[pulseaudio-discuss] PulseAudio automatically shutting down?
Hi, I am running pulseaudio 0.9.21-63-gd3efa-dirty from ubuntu 10.10 packages. I need to use pulseaudio in the server environment to perform audio routing tasks. I run it with minimal configuration: load-module module-null-sink sink_name=tuned_fake load-module module-rescue-streams load-module module-always-sink load-module module-intended-roles set-default-source tuned_fake.monitor set-default-sink tuned_fake It seems that if no client is connected, daemon exits after some (short, +/- 1 min) time. Is it correct behaviour? Thank you in advance, m. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio automatically shutting down?
'Twas brillig, and mar...@saepia.net at 10/05/11 12:35 did gyre and gimble: Hi, I am running pulseaudio 0.9.21-63-gd3efa-dirty from ubuntu 10.10 packages. I need to use pulseaudio in the server environment to perform audio routing tasks. I run it with minimal configuration: load-module module-null-sink sink_name=tuned_fake load-module module-rescue-streams load-module module-always-sink load-module module-intended-roles set-default-source tuned_fake.monitor set-default-sink tuned_fake It seems that if no client is connected, daemon exits after some (short, +/- 1 min) time. Is it correct behaviour? Thank you in advance, Yup! man pulse-daemon.conf and see exit-idle-time :D HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio automatically shutting down?
whoops, it so obvious that I missed that :P thank you! m. 2011/5/10 Colin Guthrie gm...@colin.guthr.ie: 'Twas brillig, and mar...@saepia.net at 10/05/11 12:35 did gyre and gimble: Hi, I am running pulseaudio 0.9.21-63-gd3efa-dirty from ubuntu 10.10 packages. I need to use pulseaudio in the server environment to perform audio routing tasks. I run it with minimal configuration: load-module module-null-sink sink_name=tuned_fake load-module module-rescue-streams load-module module-always-sink load-module module-intended-roles set-default-source tuned_fake.monitor set-default-sink tuned_fake It seems that if no client is connected, daemon exits after some (short, +/- 1 min) time. Is it correct behaviour? Thank you in advance, Yup! man pulse-daemon.conf and see exit-idle-time :D HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
Thanks Sean and Marten, this has answered my question, very much appreciated. I had hoped to send my local machine's microphone (for voice) over the Internet to the remote machine, but through all the various configuration I've done the delay/latency problem persists. Ah well, I had fun trying! Many thanks again, Cheers, Nick -Original Message- From: pulseaudio-discuss-boun...@mail.0pointer.de [mailto:pulseaudio-discuss-boun...@mail.0pointer.de] On Behalf Of Maarten Bosmans Sent: Sunday, April 17, 2011 9:24 PM To: General PulseAudio Discussion Subject: Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem 2011/4/17 Sean McNamara smc...@gmail.com: I don't know if pulseaudio supports any kind of protocol compression these days, but traditionally it does not. And due to that, it is generally unsuitable for use over the public internet. Lossy compression such as mp3, and protocols such as RTP and Icecast, exist for this purpose. Even if both nodes are dedicated servers with symmetric 100mbps, your transport latency over the internet is too high for most uses of PA. Indeed, it is still the case that only raw PCM audio streaming is supported. For CD quality audio this means a 44100Hz x 16bit x 2ch = 1.4 Mbit/s bandwidth requirement. I've heard about lots of interest in extending the PA protocol to support lossy, non-PCM formats, but I don't *think* that has been added quite yet. The CELT codec would be the best candidate for that, as it is specifically designed for low-latency requirements. Maarten ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
2011/4/18 Nick Holloway ourm...@hotmail.com: Thanks Sean and Marten, this has answered my question, very much appreciated. I had hoped to send my local machine's microphone (for voice) over the Internet to the remote machine, but through all the various configuration I've done the delay/latency problem persists. Ah well, I had fun trying! Then either use some VOIP application or setup a icecast server. Many thanks again, Cheers, Nick ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
Thanks again Marteen, I'll follow your advice. My goal has been to take the microphone output from the local machine and relay it to voice recognition software on the remote server... the voice recognition software (Dragon Naturally Speaking) insists on sampling the speech from the mic-in on that machine (the remote machine)... so I'll certainly investigate options around Icecast and VOIP options, to see whether they can relay sound from the local mic so that it appears to be coming from the sound card on the remote machine (ie that it looks like the local mic to the voice recognition software). Thanks for casting your mind across this, many thanks. cheers, Nick Date: Mon, 18 Apr 2011 14:10:37 +0200 From: mkbosm...@gmail.com To: pulseaudio-discuss@mail.0pointer.de Subject: Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem 2011/4/18 Nick Holloway ourm...@hotmail.com: Thanks Sean and Marten, this has answered my question, very much appreciated. I had hoped to send my local machine's microphone (for voice) over the Internet to the remote machine, but through all the various configuration I've done the delay/latency problem persists. Ah well, I had fun trying! Then either use some VOIP application or setup a icecast server. Many thanks again, Cheers, Nick ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
Hi, I'm very new to Pulseaudio (and linux generally), so apologies if my query is a bit muddled . I have spent the past few days successfully setting up Pulseaudio to relay bi-directional sound from my Debian machine with sound hardware to a virtual machine without sound hardware (Virtualbox), all on my LAN. I did this by installing Pulseaudio on both machines, then running padevchooser on the virtual machine and configuring the server, sink and source to point towards my Debian machine with sound hardware. All works perfectly, both playing and recording sound. Having achieved that I then replicated exactly the same setup, but this time from my local Debian machine to a hosted server on the Internet, over my cable broadband connection (10 Mb downstream, 512K upstream). The sound plays fine FROM the remote hosted server on my local Debian machine's speakers (via that fast 10 Mb connection) , but when I try to record TO the server (using my local machine's recording device relayed by Pulseaudio) then I hit a latency problem (via that much slower 512K upstream). In this scenario, if I open the recording device volume monitor on the remote server, it picks up a very small amount of audio initially, then the volume bursts after a short period, then slowly fails to zero, and then I receive an error message detailing the latency problem, and it crashes. Presumably this is related to bandwidth. My question is: Is there a way to configure Pulseaudio so that it perhaps compresses the stream from the local recording device *before* sending it over the Internet to the remote server? Might this get around the bandwidth/latency problem? Or perhaps there is another way of resolving this kind of problem? The default for the sampling rate on my local machine's recording device is 2 channel, 16 bit, 44100Hz... (I looked for a way to reduce that sample rate down as a possible solution in the first instance, but haven't yet worked out how to do that). I'll provide any other information of the machines, configuration etc... if that will assist. Thanks for any help offered. Cheers, Nick ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
Hi, On Apr 17, 2011 8:10 AM, Nick Holloway ourm...@hotmail.com wrote: Hi, I'm very new to Pulseaudio (and linux generally), so apologies if my query is a bit muddled . I have spent the past few days successfully setting up Pulseaudio to relay bi-directional sound from my Debian machine with sound hardware to a virtual machine without sound hardware (Virtualbox), all on my LAN. I did this by installing Pulseaudio on both machines, then running padevchooser on the virtual machine and configuring the server, sink and source to point towards my Debian machine with sound hardware. All works perfectly, both playing and recording sound. Having achieved that I then replicated exactly the same setup, but this time from my local Debian machine to a hosted server on the Internet, over my cable broadband connection (10 Mb downstream, 512K upstream). The sound plays fine FROM the remote hosted server on my local Debian machine's speakers (via that fast 10 Mb connection) , but when I try to record TO the server (using my local machine's recording device relayed by Pulseaudio) then I hit a latency problem (via that much slower 512K upstream). In this scenario, if I open the recording device volume monitor on the remote server, it picks up a very small amount of audio initially, then the volume bursts after a short period, then slowly fails to zero, and then I receive an error message detailing the latency problem, and it crashes. Presumably this is related to bandwidth. My question is: Is there a way to configure Pulseaudio so that it perhaps compresses the stream from the local recording device *before* sending it over the Internet to the remote server? Might this get around the bandwidth/latency problem? Or perhaps there is another way of resolving this kind of problem? The default for the sampling rate on my local machine's recording device is 2 channel, 16 bit, 44100Hz... (I looked for a way to reduce that sample rate down as a possible solution in the first instance, but haven't yet worked out how to do that). I don't know if pulseaudio supports any kind of protocol compression these days, but traditionally it does not. And due to that, it is generally unsuitable for use over the public internet. Lossy compression such as mp3, and protocols such as RTP and Icecast, exist for this purpose. Even if both nodes are dedicated servers with symmetric 100mbps, your transport latency over the internet is too high for most uses of PA. I've heard about lots of interest in extending the PA protocol to support lossy, non-PCM formats, but I don't *think* that has been added quite yet. PA shouldn't have crashed for you though, so if you can get a backtrace from gdb with debugging symbols and provide steps to reproduce, I'm sure someone could triage why it happened. I'll provide any other information of the machines, configuration etc... if that will assist. Thanks for any help offered. Cheers, Nick ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio recording device re-direct from local machine over cable broadband to remote server / latency problem
2011/4/17 Sean McNamara smc...@gmail.com: I don't know if pulseaudio supports any kind of protocol compression these days, but traditionally it does not. And due to that, it is generally unsuitable for use over the public internet. Lossy compression such as mp3, and protocols such as RTP and Icecast, exist for this purpose. Even if both nodes are dedicated servers with symmetric 100mbps, your transport latency over the internet is too high for most uses of PA. Indeed, it is still the case that only raw PCM audio streaming is supported. For CD quality audio this means a 44100Hz x 16bit x 2ch = 1.4 Mbit/s bandwidth requirement. I've heard about lots of interest in extending the PA protocol to support lossy, non-PCM formats, but I don't *think* that has been added quite yet. The CELT codec would be the best candidate for that, as it is specifically designed for low-latency requirements. Maarten ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] PulseAudio Developer's Meetup at LAC
(Reposting due to this being embedded in an old thread) I have worked out a separate room for us to meet at LAC from 2:00-4:00 pm on Saturday. It's official, we're on the agenda! http://lac.linuxaudio.org/2011/?page=programmode=tableday=2 I need to get back to the organizers with a purpose and agenda. I am proposing the following: PulseAudio Developer's Meetup and Working Session (2-4pm Sat, May 7th, LAC - O'Callaghan Room) - Meetup and brief introductions - Release content and schedule - Technical discussions - Coding breakouts (if time permits) Please respond with comments and suggestions as soon as possible. And, if you haven't already, let me know if you are planning on attending. Thanks! Kurt Taylor (irc krtaylor) ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio Developer's Meetup at LAC
'Twas brillig, and Kurt Taylor at 15/04/11 15:06 did gyre and gimble: (Reposting due to this being embedded in an old thread) I have worked out a separate room for us to meet at LAC from 2:00-4:00 pm on Saturday. It's official, we're on the agenda! http://lac.linuxaudio.org/2011/?page=programmode=tableday=2 http://lac.linuxaudio.org/2011/?page=programmode=tableday=2 I need to get back to the organizers with a purpose and agenda. I am proposing the following: PulseAudio Developer's Meetup and Working Session (2-4pm Sat, May 7th, LAC - O'Callaghan Room) - Meetup and brief introductions - Release content and schedule - Technical discussions - Coding breakouts (if time permits) Please respond with comments and suggestions as soon as possible. And, if you haven't already, let me know if you are planning on attending. Great! That's ideal. I think the agenda you have there is fine. I think keeping it relatively free form is fine :) Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio
Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio
Hi, On Tue, Apr 12, 2011 at 12:15 PM, duportail po...@telenet.be wrote: Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? This is the classic multi-user problem: you're using a soundcard without hardware mixing, so only one ALSA application can take direct control of the hardware device at a time. user1's pulseaudio daemon takes control of it, so obviously user2's PA daemon will just get Device or resource busy when trying to snd_pcm_open() on hw:0 (for example). This problem has existed since ALSA existed. There's no direct fix without scrapping the API and rewriting it. I guess there is no solution yet (still) in 11.04, but you can see that folks in the Ubuntu community are trying to resolve the issue with a number of possible approaches: https://wiki.edubuntu.org/BluePrints/multiuser-soundcards-pulseaudio Running as system-wide is one possible approach... http://www.pulseaudio.org/wiki/SystemWideInstance Not sure if there's yet a solution allowing the use of shm and module-native-protocol-unix for multiple users (this is ideally what you'd want for maximum performance / minimum latency), but it's quite easy to set up module-native-protocol-tcp and connect to localhost using the `default-server' parameter in client.conf, at the cost of latency. Let me know if you need more details on this particular approach. BTW, search the archives of this list at gmane http://blog.gmane.org/gmane.comp.audio.pulseaudio.general -- there are many many historical posts about multi-user setups. Sean gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio
Op 12-4-2011 18:30, Sean McNamara schreef: Hi, On Tue, Apr 12, 2011 at 12:15 PM, duportailpo...@telenet.be wrote: Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? This is the classic multi-user problem: you're using a soundcard without hardware mixing, so only one ALSA application can take direct control of the hardware device at a time. user1's pulseaudio daemon takes control of it, so obviously user2's PA daemon will just get Device or resource busy when trying to snd_pcm_open() on hw:0 (for example). This problem has existed since ALSA existed. There's no direct fix without scrapping the API and rewriting it. I guess there is no solution yet (still) in 11.04, but you can see that folks in the Ubuntu community are trying to resolve the issue with a number of possible approaches: https://wiki.edubuntu.org/BluePrints/multiuser-soundcards-pulseaudio Running as system-wide is one possible approach... http://www.pulseaudio.org/wiki/SystemWideInstance Not sure if there's yet a solution allowing the use of shm and module-native-protocol-unix for multiple users (this is ideally what you'd want for maximum performance / minimum latency), but it's quite easy to set up module-native-protocol-tcp and connect to localhost using the `default-server' parameter in client.conf, at the cost of latency. Let me know if you need more details on this particular approach. BTW, search the archives of this list at gmane http://blog.gmane.org/gmane.comp.audio.pulseaudio.general -- there are many many historical posts about multi-user setups. Sean gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss Sorry, forgot to mention: there multiple (two) usb soundcards on the system.It is working good on xubuntu 10.10. As you said, the second user gets a Device or resource busy.When I logout the first user, the second user can get a default sink. gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio
On Tue, 2011-04-12 at 12:30 -0400, Sean McNamara wrote: Hi, On Tue, Apr 12, 2011 at 12:15 PM, duportail po...@telenet.be wrote: Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? This is the classic multi-user problem: you're using a soundcard without hardware mixing, so only one ALSA application can take direct control of the hardware device at a time. user1's pulseaudio daemon takes control of it, so obviously user2's PA daemon will just get Device or resource busy when trying to snd_pcm_open() on hw:0 (for example). This problem has existed since ALSA existed. There's no direct fix without scrapping the API and rewriting it. Actually, this situation is supposed to be handled just fine (unless duportail expects both users to be able to use the audio hw simultaneously, which he doesn't mention). If this doesn't work, xubuntu is broken. Some things that can cause this problem: the users are in the audio group, consolekit isn't running or consolekit's udev-acl isn't in use. -- Tanu ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio
Op 12-4-2011 18:53, Tanu Kaskinen schreef: On Tue, 2011-04-12 at 12:30 -0400, Sean McNamara wrote: Hi, On Tue, Apr 12, 2011 at 12:15 PM, duportailpo...@telenet.be wrote: Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? This is the classic multi-user problem: you're using a soundcard without hardware mixing, so only one ALSA application can take direct control of the hardware device at a time. user1's pulseaudio daemon takes control of it, so obviously user2's PA daemon will just get Device or resource busy when trying to snd_pcm_open() on hw:0 (for example). This problem has existed since ALSA existed. There's no direct fix without scrapping the API and rewriting it. Actually, this situation is supposed to be handled just fine (unless duportail expects both users to be able to use the audio hw simultaneously, which he doesn't mention). If this doesn't work, xubuntu is broken. Some things that can cause this problem: the users are in the audio group, consolekit isn't running or consolekit's udev-acl isn't in use. Maybe xubuntu is broken.I will wait until further releases gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio
'Twas brillig, and duportail at 12/04/11 19:20 did gyre and gimble: Op 12-4-2011 18:53, Tanu Kaskinen schreef: On Tue, 2011-04-12 at 12:30 -0400, Sean McNamara wrote: Hi, On Tue, Apr 12, 2011 at 12:15 PM, duportailpo...@telenet.be wrote: Try to test the new xubuntu 11.04 with kde on a multi-user system. The first user (for example user1) can log in and gets a default sink from pulse. The second user(for example user2) that will log in gets a auto_null because pulse did not found any sinks. If i log out user1, than user2 can get a default sink by kill and start pulse. Where should be the reason? This is the classic multi-user problem: you're using a soundcard without hardware mixing, so only one ALSA application can take direct control of the hardware device at a time. user1's pulseaudio daemon takes control of it, so obviously user2's PA daemon will just get Device or resource busy when trying to snd_pcm_open() on hw:0 (for example). This problem has existed since ALSA existed. There's no direct fix without scrapping the API and rewriting it. Actually, this situation is supposed to be handled just fine (unless duportail expects both users to be able to use the audio hw simultaneously, which he doesn't mention). If this doesn't work, xubuntu is broken. Some things that can cause this problem: the users are in the audio group, consolekit isn't running or consolekit's udev-acl isn't in use. Maybe xubuntu is broken.I will wait until further releases Make sure that ck-list-sessions reports correctly that only the active user is actually active. Also check the user groups as Tanu reported. But all in all, this *should* be handled correctly (and certainly is on other systems) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio 0.9.22, after system suspends and resumes, alsasink is blocked
Running into a weird issue. I am doing the following on 0.9.22 version of pulseaudio start pulseaudio, and play and audio file. when the file is done playing, after 5 seconds, the alsa sink suspends. then I let the system go to power collapse. When I resume and play an audio file again, alsa sink is blocked somewhere, I do not see the message I: alsa-sink.c: Trying resume... The PA_SINK_SET_STATE message doesn't seem to be called to the alsa sink. Alsa is not blocked at this point, I can do an aplay to hw:0 and hear the output, bypassing pulseaudio. here is an example from pulseaudio with debug messages enabled, after suspend/resume: aplay -D media /usr/palm/sounds/phone.wav -vvv root@palm-webos:/var/home/root# I: client.c: Created 2 Native client (UNIX socket client) D: protocol-native.c: Protocol version: remote 19, local 19 I: protocol-native.c: Got credentials: uid=0 gid=0 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: no Playing WAVE '/usr/palm/sounds/phone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo D: module-suspend-on-idle.c: Sink pcm_output becomes busy. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 0 ALSA Playback on pcm_output with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: sink-input.c: media.name = ALSA Playback I: sink-input.c: application.name = ALSA plug-in [aplay] I: sink-input.c: native-protocol.peer = UNIX socket client I: sink-input.c: native-protocol.version = 19 I: sink-input.c: application.process.id = 3593 I: sink-input.c: application.process.user = root I: sink-input.c: application.process.host = palm-webos I: sink-input.c: application.process.binary = aplay I: sink-input.c: application.language = C I: sink-input.c: application.process.machine_id = palm-webos I: protocol-native.c: Requested tlength=500.00 ms, minreq=124.99 ms D: protocol-native.c: Early requests mode enabled, configuring sink latency to minreq. D: memblockq.c: memblockq requested: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=9596 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=9596 maxrewind=0 I: protocol-native.c: Final latency 554.42 ms = 391.20 ms + 2*54.40 ms + 54.42 ms D: sink-input.c: SetVolumeWithRamping: Virtual Volume From 1646=0.16 to 41160=0.247734 D: sink-input.c: Sink input's soft volume is 41160= 0.247734 D: sink-input.c: Volume Ramping: Point 1 is 1=0.15, Point 2 is 16236=0.247742 I: module-palm-policy.c: parse_message: ramp command received, sink is 4, volumetoset:70, headphones:0 ALSA - PulseAudio PCM I/O Plugin Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat: STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min: 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Max peak (11024 samples): 0x2dd0 35% Max peak (11024 samples): 0x332c 39% Max peak (11024 samples): 0x3741 #43% D: protocol-native.c: Requesting rewind due to end of underrun. D: protocol-native.c: Requesting rewind due to end of underrun. Max peak (11024 samples): 0x353f #41% D: protocol-native.c: Requesting rewind due to end of underrun. D: protocol-native.c: Requesting rewind due to end of underrun. Just waits here forever. Sink doesnt get resume call, and alsasink doesn't get resumed either Any ideas? -- -baeksanchang ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio start stop
I am setting the default sink for the user at log-in according the display he is on. First time login is ok, but if the user logs out and logs in at another display, pulse does not start with error (sorry, in dutch) E: socket-server.c: bind(): Adres is al in gebruik E: module.c: Failed to load module module-esound-protocol-unix (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Initialiseren van de daemon mislukt. Can I do something at the user logout? gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio start stop
Op 6-4-2011 13:14, duportail schreef: I am setting the default sink for the user at log-in according the display he is on. First time login is ok, but if the user logs out and logs in at another display, pulse does not start with error (sorry, in dutch) E: socket-server.c: bind(): Adres is al in gebruik E: module.c: Failed to load module module-esound-protocol-unix (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Initialiseren van de daemon mislukt. Can I do something at the user logout? gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss Ok, found it.Removing the /tmp/.esd(uid) did it.But why is this left ,after the user is logged out? gd ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio start stop
'Twas brillig, and duportail at 06/04/11 12:36 did gyre and gimble: Op 6-4-2011 13:14, duportail schreef: I am setting the default sink for the user at log-in according the display he is on. First time login is ok, but if the user logs out and logs in at another display, pulse does not start with error (sorry, in dutch) E: socket-server.c: bind(): Adres is al in gebruik E: module.c: Failed to load module module-esound-protocol-unix (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Initialiseren van de daemon mislukt. Can I do something at the user logout? gd Ok, found it.Removing the /tmp/.esd(uid) did it.But why is this left ,after the user is logged out? When you say /tmp/.esd(uid) what do you mean? On my system we use /tmp/.esd-uid/. This prevents one user from clashing with another (although that doesn't prevent malicious DoS by another user just creating lots of folder names - this is one of the problems with ESD socket path - we solve that in PA and now in systemd with some clever logic). Anyway, in the very old days, the ESD socket was just /tmp/.esd which totally sucked in a multi-user environment. Most distros patch libesound to make it /tmp/.esd-$UID but that's still sucky for deliberate DoS as noted above. PA has the option to emulate either structure of socket paths via configure switches. I suggest you ensure that your libesound and pulseaudio packages are in sync in this regard. As your distro maintainer to speak with us on IRC or via replying to this thread if you like. Alternatively, this problem could be simply that you are not using PID files and you're trying to start two pulseaudio daemons for the same user. This is not supported. PA is a per-user process. The user should only run one instance of PA. PID files typically prevent this, but this can be disabled. Either way something is not quite right on your system to be causing this problem and I hope I've pointed you in the right direction for debugging. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio start stop
[Please keep the list CC'ed on replies so other people can help and benefit form the advice and debugging process!] 'Twas brillig, and duportail at 06/04/11 14:29 did gyre and gimble: Op 6-4-2011 15:18, Colin Guthrie schreef: 'Twas brillig, and duportail at 06/04/11 12:36 did gyre and gimble: Op 6-4-2011 13:14, duportail schreef: I am setting the default sink for the user at log-in according the display he is on. First time login is ok, but if the user logs out and logs in at another display, pulse does not start with error (sorry, in dutch) E: socket-server.c: bind(): Adres is al in gebruik E: module.c: Failed to load module module-esound-protocol-unix (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Initialiseren van de daemon mislukt. Can I do something at the user logout? gd Ok, found it.Removing the /tmp/.esd(uid) did it.But why is this left ,after the user is logged out? When you say /tmp/.esd(uid) what do you mean? On my system we use /tmp/.esd-uid/. This prevents one user from clashing with another (although that doesn't prevent malicious DoS by another user just creating lots of folder names - this is one of the problems with ESD socket path - we solve that in PA and now in systemd with some clever logic). Anyway, in the very old days, the ESD socket was just /tmp/.esd which totally sucked in a multi-user environment. Most distros patch libesound to make it /tmp/.esd-$UID but that's still sucky for deliberate DoS as noted above. PA has the option to emulate either structure of socket paths via configure switches. I suggest you ensure that your libesound and pulseaudio packages are in sync in this regard. As your distro maintainer to speak with us on IRC or via replying to this thread if you like. Alternatively, this problem could be simply that you are not using PID files and you're trying to start two pulseaudio daemons for the same user. This is not supported. PA is a per-user process. The user should only run one instance of PA. PID files typically prevent this, but this can be disabled. Either way something is not quite right on your system to be causing this problem and I hope I've pointed you in the right direction for debugging. Correct,it's /tmp/.esd-$UID/socket Removing this upon re-login of the user at an other display,makes the pulseaudio starting. So the same user is logging in on separate displays? OK, there are some fundamental things here you need to appreciate: 1. PA is per-user, not per-display, there should only ever be one pulseaudio daemon running. Please confirm this. 2. You should not have to remove the socket (indeed another instance of pulseaudio will already be running and have this socket open so it's generally bad practice to remove it anyway (it'll still be kept as an open file handle until the PA process is stopped)). If another process is using the socket, find out which process it is and work out how to prevent it running. Ensure that only one PA process is running, but try and find out what is causing your esound error... How do you start PA? Is there any special script you use? 3. pacmd (which you use on your script) works with a daemon, not a display. As you should only have one daemon running per-user, running pacmd when the same user logs in on another display will change the default for all displays. You should really set PULSE_SINK property on the X11 root window instead: e.g. xprop -root -f PULSE_SINK 8s -set PULSE_SINK alsa_output.usb-0d8c_C-Media_USB_Audio_Device-00-default_1.analog-stereo 4. Multiple logins for the same user on different displays is something that is generally being deprecated, so you should likely avoid this in your setup. 5. I'm not sure what you are trying to acheive, but it seems like some kind of multi-seat setup? If this is the case you should configure you seats better. IMO, what you should do is allocate a (different) user to each seat, then write your udev rules such that certain USB ports are allocated to each user (or rather each seat, but with a 1:1 mapping this doesn't matter). Then start a separate PA daemon for each user (and thus each seat). Due to the udev permissions, each individual user should only have access to their own USB port and thus they will each only ever see one USB sound card each. All the problems are solved! I've not gone into too much detail as I don't know the intricate details, but hopefully this helps you in some small way. Col [Original reply left below for the benefit of the list] I use such bash code to assign sink to a display: if [ $DISPLAY ] then if [ $DISPLAY = :1.0 ] then /usr/local/sbin/uid.sh pacmd set-default-sink alsa_output.usb-0d8c_C-Media_USB_Audio_Device-00-default.analog-stereo fi if [ $DISPLAY = :2.0 ] then /usr/local/sbin/uid.sh pacmd set-default-sink alsa_output.usb-1130_USB_AUDIO-00-default_3.analog-stereo fi if [ $DISPLAY =
Re: [pulseaudio-discuss] pulseaudio start stop
Hello, *THIS IS IMPORTANT* As I stated last time, please keep the list CC'ed on replies. 'Twas brillig, and duportail at 06/04/11 15:24 did gyre and gimble: Correct,this is for a multiseat setup.However,the same user cannot login twice on the same computer. What I mean is:when I log in a user on ,say, display 1,(this with kde),pulseaudio runs and the deault sink is set as in the script. If I logout this user, and log him in on another display, pulse is not starting because of the /temp/.esd-$UID. Does the same user have a different home directory, or is their home directory cleaned out by some other script? If this is the case, then I suspect that PulseAudio is *still running* from the previous session, but you've trashed it's runtime dir and pid files, thus it doesn't know it's still running and it tries to start again, but the old process still has /tmp/.esd-$UID/socket open and thus prevents the other PA process from using it. PA will not exit immediately after logout but will adhere to it's exit-idle-timeout value (see /etc/pulse/daemon.conf). If you set this to a suitably low value, then this should avoid the issue. When I remove this, just before the user logs in on another display, all runs well. If you prefer, you can comment out module-esound-protocol in /etc/pulse/default.pa as this will prevent the loading of esound support and thus the opening of this file. However, if my suspicion (above) is correct, then there could be other problems (like the old pulseaudio still having the h/w open and thus hogging it from the new one) and you should probably ensure PA is properly killed on logout (or before /home/$USER is reset). HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio start stop
Op 6-4-2011 16:50, Colin Guthrie schreef: Hello, *THIS IS IMPORTANT* As I stated last time, please keep the list CC'ed on replies. 'Twas brillig, and duportail at 06/04/11 15:24 did gyre and gimble: Correct,this is for a multiseat setup.However,the same user cannot login twice on the same computer. What I mean is:when I log in a user on ,say, display 1,(this with kde),pulseaudio runs and the deault sink is set as in the script. If I logout this user, and log him in on another display, pulse is not starting because of the /temp/.esd-$UID. Does the same user have a different home directory, or is their home directory cleaned out by some other script? If this is the case, then I suspect that PulseAudio is *still running* from the previous session, but you've trashed it's runtime dir and pid files, thus it doesn't know it's still running and it tries to start again, but the old process still has /tmp/.esd-$UID/socket open and thus prevents the other PA process from using it. PA will not exit immediately after logout but will adhere to it's exit-idle-timeout value (see /etc/pulse/daemon.conf). If you set this to a suitably low value, then this should avoid the issue. When I remove this, just before the user logs in on another display, all runs well. If you prefer, you can comment out module-esound-protocol in /etc/pulse/default.pa as this will prevent the loading of esound support and thus the opening of this file. However, if my suspicion (above) is correct, then there could be other problems (like the old pulseaudio still having the h/w open and thus hogging it from the new one) and you should probably ensure PA is properly killed on logout (or before /home/$USER is reset). HTHs Col Yes, this is correct, after a time the /tmp/.esd-$UID/socket is going away. The home dir of the user is cleaned (reset) just before login (with gdm).So the /home/$USER/.pulse dir is empty. The users' KDE autostart function runs the pulse script and the default sink is set. This way the user can take place at no matter what display he chooses ,log in and have sound. If I assign a pulse-audio-sink(usb-audio) per user, than the user must always login at the same display(thus where this usb-sound card is) It is very important to assing a sink to a display: in a school with 500 students and multiple multiseatcomputers, the students must be able to take place at no matter what display and no matter what multiseatcomputer and have sound.. I just have to find a way that pulse not choose the default soundcard,when a soundcard is defective. gd, ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
I have successfully bitstreamed DTS-HD from xbmc through PA but in using the passthrough git branch it doesn't read that config file which causes problems for me forcing a sink. How can I fix this since I have to kill pulseaudio then restart it with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get hdmi audio working I am running Ubuntu 11.04 Alpha. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
On Sun, Mar 20, 2011 at 5:58 PM, Dark Shadow shadowofdarkn...@gmail.com wrote: I have successfully bitstreamed DTS-HD from xbmc through PA but in using the passthrough git branch it doesn't read that config file which causes problems for me forcing a sink. How can I fix this since I have to kill pulseaudio then restart it with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get hdmi audio working I am running Ubuntu 11.04 Alpha. I figured it out, I copied default.pa into ~/.pulse and it works now. Not preferential in a multi-user environment but it is fine for now. With the feature of lasting through a distro upgrade. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
'Twas brillig, and Maarten Bosmans at 23/03/11 18:36 did gyre and gimble: 2011/3/21 Dark Shadow shadowofdarkn...@gmail.com: On Sun, Mar 20, 2011 at 5:58 PM, Dark Shadow shadowofdarkn...@gmail.com wrote: I have successfully bitstreamed DTS-HD from xbmc through PA but in using the passthrough git branch it doesn't read that config file which causes problems for me forcing a sink. How can I fix this since I have to kill pulseaudio then restart it with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get hdmi audio working I am running Ubuntu 11.04 Alpha. I figured it out, I copied default.pa into ~/.pulse and it works now. Not preferential in a multi-user environment but it is fine for now. With the feature of lasting through a distro upgrade. That's a workaround, but pulse still should read the system-wide startup script (note that that is different from the config file) What does pulseaudio --dump-config give? Am I missing something here... default.pa is the startup script and if it exists in ~/.pulse it overrides the one in /etc/pulse/. The system-wide one won't be read at all. Even with the daemon.conf file, the same logic is true (even if it would in theory be possible to read both, with the local file overriding the system wide one on a setting-by-setting basis) Perhaps I'm misunderstanding tho' Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
On 03/23/11 17:20, Colin Guthrie wrote: 'Twas brillig, and Maarten Bosmans at 23/03/11 18:36 did gyre and gimble: 2011/3/21 Dark Shadowshadowofdarkn...@gmail.com: On Sun, Mar 20, 2011 at 5:58 PM, Dark Shadowshadowofdarkn...@gmail.com wrote: I have successfully bitstreamed DTS-HD from xbmc through PA but in using the passthrough git branch it doesn't read that config file which causes problems for me forcing a sink. How can I fix this since I have to kill pulseaudio then restart it with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get hdmi audio working I am running Ubuntu 11.04 Alpha. I figured it out, I copied default.pa into ~/.pulse and it works now. Not preferential in a multi-user environment but it is fine for now. With the feature of lasting through a distro upgrade. That's a workaround, but pulse still should read the system-wide startup script (note that that is different from the config file) What does pulseaudio --dump-config give? Am I missing something here... default.pa is the startup script and if it exists in ~/.pulse it overrides the one in /etc/pulse/. The system-wide one won't be read at all. Even with the daemon.conf file, the same logic is true (even if it would in theory be possible to read both, with the local file overriding the system wide one on a setting-by-setting basis) Perhaps I'm misunderstanding tho' Col I'm pretty sure what he means is that daemon.conf isn't being read from /etc/pulse when he doesn't have a ~.pulse/daemon.conf. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
On Wed, Mar 23, 2011 at 5:20 PM, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and Maarten Bosmans at 23/03/11 18:36 did gyre and gimble: 2011/3/21 Dark Shadow shadowofdarkn...@gmail.com: On Sun, Mar 20, 2011 at 5:58 PM, Dark Shadow shadowofdarkn...@gmail.com wrote: I have successfully bitstreamed DTS-HD from xbmc through PA but in using the passthrough git branch it doesn't read that config file which causes problems for me forcing a sink. How can I fix this since I have to kill pulseaudio then restart it with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get hdmi audio working I am running Ubuntu 11.04 Alpha. I figured it out, I copied default.pa into ~/.pulse and it works now. Not preferential in a multi-user environment but it is fine for now. With the feature of lasting through a distro upgrade. That's a workaround, but pulse still should read the system-wide startup script (note that that is different from the config file) What does pulseaudio --dump-config give? Am I missing something here... default.pa is the startup script and if it exists in ~/.pulse it overrides the one in /etc/pulse/. The system-wide one won't be read at all. Even with the daemon.conf file, the same logic is true (even if it would in theory be possible to read both, with the local file overriding the system wide one on a setting-by-setting basis) Perhaps I'm misunderstanding tho' Col Pulse stopped reading /etc/pulse before I put the file into ~/.pulse. Here is everything I did Install Ubuntu edit /etc/pulse/default.pa to add my hdmi sink ...everything works for awhile... update to the passthrough git branch to get dts-hd passthrough with xbmc my sink is gone and /etc/pulse/default.pa isn't being read copy just that file into ~/.pulse and the sink starts working again I don't even have a daemon.conf in ~/.pulse ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
'Twas brillig, and Dark Shadow at 23/03/11 23:30 did gyre and gimble: Pulse stopped reading /etc/pulse before I put the file into ~/.pulse. Here is everything I did Install Ubuntu edit /etc/pulse/default.pa to add my hdmi sink ...everything works for awhile... update to the passthrough git branch to get dts-hd passthrough with xbmc my sink is gone and /etc/pulse/default.pa isn't being read copy just that file into ~/.pulse and the sink starts working again I don't even have a daemon.conf in ~/.pulse Ahh that's probably because you compiled the branch that way. You have to pass --sysconfdir=/etc to configure for it to look in that folder. By default (i.e. no arguments to configure) it will be /usr/local/etc Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa
On Wed, Mar 23, 2011 at 5:44 PM, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and Dark Shadow at 23/03/11 23:30 did gyre and gimble: Pulse stopped reading /etc/pulse before I put the file into ~/.pulse. Here is everything I did Install Ubuntu edit /etc/pulse/default.pa to add my hdmi sink ...everything works for awhile... update to the passthrough git branch to get dts-hd passthrough with xbmc my sink is gone and /etc/pulse/default.pa isn't being read copy just that file into ~/.pulse and the sink starts working again I don't even have a daemon.conf in ~/.pulse Ahh that's probably because you compiled the branch that way. You have to pass --sysconfdir=/etc to configure for it to look in that folder. By default (i.e. no arguments to configure) it will be /usr/local/etc Col Thanks that must be it since I just used the basic --prefix=/usr ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
'Twas brillig, and Kelly Anderson at 18/03/11 21:10 did gyre and gimble: Good news :) I've just watched District 9 and Master and Commander in full dts-hd. A few minor glitches (a couple of pops, I think it might be buffer underruns), other than that it worked great! One thing I did notice with Master and Commander was that with standard dts the dialog was not nearly as clear as it was with dts-hd. dts-hd has about 4 times the bandwidth. I'll be putting up the newer code sometime tomorrow. I've been working pretty long hours on it and now I just want to relax a bit. Great news! Enjoy the relax :) -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
On Sun, 2011-03-20 at 11:09 +, Colin Guthrie wrote: 'Twas brillig, and Kelly Anderson at 18/03/11 21:10 did gyre and gimble: Good news :) I've just watched District 9 and Master and Commander in full dts-hd. A few minor glitches (a couple of pops, I think it might be buffer underruns), other than that it worked great! One thing I did notice with Master and Commander was that with standard dts the dialog was not nearly as clear as it was with dts-hd. dts-hd has about 4 times the bandwidth. I'll be putting up the newer code sometime tomorrow. I've been working pretty long hours on it and now I just want to relax a bit. Great news! Enjoy the relax :) OT: I'd call watching District 9 and Master and Commander relaxing, wouldn't you? =p ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
Dnia 2011-03-20, nie o godzinie 19:38 +0800, Ng Oon-Ee pisze: OT: I'd call watching District 9 and Master and Commander relaxing, wouldn't you? =p ReOT: No, watching District 9 was torture. :P -- Michał Sawicz mic...@sawicz.net signature.asc Description: This is a digitally signed message part ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
Good news :) I've just watched District 9 and Master and Commander in full dts-hd. A few minor glitches (a couple of pops, I think it might be buffer underruns), other than that it worked great! One thing I did notice with Master and Commander was that with standard dts the dialog was not nearly as clear as it was with dts-hd. dts-hd has about 4 times the bandwidth. I'll be putting up the newer code sometime tomorrow. I've been working pretty long hours on it and now I just want to relax a bit. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
On 03/06/11 14:54, pl bossart wrote: I've got pulseaudio xbmc passthrough working. I've rolled it together with a previously windows c++ library called Ac3Filter. I've been tweeking up Ac3Filter quite a bit. I'm using the Ac3Filter library to Mux spdif. It was a well written library/tools that was relatively easy to get ported to linux. I'm watching Avatar in wonderful DTS surround sound now. All you need is a 3D TV now... Seriously, this is good. Can you send a link to this ported AC3Filter code? I thought it was too Windows/DirectX-oriented to be used. I didn't look too much since it's GPL, and LGPL is preferred in terms of integration with proprietary components. Thanks, -Pierre I've update AudioFilter and the Xbmc AudiFilter patch. It now passes the core dts-5.1 portion of dts-hd audio streams. Supposedly ffmpeg can mux a dts-hd stream that passes through the full dts-hd stream. I want to analyse what ffmpeg is doing to the hd portion of the stream so I can implement it. If anyone knows of the ffmpeg command that will mux a dts-hd stream for spdif let me know. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
'Twas brillig, and Kelly Anderson at 09/03/11 06:59 did gyre and gimble: OK, I've just put up the patches necessary to get passthrough working with Nvidia hdmi on XBMC. This is awesome work Kelly, Please keep in mind that the PA API side of things will be changing very soon... there may be a preview branch from Arun later today of how it will work. I suspect the changes to the XBMC patch will be rather minimal tho' :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
On 03/06/2011 02:54 PM, pl bossart wrote:
All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre
Pierre,
OK, I've cleaned the code up and it works really well now (for my
purposes). It's really efficient with processing DTS files, I haven't
profiled Ac3 yet. I've rolled it into my copy of xbmc and all seems to
be working really well. I'll be putting patches for xbmc up as well as
soon as I've prep'd them.
You could do me a favor if you would complete sink_pulse.{cpp,h}. I've
been working on other things and that's really on the back burner for
me. Once the SinkPulse is done, we'll be able to play raw DTS/AC3 files
pretty much the same way paplay does for other formats. At some point
we might even want to roll AudioFilter into paplay.
I renamed Ac3Filter to AudioFilter. It really does a bunch more than
just Ac3 and I've renamed interfaces, etc.
I'm still renaming files and moving the around a bit, but the headers
should be pretty stable.
http://silka.with-linux.com/audiofilter/
I'll probably put the code in a git repository once I've got the file
renaming out of the way.
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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
On 03/08/2011 03:32 PM, Kelly Anderson wrote:
On 03/06/2011 02:54 PM, pl bossart wrote:
All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre
Pierre,
OK, I've cleaned the code up and it works really well now (for my
purposes). It's really efficient with processing DTS files, I haven't
profiled Ac3 yet. I've rolled it into my copy of xbmc and all seems
to be working really well. I'll be putting patches for xbmc up as
well as soon as I've prep'd them.
You could do me a favor if you would complete sink_pulse.{cpp,h}.
I've been working on other things and that's really on the back burner
for me. Once the SinkPulse is done, we'll be able to play raw DTS/AC3
files pretty much the same way paplay does for other formats. At some
point we might even want to roll AudioFilter into paplay.
I renamed Ac3Filter to AudioFilter. It really does a bunch more than
just Ac3 and I've renamed interfaces, etc.
I'm still renaming files and moving the around a bit, but the headers
should be pretty stable.
http://silka.with-linux.com/audiofilter/
I'll probably put the code in a git repository once I've got the file
renaming out of the way.
OK, I've just put up the patches necessary to get passthrough working
with Nvidia hdmi on XBMC.
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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
I've got pulseaudio xbmc passthrough working. I've rolled it together with a previously windows c++ library called Ac3Filter. I've been tweeking up Ac3Filter quite a bit. I'm using the Ac3Filter library to Mux spdif. It was a well written library/tools that was relatively easy to get ported to linux. I'm watching Avatar in wonderful DTS surround sound now. All you need is a 3D TV now... Seriously, this is good. Can you send a link to this ported AC3Filter code? I thought it was too Windows/DirectX-oriented to be used. I didn't look too much since it's GPL, and LGPL is preferred in terms of integration with proprietary components. Thanks, -Pierre ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
On 03/06/2011 02:54 PM, pl bossart wrote: I've got pulseaudio xbmc passthrough working. I've rolled it together with a previously windows c++ library called Ac3Filter. I've been tweeking up Ac3Filter quite a bit. I'm using the Ac3Filter library to Mux spdif. It was a well written library/tools that was relatively easy to get ported to linux. I'm watching Avatar in wonderful DTS surround sound now. All you need is a 3D TV now... Seriously, this is good. Can you send a link to this ported AC3Filter code? I thought it was too Windows/DirectX-oriented to be used. I didn't look too much since it's GPL, and LGPL is preferred in terms of integration with proprietary components. Thanks, -Pierre I'm putting the finishing touches on it, give me a few days. I'm cleaning up the code and putting it in it's own namespace so it'll integrate with large projects without conflict. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
'Twas brillig, and Kelly Anderson at 03/03/11 22:04 did gyre and gimble: Forgot to mention. I'm getting Hi-def video/audio and only using 8% cpu. That's just awesome. Vdpau is handling the video decode. Very nice :) Looking forward to getting this setup on my system :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio passthrough samplerate
Hey, I've been hacking on xbmc - pulseaudio passthrough and I'm getting there a little bit at a time. It seems that at the moment that if the daemon.conf default-sample-rate is set to 44100 then all I can passthrough is 44100 and if it's 48000 all that works are 48000 samples. Is that a limitation in the current pulse implementation? I expected that this would be set from the pa_sample_spec, but that seems to be ignored currently. I'm using paplay --rate ? --raw --passthrough for testing. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio passthrough samplerate
It seems that at the moment that if the daemon.conf default-sample-rate is set to 44100 then all I can passthrough is 44100 and if it's 48000 all that works are 48000 samples. Is that a limitation in the current pulse implementation? I expected that this would be set from the pa_sample_spec, but that seems to be ignored currently. I'm using paplay --rate ? --raw --passthrough for testing. Yes it's a know limitation I mentioned in my previous emails. Somehow we will need to reconfigure the passthrough sink frequency. -Pierre ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio xbmc passthrough success
Yahoo! I've got pulseaudio xbmc passthrough working. I've rolled it together with a previously windows c++ library called Ac3Filter. I've been tweeking up Ac3Filter quite a bit. I'm using the Ac3Filter library to Mux spdif. It was a well written library/tools that was relatively easy to get ported to linux. I'm watching Avatar in wonderful DTS surround sound now. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success
Forgot to mention. I'm getting Hi-def video/audio and only using 8% cpu. That's just awesome. Vdpau is handling the video decode. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio: module.c: module-detect is deprecated: Please use module-udev-detect instead of module-detect!
2011/3/2 Andriy Gapon a...@freebsd.org: on 02/03/2011 17:18 Michal Varga said the following: On Wed, 2011-03-02 at 16:13 +0100, Michal Varga wrote: So instead of fixing things like these in ports, I'd say the issue should be reported/fix submitted to upstream, it's pretty much possible that they don't even know that there are systems that don't run evdev (note that I don't mean it offensively, more like speaking generally from experience, it simply happens). Eh, I meant udev of course, I guess that's enough coffee for me today. That's possible, of course. Another possibility is that they think that everything is Linux, or everything is like Linux, or they don't care about not Linux. Development of pulseaudio is certainly a bit of a linux-focused affair. But we very much want other platforms to be included as well, as can be seen by the OSX, *BSD, Solaris and win32 support currently in the tree. I am all for contacting them, for sure. Just want to point out that the proposed patch would not be against the nature of the ports, because it doesn't affect functionality, but addresses OS differences. There is already some FreeBSD specific code in PulseAudio upstream. Especially considering some changes required to make pulse run on freebsd and netbsd are quite similar, I think it makes sense to submit your patches. -- Andriy Gapon Maarten ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio module-rtp-recv
Please use the PulseAudio mailing list for these kind of questions. I'll be reading and answering there. 2011/2/23 ericlesoll ericles...@gmail.com: Hi Maarten, Thanks for your job, I reported some bugs on Launchpad about PulseAudio module-rtp-recv, and I saw that you have commited 6 weeks ago some modification. The bug is that the sound go transforming after 55s. We use this module in association with module-rtp-send to make a chat audio piloting by itALC in our project LLSOLL / LESOLL that is open source digital language laboratory. We don't use loopack module due to the delay to big, we use HAL for loopback. Do you have a link for this bug? It sound like the problem my patches should solve. I download the git://git.0pointer.de/pulseaudio.git on Ubuntu 10.10 and I compiled without error output. I replaced my original module-rtp-recv.so by the fresh one then I run ldconfig. When I launch $ pactl load-module module-rtp-recv, I get a number ID without error but I ear no sound. I replace by the old one it's working until 55s then the sound go transforming. I'm not sure that's the way it's supposed to work. You better do a make install and then run /usr/local/bin/pulseaudio - and then LD_LIBRARY_PATH=/usr/local/lib /usr/local/bin/pactl load-module module-rtp-recv, or something similar. In general you can't expect modules compiled for one version of pulse to work in another version. Have ou some idea ? Have to upgrade others modules ? Can I send you log file when I launch module rtp-recv (I don't know how to log it, because -v option is not available) ? Start pulseaudio manually, like I wrote above. You probably mean that the -v options is not available for the /etc/init.d/pulseaudio script. Best regards Eric Maarten ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio duplicate trace in /var/log/syslog
Hello Tanu Thank you for your repy Tanu Kaskinen wrote, On 20/02/11 07:19: On Sat, 2011-02-19 at 12:33 +, Jon Grant wrote: Hello I have the following occurring every 5 seconds in syslog. I wanted to offer to submit a patch for review to remove some of the duplicate lines. Lines 4 and 5 are not duplicates. If you look at the PIDs, there are two pulseaudio processes. The daemon launching works so that the initial process (29468) forks a new process (29470) that becomes the actual daemon. The failed to initialise daemon message comes from the child process, and it's useful information for debugging, since it indicates that the execution reached that particular position in the code. The daemon startup failed message comes from the parent process. It's printed always if the parent process doesn't get a message from the child process telling that startup was successful. It's not a good idea to remove that message. Ok. I understand now. Perhaps the daemon could be named pulseaudiod to distinguish it. Line 3 looks quite redundant, though. But I checked the code, and the place where it's printed is such that the message is printed whenever loading the startup script fails for any reason, so just removing the line probably isn't a good idea, otherwise other error situations may not get logged adequately. So, since there are no simple fixes, I don't think it makes sense to spend effort to try to change the logging to be less repetitive. I can't see this being a big problem. A bigger problem is that pulseaudio is trying to start again and again, and always fails. The problem appears to be that loading module-esound-protocol-unix fails, preventing the whole daemon from starting. Apparently the esound socket is being used by some other process. To find out which process, run netstat -l --unix -p | grep esd Thank you for this suggestion. The problem is not visible so far today, so I will keep the command line and use it next time it occurs. I do see the following log today: Feb 20 10:44:25 note pulseaudio[1722]: ratelimit.c: 107 events suppressed This may not be related. However this message is output even when I am not playing any audio on idle system. Any idea which kind of events were suppressed? In addition, https://tango.0pointer.de does not have a signed SSL cert. Could it get a CAcert one? Or even a verisign SSL cert. Currently we get a firefox warning when connecting. It's up to Lennart. When the web server was set up, I believe it was a conscious decision to go with a self-signed cert, but I don't know if the reasons for the decision are still valid today... Ok, maybe Lennart will jump in and reply.. If donation is need for SSL cert, I can donate to cover the cost. or CAcert option. Best regards, Jon ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio duplicate trace in /var/log/syslog
On Sat, 2011-02-19 at 12:33 +, Jon Grant wrote: Hello I have the following occurring every 5 seconds in syslog. I wanted to offer to submit a patch for review to remove some of the duplicate lines. Lines 4 and 5 are not duplicates. If you look at the PIDs, there are two pulseaudio processes. The daemon launching works so that the initial process (29468) forks a new process (29470) that becomes the actual daemon. The failed to initialise daemon message comes from the child process, and it's useful information for debugging, since it indicates that the execution reached that particular position in the code. The daemon startup failed message comes from the parent process. It's printed always if the parent process doesn't get a message from the child process telling that startup was successful. It's not a good idea to remove that message. Line 3 looks quite redundant, though. But I checked the code, and the place where it's printed is such that the message is printed whenever loading the startup script fails for any reason, so just removing the line probably isn't a good idea, otherwise other error situations may not get logged adequately. So, since there are no simple fixes, I don't think it makes sense to spend effort to try to change the logging to be less repetitive. I can't see this being a big problem. A bigger problem is that pulseaudio is trying to start again and again, and always fails. The problem appears to be that loading module-esound-protocol-unix fails, preventing the whole daemon from starting. Apparently the esound socket is being used by some other process. To find out which process, run netstat -l --unix -p | grep esd In addition, https://tango.0pointer.de does not have a signed SSL cert. Could it get a CAcert one? Or even a verisign SSL cert. Currently we get a firefox warning when connecting. It's up to Lennart. When the web server was set up, I believe it was a conscious decision to go with a self-signed cert, but I don't know if the reasons for the decision are still valid today... -- Tanu ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] PulseAudio and NX
Hi - We have a Fedora and KDE4 system. If someone logs into a system using NX - http://www.nomachine.com/ - while they have a local session running, the sound from the NX session comes out of the local machine. When they log out of the remote system the pulseaudio server is killed on the local session. Is there any way to stop the remote NX sessions talking to the local pulseaudio session? We don't care about sound forwarding. Setting PULSE_SERVER to an invalid value didn't seem to help. Is there something like a PULSE_DISABLE environment variable? Thanks Jeremy ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio and NX
'Twas brillig, and Jeremy Sanders at 16/02/11 12:59 did gyre and gimble: Hi - We have a Fedora and KDE4 system. If someone logs into a system using NX - http://www.nomachine.com/ - while they have a local session running, the sound from the NX session comes out of the local machine. When they log out of the remote system the pulseaudio server is killed on the local session. Is there any way to stop the remote NX sessions talking to the local pulseaudio session? We don't care about sound forwarding. You can either: 1. Edit /usr/bin/start-pulseaudio-x11 and comment out the load-module line for module-x11-publish or 2. Run pax11publish -r This removes the configuration variables pushed into the X11 root window as properties and should do what you want. HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio and NX
Colin Guthrie wrote: You can either: 1. Edit /usr/bin/start-pulseaudio-x11 and comment out the load-module line for module-x11-publish or 2. Run pax11publish -r This removes the configuration variables pushed into the X11 root window as properties and should do what you want. Thanks very much for your answer - we'll give your suggestions a try... Jeremy ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio network audio server perfect setup
Hello, I am trying to get the perfect pulseaudio network audio server setup. So far, I have pulseaudio daemon running on my sound server. I am able to access it from my clients using padevchooser. I have two questions: 1/ padevchooser is tagged as obsolete in the pulseaudio documentation. What sould I use instead ? 2/ The last thing I miss is the use of my multimedia keyboard to be able to control the server's volume. This is not a pulseaudio issue, rather a GNOME one (yes, I am using GNOME) But I figured that someone from pulseaudio could better understand this problem: So, If I use the volume control applet, it is totaly transparent for me (when it comes to adjusting the volume levels) weather I use my local server or the networked one. However, the actions of my volume up/down keys only affects my local sound server...too bad ! Thanks in advance for any kind of help. Regards, Thomas ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio network audio server perfect setup
'Twas brillig, and Thomas Riché at 30/01/11 10:02 did gyre and gimble: Hello, I am trying to get the perfect pulseaudio network audio server setup. So far, I have pulseaudio daemon running on my sound server. I am able to access it from my clients using padevchooser. I have two questions: 1/ padevchooser is tagged as obsolete in the pulseaudio documentation. What sould I use instead ? Generally we'd recommend pavucontrol as a mixer app. padevchooser just sets properties on the X11 root window and doesn't actually do anything live with the pulseaudio server. The best approach generally is to use tunnels from your local PA server to the network server. This way you can move your streams from local devices to remote ones without having to stop and restart them. 2/ The last thing I miss is the use of my multimedia keyboard to be able to control the server's volume. This is not a pulseaudio issue, rather a GNOME one (yes, I am using GNOME) But I figured that someone from pulseaudio could better understand this problem: So, If I use the volume control applet, it is totaly transparent for me (when it comes to adjusting the volume levels) weather I use my local server or the networked one. However, the actions of my volume up/down keys only affects my local sound server...too bad ! IIRC the volume keys affect the default sink. Therefore to change the remote audio, just change the default sink to be the one that is the tunnel to the remote server :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio network audio server perfect setup
On Sun, Jan 30, 2011 at 12:25 PM, Colin Guthrie gm...@colin.guthr.iewrote: 'Twas brillig, and Thomas Riché at 30/01/11 10:02 did gyre and gimble: Hello, I am trying to get the perfect pulseaudio network audio server setup. So far, I have pulseaudio daemon running on my sound server. I am able to access it from my clients using padevchooser. I have two questions: 1/ padevchooser is tagged as obsolete in the pulseaudio documentation. What sould I use instead ? Generally we'd recommend pavucontrol as a mixer app. padevchooser just sets properties on the X11 root window and doesn't actually do anything live with the pulseaudio server. The best approach generally is to use tunnels from your local PA server to the network server. This way you can move your streams from local devices to remote ones without having to stop and restart them. 2/ The last thing I miss is the use of my multimedia keyboard to be able to control the server's volume. This is not a pulseaudio issue, rather a GNOME one (yes, I am using GNOME) But I figured that someone from pulseaudio could better understand this problem: So, If I use the volume control applet, it is totaly transparent for me (when it comes to adjusting the volume levels) weather I use my local server or the networked one. However, the actions of my volume up/down keys only affects my local sound server...too bad ! IIRC the volume keys affect the default sink. Therefore to change the remote audio, just change the default sink to be the one that is the tunnel to the remote server :) Ok, I've set up a tunnel. It is working as expected, the volume keys now affects the selected sink !! Thank you. However, I noticed a degradation in sound quality: - the sound is sometimes missing - when i change track in vlc, the beginning of the sound is chopping do you have any idea what could be causing this ? thanks again Thomas Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio problem when using x2go and likewise-open
Hi, I have read all the related documentation on http://www.pulseaudio.org/wiki/ServerStrings and related wikis, and I understood what it is written there, still I need some more information how can I debug or start manually pulseaudio SSH forward. What is this thread about: Ubuntu 10.10 with Likewise-open connected to corporate LDAP. X2go server installed and configured using pulseaudio forwarding sound from the server to the clients. Problem: x2goclient can't make ssh pulseaudio tunnel when using LDAP user. x2goclient can make ssh pulseaudio tunnel when using local ubuntu user. It is nothing with group permissions, as they are not needed, because I do have configured, and working system wide mode ( http://www.pulseaudio.org/wiki/SystemWideInstance). Logging physically on the Ubuntu x2goserver with LDAP user does have sound, but the access to the soundcard is through the physical hardware. I am in touch with x2go-developers, and there have live discussion with them on their mailing list as well: https://lists.berlios.de/pipermail/x2go-dev/2010-December/thread.html#start and more specific: the thread: [X2go-dev] x2go with likewise-open (ldap) questions https://lists.berlios.de/pipermail/x2go-dev/2010-December/001361.html Currently I do have 2 remote open x2go sessions. One using LDAP user, the other Local Ubuntu user. I don't have anything in hardware and aplay -l or aplay -L or lspci -v are absolutely same. The difference from gnome-volume-control is the Output Tab. The LDAP user has Dummy Output, while the Local Ubuntu user has Windows waveOut PCM My questions are: 1. In this situation, can I try to start manually pulseaudio server forward in the LDAP user? 2. Is there a way do make some deep debugs on the LDAP user, so to get what exactly is getting wrong P.S. My assumption is that x2go side has un-escaped string for username, which now because of LDAP include @ and converted to \. I'm checking the sources, still that will take its time. My x2goclients are both Windows 7(for all my experience up is used), and another Ubuntu 10.10 (on VMware). The VM has sound, but the x2goclient does not have in all cases (doesn't matter if the user is LDAP or Local Ubuntu). Bottom line: can you give me more information how this forward happens exactly, or help me with deep investigation from pulseaudio side. Thanks and Best Regards Ivan ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
'Twas brillig, and Noel David Torres Taño at 14/12/10 17:53 did gyre and gimble: (continues here) I've deleted .pulse and .pulse-cookie to be sure, removed both accounts from audio group and set them both in pulse-access group. Still first session opened gets sound (even when changing sessions, so ConsoleKit is not in the way) and secondly opened one gets no sound. May it be that extra PA processes are opened? How to avoid that and get each session connect to the system-wide PA? $ ps -A u | grep pulse pulse12594 0.0 0.1 273444 5132 ?Ssl 16:52 0:01 /usr/bin/pulseaudio --system --daemonize --high-priority --log-target=syslog --disallow-module-loading=1 user115671 0.0 0.1 218140 5144 ?Ssl 17:41 0:00 /usr/bin/pulseaudio --start user216193 0.4 0.0 335412 3868 ?Ssl 17:45 0:00 /usr/bin/pulseaudio --start user1 16306 0.0 0.0 9612 892 pts/1S+ 17:46 0:00 grep pulse Hmm, interesting. pulseaudio --start should basically exit when a system-wide instance is present IIRC. I'll dig through the code and do some experiments to try and work out what's going on. As I don't use system wide myself, I'm maybe missing something. I'll try and get back to you soon. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio problem when using x2go and likewise-open
As I said above, we do not do any forward at all (well, it would be SSH that does this if anything, but it doesn't - yet). All we do is piggy back onto X11 forwarding built in to SSH. Overall, I'm not sure where you are running PA and why it needs to be system-wide at all (generally I would not recommend this), but come on IRC and then we can discuss and I can explain how things should be setup. I'm coling in #pulseaudio on Freenode. FWIW, I describe how the Remote X11 case works here: http://colin.guthr.ie/2009/08/sound-on-linux-is-confusing-defuzzing-part-2-pulseaudio/ which may be useful reading. Col -- Colin Guthrie Hi Colin, thanks for your reply, I have found the root cause, and it is in x2go side You can check their mailing list for more information: https://lists.berlios.de/pipermail/x2go-dev/2010-December/001388.html Best Regards Ivan ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio problem when using x2go and likewise-open
'Twas brillig, and Colin Guthrie at 15/12/10 14:22 did gyre and gimble: 'Twas brillig, and Ivan Boyadzhiev at 15/12/10 09:48 did gyre and gimble: Hi, 8 snip! Just for the record, I spoke with Ivan and it seems the problem relates to x2go in some capacity. https://lists.berlios.de/pipermail/x2go-dev/2010-December/001388.html -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
[...] I'll dig through the code and do some experiments to try and work out what's going on. As I don't use system wide myself, I'm maybe missing something. I'll try and get back to you soon. Col Thanks Noel er Envite ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
On Domingo 12 Diciembre 2010 20:17:47 Colin Guthrie escribió: 'Twas brillig, and Noel David Torres Taño at 10/12/10 08:48 did gyre and gimble: On Jueves 09 Diciembre 2010 10:06:00 Colin Guthrie escribió: HTHs It helped :) Now we both can have half-time sound, and that is better than one getting no sound :) How can this be accomplished using a server-wide PA? (Note: yes, I know WhatIsWrongWithSystemMode but it is not true that either my or my wife's sessions are always open, nor that it is always the same session the first to be opened, so, strage as it can be, I need the equivalent solution using a server wide instance. many many thanks for PA itself and this answer Yeah there are always situations where people want to do things slightly differently and the disadvantages of System Wide are not always problematic, so as always YMMV. For yourself, perhaps system wide is the best option. Take Care. Col Yes, thanks, but... How to do it? Is it just to set a system-wide server or is there something to do in its config and/or in both $HOMEs and/or UNIX groups? Thx again er Envite ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
'Twas brillig, and Noel David Torres Taño at 14/12/10 16:41 did gyre and gimble: On Martes 14 Diciembre 2010 12:22:17 Michał Sawicz escribió: Dnia 2010-12-14, wto o godzinie 12:11 +, Noel David Torres Taño pisze: Yes, thanks, but... How to do it? Is it just to set a system-wide server or is there something to do in its config and/or in both $HOMEs and/or UNIX groups? See http://www.pulseaudio.org/wiki/SystemWideInstance First google hit for 'pulseaudio system wide' That does NOT answer my question about audio group access, config on the $HOMEs and the like. Thanks, but I learned to read when I was quite young. No need to be rude to people who post helpful links. That page only says that both users should be on pulse-access group, but does not say if, for example, .pulse-cookie must be the same, or if .pulse/client.conf has to be tweaked. It also doesn't say how much RAM you need nor whether it works if the day has the letter 'T' in it... because they are not relevant :p The fact you know about these additional parts of PA is actually working against you here - you know too much and it means you try to push them into a situation where they are not mentioned because they are not important! But cheekyness aside, the cookie is not used in system wide mode, the access is controlled via the membership of the pulse-access group. There should be no tweaks needed on .pulse/client.conf (the file should not even exist unless something manual has been done). HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
On Martes 14 Diciembre 2010 17:20:37 Colin Guthrie escribió: 'Twas brillig, and Noel David Torres Taño at 14/12/10 16:41 did gyre and gimble: On Martes 14 Diciembre 2010 12:22:17 Michał Sawicz escribió: Dnia 2010-12-14, wto o godzinie 12:11 +, Noel David Torres Taño pisze: Yes, thanks, but... How to do it? Is it just to set a system-wide server or is there something to do in its config and/or in both $HOMEs and/or UNIX groups? See http://www.pulseaudio.org/wiki/SystemWideInstance First google hit for 'pulseaudio system wide' That does NOT answer my question about audio group access, config on the $HOMEs and the like. Thanks, but I learned to read when I was quite young. No need to be rude to people who post helpful links. True, my fault (even when not helpful for me). That page only says that both users should be on pulse-access group, but does not say if, for example, .pulse-cookie must be the same, or if .pulse/client.conf has to be tweaked. It also doesn't say how much RAM you need nor whether it works if the day has the letter 'T' in it... because they are not relevant :p The fact you know about these additional parts of PA is actually working against you here - you know too much and it means you try to push them into a situation where they are not mentioned because they are not important! But cheekyness aside, the cookie is not used in system wide mode, the access is controlled via the membership of the pulse-access group. Ok. Tried but... (continues below) There should be no tweaks needed on .pulse/client.conf (the file should not even exist unless something manual has been done). Ok too HTHs Col (continues here) I've deleted .pulse and .pulse-cookie to be sure, removed both accounts from audio group and set them both in pulse-access group. Still first session opened gets sound (even when changing sessions, so ConsoleKit is not in the way) and secondly opened one gets no sound. May it be that extra PA processes are opened? How to avoid that and get each session connect to the system-wide PA? $ ps -A u | grep pulse pulse12594 0.0 0.1 273444 5132 ?Ssl 16:52 0:01 /usr/bin/pulseaudio --system --daemonize --high-priority --log-target=syslog --disallow-module-loading=1 user115671 0.0 0.1 218140 5144 ?Ssl 17:41 0:00 /usr/bin/pulseaudio --start user216193 0.4 0.0 335412 3868 ?Ssl 17:45 0:00 /usr/bin/pulseaudio --start user1 16306 0.0 0.0 9612 892 pts/1S+ 17:46 0:00 grep pulse Thanks Noel er Envite ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
'Twas brillig, and Noel David Torres Taño at 10/12/10 08:48 did gyre and gimble: On Jueves 09 Diciembre 2010 10:06:00 Colin Guthrie escribió: HTHs It helped :) Now we both can have half-time sound, and that is better than one getting no sound :) How can this be accomplished using a server-wide PA? (Note: yes, I know WhatIsWrongWithSystemMode but it is not true that either my or my wife's sessions are always open, nor that it is always the same session the first to be opened, so, strage as it can be, I need the equivalent solution using a server wide instance. many many thanks for PA itself and this answer Yeah there are always situations where people want to do things slightly differently and the disadvantages of System Wide are not always problematic, so as always YMMV. For yourself, perhaps system wide is the best option. Take Care. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio for two sessions at the same time
Hello all: I've a computer at home which uses to have two sessions opened at the same time: mine and my wife's. We use to leave both sessions opened instead of opening and closing sessions with each seat change. But the problem is that the first openes session gets sound and the other one does not: it is absolutely mute. How can we get session working in both sessions at the same time? Thanks Noel T. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
Hi, 'Twas brillig, and Noel David Torres Taño at 09/12/10 09:35 did gyre and gimble: I've a computer at home which uses to have two sessions opened at the same time: mine and my wife's. We use to leave both sessions opened instead of opening and closing sessions with each seat change. But the problem is that the first openes session gets sound and the other one does not: it is absolutely mute. This should not generally happen. PulseAudio is designed to work at a session level and ensure that when you switch between sessions, control of the audio device is handed gracefully over to the other user. All of this is actually handled at a lower level than PA. It's dealt with by a system called Console Kit. Console Kit maintains a record of who the active user is on a system (you can check via ck-list-sessions command in a terminal). Console Kit will instruct udev to write ACLs (access control lists) on various bits of hardware (including the sound card) so that only the active user has permission to use them at any one time (otherwise there could be security considerations - e.g. spying on voip calls etc. etc.) PA simply honours this lower level system. Now what is happening in your case is one of three things (the last is the most likely): 1. Console Kit is not working properly. To test this, open a session for both users and open a terminal and type ck-list-sessions. As you switch between the sessions, the active user should change. 2. Console Kit is not writing the ACLs properly. Use getfacl /dev/snd/* in each session to ensure that the relevant user appears in the ACLs for the sound. 3. One or both of your users is in the audio group. This bypasses all the nice ACL and session switching logic, but only really works if your sound hardware supports hardware mixing or you have specific reason to do something non-standard (see below). Just type groups in a terminal to see if you are in the audio group and if so, use the appropriate tools to remove this and then reboot (make sure you do this for both users) and you should get smooth switching of users. How can we get session working in both sessions at the same time? Well that's the important question. Do you *really* want it to work at the same time, or do you want it to hand over gracefully when you switch sessions. Most systems (including OSX and Windows etc. - although I've had odd experience with Windows...) do the latter but some users want the former. If you fix/debug the above mentioned issues, then you'll get a nice handover, but if you really do want both at the same time output, then the simplest way (if you are generally always logged in) is as follows: 1. Add your user to the audio group, but not your wife. 2. Login as you. 3. Start paprefs and tick the Enable Network Access box. 4. Copy the file ~/.pulse-cookie to your wife's home directory (so that you both have the same cookie file). 5. Edit/create the file ~/.pulse/client.conf in your wife's home directory and put the line default-server = localhost This will mean you run the PA daemon and your wife connects to your daemon. You can also use a system-wide daemon but this is probably easier and at least means one user can benefit from SHM IPC whereas with system-wide no users can. HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio, pcHDTV hd-5500 card, tvtime, and mythtv
Hi Trevan,
'Twas brillig, and Trevan Richins at 06/12/10 01:01 did gyre and gimble:
Hello,
I'm attempting to use pulseaudio to play back sound from my hd-5500 tv
card (I use the analogue part of it) and I'm hitting a brick wall.
My /etc/asound.conf file is:
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
If I run the command sox -c 1 -t alsa default -t alsa default, I hear
the TV sound correctly. Opening up pavucontrol shows the sound bar
moving back and forth indicating that pulse is getting the sound. But
if I run tvtime or MythTV, I don't get any sound.
MythTV in particular does very silly things with pulseaudio. I've tried
several times to argue with them about it, but sadly they've not headed
me yet.
How is your sound configured in MythTV? Do you get sound from e.g. a
pre-recorded video file or music etc.?
To explain MythTV will automatically try to outsmart your system and
deliberately take action to suspend PulseAudio unless you configure
MythTV correctly. If you use ALSA:pulse or PulseAudio:default then
MythTV should let PA do the audio but if you set ALSA:default then
MythTV will do something really dumb, which is to suspend PA even though
it will ultimately try and push audio to it. I've always patched this
behaviour in the Mandriva MythTV packages so I would suggest using the
patches there on your build of MythTV (or point your distro maintainer
at it).
I don't know if the problem is with pulseaudio, the card, or
tvtime/mythtv. Any ideas on what I should do?
Well the problems are actually quite different. tvtime and MythTV
operate quite differently.
MythTV should properly read the audio and play it back through the sound
card of your choice, but tvtime expects a loopback to be used. e.g. a
physical cable coming out of the TV card and into the line in of your
sound card.
For the tvtime case you can likely use module-loopback in PA (just type:
pactl load-module module-loopback, then use pavucontrol on the Recording
and Playback tabs to move the loopback streams to the correct recording
device (i.e. the tv card) and the correct playback device (i.e. your
sound card))
For MythTV, they need to fix their braindead configuration, but you can
maybe get it working by setting the audio device to either ALSA:pulse
or PulseAudio:default
HTHs
Feel free to ask on IRC. I think I saw you drop by and ask this question
the other day, but didn't get a change to reply before you logged out -
you need to stick around for a while :)
Take care
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
Mageia Contributor [http://www.mageia.org/]
PulseAudio Hacker [http://www.pulseaudio.org/]
Trac Hacker [http://trac.edgewall.org/]
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[pulseaudio-discuss] PulseAudio, pcHDTV hd-5500 card, tvtime, and mythtv
Hello,
I'm attempting to use pulseaudio to play back sound from my hd-5500 tv
card (I use the analogue part of it) and I'm hitting a brick wall.
My /etc/asound.conf file is:
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
If I run the command sox -c 1 -t alsa default -t alsa default, I hear
the TV sound correctly. Opening up pavucontrol shows the sound bar
moving back and forth indicating that pulse is getting the sound. But
if I run tvtime or MythTV, I don't get any sound.
I don't know if the problem is with pulseaudio, the card, or
tvtime/mythtv. Any ideas on what I should do?
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Re: [pulseaudio-discuss] PulseAudio, pcHDTV hd-5500 card, tvtime, and mythtv
On Sun, Dec 5, 2010 at 8:01 PM, Trevan Richins trevan_pu...@therichins.net wrote: If I run the command sox -c 1 -t alsa default -t alsa default, I hear the TV sound correctly. Right; you've routed 'default' through the pulse alsa-lib plugin. But if I run tvtime or MythTV, I don't get any sound. A couple distros don't route either of those through 'default' but through the older OSS API (/dev/dsp, /dev/audio), and at least a couple route the latter through ALSA's plughw, all of which generally collide with pulse on modern integrated HDA. I don't know if the problem is with pulseaudio, the card, or tvtime/mythtv. Any ideas on what I should do? Ideally you would route them through pulse; some distros patch them to allow limited functionality through pulse's alsa-lib plugin or the alsa mixer interface. Best, -Dan ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio version
[In the future please start a new message rather than replying and changing the subject as it messes up threading] 'Twas brillig, and Pichon, SylvainX at 10/11/10 10:16 did gyre and gimble: I know this topic was already discussed but I 'd like to know if are there some updates. 0.9.21 branches seems to be dead and current development branch is based on 0.9.19 and is newer. The branch for 0.9.21 is certainly not dead. I just pushed some fixes to it the other day and have regularly pushed updates to it ever since 0.9.21 was released. See here: http://git.0pointer.de/?p=pulseaudio.git;a=shortlog;h=refs/heads/stable-queue The fact that master is based of 0.9.19 is just a quirk of how things worked out. Everything that is in 0.9.21 (and the stable-queue branch) are also in master, so it's very much newer than 0.9.21. This begin to be a big problem since it is not easy to convince people to use our developments based on 0.9.19 instead 0.9.21. Even if it is newer... Yeah don't worry, it's just a quirk of how things are orgianised. I'm planning to push a tag to master to make master compile as 0.10.0-devel or at least 0.9.23) but need to get approval from Lennart before I do that (the version used inside configure.ac is actually generated from git-version-gen script, so it's labelled as 0.9.19 due to how the git tree was organised). If you are doing any work, then master is likely a good place to do it. If you prefer to do it against stable-queue, then that is likely also fine. Rest assured both master and stable-queue are quite active and despite it's lower version number master is newer and contains more (have a look at git cherry if you want to confirm :D) Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio-0.9.14 on Slackware 12.2 with Droplinegome 2.26
Basically the console kit module is failing to load due to a dbus problem. Also seems that some other application has your Alsa device open (Device or resource busy message indicates this), perhaps this means that you simply can't use your h/w in full duplex mode (e.g. opening the sink prevents the source from actually working?) Anyway, actually two bugs there. Do you think they are both PA bugs, or could the second one be hardware limitation of my old soundcard? FWIW 0.9.14 is a bit of a strange release to use. AFAIK 0.9.16 was the stable version for a while (fixing bugs in 0.9.12 through 0.9.15), but I could be wrong. The Droplinegnome devs probably decided to use PA 0.9.14 so they didn't need to upgrade the original libtool (and others). They try to upgrade a minimum of original Slackware packages. I'd strongly recommend using 0.9.21 (or rather stable-queue git branch) as many bug fixes have been included since 0.9.14. I'd also recommend newer alsa kernel and libs too. Thanks for your reply. I will see if I can upgrade to 0.9.21 without to many issues. Regards, Ron Hermsen Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio-0.9.14 on Slackware 12.2 with Droplinegome 2.26
'Twas brillig, and Ron Hermsen at 20/10/10 21:55 did gyre and gimble: Basically the console kit module is failing to load due to a dbus problem. Also seems that some other application has your Alsa device open (Device or resource busy message indicates this), perhaps this means that you simply can't use your h/w in full duplex mode (e.g. opening the sink prevents the source from actually working?) Anyway, actually two bugs there. Do you think they are both PA bugs, or could the second one be hardware limitation of my old soundcard? I suspect neither are pulseaudio bugs! The first is just a problem with your dbus or console-kit setup. I think the latter could very easily be a due to old h/w like you say. This is handled differently in newer PA versions by module-alsa-card, that implements a probing mechanism to determine how the h/w can be used. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio-0.9.14 on Slackware 12.2 with Droplinegome 2.26
Dear Reader, I have problems with pulseaudio not starting up. After changing default.pa, module-alsa-sink did found my soundcard, but PA still fails. I hope someone can help me to get it working. Audio details: “ESS 1869 Plug and Play Audio Drive”. The module that is loaded: snd_es1688 Complete lsmod output: http://pastebin.com/B0rzSrJT Modprobe and lsdev info: http://pastebin.com/rj0xqL01 If I use pulseaudio - --daemonize=false with the default “default.pa” the output I get the following failure messages: E: polkit.c: Cannot set UID on session object. I: main.c: PolicyKit refuses acquire-high-priority privilege. D: module-always-sink.c: Autoloading null-sink as no other sinks detected. E: module-console-kit.c: GetSessionsForUnixUser() call failed: org.freedesktop.DBus.Error.NoReply: Message did not receive a reply (timeout by message bus) E: module.c: Failed to load module module-console-kit (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Failed to initialize daemon. Complete output: http://pastebin.com/MuxxuLmt From the null-sink I understand that no soundcards is found. aplay -v --device=hw:0 /usr/share/sounds/shutdown.wav works Complete output: http://pastebin.com/Rdp0Ef6t Tried to change “default.pa” with the following lines: load-module module-alsa-sink device=hw:0 load-module module-alsa-source device=hw:0 With this the module-alsa-sink seems to be ok, but module-alsa-source failes. pulseaudio - --daemonize=false output: E: polkit.c: Cannot set UID on session object. I: main.c: PolicyKit refuses acquire-high-priority privilege. D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed E: alsa-util.c: Error opening PCM device hw:0: Device or resource busy E: module.c: Failed to load module module-alsa-source (argument: device=hw:0): initialization failed. E: main.c: Module load failed. E: main.c: Failed to initialize daemon. Complete output: http://pastebin.com/XaKz4ZN7 Tried to use “module-null-source”, but it is not installed on my pc. # slocate module-null-source # Without a source module specified: http://pastebin.com/aUT5bxrp I'm not sure where to go from here... ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio-0.9.14 on Slackware 12.2 with Droplinegome 2.26
'Twas brillig, and Ron Hermsen at 19/10/10 21:41 did gyre and gimble: Dear Reader, I have problems with pulseaudio not starting up. After changing default.pa, module-alsa-sink did found my soundcard, but PA still fails. I hope someone can help me to get it working. Audio details: “ESS 1869 Plug and Play Audio Drive”. The module that is loaded: snd_es1688 Complete lsmod output: http://pastebin.com/B0rzSrJT Modprobe and lsdev info: http://pastebin.com/rj0xqL01 If I use pulseaudio - --daemonize=false with the default “default.pa” the output I get the following failure messages: E: polkit.c: Cannot set UID on session object. I: main.c: PolicyKit refuses acquire-high-priority privilege. D: module-always-sink.c: Autoloading null-sink as no other sinks detected. E: module-console-kit.c: GetSessionsForUnixUser() call failed: org.freedesktop.DBus.Error.NoReply: Message did not receive a reply (timeout by message bus) E: module.c: Failed to load module module-console-kit (argument: ): initialization failed. E: main.c: Module load failed. E: main.c: Failed to initialize daemon. Complete output: http://pastebin.com/MuxxuLmt From the null-sink I understand that no soundcards is found. aplay -v --device=hw:0 /usr/share/sounds/shutdown.wav works Complete output: http://pastebin.com/Rdp0Ef6t Tried to change “default.pa” with the following lines: load-module module-alsa-sink device=hw:0 load-module module-alsa-source device=hw:0 With this the module-alsa-sink seems to be ok, but module-alsa-source failes. pulseaudio - --daemonize=false output: E: polkit.c: Cannot set UID on session object. I: main.c: PolicyKit refuses acquire-high-priority privilege. D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed E: alsa-util.c: Error opening PCM device hw:0: Device or resource busy E: module.c: Failed to load module module-alsa-source (argument: device=hw:0): initialization failed. E: main.c: Module load failed. E: main.c: Failed to initialize daemon. Complete output: http://pastebin.com/XaKz4ZN7 Tried to use “module-null-source”, but it is not installed on my pc. # slocate module-null-source # Without a source module specified: http://pastebin.com/aUT5bxrp I'm not sure where to go from here... Basically the console kit module is failing to load due to a dbus problem. Also seems that some other application has your Alsa device open (Device or resource busy message indicates this), perhaps this means that you simply can't use your h/w in full duplex mode (e.g. opening the sink prevents the source from actually working?) Anyway, actually two bugs there. FWIW 0.9.14 is a bit of a strange release to use. AFAIK 0.9.16 was the stable version for a while (fixing bugs in 0.9.12 through 0.9.15), but I could be wrong. I'd strongly recommend using 0.9.21 (or rather stable-queue git branch) as many bug fixes have been included since 0.9.14. I'd also recommend newer alsa kernel and libs too. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio + USB Microphone, how use it without recording?
'Twas brillig, and Miguel Beltran R. at 15/10/10 04:13 did gyre and gimble: Hello List I have some music videos with lyrics to use it as a karaoke, but when I reproduce the video and try to sing using the microphone don't work like I expected. There no sound out of mic. Using pavucontrol I saw in Input Devices that appeared a Logitech USB Microphone Analog Mono and the meter is moving, just haven't sound out. What I must do? I want heard the video and Mic What you technically need to do is record from the mic and then play it back through some speakers. PA ships with a convenient way to do this in the form of module-loopback. It basically does this and will also resample as needed to deal with slight variations in the different device clocks involved (e.g. your mic may record at 441002 Hz and your output device may consume at 440998 Hz). Because it has to resample there may be a slight delay. But: pactl load-module module-loopback Then use pavucontrol and under the recording and playback tabs you will see a new stream. Simply move this stream to the right devices for input and output and you will be able to hear the mic on the appropriate output. Hope that helps Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] PulseAudio + USB Microphone, how use it without recording?
2010/10/16 Colin Guthrie gm...@colin.guthr.ie 'Twas brillig, and Miguel Beltran R. at 15/10/10 04:13 did gyre and gimble: Hello List I have some music videos with lyrics to use it as a karaoke, but when I reproduce the video and try to sing using the microphone don't work like I expected. There no sound out of mic. Using pavucontrol I saw in Input Devices that appeared a Logitech USB Microphone Analog Mono and the meter is moving, just haven't sound out. What I must do? I want heard the video and Mic What you technically need to do is record from the mic and then play it back through some speakers. PA ships with a convenient way to do this in the form of module-loopback. It basically does this and will also resample as needed to deal with slight variations in the different device clocks involved (e.g. your mic may record at 441002 Hz and your output device may consume at 440998 Hz). Because it has to resample there may be a slight delay. But: pactl load-module module-loopback Then use pavucontrol and under the recording and playback tabs you will see a new stream. Simply move this stream to the right devices for input and output and you will be able to hear the mic on the appropriate output. Hope that helps Col Thanks Col, it worked ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] PulseAudio + USB Microphone, how use it without recording?
Hello List I have some music videos with lyrics to use it as a karaoke, but when I reproduce the video and try to sing using the microphone don't work like I expected. There no sound out of mic. Using pavucontrol I saw in Input Devices that appeared a Logitech USB Microphone Analog Mono and the meter is moving, just haven't sound out. What I must do? I want heard the video and Mic My current system: -Slackware 13.1 -Gnome Slack build 2.28 -and Rockband Usb Microphone -pulseaudio 0.9.21 lsusb: bash-4.1# lsusb Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 003: ID 05e3:0716 Genesys Logic, Inc. USB 2.0 Multislot Card Reader/Writer Bus 002 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 002 Device 004: ID 046d:0a03 Logitech, Inc. Logitech USB Microphone Bus 002 Device 002: ID 0bc7:0004 X10 Wireless Technology, Inc. X10 Receiver I attach two files with info of my system do you need more info? Thanks. -- Lo bueno de vivir un dia mas es saber que nos queda un dia menos de vida Module #0 Name: module-device-restore Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute state of devices module.version = 0.9.21 Module #1 Name: module-stream-restore Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute/device state of streams module.version = 0.9.21 Module #2 Name: module-card-restore Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Automatically restore profile of cards module.version = 0.9.21 Module #3 Name: module-augment-properties Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Augment the property sets of streams with additional static information module.version = 0.9.21 Module #4 Name: module-alsa-card Argument: device_id=0 name=pci-_04_05.0 card_name=alsa_card.pci-_04_05.0 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1 Usage counter: 12 Properties: module.author = Lennart Poettering module.description = ALSA Card module.version = 0.9.21 Module #5 Name: module-udev-detect Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Detect available audio hardware and load matching drivers module.version = 0.9.21 Module #6 Name: module-bluetooth-discover Argument: Usage counter: n/a Properties: module.author = Joao Paulo Rechi Vita module.description = Detect available bluetooth audio devices and load bluetooth audio drivers module.version = 0.9.21 Module #7 Name: module-esound-protocol-unix Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = ESOUND protocol (UNIX sockets) module.version = 0.9.21 Module #8 Name: module-native-protocol-unix Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Native protocol (UNIX sockets) module.version = 0.9.21 Module #9 Name: module-combine Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Combine multiple sinks to one module.version = 0.9.21 Module #10 Name: module-gconf Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = GConf Adapter module.version = 0.9.21 Module #11 Name: module-default-device-restore Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = Automatically restore the default sink and source module.version = 0.9.21 Module #12 Name: module-rescue-streams Argument: Usage counter: n/a Properties: module.author = Lennart Poettering module.description = When a sink/source is removed, try to move their streams to the default sink/source module.version = 0.9.21 Module #13
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 22/09/10 02:04 did gyre and gimble: iii---I I recompiled kdemultimedia also. On restart I obtained sound on the second login but none subsequenty and cdrom sound is practicly mute ~ a tenth as loud as a hungry mosquito. FWIW, unless you are wroking on new features, I'd recommend using the stable-queue branch upstream rather than master. It's effectively 0.9.21 with ~180 odd fixes on top. I'm hoping to release this as 0.9.22 pretty soon. Anyway, if the volume changes, it's typically due to some client changing it. In the case of logins this can be a bit confusing. Typically your user's PA daemon will ignore any volume changes when it is suspended but it will not actually restore them when it is unsuspended (I think, I need to double check this). Changes are what happens is this: 1. You log in and startup a PA process. 2. You log out, but your PA daemon sticks around for some time. 3. KDM or something similar sets the volume either by running it's own PA or via alsa etc. doesn't really matter. 4. You log in again, before your PA process has died. We look at the system volume and see it is lower than before and adjust ourselves to fit. Ironically, I believe if you wait long enough before logging back in, your PA process will die and when you login it will be restarted and will ultimately restore the volume properly. I *think* that's what's going on, but not 100% sure. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On Monday 20 September 2010 08:37:02 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 20/09/10 00:41 did gyre and gimble: On Tuesday 14 September 2010 13:58:08 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 14/09/10 13:33 did gyre and gimble: On Monday 13 September 2010 11:59:14 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: First of all thanks (or shall I say 'kallu kalay') for the help. but you really do want to use all the patches from this branch since v0.9.21 tag (there will likely be another 10 fixes landing later today) I downloaded the stuff from GIT. The bundle was labelled as [pulseaudio-0.9.19 which is strange considering the released stuff is labelled 0.9.21. I did not clean out the original installation before I compiled/installed the git stuff so a couple of the libraries had 0.9.21 extensions (I merely labelled them as _X). I then recompiled alsa-plugins and so far I have sound on all logins and logouts as well as the cli. So much progress has been made. Yeah this is just a quirk of how the versions are labelled directly from git... I really do need to fix it at some point. ---e) audio CDROMs do not play neither with KsCD nor Kaffeine for any logins This is likely an issue to do with using analog playback rather than digital extraction (DAE) and subsequent software playback. You can probably use a low level alsamixer (alsamixer -c0) to unmute the CDROM element, but really most applications should be using digital extraction and software playback these days AFAIK. I'd ignore this particular problem until all others are addressed :) I still do not have CDROM sound nither with Kaffeine nor KsCD, I tried an internet search but came up with nowt so any pointers you can give will be grtefully received. Hmm, interesting. I'll try and see what I can find out for you. I managed to fix the problem. It was a udev problem. After I updated to udev-162, the /dev/cdrom and /dev/dvd symlinks to my ide cdrom disappeared. I used a simple udev rule to re-create these and now cdrom sound plays in KsCD Kaffeine et al. Thanks again for your help. and if ever you want to try an already made sample of cblfs on you x86_64-pc please let me know. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 21/09/10 11:12 did gyre and gimble: I managed to fix the problem. It was a udev problem. After I updated to udev-162, the /dev/cdrom and /dev/dvd symlinks to my ide cdrom disappeared. I used a simple udev rule to re-create these and now cdrom sound plays in KsCD Kaffeine et al. Ahh good old udev messing things up for us! Excellent. Glad you got it all working :D Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On Tuesday 21 September 2010 12:28:04 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 21/09/10 11:12 did gyre and gimble: I managed to fix the problem. It was a udev problem. After I updated to udev-162, the /dev/cdrom and /dev/dvd symlinks to my ide cdrom disappeared. I used a simple udev rule to re-create these and now cdrom sound plays in KsCD Kaffeine et al. Ahh good old udev messing things up for us! Excellent. Glad you got it all working :D oops had a related problem. I had another machine not on the internet so I copied a download of pulse from GIT and did an installation.:- i beforehand I purged it if pulse-0.9.21/ consolekit and hal) iiI noticed that symlinks to the Makefile in the base direcory pf pulse were missing so I replace these iii---I I recompiled kdemultimedia also. On restart I obtained sound on the second login but none subsequenty and cdrom sound is practicly mute ~ a tenth as loud as a hungry mosquito. pacmd ls is below Welcome to PulseAudio! Use help for usage information. Memory blocks currently allocated: 1, size: 63.9 KiB. Memory blocks allocated during the whole lifetime: 1641, size: 7.8 MiB. Memory blocks imported from other processes: 0, size: 0 B. Memory blocks exported to other processes: 0, size: 0 B. Total sample cache size: 0 B. Default sample spec: s16le 2ch 44100Hz Default channel map: front-left,front-right Default sink name: alsa_output.pci-_00_07.0.analog-stereo Default source name: alsa_input.pci-_00_07.0.analog-stereo Memory blocks of type POOL: 1 allocated/785 accumulated. Memory blocks of type POOL_EXTERNAL: 0 allocated/0 accumulated. Memory blocks of type APPENDED: 0 allocated/0 accumulated. Memory blocks of type USER: 0 allocated/0 accumulated. Memory blocks of type FIXED: 0 allocated/154 accumulated. Memory blocks of type IMPORTED: 0 allocated/702 accumulated. 20 module(s) loaded. index: 0 name: module-device-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute state of devices module.version = 0.9.19-562-g395da-dirty index: 1 name: module-stream-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute/device state of streams module.version = 0.9.19-562-g395da-dirty index: 2 name: module-card-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore profile of cards module.version = 0.9.19-562-g395da-dirty index: 3 name: module-augment-properties argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Augment the property sets of streams with additional static information module.version = 0.9.19-562-g395da-dirty index: 4 name: module-alsa-card argument: device_id=3 name=pci-_00_07.0 card_name=alsa_card.pci-_00_07.0 namereg_fail=false tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1 used: 0 load once: no properties: module.author = Lennart Poettering module.description = ALSA Card module.version = 0.9.19-562-g395da-dirty index: 5 name: module-udev-detect argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Detect available audio hardware and load matching drivers module.version = 0.9.19-562-g395da-dirty index: 6 name: module-bluetooth-discover argument: used: -1 load once: yes properties: module.author = Joao Paulo Rechi Vita module.description = Detect available bluetooth audio devices and load bluetooth audio drivers module.version = 0.9.19-562-g395da-dirty index: 7 name: module-esound-protocol-unix argument: used: -1 load once: no properties: module.author = Lennart Poettering module.description = ESOUND protocol (UNIX sockets) module.version = 0.9.19-562-g395da-dirty index: 8 name: module-dbus-protocol argument: used: -1 load once: yes properties: module.author = Tanu Kaskinen module.description = D-Bus interface module.version = 0.9.19-562-g395da-dirty index:
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 20/09/10 00:41 did gyre and gimble: On Tuesday 14 September 2010 13:58:08 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 14/09/10 13:33 did gyre and gimble: On Monday 13 September 2010 11:59:14 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: First of all thanks (or shall I say 'kallu kalay') for the help. but you really do want to use all the patches from this branch since v0.9.21 tag (there will likely be another 10 fixes landing later today) I downloaded the stuff from GIT. The bundle was labelled as [pulseaudio-0.9.19 which is strange considering the released stuff is labelled 0.9.21. I did not clean out the original installation before I compiled/installed the git stuff so a couple of the libraries had 0.9.21 extensions (I merely labelled them as _X). I then recompiled alsa-plugins and so far I have sound on all logins and logouts as well as the cli. So much progress has been made. Yeah this is just a quirk of how the versions are labelled directly from git... I really do need to fix it at some point. ---e) audio CDROMs do not play neither with KsCD nor Kaffeine for any logins This is likely an issue to do with using analog playback rather than digital extraction (DAE) and subsequent software playback. You can probably use a low level alsamixer (alsamixer -c0) to unmute the CDROM element, but really most applications should be using digital extraction and software playback these days AFAIK. I'd ignore this particular problem until all others are addressed :) I still do not have CDROM sound nither with Kaffeine nor KsCD, I tried an internet search but came up with nowt so any pointers you can give will be grtefully received. Hmm, interesting. I'll try and see what I can find out for you. When playing via KsCD, does everything look as if it *is* playing but you just cannot hear sound? If so, when something is playing, can you run pacmd list and then attach the output here? Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On Tuesday 14 September 2010 13:58:08 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 14/09/10 13:33 did gyre and gimble: On Monday 13 September 2010 11:59:14 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: First of all thanks (or shall I say 'kallu kalay') for the help. but you really do want to use all the patches from this branch since v0.9.21 tag (there will likely be another 10 fixes landing later today) I downloaded the stuff from GIT. The bundle was labelled as [pulseaudio-0.9.19 which is strange considering the released stuff is labelled 0.9.21. I did not clean out the original installation before I compiled/installed the git stuff so a couple of the libraries had 0.9.21 extensions (I merely labelled them as _X). I then recompiled alsa-plugins and so far I have sound on all logins and logouts as well as the cli. So much progress has been made. ---e) audio CDROMs do not play neither with KsCD nor Kaffeine for any logins This is likely an issue to do with using analog playback rather than digital extraction (DAE) and subsequent software playback. You can probably use a low level alsamixer (alsamixer -c0) to unmute the CDROM element, but really most applications should be using digital extraction and software playback these days AFAIK. I'd ignore this particular problem until all others are addressed :) I still do not have CDROM sound nither with Kaffeine nor KsCD, I tried an internet search but came up with nowt so any pointers you can give will be grtefully received. Once again many thanks for your help. sincerely sibu xolo. Col ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On Monday 13 September 2010 11:59:14 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: First of all thanks (or shall I say 'kallu kalay') for the help. I'm not sure why udev-detect would cause problems. I would suggest to double check your udev setup and make sure the relevant udev rules are present on your system. udev 147 is pretty old, perhaps a newer version is needed (I can't remember the minimum system requirements), but certainly: /lib/udev/rules.d/78-sound-card.rules I updated udev to udev-162. And I can report some progress and some glitches. PROGRESS: ---a) the first login and logout via kdm gives the welcome and exit sounds ---b)the kde/~Multimedia dialog has a dummy greyed-out icon and another as 'Internal Audio analog stereo' ---c) aplay plays wav files in konsole -as well as on the cli in/before/after startkde/kdm-logout GLITCHES --- ---d) kde/sound is mute for all other logins and logouts bar the first ---e) audio CDROMs do not play neither with KsCD nor Kaffeine for any logins - 'pacmd -ls' looks like this after login via kdm Welcome to PulseAudio! Use help for usage information. Memory blocks currently allocated: 1, size: 63.9 KiB. Memory blocks allocated during the whole lifetime: 85, size: 2.6 MiB. Memory blocks imported from other processes: 0, size: 0 B. Memory blocks exported to other processes: 0, size: 0 B. Total sample cache size: 0 B. Default sample spec: s16le 2ch 44100Hz Default channel map: front-left,front-right Default sink name: alsa_output.pci-_00_07.0.analog-stereo Default source name: alsa_input.pci-_00_07.0.analog-stereo Memory blocks of type POOL: 1 allocated/28 accumulated. Memory blocks of type POOL_EXTERNAL: 0 allocated/0 accumulated. Memory blocks of type APPENDED: 0 allocated/0 accumulated. Memory blocks of type USER: 0 allocated/0 accumulated. Memory blocks of type FIXED: 0 allocated/57 accumulated. Memory blocks of type IMPORTED: 0 allocated/0 accumulated. 22 module(s) loaded. index: 0 name: module-device-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute state of devices module.version = 0.9.21 index: 1 name: module-stream-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute/device state of streams module.version = 0.9.21 index: 2 name: module-card-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore profile of cards module.version = 0.9.21 index: 3 name: module-augment-properties argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Augment the property sets of streams with additional static information module.version = 0.9.21 index: 4 name: module-alsa-card argument: device_id=0 name=pci-_00_07.0 card_name=alsa_card.pci-_00_07.0 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1 used: 0 load once: no properties: module.author = Lennart Poettering module.description = ALSA Card module.version = 0.9.21 index: 5 name: module-udev-detect argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Detect available audio hardware and load matching drivers module.version = 0.9.21 index: 6 name: module-bluetooth-discover argument: used: -1 load once: yes properties: module.author = Joao Paulo Rechi Vita module.description = Detect available bluetooth audio devices and load bluetooth audio drivers module.version = 0.9.21 index: 7 name: module-esound-protocol-unix argument: used: -1 load once: no properties: module.author = Lennart Poettering module.description = ESOUND protocol (UNIX sockets) module.version = 0.9.21 index: 8 name: module-native-protocol-unix argument: used: -1 load once: no properties: module.author = Lennart Poettering module.description = Native protocol (UNIX sockets)
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 14/09/10 13:33 did gyre and gimble: On Monday 13 September 2010 11:59:14 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: First of all thanks (or shall I say 'kallu kalay') for the help. I'm not sure why udev-detect would cause problems. I would suggest to double check your udev setup and make sure the relevant udev rules are present on your system. udev 147 is pretty old, perhaps a newer version is needed (I can't remember the minimum system requirements), but certainly: /lib/udev/rules.d/78-sound-card.rules I updated udev to udev-162. And I can report some progress and some glitches. PROGRESS: ---a) the first login and logout via kdm gives the welcome and exit sounds ---b)the kde/~Multimedia dialog has a dummy greyed-out icon and another as 'Internal Audio analog stereo' Excellent. Both are expected outcomes. Because PA has been run with the Dummy Output in the past, it will remember this for you and keep it in the list as a previously seen sink (the same view would be presented for USB devices you plug in and subsequently remove). You can effectively ignore it or remove the device-manager tdb file in the ~/.pulse folder to purge it (one of these days I'll implement some kind of GUI to remove it). ---c) aplay plays wav files in konsole -as well as on the cli in/before/after startkde/kdm-logout Cool. GLITCHES --- ---d) kde/sound is mute for all other logins and logouts bar the first Hmm, is that with different users or logging out and back in with the same user? When you say mute do you really mean mute or do you just mean that you cannot hear anything. Does the device list in the kde/~Multimedia dialog look the same? Also are you using a Display Manager and if so which one (KDM?), and are you using the stable-queue PA from git or the plain 0.9.21 tarball? The reason I ask is that if it really is muting then it's probably related to a volume adjustment race condition that may be getting in the way. It's been fixed a long time ago in the stable-queue branch and all the distros I know are shipping this fix (well they ship all the patches from stable-queue). The particular fix is: http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=540ec7b961256d6c7702448ca995f61268064190 but you really do want to use all the patches from this branch since v0.9.21 tag (there will likely be another 10 fixes landing later today) More details on the specific bug here: http://pulseaudio.org/ticket/572#comment:19 ---e) audio CDROMs do not play neither with KsCD nor Kaffeine for any logins This is likely an issue to do with using analog playback rather than digital extraction (DAE) and subsequent software playback. You can probably use a low level alsamixer (alsamixer -c0) to unmute the CDROM element, but really most applications should be using digital extraction and software playback these days AFAIK. I'd ignore this particular problem until all others are addressed :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On 09/12/10 19:04, sibu xolo wrote:
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
Try the following, it should allow alsamixer to control the default device
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
#pcm.!default {
#type pulse
#}
#ctl.!default {
#type pulse
#}
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Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 13/09/10 02:04 did gyre and gimble: a) I am using system5 init and alsamixer refuses to start on the CLI at run- level 3 reporting pulseaaudio' connection refused (or somesuch) but... I guess PA cannot start on the CLI for some reason. alsamixer or aplay should autospawn PA if it's not running (which would be typical with a console login). I suspect not being able to connect to dbus/consolekit is the problem, but to find out, run pulseaudio -vvv from the cli rather than aplay and it should tell you the reasons for not starting. b) alsamixer starts after login in kde in konsole; sound devices are correctly reported (on the cli); aplay reports no errors but no sound is heard and outputs are not mute. Check the profiles in pavucontrol's configuration tab. Post the output from pacmd ls to allow others to have a look at the setup. b) Still in kde, When the ownership/permissions on /usr/lib/dbus-1.0/dbus- daemon-launch-helper file are set as shown above the KDE System Settings/Multimedia dialog reports a dummy device $ ll /lib64/dbus-1/dbus-daemon-launch-helper -rwsr-x--- 1 root messagebus 47384 2010-03-24 09:46 /lib64/dbus-1/dbus-daemon-launch-helper Same perms here... I'm not sure why dbus failure would results in a dummy device. I wonder if it's simply that PA cannot reserve the audio device via the dbus based device reservation protocol and thus can't load the sink. Even still dbus based activation seems to work for me :s Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
On Monday 13 September 2010 09:28:22 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 02:04 did gyre and gimble: a) I am using system5 init and alsamixer refuses to start on the CLI at run- level 3 reporting pulseaaudio' connection refused (or somesuch) but... I guess PA cannot start on the CLI for some reason. alsamixer or aplay should autospawn PA if it's not running (which would be typical with a console login). I suspect not being able to connect to dbus/consolekit is the problem, but to find out, run pulseaudio -vvv from the cli rather than aplay and it should tell you the reasons for not starting. here is the ooutput from running pulseaudio -vvv on the cli BEFORE kde starts I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted D: core-rtclock.c: Timer slack is set to 50 us. I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory I: main.c: This is PulseAudio 0.9.21 D: main.c: Compilation host: x86_64-unknown-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -Wall -W -Wextra -pipe -Wno-long-long - Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef - Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self - Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict- prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn - Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno- unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common - fdiagnostics-show-option D: main.c: Running on host: Linux x86_64 2.6.35.4nbf8DRIm #1 SMP PREEMPT Wed Sep 8 15:10:46 BST 2010 D: main.c: Found 2 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: yes D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimized build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 1385ab3a82291ea2800476890006. I: main.c: Using runtime directory /home/sx/.pulse/1385ab3a82291ea2800476890006-runtime. I: main.c: Using state directory /home/sx/.pulse. I: main.c: Using modules directory /opt/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Bon appetit! I: cpu-x86.c: CPU flags: MMX SSE SSE2 SSE3 MMXEXT 3DNOW 3DNOWEXT I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: database-gdbm.c: Opened GDBM database '/home/sx/.pulse/1385ab3a82291ea2800476890006-device-volumes.x86_64- unknown-linux-gnu.gdbm' I: module-device-restore.c: Sucessfully opened database file '/home/sx/.pulse/1385ab3a82291ea2800476890006-device-volumes'. I: module.c: Loaded module-device-restore (index: #0; argument: ). D: database-gdbm.c: Opened GDBM database '/home/sx/.pulse/1385ab3a82291ea2800476890006-stream-volumes.x86_64- unknown-linux-gnu.gdbm' I: module-stream-restore.c: Sucessfully opened database file '/home/sx/.pulse/1385ab3a82291ea2800476890006-stream-volumes'. I: module.c: Loaded module-stream-restore (index: #1; argument: ). D: database-gdbm.c: Opened GDBM database '/home/sx/.pulse/1385ab3a82291ea2800476890006-card-database.x86_64- unknown-linux-gnu.gdbm' I: module-card-restore.c: Sucessfully opened database file '/home/sx/.pulse/1385ab3a82291ea2800476890006-card-database'. I: module.c: Loaded module-card-restore (index: #2; argument: ). I: module.c: Loaded module-augment-properties (index: #3; argument: ). D: cli-command.c: Checking for existance of '/opt/lib/pulse-0.9.21/modules/module-udev-detect.so': success I: module-udev-detect.c: Found 0 cards. I: module.c: Loaded module-udev-detect (index: #4; argument: ). D: cli-command.c: Checking for existance of '/opt/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': success D: dbus-util.c: Successfully connected to D-Bus system bus 244284b9a8358759ce53123d0008 as :1.7 D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: bluetooth-util.c: Bluetooth daemon is apparently not available. I: module.c: Loaded module-bluetooth-discover (index: #5; argument: ). D: cli-command.c: Checking for existance of '/opt/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success I: module.c: Loaded module-esound-protocol-unix (index: #6; argument: ). I: module.c: Loaded module-native-protocol-unix (index: #7; argument: ). D: cli-command.c: Checking for existance of
Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
'Twas brillig, and sibu xolo at 13/09/10 11:10 did gyre and gimble: On Monday 13 September 2010 09:28:22 Colin Guthrie wrote: 'Twas brillig, and sibu xolo at 13/09/10 02:04 did gyre and gimble: a) I am using system5 init and alsamixer refuses to start on the CLI at run- level 3 reporting pulseaaudio' connection refused (or somesuch) but... I guess PA cannot start on the CLI for some reason. alsamixer or aplay should autospawn PA if it's not running (which would be typical with a console login). I suspect not being able to connect to dbus/consolekit is the problem, but to find out, run pulseaudio -vvv from the cli rather than aplay and it should tell you the reasons for not starting. here is the ooutput from running pulseaudio -vvv on the cli BEFORE kde starts OK, you've got two problems here: D: cli-command.c: Checking for existance of '/opt/lib/pulse-0.9.21/modules/module-udev-detect.so': success I: module-udev-detect.c: Found 0 cards. I: module.c: Loaded module-udev-detect (index: #4; argument: ). For whatever reason, udev detection has failed and is finding no cards. This means an Auto Null sink will be loaded rather than accessing your real h/w. E: x11wrap.c: XOpenDisplay() failed E: module.c: Failed to load module module-x11-bell (argument: sample=bell- windowing-system): initialization failed. E: main.c: Module load failed. E: main.c: Failed to initialize daemon. This is interesting. The directive for loading module-x11-bell was removed from default.pa a *long* time ago, but apparently your default.pa is still trying to load it. But as your default.pa tries to load module-udev-detect, which was added *after* module-x11-bell was removed, it suggests you have edited the default.pa we ship with your own version. If you want the x11-bell module to be loaded, you should really add it to the /usr/bin/start-pulseaudio-x11 script instead. You can also put the load-module line in default.pa inside a .nofail .fail block. This will mean it will not cause daemon startup to fail, but it obviously will not actually load the module when X11 login actually occurs. That's why putting it in start-pulseaudio-x11 is better. here is the ooutput from running pacmd ls on the cli AFTER kde starts 1 sink(s) available. * index: 0 name: auto_null driver: module-null-sink.c flags: DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 1000 volume: 0: 60% 1: 60% 0: -13.31 dB 1: -13.31 dB balance 0.00 base volume: 100% 0.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 1722 KiB max rewind: 1722 KiB monitor source: 0 sample spec: s16le 2ch 44100Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 configured latency: 0.00 ms; range is 0.50 .. 1.00 ms module: 10 properties: device.description = Dummy Output device.class = abstract device.icon_name = audio-card Here is the ultimately issue. The Dummy Output. As udev-detect could not detect any cards, you get only the dummy output. I'm not sure why udev-detect would cause problems. I would suggest to double check your udev setup and make sure the relevant udev rules are present on your system. udev 147 is pretty old, perhaps a newer version is needed (I can't remember the minimum system requirements), but certainly: /lib/udev/rules.d/78-sound-card.rules Should be present and should control things in this regard. I'd certainly double check that ACLs are written properly on the /dev/snd/* nodes (getfacl /dev/snd/*) and that your user session is definitely listed in ck-list-sessions as being active. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio runs, but there's no sound
hi, finally after compile-updating 3/4 of my system, i was able to compile and make pulseaudio up and running. It runs. applications such as mplayer/sype etc run, but there's no sound :( checked out mixer, unmuted everything, no success. i'm using kernel 2.6.25.9 with no support for udev and high resolution timers. here's the output of pulseaudio -v I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory I: main.c: This is PulseAudio 0.9.21 I: main.c: Page size is 4096 bytes I: main.c: Machine ID is 31be6dfec7b88bfea6702b9848d2f7aa. I: main.c: Using runtime directory /home/den/.pulse/31be6dfec7b88bfea6702b9848d2f7aa-runtime. I: main.c: Using state directory /home/den/.pulse. I: main.c: Using modules directory /usr/local/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Dude, your kernel stinks! The chef's recommendation today is Linux with high-resolution timers enabled! I: cpu-x86.c: CPU flags: MMX SSE SSE2 SSE3 MMXEXT 3DNOW 3DNOWEXT I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. I: module-device-restore.c: Sucessfully opened database file '/home/den/.pulse/31be6dfec7b88bfea6702b9848d2f7aa-device-volumes'. I: module.c: Loaded module-device-restore (index: #0; argument: ). I: module-stream-restore.c: Sucessfully opened database file '/home/den/.pulse/31be6dfec7b88bfea6702b9848d2f7aa-stream-volumes'. I: module.c: Loaded module-stream-restore (index: #1; argument: ). I: module-card-restore.c: Sucessfully opened database file '/home/den/.pulse/31be6dfec7b88bfea6702b9848d2f7aa-card-database'. I: module.c: Loaded module-card-restore (index: #2; argument: ). I: module.c: Loaded module-augment-properties (index: #3; argument: ). W: module.c: module-detect is deprecated: Please use module-udev-detect instead of module-detect! N: alsa-util.c: Disabling timer-based scheduling because high-resolution timers are not available from the kernel. I: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 48000 Hz. I: alsa-sink.c: Successfully opened device front:0. I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-sink.c: Successfully enabled mmap() mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory I: alsa-mixer.c: Successfully attached to mixer 'hw:0' I: module-device-restore.c: Restoring volume for sink alsa_output.0.analog-stereo. I: sink.c: Created sink 0 alsa_output.0.analog-stereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right I: sink.c: alsa.resolution_bits = 16 I: sink.c: device.api = alsa I: sink.c: device.class = sound I: sink.c: alsa.class = generic I: sink.c: alsa.subclass = generic-mix I: sink.c: alsa.name = ALi M5455 I: sink.c: alsa.id = Intel ICH I: sink.c: alsa.subdevice = 0 I: sink.c: alsa.subdevice_name = subdevice #0 I: sink.c: alsa.device = 0 I: sink.c: alsa.card = 0 I: sink.c: alsa.card_name = ALi M5455 I: sink.c: alsa.long_card_name = ALi M5455 with ALC655 at irq 23 I: sink.c: device.string = front:0 I: sink.c: device.buffering.buffer_size = 19188 I: sink.c: device.buffering.fragment_size = 6396 I: sink.c: device.access_mode = mmap I: sink.c: device.profile.name = analog-stereo I: sink.c: device.profile.description = Analog Stereo I: sink.c: device.description = ALi M5455 Analog Stereo I: sink.c: alsa.mixer_name = Realtek ALC655 rev 0 I: sink.c: alsa.components = AC97a:414c4760 I: sink.c: device.icon_name = audio-card-analog I: source.c: Created source 0 alsa_output.0.analog-stereo.monitor with sample spec s16le 2ch 48000Hz and channel map front-left,front-right I: source.c: device.description = Monitor of ALi M5455 Analog Stereo I: source.c: device.class = monitor I: source.c: device.icon_name = audio-input-microphone I: alsa-sink.c: Using 3.0 fragments of size 6396 bytes (33.31ms), buffer size is 19188 bytes (99.94ms) I: alsa-sink.c: Hardware volume ranges from -127.50 dB to 12.00 dB. I: alsa-sink.c: Fixing base volume to -12.00 dB I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: alsa-sink.c: Using hardware mute control. I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5. I: alsa-sink.c: Starting playback. I: module.c: Loaded module-alsa-sink (index: #4; argument: device_id=0). N: alsa-util.c: Disabling timer-based scheduling because high-resolution timers are not available from the kernel. I: alsa-util.c:
Re: [pulseaudio-discuss] Pulseaudio runs, but there's no sound
'Twas brillig, and Ozgur at 12/09/10 15:19 did gyre and gimble: hi, finally after compile-updating 3/4 of my system, i was able to compile and make pulseaudio up and running. It runs. applications such as mplayer/sype etc run, but there's no sound :( checked out mixer, unmuted everything, no success. i'm using kernel 2.6.25.9 with no support for udev and high resolution timers. here's the output of pulseaudio -v any suggestions? Some general things: 1. Make sure you're compliing from the stable-queue branch in git. It's got ~170 odd patches on top of 0.9.21 that will likely become 0.9.22 in the not too distant future. 2. I'd recommend module-hal-detect rather than module-detect. This is deprecated in favour of udev, but it's still better for your old kernel. 3. I'd recommend updating your kernel to the latest one. You'll find a lot of alsa driver updates and other fixes that could ultimate affect things. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio runs, but there's no sound
On Sun, 2010-09-12 at 14:19 +, Ozgur wrote: hi, finally after compile-updating 3/4 of my system, i was able to compile and make pulseaudio up and running. It runs. applications such as mplayer/sype etc run, but there's no sound :( checked out mixer, unmuted everything, no success. i'm using kernel 2.6.25.9 with no support for udev and high resolution timers. here's the output of pulseaudio -v I didn't find anything wrong in that output. any suggestions? Maybe the output of pulseaudio -v shows something interesting, if you try to play something. So, post a pulseaudio log with for example one mplayer run. -- Tanu Kaskinen ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus
Greetings,
I am new to this list. I have a computer with these:
-cpu-amd64-2 cores, dsp: HDA NVidia/Realtek ALC662 rev1
-o/s cblfs-linux 64-bit kernel-2.6.35.4, xorg-7.5 kde-4.4.5, udev-147
-audio-progs: alsa-1.0.23. pulse-0.9.21 (default configurations in
/etc/pulse)
my /etc/asound.conf file has these:-
==
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
my udev-rule for alsa looks like this:-
==
# Give the audio group ownership of sound devices
SUBSYSTEM==sound, GROUP=audio
SUBSYSTEM==snd, GROUP=audio
# ALSA Devices
# When a sound device is detected, restore the volume settings
KERNEL==controlC[0-9]*, NAME=snd/%k, ACTION==add,
RUN+=/usr/sbin/alsactl restore %n
# Sound devices
KERNEL==admmidi*, GROUP=audio
KERNEL==adsp*, GROUP=audio
KERNEL==aload*,GROUP=audio
KERNEL==amidi*,GROUP=audio
KERNEL==amixer*, GROUP=audio
KERNEL==audio*,GROUP=audio
KERNEL==dmfm*, GROUP=audio
KERNEL==dmmidi*, GROUP=audio
KERNEL==dsp*, GROUP=audio
KERNEL==midi*, GROUP=audio
KERNEL==mixer*,GROUP=audio
KERNEL==music, GROUP=audio
KERNEL==sequencer*,GROUP=audio
KERNEL==by-path/*, GROUP=audio
==
sound devices in /dev appear as:-
==
sx [ ~ ]# ls -l /dev/snd
total 0
drwxr-xr-x 2 root root 80 Sep 12 2010 by-path
crw-rw 1 root audio 116, 5 Sep 12 2010 controlC0
crw-rw 1 root audio 116, 12 Sep 12 2010 controlC1
crw-rw 1 root audio 116, 11 Sep 12 2010 hwC1D0
crw-rw 1 root audio 116, 4 Sep 12 2010 pcmC0D0p
crw-rw 1 root audio 116, 10 Sep 12 2010 pcmC1D0c
crw-rw 1 root audio 116, 9 Sep 12 2010 pcmC1D0p
crw-rw 1 root audio 116, 8 Sep 12 2010 pcmC1D1c
crw-rw 1 root audio 116, 7 Sep 12 2010 pcmC1D1p
crw-rw 1 root audio 116, 6 Sep 12 2010 pcmC1D2c
crw-rw 1 root audio 116, 3 Sep 12 2010 seq
crw-rw 1 root audio 116, 2 Sep 12 2010 timer
sx [ ~ ]# ls -l /dev/audio
crw-rw 1 root audio 14, 4 Sep 12 2010 /dev/audio
sx [ ~ ]# ls -l /dev/dsp
crw-rw 1 root audio 14, 3 Sep 12 2010 /dev/dsp
sx [ ~ ]# ls -l /dev/mixer*
crw-rw 1 root audio 14, 0 Sep 12 2010 /dev/mixer
crw-rw 1 root audio 14, 16 Sep 12 2010 /dev/mixer1
sx [ ~ ]# ls -l /dev/audio
crw-rw 1 root audio 14, 4 Sep 12 2010 /dev/audio
sx [ ~ ]# ls -l /usr/sbin/alsa*
-rwxr-xr-x 1 root root 35445 Sep 1 21:09 /usr/sbin/alsaconf
-rwxr-xr-x 1 root root 213034 Sep 1 21:09 /usr/sbin/alsactl
alsa plugins; appear as:-
==
sx [ ~ ]# ls -l /usr/lib/alsa-lib
total 1376
-rw-r--r-- 1 root root 11034 Sep 12 21:50 libasound_module_conf_pulse.a
-rwxr-xr-x 1 root root 1979 Sep 12 21:50 libasound_module_conf_pulse.la
-rwxr-xr-x 1 root root 12117 Sep 12 21:50 libasound_module_conf_pulse.so
-rw-r--r-- 1 root root 88426 Sep 12 21:50 libasound_module_ctl_arcam_av.a
-rwxr-xr-x 1 root root 1098 Sep 12 21:50 libasound_module_ctl_arcam_av.la
-rwxr-xr-x 1 root root 63182 Sep 12 21:50 libasound_module_ctl_arcam_av.so
-rw-r--r-- 1 root root 39098 Sep 12 21:50 libasound_module_ctl_oss.a
-rwxr-xr-x 1 root root 1063 Sep 12 21:50 libasound_module_ctl_oss.la
-rwxr-xr-x 1 root root 32449 Sep 12 21:50 libasound_module_ctl_oss.so
-rw-r--r-- 1 root root 84188 Sep 12 21:50 libasound_module_ctl_pulse.a
-rwxr-xr-x 1 root root 1972 Sep 12 21:50 libasound_module_ctl_pulse.la
-rwxr-xr-x 1 root root 60745 Sep 12 21:50 libasound_module_ctl_pulse.so
-rw-r--r-- 1 root root 88870 Sep 12 21:50 libasound_module_pcm_a52.a
-rwxr-xr-x 1 root root 1095 Sep 12 21:50 libasound_module_pcm_a52.la
-rwxr-xr-x 1 root root 63461 Sep 12 21:50 libasound_module_pcm_a52.so
-rw-r--r-- 1 root root 41592 Sep 12 21:50 libasound_module_pcm_jack.a
-rwxr-xr-x 1 root root 1132 Sep 12 21:50 libasound_module_pcm_jack.la
-rwxr-xr-x 1 root root 34058 Sep 12 21:50 libasound_module_pcm_jack.so
-rw-r--r-- 1 root root 38126 Sep 12 21:50 libasound_module_pcm_oss.a
-rwxr-xr-x 1 root root 1063 Sep 12 21:50 libasound_module_pcm_oss.la
-rwxr-xr-x 1 root root 30615 Sep 12 21:50 libasound_module_pcm_oss.so
-rw-r--r-- 1 root root 91208 Sep 12 21:50 libasound_module_pcm_pulse.a
-rwxr-xr-x 1 root root 1972 Sep 12 21:50 libasound_module_pcm_pulse.la
-rwxr-xr-x 1 root root 67065 Sep 12 21:50 libasound_module_pcm_pulse.so
-rw-r--r-- 1 root root 33066 Sep 12 21:50 libasound_module_pcm_speex.a
-rwxr-xr-x 1 root root 1122 Sep 12 21:50 libasound_module_pcm_speex.la
-rwxr-xr-x 1
Re: [pulseaudio-discuss] pulseaudio-discuss Digest, Vol 65, Issue 10
Message: 2 Date: Sat, 14 Aug 2010 07:30:16 +0300 From: Tanu Kaskinen ta...@iki.fi Subject: Re: [pulseaudio-discuss] These ideas I've had for the better part of a month, forgive me for there length. To: pulseaudio-discuss@mail.0pointer.de Message-ID: 1281760216.5451.12.ca...@jarl Content-Type: text/plain; charset=UTF-8 On Sat, 2010-08-07 at 12:47 -0500, Mike Mestnik wrote: There is a cause to add several stream types to pulse audio for the support of all the hardware's features. Current stream types can be described by ~3 dimensions (channels, bits, bit-format, rate). The first that would be necessary is raw. That is pre-mixed information that can't be further mixed. Each stream is to be given a priority ~0-15 where default for existing streams is 7 and raw streams default to 11. This makes pre-mixed sounds cause all other sounds to be silenced so they can be played. The effect is the ability to play videos(ac3/dts) while other applications are emitting data, no need to close browser/flash. The first optional extension is the ability to indicate to sources that they will not be played in the form of a warning returned to data passed and a signal indicating that play can continue. I don't know if you've followed the recent AC3 discussion, but some support for raw (aka passthrough) streams has been implemented now. Currently the logic doesn't give precedence for passthrough streams. It might make sense to do so, though. A notable improvement over the current ALSA-passthrough is the addition of framing. Currently pausing and restarting a pre-mixed stream causes loud pops and clicks, this is undesirable and as such pulse audio can offer a solution. The current code doesn't understand such frames, but I think this idea is also a good one to keep in mind. The next stream type should be a pulse audio internal type(s) used for the mixing of mp3/ac3 and perhaps dts streams. It's my understanding that ac3 uses vectors to describe 6 channels of sound much like a 2ch mp3. If I'm not mistaken these vectors can be combined mathematically as-is to represent the combination of two sounds. This would be a great achivement over any uncompress/combine/recompress senerio. Do you have any pointers to how this is done? If it's true that there really is a way to mix compressed streams without uncompressing first, then maybe volume adjustment is possible too? I'm unsure of how this would work, but it'd be a lot of vector math. Volume adjustment should be possible by setting DRC options and perhaps other general volume tags. Adjusting the volume is difficult because it would involve cos and tan plus vector math, if I understand the format correctly. What I was suggesting is a short-cut method to doing the full uncompress/recompress. The result should be a playable stream that is encoded using compression, but is likely not vary well compressed... The resulting stream may well be bigger then it's uncompressed counterpart. The idea being that space is less-precious then CPU time. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio debug mode
'Twas brillig, and Chris at 26/07/10 23:40 did gyre and gimble: On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: What is the best way to start PA for debugging and still have all the usual clients running? If you mean having all the clients connect (e.g. applications with libcanberra support or similar for sound events), then there are basically two ways. The first is as Luke suggests. These clients will automatically reconnect to PA if they need to (provided you have a vaguely recent libcanberra), after it is restarted and run in debug mode. Alternatively you can simply set debug-level to debug in daemon.conf (in /etc/pulse or ~/.pulse), and then grep pulseaudio /var/log/messages Col Colin, the link below is for some more debug output. Notice in the first section that 8 seconds after spamd starts processing a message the the Alsa error starts, 2 seconds after that the overruns start. Notice in line 145 that it took 145 seconds to process a message, that's about 125 too long. I've noticed that when I start getting the overrun errors that the processing of a message takes forever, though this doesn't happen every time, just periodically. All I know is that while this is going on the drive is constantly being accessed for minutes at a time in the first case from 9:03 to 9:08. http://pastebin.com/tZNYaqRV OK, I'll prepare some packages for you so that we can start to isolate what queue it is that is causing the problem. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio debug mode
On Tue, 2010-07-27 at 08:37 +0100, Colin Guthrie wrote: 'Twas brillig, and Chris at 26/07/10 23:40 did gyre and gimble: On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: What is the best way to start PA for debugging and still have all the usual clients running? If you mean having all the clients connect (e.g. applications with libcanberra support or similar for sound events), then there are basically two ways. The first is as Luke suggests. These clients will automatically reconnect to PA if they need to (provided you have a vaguely recent libcanberra), after it is restarted and run in debug mode. Alternatively you can simply set debug-level to debug in daemon.conf (in /etc/pulse or ~/.pulse), and then grep pulseaudio /var/log/messages Col Colin, the link below is for some more debug output. Notice in the first section that 8 seconds after spamd starts processing a message the the Alsa error starts, 2 seconds after that the overruns start. Notice in line 145 that it took 145 seconds to process a message, that's about 125 too long. I've noticed that when I start getting the overrun errors that the processing of a message takes forever, though this doesn't happen every time, just periodically. All I know is that while this is going on the drive is constantly being accessed for minutes at a time in the first case from 9:03 to 9:08. http://pastebin.com/tZNYaqRV OK, I'll prepare some packages for you so that we can start to isolate what queue it is that is causing the problem. Col Thanks Colin, looking forward to them. Chris -- Chris KeyID 0xE372A7DA98E6705C signature.asc Description: This is a digitally signed message part ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio debug mode
On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: What is the best way to start PA for debugging and still have all the usual clients running? If you mean having all the clients connect (e.g. applications with libcanberra support or similar for sound events), then there are basically two ways. The first is as Luke suggests. These clients will automatically reconnect to PA if they need to (provided you have a vaguely recent libcanberra), after it is restarted and run in debug mode. Alternatively you can simply set debug-level to debug in daemon.conf (in /etc/pulse or ~/.pulse), and then grep pulseaudio /var/log/messages Col Colin, the link below is for some more debug output. Notice in the first section that 8 seconds after spamd starts processing a message the the Alsa error starts, 2 seconds after that the overruns start. Notice in line 145 that it took 145 seconds to process a message, that's about 125 too long. I've noticed that when I start getting the overrun errors that the processing of a message takes forever, though this doesn't happen every time, just periodically. All I know is that while this is going on the drive is constantly being accessed for minutes at a time in the first case from 9:03 to 9:08. http://pastebin.com/tZNYaqRV -- Chris KeyID 0xE372A7DA98E6705C signature.asc Description: This is a digitally signed message part ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio debug mode
On Fri, Jul 23, 2010 at 03:04:01AM CEST, Chris wrote: What is the best way to start PA for debugging and still have all the usual clients running? You are probably best off to run pulseaudio in a terminal with - specified on the command-line. This will give you a rather verbose log of what pulseaudio is doing. Then just pipe the data to a file for later analysis. You may also need to turn pulseaudio autospawning off to prevent pulseaudio automatically starting again when you kill it to be run in a terminal. Luke ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio debug mode
On 2010-07-23 03:04, Chris wrote: What is the best way to start PA for debugging and still have all the usual clients running? pacmd set-log-level 4 ...if your log target is set to syslog, just watch it grow :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
