Re: [pulseaudio-discuss] Two identical USB sound-cards - second card fails to load because card-name found in hashmap.
OK, well, thanks for not jumping down my throat. As I said, I really don't have a clue, but figured that could possibly be the case. If that isn't it, then I'm sorry, I don't have a slightest idea. Again, sorry about that. I figured I'd at least take a stab, though. Chris. - Original Message - From: Ivar Mossin To: General PulseAudio Discussion Sent: Saturday, August 21, 2010 8:40 PM Subject: Re: [pulseaudio-discuss] Two identical USB sound-cards - second card fails to load because card-name found in hashmap. Thanks for the reply, but no, that is not the case. Both cards are working in ALSA. They are both USB cards, and all USB devices share 1 single IRQ assigned to the USB-Controller, AFAIK. Ivar. On Sat, Aug 21, 2010 at 11:54 PM, Chris Gilland cgilla...@carolina.rr.com wrote: Is it possible that sense you have two of the exact same cards, maybe one of them isn't working as there is an I R Q conflict between both of them? I'll admit my expertees on Pulse Audio are not very good at all, and probably I shouldn't be making this assumption being I really don't totally know what I'm doing, admittedly, but it is a thought. Chris. -- ___ 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] Two identical USB sound-cards - second card fails to load because card-name found in hashmap.
Is it possible that sense you have two of the exact same cards, maybe one of them isn't working as there is an I R Q conflict between both of them? I'll admit my expertees on Pulse Audio are not very good at all, and probably I shouldn't be making this assumption being I really don't totally know what I'm doing, admittedly, but it is a thought. Chris. - Original Message - From: Ivar Mossin To: pulseaudio-discuss@mail.0pointer.de Sent: Saturday, August 21, 2010 3:51 PM Subject: [pulseaudio-discuss] Two identical USB sound-cards - second card fails to load because card-name found in hashmap. Hello. I'm having problems loading two identical sound-cards in my computer. I'm using Ubuntu 9.10 and PulseAudio 0.9.19 coming with this release. I have tried looking at the logs, and when using loglevel 4 I find these lines: (first card - failing scenario): Aug 9 22:53:08 ivar-laptop pulseaudio[2573]: module.c: Loaded module-alsa-card (index: #24; argument: device_id=1 name=usb-BeAutiful_Qing_Audioengine_AW1-00 card_name=alsa_card.usb-BeAutiful_Qing_Audioengine_AW1-00 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1). (second card - failing scenario): Aug 9 22:58:32 ivar-laptop pulseaudio[2573]: module.c: Failed to load module module-alsa-card (argument: device_id=2 name=usb-BeAutiful_Qing_Audioengine_AW1-00 card_name=alsa_card.usb-BeAutiful_Qing_Audioengine_AW1-00 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1): initialization failed. I downloaded the source-code using 'apt-get source pulseaudio' and was looking around a bit. What I found was that the pa__init() function in modules/alsa/module-alsa-card.c called a function pa_card_new() located in pulsecore/card.c which returned a null-pointer. This function again called pa_namereg_register() in pulsecore/namereg.c which returned NULL because it could find the card-name in the hashmap and the fail argument was set to TRUE. As a simple test, I changed the fail argument to FALSE, and the second module loaded as well: (first card - working scenario): Aug 21 18:10:03 ivar-laptop pulseaudio[3835]: module.c: Loaded module-alsa-card (index: #18; argument: device_id=2 name=usb-BeAutiful_Qing_Audioengine_AW1-00 card_name=alsa_card.usb-BeAutiful_Qing_Audioengine_AW1-00 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1). (second card - working scenario): Aug 21 18:12:37 ivar-laptop pulseaudio[3835]: module.c: Loaded module-alsa-card (index: #19; argument: device_id=3 name=usb-BeAutiful_Qing_Audioengine_AW1-00 card_name=alsa_card.usb-BeAutiful_Qing_Audioengine_AW1-00 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1). Having this flag set to TRUE basically disables the functionality further down in the pa_namereg_register() which tries to add a .%u to the card-name, where %u starts at 2 and gives up at 99. Trying to figure out where this flag was set, I found that it had been set by a function set_card_name() called further up in pa__init(). If the module being loaded has the argument card_name or name, then data-namereg_fail is set to TRUE. Looking at the arguments given in the logfile, it actually provides both of these arguments, also at load-time: (second card - failing scenario): Aug 9 22:58:32 ivar-laptop pulseaudio[2573]: module-udev-detect.c: Loading module-alsa-card with arguments 'device_id=2 name=usb-BeAutiful_Qing_Audioengine_AW1-00 card_name=alsa_card.usb-BeAutiful_Qing_Audioengine_AW1-00 tsched=yes ignore_dB=no card_properties=module-udev-detect.discovered=1' So my questions are: Is there a way to load both these cards without modifying the source? Can I configure something to make this work? What is the reasoning behind setting this flag at all? Which side-effect will I experience by simply ignoring this flag, and trying to add a .%u to the card-name anyway? Thanks for any help provided. Kind Regards, Ivar Mossin -- ___ 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 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
[pulseaudio-discuss] pulseaudio debug mode
What is the best way to start PA for debugging and still have all the usual clients running? -- 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[24553]: ratelimit.c: 497 events suppressed
On Wed, 2010-07-14 at 08:50 +0100, Colin Guthrie wrote: 'Twas brillig, and Chris at 14/07/10 01:23 did gyre and gimble: I've just recently upgraded to Mandriva 2010.1 hoping that the above problems would go away. They haven't. I'm still seeing syslog entries like this: pulseaudio[24553]: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -161920 bytes (-917 ms). localhost pulseaudio[24553]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers. Is this an Alsa issue or a Pulseaudio issue. Other than disabling pulseaudio how can I attempt to correct this? The clue is on the line above. The issue is that the driver (i.e. from the kernel) is giving bogus information to PA. You can perhaps work around the problem by disabling Glitch Free mode via draksound, but the problem should really be reported to the Alsa guys via alsa-dev mailing list (I find it's more productive than their bug tracker). Col I've always had 'Glitch Free' mode disabled after our conversation on this when in happened in 2010.0. I've reported this to the alsa-dev list and the only one who responded way Raymond Yau who, as before, replied with I need to run pulseaudio -k ; pulseaudio -v. I've done that and have a bzip2 file of 9k, was 129k. I don't know what to do with the log file now though. Is this a Mandriva issue? Thanks 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
[pulseaudio-discuss] pulseaudio[24553]: ratelimit.c: 497 events suppressed
I've just recently upgraded to Mandriva 2010.1 hoping that the above problems would go away. They haven't. I'm still seeing syslog entries like this: pulseaudio[24553]: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -161920 bytes (-917 ms). localhost pulseaudio[24553]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers. localhost pulseaudio[24553]: alsa-util.c: snd_pcm_dump(): localhost pulseaudio[24553]: alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0 localhost pulseaudio[24553]: alsa-util.c: Its setup is: localhost pulseaudio[24553]: alsa-util.c: stream : PLAYBACK localhost pulseaudio[24553]: alsa-util.c: access : MMAP_INTERLEAVED localhost pulseaudio[24553]: alsa-util.c: format : S16_LE localhost pulseaudio[24553]: alsa-util.c: subformat: STD localhost pulseaudio[24553]: alsa-util.c: channels : 2 localhost pulseaudio[24553]: alsa-util.c: rate : 44100 localhost pulseaudio[24553]: alsa-util.c: exact rate : 44101 (144510/32768) localhost pulseaudio[24553]: alsa-util.c: msbits : 16 localhost pulseaudio[24553]: alsa-util.c: buffer_size : 4408 localhost pulseaudio[24553]: alsa-util.c: period_size : 1102 localhost pulseaudio[24553]: alsa-util.c: period_time : 24988 localhost pulseaudio[24553]: alsa-util.c: tstamp_mode : ENABLE localhost pulseaudio[24553]: alsa-util.c: period_step : 1 localhost pulseaudio[24553]: alsa-util.c: avail_min: 1102 localhost pulseaudio[24553]: alsa-util.c: period_event : 1 localhost pulseaudio[24553]: alsa-util.c: start_threshold : -1 localhost pulseaudio[24553]: alsa-util.c: stop_threshold : 1155530752 localhost pulseaudio[24553]: alsa-util.c: silence_threshold: 0 localhost pulseaudio[24553]: alsa-util.c: silence_size : 0 localhost pulseaudio[24553]: alsa-util.c: boundary : 1155530752 localhost pulseaudio[24553]: alsa-util.c: appl_ptr : 735801008 localhost pulseaudio[24553]: alsa-util.c: hw_ptr : 735846184 localhost pulseaudio[24553]: ratelimit.c: 280 events suppressed localhost pulseaudio[24553]: asyncq.c: q overrun, queuing locally Because of the constant disk access when this is happening my system becomes almost unusable. Is this an Alsa issue or a Pulseaudio issue. Other than disabling pulseaudio how can I attempt to correct this? -- 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
[pulseaudio-discuss] Audio Capture ( recording a Skype conversation )
Dear PulseAudio community, My apologies if this is either a double post or simply a stupid question. As the subject line suggests I'm interested in recording something like a Skype conversation. I say something like a Skype conversation because I work with some similar voip teleconferencing programs other than Skype. I've been looking in several places for some information. Hopefully I'm close to having a solution, but I'm not quite there. pacat -r -d alsa_output.pci-_00_14.2.analog-stereo.monitor I've used the above command to capture audio passing through my sound server. I've also added a virtual device in my /etc/alas/pulse-default.conf file. So I can also capture that audio with other programs such as Audacity. So I need to include my microphone to the recording. I've found the following command, which doesn't work for me, to pass two streams: pacat -r -d alsa_input.pci-_00_14.2.analog-stereo | pacat -p -d alsa_output.pci-_00_14.2.analog-stereo.monitor The above results in: Stream error: No such entity write() failed: Broken pipe for me. So, is there a way to combine my two devices? I'm assuming something like a pacmd script will do the trick, but I don't yet understand how the module-combine statements work. Many Thanks. _ http://clk.atdmt.com/UKM/go/195013117/direct/01/ ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Audio Capture ( recording a Skype conversation )
Thanks for your response. I will attempt your recommendation. pacat -r -d alsa_input.pci-_00_14.2.analog-stereo | pacat -p -d alsa_output.pci-_00_14.2.analog-stereo.monitor This will certainly not work. YOu are trying to record from your input device (mic) and then play it back via another input device (the monitor of you output is an input!). That's what I thought that command was doing, but I didn't have the confidence of a good knowledge base to rule it out. This means that you will effectively be recording both your mic and the output of your sound card and playing back both streams on the null sink. As it is a null sink you wont be able to hear it, but you should be able to see the VU meter in the playback tab of pavucontrol. When you say I won't be able to hear it, do you mean the recorded stream, or do you mean everything even my VoIP conversation? Thanks. Dear PulseAudio community, My apologies if this is either a double post or simply a stupid question. As the subject line suggests I'm interested in recording something like a Skype conversation. I say something like a Skype conversation because I work with some similar voip teleconferencing programs other than Skype. I've been looking in several places for some information. Hopefully I'm close to having a solution, but I'm not quite there. pacat -r -d alsa_output.pci-_00_14.2.analog-stereo.monitor I've used the above command to capture audio passing through my sound server. Good so far. I've also added a virtual device in my /etc/alas/pulse-default.conf file. So I can also capture that audio with other programs such as Audacity. THis is not really correct. For one you should not really edit the system files, but instead edit your own users' ~/.asoundrc file. Secondly, what definition did you add here? In theory you should need none. You generally should just tell audacity to use pulse then when it is recording fire up an application like pavucontrol, go to the Recording tab and locate the Audacity recroding stream, then move it so that it is recording from the monitor device you want. This choice will be remembered for next time. So I need to include my microphone to the recording. I've found the following command, which doesn't work for me, to pass two streams: pacat -r -d alsa_input.pci-_00_14.2.analog-stereo | pacat -p -d alsa_output.pci-_00_14.2.analog-stereo.monitor This will certainly not work. YOu are trying to record from your input device (mic) and then play it back via another input device (the monitor of you output is an input!). If you were to play it back properly, it would sound awful to the person speaking on the VoIP app in question. I'll outline my recommendation below. The above results in: Stream error: No such entity write() failed: Broken pipe for me. So, is there a way to combine my two devices? I'm assuming something like a pacmd script will do the trick, but I don't yet understand how the module-combine statements work. Well module-combine is technially module-combine-sink. i.e. is allows you to combine multiple outputs, but not multiple inputs. What you want to do is take two inputs (your mic and the monitor of your output) and combine then and record them. There is not (currently) a module-combine-source so this is impossible. However you can do soemthing that approximates this. Firstly load a null sink: pactl load-module module-null-sink sink_name='foo' Then load two module loops backs: pactl load-module module-loopback pactl load-module module-loopback Using pavucontrol, connect the output of the two loopbacks to the null sink. Connect the input of the two loopbacks to: alsa_output.pci-_00_14.2.analog-stereo.monitor and alsa_input.pci-_00_14.2.analog-stereo This means that you will effectively be recording both your mic and the output of your sound card and playing back both streams on the null sink. As it is a null sink you wont be able to hear it, but you should be able to see the VU meter in the playback tab of pavucontrol. Then in order to record the combined result, you will simply select foo.monitor. pacat -r -d foo.monitor Jobs a good 'un! :p HTHs 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] asyncq.c: q overrun, queuing locally
I noticed tonight that when I was copying a 1.2Gb file to my thumb drive that it took forever. From a suggestion in another list I ran in a terminal pulseaudio -k ; pulseaudio -. I've attached the output. One other thing of note, after disabling pulseaudio I was able to copy the file in around 3minutes. In the previous example I had to stop the copy process after 15 minutes at only 50%. This is on: Mandriva 2010 pulseaudio 0.9.19-7mdv2010.0 Gnome 2.28.0 Any other information I need to provide please let me know. Chris -- KeyID 0xE372A7DA98E6705C palogfilecopy122809.txt.bz2 Description: application/bzip 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
[pulseaudio-discuss] 3 cards, use all w/ PA AND one of them with skype too ...
Hello,
i try to setup my 3 cards in a way so that it works like the following:
1st device onboard (Intel 82801H 8086:284b) headset
2nd device pci (ES1969 Solo-1 125d:1969) room1
3rd device pci (Ensoniq ES1371 1274:1371) room2
i want to use all 3 devices with PA for output and only the first device
for input. i managed to combine them to be a simultaneous device for
outputting to two at the same time (room1 and room2), simultaneous output
to all three is default in PA, so this is trivial ;-)
The problems i have is with the input of the headset and getting it
running with skype. I thought i could use .asoundrc to redirect all alsa
based applications to pulse, but that only works for output ...
.asoundrc:
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
So, how can i tell alsa/pulse to let the mic of the headset be used from
within alsa based applications?
thanks, Chris
--
Now playing: Duca - Technology | Psychedelic 2009
___
pulseaudio-discuss mailing list
pulseaudio-discuss@mail.0pointer.de
https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] can't open windows cli
Hi Fred, i can not tell how to use native remote sink because i didn't use it, but i used the cli module by enabling module-cli-protocol-tcp in the file default.pa (don't know where you can find this file on windows) and then telnetting into the pulseaudiop cli using telnet locahost 4712. i think there are issues with module-cli-protocol-tcp binding to the loopback device thus it might not be accessible from a remote host, but, however, from the machine itself you should be able to establish a connection to the local pulseaudio daemon. ch...@joysn:~$ cat /etc/pulse/default.pa | grep cli load-module module-cli-protocol-tcp ch...@joysn:~$ telnet localhost 4712 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Welcome to PulseAudio! Use help for usage information. and maybe this helps with remoting your sink? http://en.wikibooks.org/wiki/Configuring_Sound_on_Linux/Pulse_Audio/Remote_server hth On Mon, Jul 06, 2009, LECONTE Frédéric wrote: Hi everyone, I'm using PA to hear music from one linux box. I achieve to send sound from one other linux machine with module-esound-sink I tried to make same thing from windows but it doesn't seem to work I have 2 questions: how to use native remote sink from my linux machine(since esound is deprecated) ? how to open PA cli on windows( pulseaudio.exe -nC doesn't open cli) ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Stopping pulse in Fedora 11
2009/6/12 Lennart Poettering lenn...@poettering.net: It's simply started by the first app that needs it. Place autospawn = no in ~/.pulse/client.conf (or /etc/pulse/client.conf). Great, that did the trick, thanks. Shouldn't be too hard to fix for someone who cares to fix this. All that is needed is probably some fragment in /usr/share/alsa/cards/ that maps front:xxx in some sensible way to your card. So is that how pulse picks up it's device configuration, via the ALSA config? I'm trying to do some investigation but the config file format is a little confusing at first. I'll keep working on it anyway - I've found a few hints via the wonder of google. If pulse does use the alsa config to pick up it's devices, this thread is probably quite pertinent: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-January/014029.html Thanks again, you must get through a huge pile of emails per day! Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Stopping pulse in Fedora 11
Hi, First of all, pulse is looking great generally - on my office PC it works beautifully, however I've had lots of problems on my home PC. I'm trying to do some diagnosis so I want to stop pulse on Fedora 11 which I installed a couple of days ago. The problem is it keeps respawning and I can't figure out what mechanism is causing it. I've tried pulseaudio -k, kill -9 pulses-pid, I've moved /etc/alsa/pulse-default.conf out of the way (on the advice of someone on the fedora mailing list. I've looked in /etc/event.d and /etc/init.d to see if I can figure out how pulse is started but I just can't figure it out. It always restarts itself. I'd appreciate any ideas on how to stop it, just so I can do some more investigation into what might be causing my lack of audio. For the record, I've got an M-Audio Audiophile 24/96 card. The bug report here is what I seem to be experiencing: https://bugzilla.redhat.com/show_bug.cgi?id=499435 TIA, Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] hopelessly newbish question
I beg a thousand pardons for asking such a simple question, but I have been googling for hours to no avail. I have a moderately simple two computer PulseAudio setup. I have a workstation that hosts all of my audio generating applications and a low power laptop connected to my stereo that acts as PulseAudio server. The laptop actually sits between the audio out of my TV and the TV input on my amp/receiver. This lets me use the laptop soundcard to mix the output of my workstation and TV for playback. That all works well enough. I am encounting problems when I try to work jack and qsynth into the mix. I have a midi keyboard attached to my workstation. I'd like to play it with fluidsynth and have the output sent to the pulseaudio server on the laptop. I feel like I am most of the way there. I have pulseaudio-module-jack installed on the workstation. I have jackd running with the output of qsynth being sent to PulseAudio JACK source. With PA Volume Control connected to the laptop's sound server, Jack source shows in the Input Devices list. Now, how do I go about listening to that source? -- Chris Ribe IT Specialist Pandion Systems, Inc. www.pandionsystems.com (352) 505 1829 cr...@pandionsystems.com ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] pulseaudio[1495]: protocol-esound.c: recieved unimplemented request #22
Mandriva 2009(Free), Gnome Desktop, Pulseaudio 0.9.10. Last Wednesday night gstreamer was updated on an auto update, since then I've been seeing the above in my syslog. I've only been able to find one reference to this in google and it dealt with ogg files not being played. I don't have that problem though. In fact I can really find no issues as all file types are being played through XMMS, Amarok and XMMS2. RealPlayer crashes when I try to play a stream but I don't use it much anyway. Here are the files that were updated: gstreamer0.10-plugins-good-0.10.10-2.1mdv2009.0 installed gstreamer0.10-flac-0.10.10-2.1mdv2009.0 installed gstreamer0.10-pulse-0.10.10-2.1mdv2009.0 installed I see this entry in my syslog about every minute or so. Any advice on what to do about this would be appreciated. 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] Very crackly audio and crashing from cpu overload
2009/1/9 Mark Greenwood extreme...@ntlworld.com: Just a thought, do you have any VIA chipsets on your motherboard? If you do, throw it away and buy another one. It'll save you hours of fruitlessly trying to find a software problem that doesn't exist. Don't ask me how I know this unless you want a rant... ;) Mark Thankfully not. It's an ALi (or ULi - same company I think) chipset. Although it could still be the problem. I'll probably tinker some more tonight. Thanks for the idea though! Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Very crackly audio and crashing from cpu overload
Hi, I've been having lots of problems with audio in Fedora 10. At first I was having problems playing sound in youtube via firefox. I found a tutorial which suggested I change a line in default.pa from: load-module module-hal-detect to load-module module-hal-detect tsched=0 That did seem to improve things. However amarok is really not happy. I can play mp3s for a minute or two but it is very crackly. Then eventually the audio stops and amarok locks up. /var/log/messages shows messages like this: Jan 8 21:26:46 localhost pulseaudio[3188]: main.c: High-priority scheduling enabled in configuration but not allowed by policy. Jan 8 21:26:46 localhost pulseaudio[3188]: core-util.c: setpriority(): Permission denied Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Device hw:0 doesn't support 2 channels, changed to 10. Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Device hw:0 doesn't support sample format s16le, changed to s32le. Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Cannot find fallback mixer control PCM. Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Device hw:0 doesn't support 2 channels, changed to 12. Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Device hw:0 doesn't support sample format s16le, changed to s32le. Jan 8 21:26:46 localhost pulseaudio[3191]: alsa-util.c: Cannot find fallback mixer control Mic. Jan 8 21:33:29 localhost pulseaudio[3191]: cpulimit.c: Recevied request to terminate due to CPU overload. I've running an AMD Athlon 64 x2 (can't remember the exact model) although it's 32 bit Fedora 10. The soundcard is an M-Audio Audiophile 2496 (ICE1712 chip I believe). I never got pulseaudio working with this setup in Fedora 9 and had to remove all pulse packages. Now it's almost working but there are clearly still issues. Any ideas? (This is my home PC so I can only investigate in the evenings, UK time, so bear with me! ) Thanks, Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Very crackly audio and crashing from cpu overload
2009/1/8 Lennart Poettering lenn...@poettering.net: The fragment stuff is not used when tsched is enabled. So playing around with the frag stuff is pointless when tsched=1. At least I'm not going mad. I didn't think it was doing anything but it's somewhat subjective. Any thoughts on the wakeup watermark though? It seems that whenever there is a glitch, the watermark is increased, and whenever the watermark is increased, the frequency of glitching increases until it hits the limit - in my case 71.27ms. Is there anyway to force this wakeup watermark setting? Everything works better for me, although still not perfect, when it is a low number. Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Very crackly audio and crashing from cpu overload
2009/1/8 Lennart Poettering lenn...@poettering.net: On Thu, 08.01.09 23:03, Chris (chris1.nore...@googlemail.com) wrote: 2009/1/8 Chris chris1.nore...@googlemail.com: 2009/1/8 Lennart Poettering lenn...@poettering.net: I've been playing with the default-fragments and default-fragment-size-msec settings but they don't seem to be having any effect. I'm just getting occasional skipping now - maybe one skip of a fraction of a second every 3 to 5 seconds. The bad crackling went when I set tsched=1. Think I've figured out something interesting... when I first start playing music, it works reasonably well, but after a while, maybe 30 seconds, there is a glitch. At soon as there is a glitch, a message appears in /var/log/messages about increasing the wakeup watermark. Then the next glitch occurs after about 15 seconds and the watermark is increased. Basically, the glitches become more and more frequent as the watermark is increased. Here's the log output... Jan 8 22:58:20 localhost pulseaudio[2772]: module-alsa-sink.c: Increasing wakeup watermark to 5.99 ms Jan 8 22:58:45 localhost pulseaudio[2772]: module-alsa-sink.c: Increasing wakeup watermark to 11.97 ms Jan 8 22:59:15 localhost pulseaudio[2772]: module-alsa-sink.c: Increasing wakeup watermark to 23.95 ms Jan 8 22:59:17 localhost pulseaudio[2772]: module-alsa-sink.c: Increasing wakeup watermark to 47.89 ms Jan 8 22:59:24 localhost pulseaudio[2772]: module-alsa-sink.c: Increasing wakeup watermark to 71.27 ms It never goes above that 71.27ms point but the glitches are frequent from that point on until I restart pulse. When PA detects an underrun it thinks: ah, so we didn't wake up in time to fill the playback buffer again, so next time lets wake up a bit earlier to increase the chance that we can fill up the playback buffer in time. That means the wakeup watermark (i.e. the time PA estimnates it needs to fill up the buffer again before the buffer might run empty) is doubled on each drop out. However we don't increase the wakeup boundary without limits -- instead we make sure we will still sleep for at least half the buffer size. That's why the watermark doubles on each drop-out until an upper limit is reached. Now the question is of course why you get those drop outs in the first place. There might be three reasons for that: 1) The driver is broken and lies about the timing parameters (i.e. the playback position of the hardware). PA relies on correct timing information from the driver to estimate when to wake up next. This would need to be fixed in the ALSA driver. 2) The estimnation logic in PA is broken. i.e. although we got correct timing information from the driver we miscalculate the wakeup time because the retard who wrote that code (read: me) made a mistake when he coded it because that stuff is a bit complex and he is not. This would need fixing in PA itself. 3) We estimate everything correctly and set up our timers correctly, but the kernel doesn't obey and doesn't wake us up in time. Bad bad kernel. There might be a lot of reasons for that. Usually this has todo with drivers that block the CPU for too long and thus bar userspace from getting scheduled in time. Particularly bad at this are closed-source drivers (ndis, nvidia). The tool latencytop may be used to figure out what is going on. I am tempted to blame #1 an #3 for most incarnations of this bug, because it is only triggered by very specific hardware setups -- and of course because software I write doesn't have any bugs! I hope this helps a bit. Very useful info, thanks. I'm probably gonna call it a night now but I'll do my best to investigate each of those options in the next few days. For me, #3 sounds most probable since I'm running the dreaded nvidia blob. I wanted compiz eye-candy although I've not managed to get it working yet and Xorg still seems to be using a lot of CPU so it's probably the nvidia driver to blame. I'll be sure to let you know how it pans out because you've been very helpful. Keep up the good work! Cheers, Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Very crackly audio and crashing from cpu overload
2009/1/8 Renke Brausse rbrau...@gmx.net: sorry, off topic - but I can't resist... Am Freitag, den 09.01.2009, 00:50 +0100 schrieb Lennart Poettering: and of course because software I write doesn't have any bugs! do you need a new job? just contact me :) [:p] If you need a programmer who can produce obscure bugs out of thin air, I'm your man, otherwise go for Lennart! Anyway, I tried the nv driver in case it was the nvidia taint messing up things but no change in behaviour. Looking at Lennart's options, I doubt that the sound card driver is bad. The ICE1712 chip has been around for at least 6 years in my experience. My employer makes embedded Linux devices and we've used various M-Audio devices, all using the ICE1712 since RedHat 7.3! Doesn't make it impossible but quite unlikely. Whilst I don't know much about the driver in detail, and I guess it hasn't been touched for a long time, I was under the impression it was a well documented and supported device. So I'm still thinking that the problem is related to my XOrg CPU usage and some kind of interrupt issue. When I was using the nvidia driver, CPU usage by Xorg was around 30-40%! Now I'm on the OSS nv driver and it's about 15-20% (think that's for 1 core only). Doesn't seem that much but maybe it's a timing issue related to interrupts. I'll have a play with latencytop in the next few days. Thanks again for the help. I'm getting closer to sorting this out and if I do I'll try to post it web-wide so googlers can find it. Cheers, Chris. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] module-tunnel-sink: where to start?
Alright... so, I've finally got an environment with bleeding edge everything, and am able to build and run pulseaudio from GIT. However, the GIT version dies right away with a PA_SINK_INPUT_IS_LINKED assertion. I saw a mention on trac that this is known issue, and that there are a few workarounds, but I didn't get that far ;) For the time being, I elected to stick with the last 'stable' release of 0.9.13. With 0.9.10, I was using esound tunnels to play via remote servers. However, with 0.9.13 these die quite quickly when calling PA_SINK_MESSAGE_GET_LATENCY. (I thought the fact that esound tunnels don't have latency measurement was expected behaviour?) Anyhow, I'd really like to use native tunnels, because of their latency measurement, so that I can play audio simultaneously across all of the systems in my house. My one server (the sink) is running 0.9.10, while my laptop (the source) is running 0.9.13. If I play audio over a tunnel only, 9 times out of 10, everything is just fine. Occasionally, I'll have a glitch. If I go to a combined sink (local ALSA and remote tunnel), 9 times out of 10, all hell breaks loose. Things start out well, but a few seconds into playback, I start getting glitches on the remote machine, which get worse and worse. PA report a remote sample rate estimation that is way out of whack ( 80,000 Hz) and the audio pretty much drops out altogether. Is this an issue that would likely go away if both ends were running bleeding edge ALSA and PA? Or is this something that can be addressed in module-tunnel? If the latter, then where would be a good place to start? (And I still want to mock up some audio-panel pavucontrol-on-steroids type application, but I spent 2 days this week restoring my server after a hard-drive crash... alas, haven't had any tinkering time!) Cheers, Chris ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] module-tunnel-sink: where to start?
I apologize for the previous thread hijack... an inadvertant 'reply to' instead of 'compose to'. Anyhow, the same message, but now with new thread goodness! - Alright... so, I've finally got an environment with bleeding edge everything, and am able to build and run pulseaudio from GIT. However, the GIT version dies right away with a PA_SINK_INPUT_IS_LINKED assertion. I saw a mention on trac that this is known issue, and that there are a few workarounds, but I didn't get that far ;) For the time being, I elected to stick with the last 'stable' release of 0.9.13. With 0.9.10, I was using esound tunnels to play via remote servers. However, with 0.9.13 these die quite quickly when calling PA_SINK_MESSAGE_GET_LATENCY. (I thought the fact that esound tunnels don't have latency measurement was expected behaviour?) Anyhow, I'd really like to use native tunnels, because of their latency measurement, so that I can play audio simultaneously across all of the systems in my house. My one server (the sink) is running 0.9.10, while my laptop (the source) is running 0.9.13. If I play audio over a tunnel only, 9 times out of 10, everything is just fine. Occasionally, I'll have a glitch. If I go to a combined sink (local ALSA and remote tunnel), 9 times out of 10, all hell breaks loose. Things start out well, but a few seconds into playback, I start getting glitches on the remote machine, which get worse and worse. PA report a remote sample rate estimation that is way out of whack ( 80,000 Hz) and the audio pretty much drops out altogether. Is this an issue that would likely go away if both ends were running bleeding edge ALSA and PA? Or is this something that can be addressed in module-tunnel? If the latter, then where would be a good place to start? (And I still want to mock up some audio-panel pavucontrol-on-steroids type application, but I spent 2 days this week restoring my server after a hard-drive crash... alas, haven't had any tinkering time!) Cheers, Chris ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] pulseaudio elusive
Yeah, I think more needs to be done here to make it more obvious. In actual fact a usability improvement was to default to the Sinks tab if no streams are running, but in actual fact if it started with a blank pane and some text saying No active streams. This pane will list blah, blah blah would actually be better for first time users etc. To me, the PA metaphor seems really suited to a directed graph visualization -- boxes and arrows kind of stuff. Sources and sinks all get to be boxes, and links between them between them get to be ... arrows. (Someone has to have thought of this before.) I think Conduit implements a similar GUI. This was mentioned in an earlier thread, and I too think this is a great idea. I was planning on putting together some mockups to start working through the details of what this would look like. Once I have something, I'll be sure to post it here and would definitely appreciate any feedback. Cheers, Chris Hamilton ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Potential module: persistent-sink
Is it possible to have a network sink and a local sink synchronized? Or multiple network sinks? (In all cases, they have been combined using module-combine). I'm running 0.9.10, and use a module-esound-sink because the module-tunnel-sink is extremely flaky (immediately get buffer underrun complaints, and choppy sound out of the remote system). Using the esound sink, there is a noticeable lag between the two systems, which one would think could be at least synchronized. Any feedback on this would be great. Sadly the correct fix here is to make the module-tunnel sink work nicer! In theory in 0.9.13+, it should be possibly to dynamically change the buffering of the tunnel on the fly to cope with underruns etc. (not 100% sure if it really needs 0.9.13 here but it certainly fits with the glitch-free, watermark level stuff). The esound protocol does not contain any information about timing/latency to the best of my knowledge so it will be very hard to get synchronised correctly. I figured that would be the response I'd get ;) I tried upgrading to 0.9.13 the other day, but it turned out there were too many dependencies I was going to have to drag in (newer version of alsa, libspeex, etc) that weren't in the Hardy repositories (my laptop currently runs hardy). I'd like to get my hands dirty with some development work, so I suppose I'm going have to bite the bullet and perform all of the necessary upgrades! Something that I would love to see one day (sooner rather than later): A virtual 'switchboard' type system for configuring pipes, where connections are drawn with the mouse, etc. This could have a nice graphical and intuitive switchboard (with cables, etc) view, and a simpler more compact matrix view (inputs on one side, outputs on the other, combiners living as both inputs and outputs). Ideally, this would keep information about all past clients (even though they may not be presently active), as well as past sinks (even through they may be down temporarily). Status of the sinks/sources could be indicated through color. A 'recursive' behaviour would be nice, wherein if the local user had access, they could zoom into the switchboard panel of a remote server as well. Does this sound like something anybody else would like to see? I can easily put together a UI mock-up illustrating exactly what I envision... Please do some mockups. It's important to realise that pulse is not jack, and this it's not really meant to be an all singing interconnect type app, but IMO this kind of visualisation is quite nice. I presume this interface would kinda be like a pavucontrol on steroids? Viewing remote servers should be easy enough (effectively the same as running pavucontrol via PULSE_SERVER=remote pavucontrol). Again, (as per my other mail) seeing non-active sinks and apps will not currently be supported by the protocol (to the best of my knowledge) so to get full feature out of this interface. Yes, I was imagining a pavucontrol steroids, with a heavy focus on a visual representation of what's going on (to make things quite easy for beginners), and with the ability to drill as deep down into the guts of things as possible. I'll try and put together some mock-ups over the next week to show what I mean. And yes, modifying the protocol to be able to query for previous (but not active) sinks and apps would be ideal, but can always come later. Does the API allow querying for a list of plugins that are installed on a server, with all of their metadata? (ie: name, description, parameters plus default and valid values, etc) This would be something else that would be quite useful in such an application. Cheers, Chris ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Potential module: persistent-sink
Is it possible to have a stream stay connected to a sink, be rescued, but then be reconnected if the sink that disappeared becomes available once again? Something similar to rescue-stream and always-sink, but that will try to reconstruct the previously existing sink in the meantime. (For example, if it was a network sink, it'll keep trying to reconnect to the host if it went down). I was thinking of a use case something like: load-module module-tunnel-sink server=someserver \ sink_name=someserver_tunnel load-module module-persistent-sink child=someserver_tunnel \ sink_name=someserver Then, streams could be directed to the 'someserver' sink, which thinly wraps the 'someserver_tunnel'. If the server is unable to be brought up, or goes down, then the persistent-sink will rescue the stream by pointing it either at the default sink or a null sink (configurable). In the meantime, it'll continue to try and rebuild the sink that went down. Would this be useful? Having not really touched the PulseAudio source, is it something that would be (somewhat easily) possible in the current codebase? I'm not afraid to get my hands dirty, but would appreciate a push in the right direction. Another unrelated question: Is it possible to have a network sink and a local sink synchronized? Or multiple network sinks? (In all cases, they have been combined using module-combine). I'm running 0.9.10, and use a module-esound-sink because the module-tunnel-sink is extremely flaky (immediately get buffer underrun complaints, and choppy sound out of the remote system). Using the esound sink, there is a noticeable lag between the two systems, which one would think could be at least synchronized. Any feedback on this would be great. Something that I would love to see one day (sooner rather than later): A virtual 'switchboard' type system for configuring pipes, where connections are drawn with the mouse, etc. This could have a nice graphical and intuitive switchboard (with cables, etc) view, and a simpler more compact matrix view (inputs on one side, outputs on the other, combiners living as both inputs and outputs). Ideally, this would keep information about all past clients (even though they may not be presently active), as well as past sinks (even through they may be down temporarily). Status of the sinks/sources could be indicated through color. A 'recursive' behaviour would be nice, wherein if the local user had access, they could zoom into the switchboard panel of a remote server as well. Does this sound like something anybody else would like to see? I can easily put together a UI mock-up illustrating exactly what I envision... Regards, Chris Hamilton (a recent but very enthusiastic PulseAudio convert) ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Controlling where module-rtp-send sends multicast packets?
I'm more intrigued by the possibility of turning my dumpster bound PC hardware into something useful using PulseAudio. Work out the electricity and AC costs and you won't think they are so cheap. This node I working on right now is a 1.1Ghz Celeron laptop. Power consumption is probably about 50w. It is destined for an uncooled part of the house (screened in porch) and will be powered on for about 100 hours / year. That means the yearly electricity and AC costs will be approximately 30 *cents*. -chris -- TV/IT Engineer WCJB-TV Gainesville, FL (352) 416 0648 [EMAIL PROTECTED] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
