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

[alsa-devel] HDA Intel - Conexant CX20549 loaded but not working
by S Boucher 14 Apr '07

14 Apr '07

Using alsa-driver-1.0.14rc3 I have the following problems: - cannot record from internal mic (of an hp dv6245) - spdif not working - using jack I get lots of xruns (I don't get any xrun on the exact same system if I use an M-Audio Transit, so I'm assuming something ain't kosher in the conexant driver) - I was able to record cleanly from ext mic (although I had to select int mic) I'm with Ubuntu Feisty. N E Waze, if I can be of assistance let me know soon. The nice HP DV6245 laptop is heading back to the store in 1-2 days (nice system, except I don't have time to learn sign language :-)). __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Kernel crash with 2.6.20.x : snd-usb-audio implicated...
by Chris Rankin 14 Apr '07

14 Apr '07

Hi, The 2.6.20.x kernels all crash on my SMP-HT machine, and the oops report always shows that udev was trying to access an endpoint belonging to the snd-usb-audio driver. The frustrating part is that this particular endpoint shouldn't even exist! (It doesn't exist whenever I check the /sys/class/usb_endpoint directory.) Is there any way that userspace ("winecfg" in my case) can cause the snd-usb-audio driver to switch from (If# = 2, Alt = 0) to (If# = 2, Alt > 0) please? Because that is currently the only way that I can see this endpoint ever being created. Thanks, Chris --- bugme-daemon(a)bugzilla.kernel.org wrote: > Date: Sat, 14 Apr 2007 05:22:23 -0700 > From: bugme-daemon(a)bugzilla.kernel.org > To: rankincj(a)yahoo.com > Subject: [Bug 8198] udevd triggers oops when walking through /sys > > http://bugzilla.kernel.org/show_bug.cgi?id=8198 > > ------- Additional Comments From rankincj(a)yahoo.com 2007-04-14 05:22 ------- > Created an attachment (id=11144) > --> (http://bugzilla.kernel.org/attachment.cgi?id=11144&action=view) > dmesg output from 2.6.20.7 oops > > I don't know if this output is any more useful than any of the others. And I > can't see where the "single bit error" comes from when the output is full of > "use after free" poison. However, this much at least is different: > > EIP: [<6b6b6b6b>] 0x6b6b6b6b SS:ESP 0068:ec9eaf48 > <3>Slab corruption: start=eceea92c, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 080: 6b 6b 6b 6b 6a 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > Single bit error detected. Probably bad RAM. > Run memtest86+ or a similar memory test tool. > Prev obj: start=eceea720, len=512 > Redzone: 0x170fc2a5/0x170fc2a5. > Last user: [<f886eda5>](usb_create_ep_files+0xf1/0x2c2 [usbcore]) > 000: f8 fd 1b c2 e0 ee 4e f7 01 00 00 00 00 00 00 00 > 010: 30 a7 ee ec 30 a7 ee ec f4 91 1e c0 96 90 1e c0 > Next obj: start=eceeab38, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 000: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > 010: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > > Again, it's claiming that the last sysfs file accessed was the non-existent > /class/usb_endpoint/usbdev1.3_ep84/dev, which I simply cannot explain because > this endpoint belongs to If# = 2, Alt > 0: > > I: If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 90 Ivl=1ms > I: If#= 2 Alt= 2 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 46 Ivl=1ms > I: If#= 2 Alt= 3 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 25 Ivl=1ms > I: If#= 2 Alt= 4 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 16 Ivl=1ms > > Could the ALSA USB audio driver be changing the "Alt" on the fly somehow, > causing usbcore to need to create the extra endpoint in /sys? ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

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[alsa-devel] [2/3] 2.6.21-rc6: known regressions
by Adrian Bunk 14 Apr '07

14 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 : 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> 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 : USB: iPod doesn't work (CONFIG_USB_SUSPEND) References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Caused-By : Marcelo Tosatti <marcelo(a)kvack.org> commit 1d619f128ba911cd3e6d6ad3475f146eb92f5c27 Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged Subject : USB: Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Patch : http://lkml.org/lkml/2007/4/5/154 Status : patches available

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[alsa-devel] function location?
by Ashlesha Shintre 13 Apr '07

13 Apr '07

Hi, I am trying to write an application to just play a wave file and record sound using ALSA -- I looked at the pcm_min.c file and tried to run it - I am using UBUNTU and 1.0.11-7ubuntu3 sound library called libasound2. I cant seem to find the pcm.c file which has all the snd_pcm_open and other functions used in the pcm_min.c file! I am using : /usr/include/alsa/asoundlib.h header -- but i m not sure where the functions defined in this header are -- and, find / -name pcm.c yields nothing --nor does find / -name snd_pcm_open Any pointers please! Thanks, Ashlesha.

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[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

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[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

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[alsa-devel] Playback Devices in PREPARED State Throw Off POLLOUT Events?
by Jeff Rush 13 Apr '07

13 Apr '07

A question re normal internal operation of ALSA. I'm using the poll() interface to wait for a audio playback device to ask for more samples. The device is in the PREPARED state, not RUNNING state. At the beginning of my program, _before_ I write any samples to it, I'm getting repeated POLLOUT events, when I do not yet have data available to send. They seem to come frequent enough to load down my CPU, rather than every N ticks, so it seems. I thought a device was idle in the PREPARED state, and you had to either set the start conditions and write data, or explicitly invoke snd_pcm_start(). In my case, I wanted to use snd_pcm_start() once I had data to send, and also used the start conditions, set at 50%, to cause the audio output to them proceed to the speaker. The output snd_pcm_status_dump() right after exiting from the very first poll() ever in the program shows the following, and that I have a POLLOUT event pending. --- Status of Headphones Player --- state : PREPARED trigger_time: 0.000000 tstamp : 1176119509.658821000 delay : 0 avail : 2400 avail_max : 0 POLLOUT fired, calling ready_for_write() I suppose I could force it back to an earlier state like SND_PCM_STATE_OPEN or SND_PCM_STATE_SETUP, but that seems a bit harsh, since the snd_pcm_hw_params() takes it to the SND_PCM_STATE_PREPARED itself. Thanks for any tips, -Jeff

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 13 Apr '07

13 Apr '07

On Fri, 13 Apr 2007, Johannes Berg wrote: > On Thu, 2007-04-12 at 13:15 +0200, Rene Herman wrote: > > > > With Johannes don't know what kind of sound the chip produces using > > > OSS I'm stuck again. I have attached the samples to this e-mail. What > > > would be a reasonable debug step here? > > > > This message got stuck in the moderation queue due to being too big to > > pass through. The list limit is set at 60 kB and while that may be too > > low a limit, the 1.5 MB that was attached to this one was rather big. > > Too big for public lists. > > > > I approved it anyway, and then the attachment got scrubbed none the less > > by the list software it seems. > > I copied the files: > http://johannes.sipsolutions.net/files/stefan-noise.wav > http://johannes.sipsolutions.net/files/stefan-aplay.wav > > The interesting thing to note is that aoss and snd-pcm-oss have the > exact same effect, which is totally strange unless... hey! does aoss > maybe fall back to real oss when it is available? Point is the OSS modules aren't loaded :\ Stefan

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[alsa-devel] rawmidi timestamps
by Florian 13 Apr '07

13 Apr '07

Hi, thanks for replying the "blocking read" question - it works fine with poll(). I'm using its timeout mechanism to bail out every 200msecs of inactivity. I wondered what snd_rawmidi_status_get_tstamp() is meant to return? Briefly checking out the driver midlevel, the timestamp is just ignored and setmem'ed to 0, or did I miss anything? 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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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 12 Apr '07

12 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi, With Johannes don't know what kind of sound the chip produces using OSS I'm stuck again. I have attached the samples to this e-mail. What would be a reasonable debug step here? Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHgiuYH1+F2Rqwn0RCidgAJ44R70RyxNLoWOh681UGFbP+rrVFACfROLN bVFdMfHcIP4nLUDfaqrsbbQ= =MRRf -----END PGP SIGNATURE-----

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