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[alsa-devel] seg fault in libasound
by Jerry Geis 26 Jun '08

26 Jun '08

I am running asterisk on debian Etch , i486 with math emulation in kernel. Verified by /boot/config-file. I am getting a seg fault in alsa. First I ran straight debian with alsa install. I got the same seg fault. Second I installed 1.0.16 and I still get the seg fault. Third I installed 1.0.17rc2 and I still get the seg fault. Posted to users first and stan suggested I post to devel. Below is the trace. I am only using the Console/Dsp from asterisk at this point. Which is the interface to the soundcard and alsa. What shall I do? Jerry -------------------------- Program received signal SIGABRT, Aborted. [Switching to Thread -1224033360 (LWP 2440)] 0xb7d77947 in raise () from /lib/tls/libc.so.6 (gdb) where #0 0xb7d77947 in raise () from /lib/tls/libc.so.6 #1 0xb7d790c9 in abort () from /lib/tls/libc.so.6 #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6 #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0, src_area=0x81cb7ec, src_offset=5223, samples=816, format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499 #4 0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at pcm_dsnoop.c:77 #5 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932 #6 0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at pcm_generic.c:143 #7 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932 #8 0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624 #9 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932 #10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8, areas=0xb70a5b70, offset=0, size=160, func=0xb7421e10 <snd_pcm_mmap_read_areas>) at pcm.c:6376 #11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8, buffer=0xb74a5a40, size=160) at pcm_mmap.c:236 #12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40, size=2440) at pcm_local.h:521 #13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683 #14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at channel.c:2100 #15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68, config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380 #16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18, config=0xb70a6eac) at res_features.c:1490 #17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8, peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773 #18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8) at app_dial.c:1827 #19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0, context=0x82129d8 "smvoice", exten=0x8212a28 "s", priority=3, label=0x0, callerid=0x8215450 "404", action=E_SPAWN) at pbx.c:537 #20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317 #21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636 #22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909 #23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0 #24 0xb7e1b01e in clone () from /lib/tls/libc.so.6

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[alsa-devel] [asoc-dev][RFC/PATCH 0/2] Generic DAPM register modifier widget
by Jarkko Nikula 26 Jun '08

26 Jun '08

Hi At least tlv320aic3x codec has several use cases where multiple register bits are needed to change when widget power state changes and where these are not worth/safe to expose into user space. Like mic bias voltage. Currently there is one use case for this multiple bit change in tlv320aic3x: SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), There are two problems with that: 1. Two widgets having the same name and thus snd_soc_dapm_add_route will add only first one them into audio path 2. Even then bits aren't modified at once and thus some fault bit configuration might be present while power state is changing. So I came with this idea of generic register modifiers widget and event handler for it. -- Jarkko

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[alsa-devel] No sound on MacBook 3.1
by Marco Vermeulen 26 Jun '08

26 Jun '08

Hi there, I hope that someone can help me, as I've tried everything conceivable to get sound working on my Macbook 3.1. I've visited many a forum, and eventually came across this list. My last hope :) I purchased a Macbook 3.1 earlier this year. On installing Linux, I haven't been able to get any sound going. I've tried several distributions: Ubuntu Hardy, Fedora 8/9, and now openSuse. I came across a script that uploads local relevant information to a paste bin, so here's a link: http://pastebin.ca/1055610 Please help, I'm desperate to get this going on my machine! Thanks & Regards, Marco.

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[alsa-devel] support for RME HDSPe RayDAT
by Matthijs ten Berge 26 Jun '08

26 Jun '08

Hello all, <intro:> As you might already know, RME has released a successor for the HDSP 9652, the HDSPe RayDAT (http://www.rme-audio.de/en_products_hdspe_raydat.php). It is based on the 9652's design, with some new/changed features (e.g. more channels and lower latencies), but the most important hardware change is that it is now a PCI-express card. I'm currently in the process of choosing which audio interface to buy, the 'old' HDSP 9652 or the new RayDAT (I haven't found any other card with 3 or more ADAT in/outs in the same price range or below, but suggestions are welcome). Of course I can stay on the safe side and opt for the HDSP 9652, which has stable and complete driver support (right, eh?). But on the other hand the RayDAT has better features and future, but without the guarantee that it is or will be supported it can only serve as a very costly paper weight (and not even a good one I suppose). I wrote an e-mail to RME, but a bit too early, the card was not released at that moment. See below for their reply. <question:> Are there any plans to add support for the RME RayDAT to the HDSP driver/tools? Or perhaps somebody is already working on it right now? It is not mentioned in the Wiki matrix atm. I wish I could just do it myself, but (concerning kernel programming) I never did more than a 'hello world' kernel module, so I don't think this would be a good idea... (I do have coding experience, but not in-kernel).

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Re: [alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Takashi Iwai 26 Jun '08

26 Jun '08

At Thu, 26 Jun 2008 14:43:30 +0800, <Ashwin.Chandra(a)corel.com> wrote: > > Hello, > I have found an issue with the mixer devices. I had been controlling > only one elem, "LineOut", but there is another elem named "iSpeaker" > which is the one that shows only left speaker active in alsamixer. I am > not sure why I would need to control both of them. I would expect > "Lineout" to be the master control. But anyway I can fix the problem in > my application by un setting mute on the right channel at the beginning > of the application. Good to hear that you found out the culprit. > Do you have any idea about this though? (or have seen this before)? There have been similar bug reports, but your finding isn't unknown (and thus important). Could you run alsa-info.sh with --no-upload option for both working and non-working cases, and attach the generated files? The script is found at http://www.alsa-project.org/alsa-info.sh thanks, Takashi > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, June 25, 2008 11:26 PM > To: Ashwin Chandra > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > speaker only > > At Thu, 26 Jun 2008 03:28:42 +0800, > <Ashwin.Chandra(a)corel.com> wrote: > > > > Hi Takashi, > > Actually now I am noticing the problem (left speaker only) with > > mplayer/smplayer in their OSS emulation as well. It seems to happen > once > > I run my player first, then after that the other players exhibit the > > problem. > > In this case, mplayer will choose the ALSA mode automatically, I > guess. So the problem appears to be ALSA-mode specific, possibly > about dmix. > > Try to define the following in ~/.asoundrc: > pcm.!default "plughw" > > then it won't use dmix as default any more. (Then you can use only > one app at the same time, of course.) Check whether the problem can > be reproduced with this setting. > > > Takashi > > > > Sometimes after muting and unmuting my player, the problem goes > > away and also away on the players. It is almost like my player is > > causing bad internal state for the audio driver. Do you have any > > suggestions to work around this? > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Wednesday, June 25, 2008 3:23 AM > > To: Ashwin Chandra > > Cc: alsa-devel(a)alsa-project.org > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > speaker only > > > > At Wed, 25 Jun 2008 01:52:16 +0800, > > <Ashwin.Chandra(a)corel.com> wrote: > > > > > > I have an Intel ALC861VD. Other players (mplayer, smplayer, aplay) > > don't > > > seem to exhibit this problem. Is there any workaround I can do in my > > > application so that upon first time sound playback, I can get it to > > play > > > on both channels? > > > > Check /proc/asound/card0/pcm0p/sub0/* files and compare between > > working and non-working states. I guess the working case uses OSS > > emulation or 16bit format. > > > > If so, try to add the following to ~/.asoundrc: > > > > defaults.pcm.dmix.format S16_LE > > > > > > Takashi > > > > > > > > -----Original Message----- > > > From: stan [mailto:ghjeold_i_mwee@cox.net] > > > Sent: Tuesday, June 24, 2008 1:25 AM > > > To: Ashwin Chandra > > > Cc: alsa-devel(a)alsa-project.org > > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > > speaker only > > > > > > Ashwin.Chandra(a)corel.com wrote: > > > > > > > > > > > > Hello, > > > > > > > > I am having an issue with my playback application where upon > > playback > > > > start, only the left speaker is outputted. But if I mute and > unmute > > > > (either through my application or using alsamixer), the sound > > playback > > > > resumes on both left and right speakers. This only happens on a > > > > particular system too (out of 5 systems). But I know it is my > issue > > > > because mplayer or smplayer works fine. > > > > > > > > > > > > > > > > I believe I am setting the correct HW params (ie. 2 channels), > etc. > > > > > > > > > > > > > > > > Any ideas? > > > > > > > > > > > > > > > > Thanks! > > > > > > > > > > > The ICE1724 driver has exactly this behavior. It is a known bug and > > has > > > > > > been there for years. > > > You don't say what chip is having the problem, but I would guess > that > > > this is it. Most people > > > just program a script to adjust the volume at startup to cure it. > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > >

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[alsa-devel] ALSA 1.0.17rc2 release
by Jaroslav Kysela 26 Jun '08

26 Jun '08

Hello all, ALSA 1.0.17rc2 packages (except tools, oss, python - no changes) were released and are available for download on ALSA server. Changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.17rc1_v1.0.17rc2 The alsa-driver compilation with 2.4 kernels should be fixed in this release. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Takashi Iwai 26 Jun '08

26 Jun '08

At Thu, 26 Jun 2008 03:28:42 +0800, <Ashwin.Chandra(a)corel.com> wrote: > > Hi Takashi, > Actually now I am noticing the problem (left speaker only) with > mplayer/smplayer in their OSS emulation as well. It seems to happen once > I run my player first, then after that the other players exhibit the > problem. In this case, mplayer will choose the ALSA mode automatically, I guess. So the problem appears to be ALSA-mode specific, possibly about dmix. Try to define the following in ~/.asoundrc: pcm.!default "plughw" then it won't use dmix as default any more. (Then you can use only one app at the same time, of course.) Check whether the problem can be reproduced with this setting. Takashi > Sometimes after muting and unmuting my player, the problem goes > away and also away on the players. It is almost like my player is > causing bad internal state for the audio driver. Do you have any > suggestions to work around this? > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, June 25, 2008 3:23 AM > To: Ashwin Chandra > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > speaker only > > At Wed, 25 Jun 2008 01:52:16 +0800, > <Ashwin.Chandra(a)corel.com> wrote: > > > > I have an Intel ALC861VD. Other players (mplayer, smplayer, aplay) > don't > > seem to exhibit this problem. Is there any workaround I can do in my > > application so that upon first time sound playback, I can get it to > play > > on both channels? > > Check /proc/asound/card0/pcm0p/sub0/* files and compare between > working and non-working states. I guess the working case uses OSS > emulation or 16bit format. > > If so, try to add the following to ~/.asoundrc: > > defaults.pcm.dmix.format S16_LE > > > Takashi > > > > > -----Original Message----- > > From: stan [mailto:ghjeold_i_mwee@cox.net] > > Sent: Tuesday, June 24, 2008 1:25 AM > > To: Ashwin Chandra > > Cc: alsa-devel(a)alsa-project.org > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > speaker only > > > > Ashwin.Chandra(a)corel.com wrote: > > > > > > > > > Hello, > > > > > > I am having an issue with my playback application where upon > playback > > > start, only the left speaker is outputted. But if I mute and unmute > > > (either through my application or using alsamixer), the sound > playback > > > resumes on both left and right speakers. This only happens on a > > > particular system too (out of 5 systems). But I know it is my issue > > > because mplayer or smplayer works fine. > > > > > > > > > > > > I believe I am setting the correct HW params (ie. 2 channels), etc. > > > > > > > > > > > > Any ideas? > > > > > > > > > > > > Thanks! > > > > > > > > The ICE1724 driver has exactly this behavior. It is a known bug and > has > > > > been there for years. > > You don't say what chip is having the problem, but I would guess that > > this is it. Most people > > just program a script to adjust the volume at startup to cure it. > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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[alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Ashwin.Chandra＠corel.com 26 Jun '08

26 Jun '08

Hello, I am having an issue with my playback application where upon playback start, only the left speaker is outputted. But if I mute and unmute (either through my application or using alsamixer), the sound playback resumes on both left and right speakers. This only happens on a particular system too (out of 5 systems). But I know it is my issue because mplayer or smplayer works fine. I believe I am setting the correct HW params (ie. 2 channels), etc. Any ideas? Thanks!

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[alsa-devel] IBM X41 laptop, no headphone detection after s2ram cycle
by Dmitry E. Mikhailov 25 Jun '08

25 Jun '08

Hi, For about a year I got a following problem. After s2ram cycle I got only sound from internal speaker, even if the headphones are plugged in. s2disk (hibernate) restores the functionality. Sometimes (rare!) I got the functionality after s2ram. If I repeat s2ram, it stops working again. I compared ALSA registers before and after: they are all the same. It was Ok when I got Fedora7 with kernel 2.6.18 installed. But even if I restore 2.6.18 now, things don't go right. Before the Switch from 'another proprietary OS' it was also Ok. With Fedora9 I got a famous 'alsasound' service. It's restart doesn't help. Reloading modules doesn't help. Unload prior to s2ram and load after it doesn't help. Rolling ALSA back doesn't seem to help but I could do it wrong way. Tested kernels: 2.6.18.something, 2.6.19, 2.6.20, 2.6.20.4, 2.6.22, 2.6.22.something, 2.6.23, 2.6.23.6, 2.6.23.12, 2.6.23.12, 2.6.25, 2.6.25.3, 2.6.25.14 (current). Or it's a hardware failure? As a last resort I'll reinstall /\/\$ \/\/in to check. Best regards, Dmitry Mikhailov

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[alsa-devel] A few general questions (my first driver): Marian MARC4
by Joshua Boyd 25 Jun '08

25 Jun '08

I have a Marian Marc4 card. I have the documentation for this card, which has been on their web site for quite some time. I want to write a driver for this card. This is my first ALSA driver, but it isn't my first kernel driver. Anyway, I can't google any mention of this family of cards on this mailing list, which seems a bit odd. I did find some mentions on linux-audio-developer, about the documentation, but no mention there of prior work on this family. However, I did find a MARC8 driver on sourceforge (http:// sourceforge.net/projects/marc8-driver). Does anyone know anything about this? I'm vary curious about why I've never seen it mentioned here before. The license is listed as GPL. The Marian Marc8 was formerly known as the Sek'd Siena, but again I can't find any real mention of that in the archives. Does anyone have any idea why the MARC8 wasn't mentioned and merged into the main ALSA sources? I did go ahead and add some Marian product information to the support matrix on the alsa wiki, including a link to the sourceforge page. My plan is to add MARC4 to the MARC8 driver. The register layout is basically the same, adjusted for the 4 versus 8 channels. Both cards use the same PLX chip. The biggest difference is in the layout of the play/capture RAM area, which is because of the 4 versus 8 channels. It seems silly to make two separate drivers. I would like to see my results eventually merged into the main ALSA. I'm hoping that if I get these into the main tree that I would then be able to get some support from Marian in the form of clearance (and a card) to do the Trace D4 and D4 SRC, and possibly even documentation and hardware for the UCON CX. Anyway, moving on to my specific questions. First, is it appropriate to put support for two device IDs in the same file? The examples I tried looking at ended up being a single device id, differentiated by a subvendor ID. Second, is it wrong to use the same structures, but with an enum field in the main snd_marc device structure that points out which card I'm talking about, then else where switching on that enum to figure out how many channels I need to operate on? Third, once I get this functional on my MARC4 card, how hard is it likely to be to get it accepted? I don't expect low standards for code that is accepted, but I'd like to have some idea how picky the process may be.

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