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

[alsa-devel] crash/reboot with rawmidi on ice1712 dual opteron
by Florian 25 Apr '07

25 Apr '07

Hi, on one particular machine, an IBM workstation (Intellistation A Pro with dual opteron 250, 4GB RAM), opening a rawmidi input port on an M-Audio 2496 card will cause an immediate reboot of the machine. I've tried with different SMP kernels and different ALSA versions (including 1.0.14rc3), but always the same. Syslog does not show anything. Using a USB MIDI interface works fine. Also, MIDI input on a similar workstation (Intellistation Z Pro, dual xeon, 1GB RAM) with M-Audio 2496 works without problems. Any ideas? Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

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[alsa-devel] Digigram ISA cards
by Ash Willis 25 Apr '07

25 Apr '07

Hi, I'm thinking about trying to write a driver for some old Digigram cards soon, namely the PCX11 and PCX9. I might be wrong but it seems these cards natively support MPEG layers I and II and only actually support PCM by re-encoding in-driver. Does alsa-lib support MPEG conversions? Are there examples of drivers that already do this type of thing? Please forgive my ignorance. I grepped the sources and didn't find too much info. Ash =

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[alsa-devel] [1/3] 2.6.21-rc7: known regressions (v2)
by Adrian Bunk 25 Apr '07

25 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : gammu no longer works References : http://lkml.org/lkml/2007/4/20/84 Submitter : Wolfgang Erig <Wolfgang.Erig(a)gmx.de> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Wrong "From" email address selected earlier. Oops.

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] ICE 1724 MIDI support still broken?
by Daniel James 24 Apr '07

24 Apr '07

Hello ALSA team, We have a user with an ESI Juli@ whose Linux box hangs when any MIDI application starts up. I read that MIDI support is broken generally on ICE 1724 cards; is this still the case in recent driver releases? If so, is there any interest in fixing this long-standing issue? The user has indicated that his box would be available by ssh for any tests required. Cheers! Daniel

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[alsa-devel] Standard config and plugin paths
by Takashi Iwai 24 Apr '07

24 Apr '07

Hi, while working on openSUSE 10.3 ALSA packaging, I had a discussion regarding the config and plugin paths. Basically, the current configs (typically in /usr/share/alsa or /etc/alsa) and binary plugins (/usr/lib/alsa-lib) are fairly strictly bound to the specific library, libasound.so.2. That is, if we publish a newer, incompatible version, libasound.so.3, these files will be very unlikely unusable, too. The suggestion was that we should keep these config and plugin files in the version-dependent paths so that no conflictions would occur at update. For example, instead of /usr/share/alsa, /usr/share/alsa2 or whatever. The fix would be easy -- just add some options to configure script of alsa-lib and alsa-plugins. So, the question is what path names are most suitable as the standard. For example, I find the following OK: - $DATADIR/alsa-1.0/ - $LIBDIR/alsa-lib-1.0/ Or, the deeper paths: - $DATADIR/alsa/1.0/ - $LIBDIR/alsa-lib/1.0/ For keeping the backward compatibility, we may implement fallback to read the old default path in addition, if the new path doesn't exist. In that case, the former suggestion would be cleaner. Any comments or suggestions? Takashi

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[alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 24 Apr '07

24 Apr '07

Greetings, I'm setting out to write an ALSA driver for a very limited device thats behind a PLD. The actual codec is a uda1380 and does 16bit MSB stereo encode/decode. The way the device is driven the sampling rate is limited to 8KHz and data transfer to/from the device I have to handle manually. I've read over the 'writing an ALSA driver' tutorial and have started the driver but have run into a few questions: - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB samples does the following snd_pcm_hardware_t look right?: static snd_pcm_hardware_t my_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? .rates = SNDRV_PCM_RATE_8000, // seems redundant? .rate_min = 8000, .rate_max = 8000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 32768, .period_bytes_min = 4096, .period_bytes_max = 32768, .periods_min = 1, .periods_max = 1024, }; - I'm a little confused at how to deal with copying data from/to the device. It looks like I would implement the copy callback and copy the passed buffer to the device in the device specific manner I need to (playback), but I'm not clear where to put data that I receive from the device (capture). I notice that when the hw_params callback is called a dma buffer has been created by ALSA in the substream->runtime struct. I assume this is created based on hte buffer_bytes_max field above and I'm to copy data to this buffer? How do I know where to copy it within the buffer and what data ALSA has already consumed from the buffer? - I might need to create an intermediate buffer between ALSA's rather large buffers and my hardware buffer. Is there any driver that I could look at that does this? The tutorial talks about the vxpocket driver but that code looks very foriegn from the rest of the ALSA drivers. - I'm not sure I understand what the 'period' is about. Its clear to me that I need to call snd_pcm_period_elapsed() periodically from either an IRQ or other timed callback, but I'm not clear what the hardware pointer is or how that converts to 'frames' and 'periods'. - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA provides a mechanism for converting data from other formats? Thanks for any pointers, Tim

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[alsa-devel] What does start_threshold mean, and what value should it contain?
by j t 24 Apr '07

24 Apr '07

Hi. I'm trying to debug a problem with xruns which happen when playing a wav file with vlc, but I can't reproduce the problems when using "aplay -M" to play the same wav file. (logged as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3031). I've checked the contents of /proc/asound/card0/pcm0p/sub0/ in both cases - they appear to be very similar except for the contents of sw_params: when I use vlc, start_threshold says 1, but when I use "aplay -M", start_threshold says 16384. So, what does start_threshold mean, is there a particular value that start_threshold "should" contain, and could this difference be the cause of the xruns? (The aplay man page doesn't seem to have any flags for controlling the value of this parameter, so it doesn't appear to be "user-configurable"). (Apologies if this is the wrong mailing list - I though that -devel would be better than -user, but I could be wrong). Thanks, Jaime

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