Re: [Alsa-devel] status of powermac sound?
On Thu, 13 May 2004 02:53, Sjoerd Simons wrote: > Using this on my albook. I get strange distorsions and a too low pitch > when playing a dvd or divx with 48Khz audio. > > For the record, the hardware is: > 0 [Snapper]: PMac Snapper - PowerMac Snapper > PowerMac Snapper (Dev 35) Sub-frame 0 yep, exactly. And normally using alsa directly does not work, and if it does it shows the same effects... mplayer with -ao alsa1x yields: alsa-play: unknown status, trying to reset soundcard works fine with -ao oss xmms doesn't work with alsa-plugin at all (well, OK, who cares... xmms seems to have died anyways...) astonishingly jackd normally works. Is there an option to force the driver to always use a buffer size of a power of 2? Have fun* Niklas --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson & Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] status of powermac sound?
Hi *, is there any progress on the status of full support for the newer devices as snapper, etc (in the AlBooks,..). I'm running the 2.6.5 benh-kernel from bitkeeper and basically the sound only works reliably when using the oss emulation. I can remember a short time under 2.6.* when alsa output actually worked, though... If you need a guinea pig, I will gladly test patches on my AlBook running gentoo. Have fun* Niklas --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson & Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] USB Audio problems
Karim Yaghmour wrote: Well, it seems that I'm going to have to answer my own self ... :) Yes, usb-audio seems to be a bit forgotten... (Takashi, those mplayer-plughw-segfaults still persist, even on x86 and even with kernel 2.4 (current cvs of drivers/lib, of course) The following is what I've been able to find using additional tracing info. Also there's a fix for usbaudio.c. hmmm, the submit_urb-error is gone, but random lockups and usb-device-disconnect until reboot have come... (bitkeeper-2.6 from just one hour ago...) with your function. Have fun* Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] usb-audio
Am Donnerstag, 30. Oktober 2003 20:46 wurde geschrieben: > At Thu, 30 Oct 2003 13:42:14 +0100, > > hmm, really weird. > > meanwhile, i rewrote snd_pcm_linear_convert() without goto trick. > could you try the attached patch? ... you don't really want to know... Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 16384 (LWP 22147)] 0x0fd41564 in snd_pcm_linear_conv_xx12_xx21 ( src=0x104e4000 , dst=0x30c28590 "", src_step=4, dst_step=4, frames=7835) at pcm_linear.c:303 303 DEF_CONV(conv_xx12_xx21,l_conv_xx12_xx21); (gdb) bt #0 0x0fd41564 in snd_pcm_linear_conv_xx12_xx21 ( src=0x104e4000 , dst=0x30c28590 "", src_step=4, dst_step=4, frames=7835) at pcm_linear.c:303 #1 0x0fd429f4 in snd_pcm_linear_convert (dst_areas=0x104d1f18, dst_offset=0, src_areas=0x7fffcf10, src_offset=0, channels=2, frames=16384, convidx=0) at pcm_local.h:362 #2 0x0fd435a0 in snd_pcm_linear_write_areas (pcm=0x0, areas=0x1, offset=2147471120, size=12, slave_areas=0x0, slave_offset=1140885572, slave_sizep=0xfdb408c) at pcm_linear.c:722 #3 0x0fd3f808 in snd_pcm_plugin_write_areas (pcm=0x2, areas=0x7fffcf10, offset=0, size=273489688) at pcm_plugin.c:365 #4 0x0fd35d94 in snd_pcm_write_areas (pcm=0x104db890, areas=0xfd41554, offset=0, size=273489688, func=0) at pcm.c:6206 #5 0x0fd3fc14 in snd_pcm_plugin_writei (pcm=0x104db600, buffer=0x30c28590, size=273489688) at pcm_plugin.c:436 #6 0x0fd2ec3c in snd_pcm_writei (pcm=0x0, buffer=0x0, size=0) at pcm_local.h:368 #7 0x10078718 in outputaudio () #8 0x1007837c in outputaudio () #9 0x1002ed74 in main () Niklas -- --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] usb-audio
Am Donnerstag, 30. Oktober 2003 13:17 schrieb Takashi Iwai: > At Wed, 29 Oct 2003 23:26:07 +0100, > > Niklas Werner wrote: > > Am Mittwoch, 29. Oktober 2003 19:24 wurde geschrieben: > > > At Tue, 28 Oct 2003 20:18:35 +0100, > > > > > > > > > hmm, it seems that a wrong label is used. the label should be > > > conv_xx12_xx21 (= conv_labels[35]). something is really broken. > > > > > > could you check stepwise the loop there? > > > > still checking (any tips on speeding this up (I'm trying "step 8" > > atm?), seems to happen after the first set of frames (=16384) is > > processed. > > in the first process, did it go to conv_xx12_xx21 properly? nope: 0x0fd422f4 184 goto *conv; (gdb) si 299 conv_1234_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 16); goto CONV_END; (gdb) si 0x0fd424c0 299 conv_1234_xx21: as_u16(dst) = bswap_16 (as_u32c(src) >> 16); goto CONV_END; (gdb) si 0x0fd424c4 299 conv_1234_xx21: as_u16(dst) = bswap_16 (as_u32c(src) >> 16); goto CONV_END; (gdb) si 0x0fd424c8 299 conv_1234_xx21: as_u16(dst) = bswap_16 (as_u32c(src) >> 16); goto CONV_END; (gdb) si 0x0fd424cc 299 conv_1234_xx21: as_u16(dst) = bswap_16 (as_u32c(src) >> 16); goto CONV_END; (gdb) si 0x0fd4235c 230 conv_xxx1_xx10: as_u16(dst) = (u_int16_t)as_u8c(src) << 8; goto CONV_END; (gdb) si 0x0fd42360 230 conv_xxx1_xx10: as_u16(dst) = (u_int16_t)as_u8c(src) << 8; goto CONV_END; (gdb) si 190 dst += dst_step; Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] usb-audio
Am Mittwoch, 29. Oktober 2003 19:24 wurde geschrieben: > At Tue, 28 Oct 2003 20:18:35 +0100, > > > hmm, it seems that a wrong label is used. the label should be > conv_xx12_xx21 (= conv_labels[35]). something is really broken. > > could you check stepwise the loop there? > still checking (any tips on speeding this up (I'm trying "step 8" atm?), seems to happen after the first set of frames (=16384) is processed. the label stays wrong throughout the whole loop, the convidx stays 35 > thanks for check. > can you hear a clear sound with aplay -M ? > Yes :-) Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] record on new Powerbooks?
Am Mittwoch, 29. Oktober 2003 19:15 wurde geschrieben: > At Wed, 29 Oct 2003 19:03:02 +0100, > > Niklas Werner wrote: > > Hi all, > > > > I can't get the internal mic or recording from either usb or line-in > > to work on a new Powerbook 15" (Aluminium). > > The driver works (detects a "snapper"), but only for playback (and > > DRC and bass/treble). > > the capture on snapper may not work. the code was added at the time i > had time to touch a powerbook for an hour and found that the capture > DMA exists. but, apparently, some codec (preamp?) initialization > would be needed errm, I found out, it sort of works... (playing with freqtweak) well: the record level is sort of controlled with the master fader: sometimes the gain is reduced, sometimes there's just static noise (strange that). All in all it does not seem to be predictably controllable. > perhaps one can see more detailed information in drawin. > maybe the darwin-sources help... > > Takashi Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] record on new Powerbooks?
Hi all, I can't get the internal mic or recording from either usb or line-in to work on a new Powerbook 15" (Aluminium). The driver works (detects a "snapper"), but only for playback (and DRC and bass/treble). Is there any configuration feature I overlooked? for the record: kernel 2.6.0-test9 with benh-patches. details on the machine at: "http://developer.apple.com/documentation/Hardware/Developer_Notes/ Macintosh_CPUs-G4/15inchPowerBookG4/index.html" I gladly test any patches .-) Have fun* Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] usb-audio
Here we go again, Am Dienstag, 28. Oktober 2003 19:25 schrieb Takashi Iwai: > > hmm, then something wrong in the converter routine... > > needs to take a deeper look. > > i found a bug regarding the plugin but it must be another bug from the > above problem. > > segfault is a bit puzzling. could you try the attached patch to see > the parameters? or, even better, check the values in gdb's > backtrace. Building audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit... Video: no video Starting playback... XXX linear_convert: dst=0x104d3dc0/0, src=0x7fffc020/0, ch=2, frames=16384, idx=35 === gdb: Starting playback... XXX linear_convert: dst=0x104d1f18/0, src=0x7fffcf40/0, ch=2, frames=16384, idx= 35 Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 16384 (LWP 14666)] snd_pcm_linear_convert (dst_areas=0x104d1f18, dst_offset=0, src_areas=0x7fffcf40, src_offset=0, channels=2, frames=16384, convidx=35) at plugin_ops.h:299 299 conv_1234_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 16); goto CONV_EN D; (gdb) bt #0 snd_pcm_linear_convert (dst_areas=0x104d1f18, dst_offset=0, src_areas=0x7fffcf40, src_offset=0, channels=2, frames=16384, convidx=35) at plugin_ops.h:299 #1 0x0fd4326c in snd_pcm_linear_convert (dst_areas=0x104d1f18, dst_offset=0, src_areas=0x7fffcf40, src_offset=0, channels=2, frames=16384, convidx=35) at pcm_linear.c:170 #2 0x0fd41838 in snd_pcm_plugin_write_areas (pcm=0x0, areas=0x7fffcf40, offset=2, size=2147471168) at pcm_plugin.c:365 #3 0x0fd37dc4 in snd_pcm_write_areas (pcm=0x23, areas=0x104db86c, offset=2, size=2147471168, func=0) at pcm.c:6206 #4 0x0fd41c44 in snd_pcm_plugin_writei (pcm=0x104d1f18, buffer=0x4, size=2147471168) at pcm_plugin.c:436 #5 0x0fd30c6c in snd_pcm_writei (pcm=0x0, buffer=0x0, size=0) at pcm_local.h:368 #6 0x10078718 in outputaudio () #7 0x1007837c in outputaudio () #8 0x1002ed74 in main () == > > > also, what happens if running mplayer with mmap option, i.e. > > % mplayer -ao alsa9:emi26:mmap > horrible, distorted sound: (sounds like computer is too slow to deliver the samples) ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: Die Dateizugriffsnummer ist ein schlechter Verfassung meaning: "File descriptor in bad state" (whoever sneaked in this translation...) Have fun Niklas --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] usb-audio
Am Dienstag, 28. Oktober 2003 13:13 wurde geschrieben: > At Tue, 28 Oct 2003 12:48:17 +0100, > > Niklas Werner wrote: > > Am Dienstag, 28. Oktober 2003 12:10 schrieb Takashi Iwai: > > > > No, I don't think it is. > > > > I get similar problems with my emi 2|6 and alsaplayer, mplayer, > > > > xmms, ... > > > > > > did you use plughw instead of hw in all cases? > > > otherwise they won't work always. > > > > plughw doesn't work at all! > > for xmms, too? oops, sorry, forgot. xmms works! except for adjusting the PCM-Volume, such is life... (this works on Intel, btw) Is there any special trick regarding the ctl.* for plughw? alsa-xmms is probably the newest one, I am using gentoo... i tried using diefferent mixer-card settings, but none did work. > > to be sure, try to start mplayer on gdb and check what is broken. > tracing via strace woule help, too. gdb: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 16384 (LWP 28218)] 0x0fd47d64 in snd_pcm_linear_convert () from /usr/lib/libasound.so.2 (gdb) bt #0 0x0fd47d64 in snd_pcm_linear_convert () from /usr/lib/libasound.so.2 #1 0x0fd47b00 in snd_pcm_linear_convert () from /usr/lib/libasound.so.2 #2 0x0fd4643c in snd_pcm_plugin_write_areas () from /usr/lib/ libasound.so.2 #3 0x0fd3ce54 in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #4 0x0fd467bc in snd_pcm_plugin_writei () from /usr/lib/libasound.so.2 #5 0x0fd3615c in snd_pcm_writei () from /usr/lib/libasound.so.2 #6 0x10078718 in outputaudio () #7 0x1007837c in outputaudio () #8 0x1002ed74 in main () strace: see attached file. have fun* Niklas mplayer-alsa9-plughw-strace.log.gz Description: GNU Zip compressed data
Re: [Alsa-devel] usb-audio
Am Dienstag, 28. Oktober 2003 12:10 schrieb Takashi Iwai:
>
> > No, I don't think it is.
> > I get similar problems with my emi 2|6 and alsaplayer, mplayer, xmms,
> > ...
>
> did you use plughw instead of hw in all cases?
> otherwise they won't work always.
plughw doesn't work at all!
> mplayer has an option to specify the device name.
> it'd be better to define a new pcm in ~/.asoundrc such as
>
> pcm.emi26 {
> type plug
> slave.pcm "hw:0"
> }
>
> (where "hw:0" should be changed to the corresponding one)
>
> and run like
>
> % mplayer -ao alsa9:emi26 ...
>
gives:
Checking audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 44100 hz, big endian signed int
AF_pre: 44100Hz 2ch Signed 16-bit (Big-Endian)
alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit
(Big-Endian)
alsa-init: soundcard set to emi26
alsa9: 44100 Hz/2 channels/2 bpf/32768 bytes buffer/Signed 16 bit Big
Endian
AO: [alsa9] 44100Hz 2ch Signed 16-bit (Big-Endian) (2 bps)
Building audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
Video: no video
Starting playback...
MPlayer interrupted by signal 11 in module: play_audio
and no sound, whereas accessing hw:1 directly at least produces output.
playing through plug-hw with aplay works, though
alsaplayer doesn't like the card with plug-layer. (alsaplayer seems to
have deeper problems, since its doesn't work on the internal snapper, as
well.
(Oh: kernel 2.6-test8-benh from bitkeeper, if that is of any interest)
Have fun*
Niklas
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Re: [Alsa-devel] usb-audio
Am Dienstag, 28. Oktober 2003 11:11 schrieb Takashi Iwai: > At Mon, 27 Oct 2003 20:53:08 +0100, > > Antonio Willy Malara wrote: > > On 2003.10.27 19:16, Takashi Iwai wrote: > > > > /* FIXME: correct endianess and sign? */ > > > > > > could you give more information: > > > which program, which device and what format doesn't it work? > > > > the system is a powermac, the device is a Griffin iMic, the app is > > jack version 0.80, the output is: > > > > Sorry. The audio interface "hw:1"doesn't support either of the two > > hardware sample formats that jack can use. > > then it's a problem of JACK, not ALSA. > the hardware doesn't support 32bit integer but only 24bit packed in 3 > bytes. No, I don't think it is. I get similar problems with my emi 2|6 and alsaplayer, mplayer, xmms, ... My impression is that ALSA assumes that the connected interfaces _always_ allow for the same Endianess as the system. This, of course, isn't always so. Mplayer: Checking audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit... AF_pre: af format: 2 bps, 2 ch, 44100 hz, big endian signed int AF_pre: 44100Hz 2ch Signed 16-bit (Big-Endian) alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Big-Endian) alsa-init: soundcard set to emi alsa-init: format Signed 16-bit (Big-Endian) are not supported by hardware, trying default alsa9: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa9] 44100Hz 2ch Signed 16-bit (Little-Endian) (2 bps) alsaplayer either dies with "FIXME: f_unsynchronization is set.Please contact alsaplayer team." or produces static noise. aplay does get this right, so it seems to use a different method of querying the device. Have fun* Niklas --- This SF.net email is sponsored by: The SF.net Donation Program. Do you like what SourceForge.net is doing for the Open Source Community? Make a contribution, and help us add new features and functionality. Click here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] USB-Audio strangeness
Hi List! I am just wondering whether anybody has a solution for following (admittedly minor) bug: When I cold-boot (doesn't matter whether PPC or Intel) with my emi 2|6 or reconnect the device the range of the mixer always is set to 0-50% while using the full range in the hardware. So 50% mixer-value is 100% volume. If the emi already has its firmware when booting (ie warm reboot), the mixer settings are right. when using the oss- audio-module this doesn't happen. I am using newest cvs-alsa (from yesterday) on Gentoo-1.4, SuSE-7.3 or debian-sarge (all the same). Thanks Niklas --- This SF.net email is sponsored by:Crypto Challenge is now open! Get cracking and register here for some mind boggling fun and the chance of winning an Apple iPod: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0031en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] oops with current ALSA CVS + gcc3
Paul Davis wrote: did you compile the kernel itself with the same gcc? the combination of gcc-2.x and gcc-3.x on the kernel space will likely cause oops. no, i knew about that issue, and thats why i had to recompile everything. i am tempted to go back to my older ALSA CVS tree (i tar'ed it before fetching current CVS) and see what happens. Just FYI. On my gentoo-1.4 everything compiles and runs fine (via82xx, usb-audio, dummy, virmidi) maybe this helps: Reading specs from /usr/lib/gcc-lib/i686-pc-linux-gnu/3.2.2/specs Configured with: /var/tmp/portage/gcc-3.2.2-r3/work/gcc-3.2.2/configure --prefix=/usr --bindir=/usr/i686-pc-linux-gnu/gcc-bin/3.2 --includedir=/usr/lib/gcc-lib/i686-pc-linux-gnu/3.2.2/include --datadir=/usr/share/gcc-data/i686-pc-linux-gnu/3.2 --mandir=/usr/share/gcc-data/i686-pc-linux-gnu/3.2/man --infodir=/usr/share/gcc-data/i686-pc-linux-gnu/3.2/info --enable-shared --host=i686-pc-linux-gnu --target=i686-pc-linux-gnu --with-system-zlib --enable-languages=c,c++,ada,f77,objc,java --enable-threads=posix --enable-long-long --disable-checking --enable-cstdio=stdio --enable-clocale=generic --enable-__cxa_atexit --enable-version-specific-runtime-libs --with-gxx-include-dir=/usr/lib/gcc-lib/i686-pc-linux-gnu/3.2.2/include/g++-v3 --with-local-prefix=/usr/local --enable-shared --enable-nls --without-included-gettext Thread model: posix gcc version 3.2.2 have fun* Niklas --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] emi 2|6 usb-audio
Hi all! Good News (especially to takashi): Emagic's emi 2|6 works with 6 channels out using a52dec from dvd! unfortunately there seems to be no way of accesing the 6 channels via the .asoundrc-trick from patrick. I can't get any pcmCxD1 or pcmCxD2 device. alsamixer only shows 2 channels in and 2 channel out. === cat /proc/asound/emi/stream0 EMAGIC GmbH Emagic EMI 2|6 : USB Audio Playback: Status: Stop Altset 1 Format: S16_LE Channels: 2 Endpoint: 10 OUT (ASYNC) Rates: 44100, 48000, 96000 Altset 2 Format: S16_LE Channels: 6 Endpoint: 10 OUT (ASYNC) Rates: 44100, 48000 Altset 3 Format: S24_3LE Channels: 6 Endpoint: 10 OUT (ASYNC) Rates: 44100, 48000 Altset 4 Format: S24_3LE Channels: 2 Endpoint: 10 OUT (ASYNC) Rates: 96000 Capture: Status: Stop Altset 1 Format: S16_LE Channels: 2 Endpoint: 12 IN (ASYNC) Rates: 44100, 48000, 96000 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 12 IN (ASYNC) Rates: 44100, 48000, 96000 But: cat /proc/asound/devices 0: [0- 0]: ctl 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 33: : timer 32: [1- 0]: ctl 48: [1- 0]: digital audio playback 56: [1- 0]: digital audio capture card0 is my ppc-screamer, card1 is the emi. Does anybody have an idea, how I must tweak my .asoundrc to get a device I can send 6 channels of appropriately formatted data to? I'm aware that a52dec seems to send sample1Ch1-sample1Ch2-sample1Ch3 ... I'd like to control the channels seperately, but I'm afraid that won't be possible... Have fun* Niklas --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] status of usb audio driver
Well, now comes the ugly things: Using the usb-audio-driver leads to system-lockups: No matter whether I access my emi26 via aplay or via oss-emulation, after some time or (more probable) some amount of data my System locks up (hard reset ;-() or at least the usb-bus locks absoulety thight up. I can't reload the driver, not even reconnecting the device helps it always gives a timeout message: Jun 13 12:26:19 Schlumpfine kernel: 0: [cd2c01e0] link (0d2c0210) e0 Stalled CRC/Timeo Length=7 MaxLen=7 DT0 EndPt=0 Dev=2, PID=2d(SETUP) (buf=0d0222c0) Jun 13 12:26:19 Schlumpfine kernel: 1: [cd2c0210] link (0d2c0240) e3 SPD Active Length=0 MaxLen=11 DT1 EndPt=0 Dev=2, PID=69(IN) (buf=08a9c000) Jun 13 12:26:19 Schlumpfine kernel: 2: [cd2c0240] link (0001) e3 IOC Active Length=0 MaxLen=7ff DT1 EndPt=0 Dev=2, PID=e1(OUT) (buf=) Jun 13 12:25:03 Schlumpfine kernel: usb_control/bulk_msg: timeout Jun 13 12:25:03 Schlumpfine kernel: emi26_load_firmware - error loading firmware : error = -110<6>IPv6 v0.8 for NET4.0 Jun 13 12:25:28 Schlumpfine kernel: usb_control/bulk_msg: timeout Jun 13 12:25:43 Schlumpfine last message repeated 149 times Jun 13 12:26:06 Schlumpfine kernel: emi26: set_reset (1) failed<3>emi26_load_fir mware - error loading firmware: error = -110<7>uhci.c: root-hub INT complete: po rt1: 580 port2: 48a data: 4 This doesn't happen with the kernel module (audio). Unfortunately I can't find any clues in the logfiles, maybe you can... At least I can reproduce this easily using xmms to playback some 7 minutes or so of 48kHz mp3s Using aplay and waves this effect tends to appear later. Using xmms' alsa-plugin crashes the sound-driver at once... right, have fun* Niklas ___ Don't miss the 2002 Sprint PCS Application Developer's Conference August 25-28 in Las Vegas - http://devcon.sprintpcs.com/adp/index.cfm?source=osdntextlink ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] status of usb audio driver
> > OK, in stream0 now my emi26 is recognised as being capable of 6 outs
> > @ 44.1 or 48 k and 2 out @ 96 kHz, but still I only get
> > /proc/asound/card0/: id pcm0p pcm0c stream0
>
> it's ok. a stream can support different formats and channels.
> if you find "Channels: 6" in stream0 proc file, then it means that the
> pcm device 'hw:0,0' (or hw:X,0 where X is the card number) can accept
> 6 channels playback/capture. if so, you can just feed 6-channels
> interleaved data to the device.
> well, this means also, that you have to access all 6 channels at the
> same time.
bugger!
Well, OK, at least that gives me an idea... but the much discussed
("multiple cards"-thread on jackit-devel) .asoundrc-fiddling for getting
more than one app to do the playback should work ok then, shouldn't it?
I'll give it a try :-)
Have fun*
Niklas
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Re: [Alsa-devel] status of usb audio driver
Am Wednesday, 12. June 2002 16:13, wurde geschrieben: > At Wed, 12 Jun 2002 23:10:19 +0900, > > Patrick Shirkey wrote: > > Takashi Iwai wrote: > > > it's on cvs. please give a try. > > > btw, now descriptor proc was removed (one can use lsusb anyway). > > > instead you'll see formats supported on each stream. OK, in stream0 now my emi26 is recognised as being capable of 6 outs @ 44.1 or 48 k and 2 out @ 96 kHz, but still I only get /proc/asound/card0/: id pcm0p pcm0c stream0 so I still cant' access the other 4 outs. Or is there a way to use stream0 directly ? Have fun* Niklas ___ Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] USB-Audio (EMI 2|6)
Hi All! the silent reader strikes again... Thanks a lot for usb-audio-support in alsa! Just to let you know: it works fine with the emagic emi2|6-interface by using the oss-firmware-loader from http://www.vtoy.fi/~tapio/emi26.html and rmmod-ing "audio" afterwards! I tried it on a powerbook pismo (G3 500), but it should work on a pc as well. (since the FW-Loader works on both) capture seems to work as well, at least the soundcard switched into record-mode... Now this would allow me to use jack for my thesis instead of portaudio. Great! :-) Have fun* Niklas ___ Don't miss the 2002 Sprint PCS Application Developer's Conference August 25-28 in Las Vegas -- http://devcon.sprintpcs.com/adp/index.cfm ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
