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works > > > well. Please add to default setting. It was written in Linux Kernel > > > documentation to report this issue here. > > > > > > Thank you, > > > Peter > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis <tdavis(a)dsl-only.net>

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Re: [alsa-devel] Intel hda ICH7 not worked anymore after using wine (SOLVED!)
by Xavi de Blas 10 Jul '07

10 Jul '07

It worked!!! Thanks a lot I thought my hardware was broken! 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>: > > Patch needs to be applied as follows: > > cd alsa-driver-1.0.14/alsa-kernel > patch -p1 < LenovoX60S.patch > cd .. > > Then you can rerun configure && make && make install. > > Tobin > > On Mon, 2007-07-09 at 17:25 +0200, Xavi de Blas wrote: > > 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>:> Sorry for missing this thread. I just woke up.>> I looked at your first pastebin output, and the actual audio codec shows> up as 0x51d41981, when it should be 0x11d41981. This is why the Analog> driver patch isn't detecting it.>> Can you try this patch against alsa-driver-1.0.14? It should detect> this codec. Still not sure why it is misconfigured, unless there is a> bios issue.> > I'm afraid, probably when i tried the Linuxant codecs for the modem it broke thebios. Not sure, but sound worked, and continued working some rebootslater. But when i tried a few things with wine and rebooted, soundnever worked again > When i aply the patch (probably i don't do it good) there are problems: > > xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14$ catLenovoX60S.patch | patch -p1patching file pci/hda/patch_analog.cHunk #1 FAILED at 3402.1 out of 1 hunk FAILED -- saving rejects to file pci/hda/patch_analog.c.rej > > xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14/pci/hda$ catpatch_analog.c.rej*************** struct hda_codec_preset snd_hda_preset_a*** 3402,3407 **** { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },--- 3402,3408 ---- { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },+ { .id = 0x51d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, > > The patch is ok? > Thanks > (Tobin sorry for sending messge two times) > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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[alsa-devel] Writing a new plugin for ALSA
by Nobin Mathew 10 Jul '07

10 Jul '07

Hi, I am trying to write a new ALSA plugin, like dmix. My intend is to give priority to streams, i will not do up sample and down sample as we do in the case of dmix. If we do up sample and down sample then I think it will take lot of cpu. In my case I have three streams audio, bluetooth sco and Navigation sound data. priority is audio < sco < Nav audio rate is 44100hz, sco rate is 8000hz when sco is up audio should pause playback and audio should continue when sco is finished. I dont want to do software mixing. Thats why I am trying to write this new plugin. Assume that audio is active first, then sco comes to active state. So what i will do is in the library i will put audio app in sleep(), and re-configure the pcm for sco. Once sco is finished i will re-configure pcm for audio again like wise. so my question is there any api in the alsa-lib to take a copy of pcm structures, so when audio comes back gain i can use these saved pcm conf to that. Is there any api in the alsa-lib to reset the pcm structure to initialized state, means state just after open. Thanks Nobin Mathew

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[alsa-devel] [PATCH] snd-usb-audio: Add another QuickCam ID
by Ritesh Raj Sarraf 10 Jul '07

10 Jul '07

Hi Clemens, Can you push this for 2.6.23 ? Or is there anything else missing in the patch ? Thanks, Ritesh =================================================== On Friday 08 June 2007, Clemens Ladisch wrote: > Ritesh Raj Sarraf wrote: > > dell_xpsm1210_quickcam.patch > > Please provide a Signed-off-by line. > > I guess this patch is for the integrated webcam of the Dell XPS M1210? > > Hi Clemens, I've regenerate the patch properly this time. And yes, this patch is for the integrated webcam shipped with M1210. PS: Request you to CC me if there are any comments. I'm not subscribed to the list. Thanks, Ritesh -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention." -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention."

1 0

[alsa-devel] ALC882 on Abit AW9D-MAX
by Dan Aloni 10 Jul '07

10 Jul '07

Hello, Yesterday I posted this bug in the bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3013 I have received no reponse yet. I'd like to cooperate on solving this. Thanks, -- Dan Aloni XIV LTD, http://www.xivstorage.com da-x (at) monatomic.org, dan (at) xiv.co.il

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[alsa-devel] Unable to find an usable access for 'default'
by Timur Tabi 10 Jul '07

10 Jul '07

I'm developing a new ASOC driver, and I get this error message: ALSA lib pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' Can someone tell me what this means? FYI, I'm in the early stages of testing my driver, and before I started I knew nothing about ALSA. This is on an embedded system, so I don't know if my ALSA installation is correct. Am I missing some kind of configuration file? -- Timur Tabi Linux Kernel Developer @ Freescale

2 7

[alsa-devel] Intel hda ICH7 not worked anymore after using wine
by Xavi de Blas 09 Jul '07

09 Jul '07

Hello, I installed Wine on my Ubuntu Feisty on IBM-lenovo X60s, and after reboot, sound never worked again. I tried to activate and to hide modem on the bios. I tried to stop machine, take off battery, wait 10s, put battery again, and boot without the power cable connected... and i tried lots of things from ubuntuforums. I reinstalled feisty and sound doesn't work. I used alsa-info.sh script, here is the result: http://pastebin.ca/610317 Should the new kernel 2.6.22 help me? http://kernelnewbies.org/Linux_2_6_22#head-0d133cda430e5d18f1c72dd3cfaa6a92… Thanks

3 12

Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 09 Jul '07

09 Jul '07

At Mon, 25 Jun 2007 15:16:41 -0500, Matt Mullins wrote: > > On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > Yes, single_cmd mode is used always. > > That's bad. > > I believe we have established that. What can I do about it? *Can* I > do anything about it? What functions should I start debugging? Has > this happened with other codecs before? What documentation will I > need or want other than the Intel HDA and STAC9205 specifications? Similar problems have been seen in some laptops, too. I vaguely remember that they need some ACPI-related boot options to fix it somehow. I think it's likely irrelevant with the codec chip, but rather a problem of the controller side. > I'm definitely willing to help work on this, as not many people have > this hardware. I'm just new to the sound system, so I'm still trying > to figure out what-calls-what. Takashi

3 10

[alsa-devel] stac9200 speaker/hp not detected
by Brian Hinz 09 Jul '07

09 Jul '07

Hi, I'm trying to get the sound working on my Gateway S-7110M (a.k.a. NX100X?). Using the ini files from the windows driver, I created the following patch (applied to both the hg tree, and 1.0.14): --- a/alsa-kernel/pci/hda/patch_sigmatel.c 2007-07-03 21:59:21.000000000 -0400 +++ b/alsa-kernel/pci/hda/patch_sigmatel.c 2007-06-04 03:28:54.000000000 -0400 @@ -39,6 +39,7 @@ enum { STAC_REF, + STAC_M4, STAC_9200_MODELS }; @@ -453,12 +452,19 @@ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; +static unsigned int stac9200_M4_pin_configs[8] = { + 0x400000fe, 0x00451030, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; + static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, + [STAC_M4] = stac9200_M4_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_REF] = "ref", + [STAC_M4] = "m4", }; static struct snd_pci_quirk stac9200_cfg_tbl[] = { @@ -494,6 +486,9 @@ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f5, "Dell Inspiron 1501", STAC_REF), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_M4), + /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF), however it still does not work. I can see in dmesg that the speaker and headphones are not detected, but I can't seem to figure out why: [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'm4' is selected [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x0, fmic=0x10, line=0x0, fline=0x0, cd=0x0, aux=0x0 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) This laptop has only an internal speaker, a headphone out, and a mic in (unless there are other outs enabled when the dock is connected). The windows INI seems to be setting some gpio bits but I don't understand how to map the ini syntax into the alsa driver (or if it needs to be...) [Models] HDAUDIO\FUNC_01&VEN_8384&DEV_7690=this Default=this [CopyFiles.32] stacgui.cpl = 0x25 stlang.dll = 0x25 stacsv.exe = 0x25 [CopyFiles.64] stacui64.cpl = 0x25 stlang64.dll = 0x25 stacsv64.exe = 0x25 [LaunchProgramsAfter.32] STacSV -r [LaunchProgramsAfter.64] STacSV64 -r [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.32] [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.64] [DelRegKeys] HKR\Settings=Yes HKCU\Software\Sigmatel\Settings=Yes HKLM\Software\Sigmatel\Presets=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel C-Major Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel High Definition Audio=Yes [HKLM\Software\Sigmatel\GlobalState] [HKR] CodecName = "STAC9200" [HKR\Settings] TraceFlags = dword: 0x89B3 CPL_SaveExtraPower = hex: 0 CPL_UserControlsFixedDigitalOutRate = hex: 0 InitVerbs = hex: 0x03, 0xa0, 0x43, 0x01 9200MixerCompensation1PcmMode = hex: 2 9200MixerCompensation2PcmMode = hex: 3 CodecAccessAttempts = dword:00000352 [HKR\Settings\GPIO] 03 = hex: 0x85 AntiPopGpo = hex: 3,2 [HKR\Settings\pin] [HKR\Settings\Pin\08] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\09] CfgDflt = dword: 0x00451030 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400100F0 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x90110010 Action = hex: 81,09 [HKR\Settings\Pin\0F] CfgDflt = dword: 0x400100F1 [HKR\Settings\Pin\10] CfgDflt = dword: 0x02A1902E [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000F2 [HKR\Settings\Pin\12] CfgDflt = dword: 0x500000F3 [HKR\Settings\Connselector] [HKR\settings\filter\Speaker] UniqueID = hex: 0 LegacyIoctlHere = hex: 1 Pins = hex: 0E RenderConverter = hex: 2 Associations = hex: 1 DigitalVolumeControl = hex: 0 EnableSetFormatSupport = hex: 1 EnableHwSpeakerConfigurationSupport = hex: 0 EnabledPcmRenderBitsAndRates = dword: 0x000FFFFE MasterControlsIndependentMixerInputs = hex: 0 EnableVendorPcBeep = hex: 1 PcBeepVolume = dword: 0 PcBeepMute = hex: 1 EnableInputMonitor = hex: 1 RecMonitorMute = hex: 1 EnabledNonPcmBitsAndRates = dword: 0x20040 BiosInterfaceAddress = dword: 0x000DF7E0 [HKR\settings\filter\SpdifOut] UniqueID = hex: 1 LegacyIoctlHere = hex: 0 Pins = hex: 09 SpdifOutConverter = hex: 5 SpdifMaster = hex: Associations = hex: 3 EnabledPcmRenderSpdifBitsAndRates = dword: 0x000FFFFF DisableSPDIFControl = hex: 0 [HKR\Settings\filter\MuxedIn] UniqueID = hex: 2 LegacyIoctlHere = hex: 0 Pins = hex: 10 CaptureConverter = hex: 3 EnableCompositeMuxAndAdcCtrl = hex: 0 Associations = hex: 2 MicBoost = hex: 1 DisableMicBoost = hex: 0 EnabledPcmCaptureBitsAndRates = dword: 0x000FFFFF MicInRecVolume = dword: 0x000a8000 [HKR\Settings\filter\MuxedIn\MaxLvlLimiters] MicBoostLevel = dword: 0x000c0000 [HKR\Settings\filter\MuxedIn\TopoNames] [HKLM\software\SigmaTel\Apo\GFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\software\SigmaTel\Apo\GFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\SysEq] Load = dword: 1 IdEffect = dword: 101 IdEffectType = dword: 1 Order = dword: 1100 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq] NumBands = dword:12 RegVersion = dword: 1 Bypass = dword: 0 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band0] idxBand = dword: 0 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 1 GroupId = dword: 0 Freq =dword: 150 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band1] idxBand = dword: 1 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 300 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band2] idxBand = dword: 2 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 500 GainDbx1000 = dword: -2000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band3] idxBand = dword: 3 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 750 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band4] idxBand = dword: 4 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 1000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band5] idxBand = dword: 5 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 2000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band6] idxBand = dword: 6 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 8000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band7] idxBand = dword: 7 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 12000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band8] idxBand = dword: 8 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 15000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band9] idxBand = dword: 9 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band10] idxBand = dword: 10 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band11] idxBand = dword: 11 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 2 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Compressor] Load = dword: 1 IdEffect = dword: 5 Order = dword: 1150 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Compressor] CompRatio= dword: 200 CompAttack= dword: 9910 CompRelease= dword: 9910 CompThreshDb= dword: -600 MakeupGainDb= dword: 200 GateRatio= dword: 600 GateAttack= dword: 9990 GateRelease= dword: 9990 GateThreshDb= dword: -9000 Flags= dword: 8 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Limiter] Load = dword: 1 IdEffect = dword: 6 Order = dword: 1155 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Limiter] CompRatio= dword: 3700 CompAttack= dword: 9900 CompRelease= dword: 9900 CompThreshDb= dword: 0 MakeupGainDb= dword: 0 GateRatio= dword: 100 GateAttack= dword: 9000 GateRelease= dword: 9000 GateThreshDb= dword: -9000 Flags= dword: 10 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions] Version=dword:1 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions\JackPresenseD] Action1="if(JackState(13) == 1,(BypassSystemEq(1)),(BypassSystemEq(0)))" Action2="if(JackState(13) == 1,(BypassSystemComp(1)),(BypassSystemComp(0)))" [HKLM\SOFTWARE\Sigmatel\STacGUI] Config_Tab = dword: 0x8 Config_Spk = dword: 0x0 Config_Adv = dword: 0x01 [HKLM\SOFTWARE\Sigmatel\STacGUI\XML] Nconfigs = dword: 0x0 here's the output of /proc/asound/card0/codec#0: Codec: SigmaTel STAC9200 Address: 0 Vendor Id: 0x83847690 Subsystem Id: 0x100 Revision Id: 0x102201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x0a Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x08 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x07 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x08 0x0a Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x400000fe: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x00451030: [Jack] SPDIF Out at Ext N/A Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x0a Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0d 0x0d] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x80 0x80] Connection: 1 0x07 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x10* 0x0f 0x0e 0x0d 0x12 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x400100f0: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90110010: [Fixed] Speaker at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x400100f1: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0b Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a1902e: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0b Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x500000f2: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x13 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500000f3: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x13 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Thanks for any assistance! -brian

2 7

[alsa-devel] Question about write asoc driver
by Roy Huang 09 Jul '07

09 Jul '07

Hi guys, I am write a codec driver under asoc frame work for ad1980 by study the sample driver wm9712.c. I find there are some control item likes "out3 switch" to control 16h:15. There are also some widget item likes "out 3 PGA" to control 24:5. How can I determine a register bit should be a control item or widget item? Thanks ahead. -Roy

2 1

Re: [alsa-devel] [Pkg-alsa-devel] [PATCH] amixer and special 'front...' channels
by Elimar Riesebieter 09 Jul '07

09 Jul '07

This isn't Ubuntu/Debian specific and should be applied upstream? On Sat, 07 Jul 2007 the mental interface of David Decotigny told: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Hi, > > Imagine you have this kind of control: > > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > ... and you want to set it to 'Front Mic': > > shell> amixer sset 'Input Source' 'Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > This does not work ! Because amixer will think that 'Front' is a > modifier for 'Mic'. > > Attached is a patch against alsa-utils-1.0.13-1ubuntu5 that solves this. > With this patch, one can bypass the modifier parsing: one simply escapes > the parameter with one or more backslash(es), and that's all: > > shell> amixer sset 'Input Source' '\Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Front Mic' > > What this patch also does, is that it adds correct identification of > items. With the original version, given the following enumeration: > foo foobar > Then the parameter 'foobar' will be identified as 'foo', not 'foobar'. > The patch should solve this potential problem (not tested though). > > Bye, > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.6 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFGj/X0ld7vhusVrCERAjIZAJ9w9GLD+QBwDCaTs32hpW6i9nVMfgCfdS5w > jhipZ2jm87fq4zx4f5Q9Vsc= > =M2KP > -----END PGP SIGNATURE----- > diff -ru alsa-utils-1.0.13/amixer/amixer.1 alsa-utils-1.0.13-mine/amixer/amixer.1 > --- alsa-utils-1.0.13/amixer/amixer.1 2006-09-29 04:53:26.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.1 2007-07-07 13:09:04.000000000 -0700 > @@ -44,11 +44,14 @@ > The parameters \fIcap, nocap, mute, unmute, toggle\fP are used to > change capture (recording) and muting for the group specified. > > -The optional modifiers can be put as extra parameters to specify > -the stream direction or channels to apply. > -The modifiers \fIplayback\fP and \fIcapture\fP specify the stream, > -and the modifiers \fIfront, rear, center, woofer\fP are used to specify > -channels to be changed. > +The optional modifiers can be put as extra parameters to specify the > +stream direction or channels to apply. The modifiers \fIplayback\fP > +and \fIcapture\fP specify the stream, and the modifiers \fIfront, > +rear, center, woofer\fP are used to specify channels to be changed. If > +there is a confusion between the parameter and these modifiers, then > +any character in the parameter may be escaped with a backslash to > +disambiguate: '\\Front Mic' will be understood as "parameter 'Front > +Mic'", and not as "modifier 'Front' applied to 'Mic' parameter". > > A simple mixer control must be specified. Only one device can be controlled > at a time. > diff -ru alsa-utils-1.0.13/amixer/amixer.c alsa-utils-1.0.13-mine/amixer/amixer.c > --- alsa-utils-1.0.13/amixer/amixer.c 2007-07-07 13:03:39.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.c 2007-07-07 12:59:19.000000000 -0700 > @@ -1333,6 +1333,16 @@ > return def; > } > > +/** Transform '\a\b c \de' into 'ab c de' */ > +static void unescape(char * arg) > +{ > + char * c; > + for (c = arg ; arg && *arg ; arg ++) > + if (*arg != '\\') > + *(c++) = *arg; > + *c = '\0'; > +} > + > static int get_enum_item_index(snd_mixer_elem_t *elem, char **ptrp) > { > char *ptr = *ptrp; > @@ -1347,7 +1357,7 @@ > if (snd_mixer_selem_get_enum_item_name(elem, i, sizeof(name)-1, name) < 0) > continue; > len = strlen(name); > - if (! strncmp(name, ptr, len)) { > + if (!strncmp(name, ptr, len) && (ptr[len] == '\0')) { > if (! ptr[len] || ptr[len] == ',' || ptr[len] == '\n') { > ptr += len; > *ptrp = ptr; > @@ -1428,6 +1438,7 @@ > dir = dir_mask(&ptr, dir); > if (*ptr == '\0') > continue; > + unescape(ptr); > multi = (strchr(ptr, ',') != NULL); > optr = ptr; > for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { > _______________________________________________ > Pkg-alsa-devel mailing list > Pkg-alsa-devel(a)lists.alioth.debian.org > http://lists.alioth.debian.org/mailman/listinfo/pkg-alsa-devel -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

2 1

[alsa-devel] [PATCH] usb-audio: another Logitech camera/microphone ID match
by Daniel Drake 09 Jul '07

09 Jul '07

From: George Shapovalov <george(a)gentoo.org> Another case of ALSA ticket #3040: we need an explicit USB ID match for another Logitech camera/microphone device. Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -57,6 +57,15 @@ USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, + .idProduct = 0x08ae, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL

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[alsa-devel] M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 09 Jul '07

09 Jul '07

Hi all ! I don't know if this is already known (?), but the M-Audio Audiophile USB seem to be possible to run in native S16_LE mode if initialized with device_setup=0x01 (possibly with 0x11 also, have not tested that). This is at least the case with my rather new box (bought a couple of weeks ago). Patch for usb/usbaudio.c (alsa-driver-1.0.14): static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) { switch (chip->usb_id) { case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ if (fp->endpoint & USB_DIR_IN) return 1; break; case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + // Is small endian if setup with 0x01 ! + if (device_setup[chip->index] != 0x01) return 1; break; } return 0; } /Håkan

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[alsa-devel] [PATCH] ALSA AICA sound on SEGA Dreamcast - fix behaviour in poor resource conditions
by Adrian McMenamin 09 Jul '07

09 Jul '07

Patch against code in alsa-kernel - tested with patched up 2.6.21-rc7 and working well Submitted by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff -ruN aicaold/aica.c aicanew/aica.c --- aicaold/aica.c 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.c 2007-07-05 23:02:02.000000000 +0100 @@ -65,10 +65,6 @@ /* Use workqueue */ -static struct spu_work_holder { - struct work_struct spu_dma_work; - void *sspointer; -} spu_working; static struct workqueue_struct *aica_queue; @@ -252,18 +248,15 @@ static void run_spu_dma(struct work_struct *work) { int buffer_size; - struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_card_aica *dreamcastcard; - struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); - substream = holder-> sspointer; - dreamcastcard = substream->pcm->private_data; - runtime = substream->runtime; + dreamcastcard = container_of(work, struct snd_card_aica, spu_dma_work); + runtime = dreamcastcard->substream->runtime; if (unlikely(dreamcastcard->dma_check == 0)) { buffer_size = frames_to_bytes(runtime, runtime->buffer_size); if (runtime->channels > 1) dreamcastcard->channel->flags |= 0x01; - aica_dma_transfer(runtime->channels, buffer_size, substream); + aica_dma_transfer(runtime->channels, buffer_size, dreamcastcard->substream); startup_aica(dreamcastcard); dreamcastcard->clicks = buffer_size / (AICA_PERIOD_SIZE * runtime->channels); @@ -271,7 +264,7 @@ } else { aica_dma_transfer(runtime->channels, AICA_PERIOD_SIZE * runtime->channels, - substream); + dreamcastcard->substream); snd_pcm_period_elapsed(dreamcastcard->substream); dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) @@ -307,7 +300,8 @@ dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(spu_working.spu_dma_work)); + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + } static void spu_begin_dma(struct snd_pcm_substream *substream) @@ -317,10 +311,8 @@ struct snd_pcm_runtime *runtime; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; - /* Use queue to do the heavy lifting */ - spu_working.sspointer = substream; - INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); - queue_work(aica_queue, &(spu_working.spu_dma_work)); + //get the queue to do the work + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); @@ -366,7 +358,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,17 +396,10 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; - case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); - break; default: return -EINVAL; } @@ -610,6 +597,8 @@ strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strcpy(dreamcastcard->card->longname, "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + //start the worker thread + INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) diff -ruN aicaold/aica.h aicanew/aica.h --- aicaold/aica.h 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.h 2007-07-05 23:02:02.000000000 +0100 @@ -69,6 +69,7 @@ }; struct snd_card_aica { + struct work_struct spu_dma_work; struct snd_card *card; struct aica_channel *channel; struct snd_pcm_substream *substream;

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[alsa-devel] [HDA Intel/STAC92xx] mic not working (retasking?)...
by Daniel J Blueman 08 Jul '07

08 Jul '07

With both 2.6.20 and 2.6.22 kernels on Ubuntu 7.04 on my Sony Vaio SZ240, I'm unable to get my mic connector working at any cost. Well, almost...reading from /dev/dsp, I do see a minute change from 127 for 126 or 128 if I tap my external mic. This would imply the jack-retasking feature of this darn STAC chipset, right? How can I change the jack retasking with the Intel HDA drivers or get further insight into this problem? Thanks in advance, Daniel --- [1] $ lspci -s0:1b.0 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ amixer info Card default 'Intel'/'HDA Intel at 0xde400000 irq 22' Mixer name : 'SigmaTel CXD9872RD/K' Components : 'HDA:83847661 HDA:14f12bfa' Controls : 6 Simple ctrls : 5 $ amixer controls numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' numid=6,iface=MIXER,name='PCM Playback Volume' numid=5,iface=MIXER,name='Capture Source' numid=4,iface=MIXER,name='Capture Switch' numid=3,iface=MIXER,name='Capture Volume' $ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'PCM',0 Simple mixer control 'Line',0 Simple mixer control 'Mic',0 Simple mixer control 'Capture',0 $ amixer contents numid=2,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=127,step=0 : values=66,66 numid=6,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=230,230 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=5,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Line' ; Item #1 'Mic' : values=1 numid=4,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=15,step=0 : values=15,15 | dBscale-min=0.00dB,step=1.50dB,mute=0 -- Daniel J Blueman

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[alsa-devel] Remove dynamic configuration from alsa-lib
by Fausto Carvalho Marques SIlva 08 Jul '07

08 Jul '07

I have a hard job...to remove dynamic configuration from alsa-lib (conf.c,confmisc.c). I found it difficult because snd_conf_* functions are spread all over *.c files. Is there am easy way of doing it ? Does anyone have some "guidelines" of doing it? Any help is wellcome... Fausto Carvalho Marques Silva

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[alsa-devel] hda-intel on macbook: unknown model for ACL882
by Steffen Schulz 07 Jul '07

07 Jul '07

Hi all, Since I didn't get much feedback on alsa-users, I figured I'd place a pointer here, too. The revision 3 macbook pro(santa-rosa) ships with a ALC882. Loading hda_intel with auto-detection results in very VERY quiet sound out of the headphones if 'front' is set to 100%, mic input seems to work too, but also very quiet. Providing an option 'model=w2jc' to modprobe however results in working(loud) front and mic. But this model does not seem to provide headphones output, so they don't work. This sounds to me like it may be easy to fix, so I hope you find this information useful. Would be great to have the sound working. Regards, Steffen (subscribed for the moment..) > The snd-hda-module loads with the following message: > | kernel: hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x18, fmic=0x0, line=0x1a, fline=0x0, cd=0x0, aux=0x0 > > aplay -l says: > > | **** List of PLAYBACK Hardware Devices **** > | card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > | card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > > lspci -vv -nn says: > > | 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) > | Subsystem: Apple Computer Inc. Unknown device [106b:00a0] > | Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- > | Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- > | Latency: 0, Cache Line Size: 256 bytes > | Interrupt: pin A routed to IRQ 20 > | Region 0: Memory at 9b500000 (64-bit, non-prefetchable) [size=16K] > | Capabilities: [50] Power Management version 2 > | Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) > | Status: D0 PME-Enable- DSel=0 DScale=0 PME- > | Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- > | Address: 0000000000000000 Data: 0000 > | Capabilities: [70] Express Unknown type IRQ 0 > | Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- > | Device: Latency L0s <64ns, L1 <1us > | Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- > | Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ > | Device: MaxPayload 128 bytes, MaxReadReq 128 bytes > | Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 > | Link: Latency L0s <64ns, L1 <1us > | Link: ASPM Disabled CommClk- ExtSynch- > | Link: Speed unknown, Width x0 > | Capabilities: [100] Virtual Channel > | Capabilities: [130] Unknown (5)

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