Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error
#871: PA compile fails on MAC OS X due to bonjour error ---+ Reporter: spitfire | Owner: lennart Type: defect| Status: closed Milestone:| Component: module-zeroconf-* Resolution: fixed |Keywords: ---+ Changes (by coling): * status: new = closed * resolution: = fixed Comment: OK, I've pushed Daniel's patch now, so will close this bug. If you still get a problem, please reopen :) -- Ticket URL: http://pulseaudio.org/ticket/871#comment:6 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #716: PA loses mono audio support when remixing is disabled (was: PA loses microphone when remixing is disabled)
#716: PA loses mono audio support when remixing is disabled +--- Reporter: Sam Stone | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: microphone, remixing, mono +--- Changes (by Captain Chaos): * keywords: microphone, remixing = microphone, remixing, mono Comment: Is anything happening on this? If I understand it correctly, a pulseaudio user with more than two loudspeakers currently has three choices: 1. Accept that all audio that plays is sent to all speakers, whether or not that's appropriate 2. Accept that mono audio does not play at all 3. Jerry rig a complicated system with virtual devices, which will only partly solve the problem (namely only for programs which allow you to choose the device they play to) To me all these options are unacceptable. People with surround sound systems should not be punished by having their audio not work at all, or be played over the wrong speakers. Instead, pulseaudio should intelligently choose which speakers to direct the audio to. I find the suggestion that remixing is needed to allow mono sound to be played on 5.1 speakers a bit odd. Surely mono sound could be played over the center speaker, no remixing required? Pulseaudio currently already correctly plays stereo sound over just the front left and right speakers when remixing is turned off, surely it would not be hard to similarly direct mono channels to the center speaker? -- Ticket URL: http://pulseaudio.org/ticket/716#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #859: racy crashes
#859: racy crashes +--- Reporter: jankratochvil | Owner: lennart Type: defect | Status: closed Milestone: 0.9.22 | Component: clients Resolution: fixed |Keywords: +--- Changes (by lennart): * status: new = closed * resolution: = fixed * milestone: = 0.9.22 Comment: Applied now to stable-queue. This will go in the next release. Sorry for the delay. -- Ticket URL: http://pulseaudio.org/ticket/859#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #843: Some PA strings are not translated in the Gnome Sound Properties window
#843: Some PA strings are not translated in the Gnome Sound Properties window ---+ Reporter: kelemeng | Owner: lennart Type: defect| Status: closed Milestone: 0.9.22| Component: module-alsa-* Resolution: fixed |Keywords: i18n ---+ Changes (by lennart): * status: new = closed * resolution: = fixed * milestone: = 0.9.22 Comment: Applied in stable-queue. This will be part of the next release. -- Ticket URL: http://pulseaudio.org/ticket/843#comment:8 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #845: PulseAudio not detecting sound card - Alienware laptop, ubuntu 10.4
#845: PulseAudio not detecting sound card - Alienware laptop, ubuntu 10.4 -+-- Reporter: pacanukeha | Owner: lennart Type: defect | Status: closed Milestone: | Component: module-detect Resolution: elsewhere |Keywords: -+-- Changes (by lennart): * status: new = closed * resolution: = elsewhere Comment: The kernel is not aware of your sound card as it appears. Please report this bug against the kernel. Also, please direct issues like this to Ubuntu first: http://pulseaudio.org/wiki/UbuntuBugs -- Ticket URL: http://pulseaudio.org/ticket/845#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs
#852: Pulseaudio doesn't map multiple inputs/outputs +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: elsewhere |Keywords: +--- Changes (by lennart): * status: new = closed * resolution: = elsewhere Comment: PA does not expose more than one input or more than one output per sound card, because we cannot properly detect how they interact. This is a limitation and unless we get a proper enumeration interface there this cannot be fixed. You can manually work-around this however, by loading module-alsa-sink with an explicit hw:x,y alsa device. Anyway, closing this since we cannot support this unless ALSA is fixed. -- Ticket URL: http://pulseaudio.org/ticket/852#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..
#855: Illegal instruction.. ---+ Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: new Milestone: 0.9.22| Component: gst-pulse Resolution:|Keywords: ---+ Comment(by lennart): What makes you think this is a PA bug? I'd guess this bug is somwhere hidden in the Gst pa driver, not in pa code itself. Please provide a backtrace for these (generate this with gdb's bt full) crashes, so that we can figure out where this is triggered. -- Ticket URL: http://pulseaudio.org/ticket/855#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values
#862: snd_pcm_avail_delay() returned strange values +--- Reporter: oniram | Owner: lennart Type: defect | Status: closed Milestone: | Component: module-alsa-* Resolution: elsewhere |Keywords: +--- Changes (by lennart): * status: new = closed * resolution: = elsewhere Comment: The message is pretty explicit and asks you to report this to the ALSA devs, not the PA devs. Please report the bug to the ALSA developers. Thank you! -- Ticket URL: http://pulseaudio.org/ticket/862#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #864: use GNU readline in pacmd
#864: use GNU readline in pacmd --+- Reporter: skierpage| Owner: lennart Type: enhancement | Status: closed Milestone: | Component: daemon Resolution: invalid |Keywords: --+- Changes (by lennart): * status: new = closed * resolution: = invalid Comment: nah, that's hardly possible to implement since pacmd is little more than a socat implementation. You are directly talking to the server there, and readline requires a TTY to work, not a socket. -- Ticket URL: http://pulseaudio.org/ticket/864#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults ---+ Reporter: rabbit64 | Owner: lennart Type: defect| Status: closed Milestone:| Component: module-combine-* Resolution: invalid |Keywords: ---+ Changes (by lennart): * status: new = closed * resolution: = invalid Comment: This is intended this way: whenever we run into an underrun we trigger a SIGTRAP so that we can debug things. This happens only if you compile things with DEBUG_TIMING. So, this is not a bug. This is intended behaviour. Don't use DEBUG_TIMING if you are not ready to use a debugger to handle its effects! -- Ticket URL: http://pulseaudio.org/ticket/865#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #877: Bluetooth headset does not appear in the sound devices list
#877: Bluetooth headset does not appear in the sound devices list -+-- Reporter: dgvirtual | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: distrospecific |Keywords: bluetooth, headset -+-- Changes (by lennart): * status: new = closed * resolution: = distrospecific Comment: Please use bluez-gnome to connect headset with the system. Also, please make sure to file ubuntu bugs in the ubuntu bugzilla: http://pulseaudio.org/wiki/UbuntuBugs -- Ticket URL: http://pulseaudio.org/ticket/877#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #834: Provide a well tested 32bit equivalent to padsp
#834: Provide a well tested 32bit equivalent to padsp --+- Reporter: alsuren | Owner: lennart Type: enhancement | Status: closed Milestone: | Component: padsp Resolution: invalid |Keywords: --+- Changes (by lennart): * status: new = closed * resolution: = invalid Comment: Uh? just compile PA twice and the padsp .so will be compiled twice, too. LD_PRELOAD will then ensure that the right version is loaded. This is a distro bug, not a PA bug. -- Ticket URL: http://pulseaudio.org/ticket/834#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs
#852: Pulseaudio doesn't map multiple inputs/outputs +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: elsewhere |Keywords: +--- Comment(by LCID Fire): I disagree. You could very well built in a virtual mapping at least for the time being - especially since there are not that many cards out there with multiple input ports. On the other hand - the devices are mostly studio cards where you'd like to work on data in realtime and for that one uses ALSA or Jack. So not urgent for me. -- Ticket URL: http://pulseaudio.org/ticket/852#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #877: Bluetooth headset does not appear in the sound devices list
#877: Bluetooth headset does not appear in the sound devices list +--- Reporter: dgvirtual | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: bluetooth, headset | +--- Pairing the headset and the computer is successful. The connection initially is also established (after a fresh reboot). But the headset does not appear in Pulseaudio devices list (as checked via pavucontrol, kmix, KDE's Phonon devices list). Once the bluetooth headset is connected, pulseaudio sometimes freezes (if I played sound on Kaffeine, it sometimes stops). The headset is ok, works with my phone, I also managed to get it to work once with pulseaudio, but do not ask me how, it was accidental and I was not able to replicate that (and it was with packages from ppa:ubuntu- audio-dev/ppa for maverick ubuntu; since those froze more often, I downgraded, and these tests are done with default ubuntu maverick packages). System information: Bluetooth headset: Sony Ericsson HBH-610a $ hcitool info 00:16:B8:0A:14:3E Requesting information ... BD Address: 00:16:B8:0A:14:3E Device Name: HBH-610a LMP Version: 2.0 (0x3) LMP Subversion: 0xe1b Manufacturer: Texas Instruments Inc. (13) Features: 0xff 0xfd 0x29 0x78 0x18 0x18 0x00 0x80 3-slot packets 5-slot packets encryption slot offset timing accuracy role switch hold mode sniff mode park state channel quality SCO link HV2 packets HV3 packets u-law log A-law log CVSD transparent SCO enhanced iscan interlaced iscan interlaced pscan inquiry with RSSI AFH cap. slave AFH class. slave AFH cap. master AFH class. master extended features OS: Kubuntu Maverick i386 $ uname -a Linux bala 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:36:48 UTC 2010 i686 GNU/Linux $ pulseaudio --version pulseaudio 0.9.21-63-gd3efa-dirty Some info about the bluetooth adapter: $ hciconfig hci0: Type: BR/EDR Bus: USB BD Address: 00:11:67:D6:35:60 ACL MTU: 1021:4 SCO MTU: 48:10 UP RUNNING PSCAN ISCAN RX bytes:4749 acl:81 sco:0 events:173 errors:0 TX bytes:2725 acl:96 sco:0 commands:48 errors:0 $ lsusb | grep Blue Bus 004 Device 002: ID 1131:1004 Integrated System Solution Corp. Bluetooth Device Likely related bugs reported on launchpad: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/530264 https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/327284 I am attaching two files. One is filtered output of /var/log/messages of a bluetooth connection session. The ohter one is pulseaudio - output of the same session. I will gladly do any tests required for this bug to be fixed. -- Ticket URL: http://pulseaudio.org/ticket/877 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel -+-- Reporter: stephe | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: fixed |Keywords: -+-- Comment(by stephe): Thanks. I will take a look at the dbus stuff when I get a chance. It sounds like you need to load module-http-protocol-tcp in PA so that Rygel can get the data stream from PA to serve up. -- Ticket URL: http://pulseaudio.org/ticket/876#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #870: USB headphone is not listed in pulseaudio
#870: USB headphone is not listed in pulseaudio +--- Reporter: JeroenGeldhof | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: fixed |Keywords: +--- Comment(by coling): I suspect it's some variant of the bug I mentioned then. We'll get that sorted eventually... once we nail down the cause... as the saying goes, it's like nailing jelly to a fence post :D -- Ticket URL: http://pulseaudio.org/ticket/870#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error
#871: PA compile fails on MAC OS X due to bonjour error ---+ Reporter: spitfire | Owner: lennart Type: defect| Status: new Milestone:| Component: module-zeroconf-* Resolution:|Keywords: ---+ Description changed by coling: Old description: I tried to compile pulseaudio from git HEAD, and got errors about bonjour (of course I don't have avahi installed): In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or directory) make[3]: *** [module-bonjour-publish.lo] Error 1 New description: I tried to compile pulseaudio from git HEAD, and got errors about bonjour (of course I don't have avahi installed): {{{ In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or directory) make[3]: *** [module-bonjour-publish.lo] Error 1 }}} -- -- Ticket URL: http://pulseaudio.org/ticket/871#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error
#871: PA compile fails on MAC OS X due to bonjour error ---+ Reporter: spitfire | Owner: lennart Type: defect| Status: new Milestone:| Component: module-zeroconf-* Resolution:|Keywords: ---+ Comment(by coling): Hi, Daniel Mack has done a lot of the OSX compatibility stuff so I suggest you contact him. I'll send him a mail to notify him on this ticket. Col -- Ticket URL: http://pulseaudio.org/ticket/871#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #868: No digital out with Audiotrak Prodigy HD2
#868: No digital out with Audiotrak Prodigy HD2 --+- Reporter: anbello | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: Audiotrak Prodigy HD2 ICE1724 Envy24 --+- Comment(by coling): Can you use the device via alsa as iec958:0 e.g. aplay -D iec958:0 PA relies on lower level alsa constructs to make it work, so even if it works with hw:0,1, then it doesn't really help us unless it's wrapped up properly. You can hack in support by editing the mixer path scripts or even writing a custom mixer profile for the device (and assigning it via a udev rule) by fiddling with the files in /usr/share/pulseaudio/alsa-mixer/. They are all hand editable (tho' be careful not to change too much that you break it!) and do not require PA to be recompiled (just restarted). If the alsa level access works fine with the iec958 device, then please attach the output of amixer -c0 and pacmd list to this bug. -- Ticket URL: http://pulseaudio.org/ticket/868#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel -+-- Reporter: stephe | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: fixed |Keywords: -+-- Changes (by coling): * status: new = closed * resolution: = fixed Comment: OK, I've merged this now. I wasn't able to get it working on my PS3 but then I'm not sure it worked before anyway (first time I tried). I get a DLNA error when I try to connect to the audio device :( But it does show up and I think that's good enough for a merge :D One comment about the patch, it seems to define some functions that are very dbus centric. If these are standard utilities, then I suggest they go instead into dbus-util.c. If you could look into that I'd be very greatful. Feel free to drop by on IRC or on the mailing list if it's easier. I'll also push this out to stable-queue branch so it'll be part of the next PA release. I think that's fair to target it at that. -- Ticket URL: http://pulseaudio.org/ticket/876#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0% -+-- Reporter: murz| Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by murz): One of the method to solve this is to store ballance value of channels when setting volume to 0%, but it solve the problem only for 2-channels sinks, but for describe volumes in 4.1-7.1 sink we must create more than one ballance value. I think that better solution is store per-channel volumes in sink unchanged, and create new value - main sink volume (something like base volume). And via increase/decrease sink commands do the change only main volume and don't touch channel volumes config. And adding main sink volume will add new feature to pulseaudio: ability to manually set maximum volume for sink (user can set 50% volume of each channel in sink and after that he been able to not be afraid making sound too loud, increasing main volume to 100%). Because very often sound systems have many decibels that was never used, because on 100% volume the sound is very loud. -- Ticket URL: http://pulseaudio.org/ticket/874#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use
#789: 100% CPU use -+-- Reporter: porton | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by porton): {{{ (gdb) thread apply all bt Thread 3 (Thread 0xb2d3cb70 (LWP 3711)): #0 0xb78b7424 in __kernel_vsyscall () #1 0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6 #2 0xb784fda0 in pa_rtpoll_run () from /usr/lib/libpulsecore-0.9.21.so #3 0xb2d5a4ac in ?? () from /usr/lib/pulse-0.9.21/modules/libalsa-util.so #4 0xb780e322 in ?? () from /usr/lib/libpulsecommon-0.9.21.so #5 0xb7554955 in start_thread () from /lib/i686/cmov/libpthread.so.0 #6 0xb7496e7e in clone () from /lib/i686/cmov/libc.so.6 Thread 2 (Thread 0xb253bb70 (LWP 3713)): #0 0xb78b7424 in __kernel_vsyscall () #1 0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6 #2 0xb784fda0 in pa_rtpoll_run () from /usr/lib/libpulsecore-0.9.21.so #3 0xb2d60c66 in ?? () from /usr/lib/pulse-0.9.21/modules/libalsa-util.so #4 0xb780e322 in ?? () from /usr/lib/libpulsecommon-0.9.21.so #5 0xb7554955 in start_thread () from /lib/i686/cmov/libpthread.so.0 #6 0xb7496e7e in clone () from /lib/i686/cmov/libc.so.6 Thread 1 (Thread 0xb7012700 (LWP 3670)): #0 0xb78b7424 in __kernel_vsyscall () #1 0xb7489775 in ppoll () from /lib/i686/cmov/libc.so.6 #2 0xb77ad4aa in pa_mainloop_poll () from /usr/lib/libpulse.so.0 #3 0xb77aec43 in pa_mainloop_iterate () from /usr/lib/libpulse.so.0 #4 0xb77aed14 in pa_mainloop_run () from /usr/lib/libpulse.so.0 #5 0x08052a2e in main () }}} -- Ticket URL: http://pulseaudio.org/ticket/789#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel
#876: [PATCH] Implement MediaServer2 D-Bus interface so PA can work with current Rygel +--- Reporter: stephe | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | +--- This patch implements the MediaServer2 D-Bus interface, http://live.gnome.org/Rygel/MediaServer2Spec, so that PA can work with Rygel version 0.7.1 and higher. Tested with Rygel 0.8.2 on Fedora 14. MediaStreamer running on a Nokia N800 and a Sony BDP-S570 Blu-ray player were used as the upnp clients. A note about testing. Something in Rygel seems to be suffering from the recent change to memcpy in glibc. The LD_PRELOAD trick in Fedora bug https://bugzilla.redhat.com/show_bug.cgi?id=638477 works around the problem. Also, the DLNA/UPnP Streaming source is very distorted with crackling and other noises and seems to run too fast. -- Ticket URL: http://pulseaudio.org/ticket/876 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices
#875: [patch] Add module for jack sense evdev input devices -+-- Reporter: joeshaw | Owner: lennart Type: enhancement | Status: new Milestone: | Component: core Keywords: | -+-- Sound modules in the Linux kernel have the ability to create input devices for sound card jacks through the snd_jack API. (See sound/core/jack.c) These devices are exported as evdev devices which notify (and can be queried about) a few different switches. I'm attaching a new PA module which maps one of those devices to a source and/or sink. Whenever the jack sense changes, a property is set on the source/sink. Clients can watch for notifications and take action (such as muting, changing the default source/sink, migrating streams to that source/sink) when the property changes. -- Ticket URL: http://pulseaudio.org/ticket/875 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices
#875: [patch] Add module for jack sense evdev input devices --+- Reporter: joeshaw | Owner: lennart Type: enhancement | Status: new Milestone: | Component: core Resolution: |Keywords: --+- Comment(by coling): Thank you very much for this. I just wish you'd discussed the implementation with us before starting development. Jack Sensing support is not something that should be handled as a module, but rather something built in to the alsa-sink/source itself. Ultimately Jack Sensing should result in a change of sink/source port e.g. changing from built in Speakers to Headphones port in the case. Clients should not need to do any watching of sink properties etc. however they may wish to take some action, however they will be made aware of this due to the port changes anyway. The main complications with regard to Jack Sensing comes from when users have specifically overridden the port manually - who do we want to take notice of, the user choice or the automatic information? I'd err on the side of the latter, but there are still some complications (e.g. some hardware has three ports - Speakers, Headphones and something else I can't remember off the top of my head, so it's not a simple binary choice. But certainly the ports system is the correct place to hook up jack sensing. I'd love it if you would come and discuss this with us on the IRC channel or via the devel list. -- Ticket URL: http://pulseaudio.org/ticket/875#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #875: [patch] Add module for jack sense evdev input devices
#875: [patch] Add module for jack sense evdev input devices --+- Reporter: joeshaw | Owner: lennart Type: enhancement | Status: new Milestone: | Component: core Resolution: |Keywords: --+- Comment(by joeshaw): Is there a concept of a null port? We would need the null port to be selected until another jack was sensed, because our clients make determinations about which sound card to use based on whether the jacks on one of them is used at all. -- Ticket URL: http://pulseaudio.org/ticket/875#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0% -+-- Reporter: murz| Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by murz): In sink I see the value Base Volume, maybe with it I can change the volume without break the balance. How can I change it via pactl or api? -- Ticket URL: http://pulseaudio.org/ticket/874#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0% -+-- Reporter: murz| Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by tanuk): Replying to [comment:3 murz]: In sink I see the value Base Volume, maybe with it I can change the volume without break the balance. How can I change it via pactl or api? Nope, base volume doesn't solve this problem. Base volume is the level that alsa reports as 0dB. I guess solving this problem requires changing the internal volume representation. Currently volume is represented as a list of individual channel volumes relative to a fixed 100% level. Maybe it should be changed so that there would be a separate reference level variable that would tell the level of the loudest channel, and other channels would be relative to this reference level. This way the balance would be preserved even when the reference level is set to 0%. This would require quite a lot of work, but it should be fairly straightforward work. -- Ticket URL: http://pulseaudio.org/ticket/874#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0% +--- Reporter: murz| Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | +--- I have setup different channel volume for left, right and other channels in output sink. When I changing volume to 0% and restore it (I try kmix in KDE and veromix plasmoid), my per-channel volume config is broken, all channels have the same volume and I need to do the config again. Here is video for kmix: http://www.youtube.com/watch?v=zjPbBd2CCzc What I can do to don't break per-channel volume when setting 0% volume? Does pulseaudio have any option in API or special function/method for solving this issue? -- Ticket URL: http://pulseaudio.org/ticket/874 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #874: Pulseaudio breaks per-channel volume config when restoring volume from 0%
#874: Pulseaudio breaks per-channel volume config when restoring volume from 0% -+-- Reporter: murz| Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by murz): gnome volume manager have the same problem, here is the bug about this in ubuntu: https://bugs.launchpad.net/ubuntu/+source/gnome-media/+bug/672420 -- Ticket URL: http://pulseaudio.org/ticket/874#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #872: Loss of sound with two sources
#872: Loss of sound with two sources +--- Reporter: AdamK | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | +--- I have one sound source with no heavy latency requirements (movie playing in VLC). Now, if another sound is played (system sound for example), the first time two sources have to be mixed there is a slight loss of sound (for less then second). It does not happen afterwards. $ pulseaudio --version pulseaudio 0.9.21-98-ga8d7-dirty (ubuntu beta) -- Ticket URL: http://pulseaudio.org/ticket/872 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #872: Loss of sound with two sink inputs (was: Loss of sound with two sources)
#872: Loss of sound with two sink inputs -+-- Reporter: AdamK | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by coling): Could this be a result of artefacts from the Flat Volumes feature? (pulseaudio will always try and use h/w volume whenever possible to reduce overhead and thus power consumption etc - and thus when there is only one stream playing, the volume of that stream will always be the same as the underlying alsa volume - although the concept of a system volume is still maintained and presented to the user even if it's not really used like that underneath!) To test this, load up alsamixer -c0 and have a look at the bars for Master and maybe PCM too. When your second sink input (source is not the right term as a source is a recording device - e.g. a mic. I've fixed the summary to reflect this.) starts (and you hear your dropout), do you see them jump about a bit? If so, you can work around this by disabling flat-volumes in daemon.conf. Git master already has support to minimise the fallout from this - which stems from there being no way to synchronise mixer changes in alsa - so if this is the problem you are seeing, it's likely fixed already :) -- Ticket URL: http://pulseaudio.org/ticket/872#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%
#821: volume mute at 15% +--- Reporter: yelo3 | Owner: lennart Type: defect | Status: closed Milestone: | Component: module-alsa-* Resolution: elsewhere |Keywords: +--- Changes (by tanuk): * status: new = closed * resolution: = elsewhere Comment: Replying to [comment:2 starT_T]: I tested, set the volume to 14%, -51.23 dB, I am sure that I can hear the sound. Great. -48 dB isn't mute. Yep. It isn't. But which value is good for mute? In Pulseaudio volume scale (0-65536) 0 means mute. In decibel scale -inf dB means mute. Did that answer your question? I'm a bit confused about your message - are you just reporting that you can't reproduce this bug? This bug is actually an alsa bug, so whether you can reproduce this or not depends on your hardware. Hmm, since this is an alsa bug, I guess nobody has anything against me resolving this ticket as elsewhere. -- Ticket URL: http://pulseaudio.org/ticket/821#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%
#821: volume mute at 15% +--- Reporter: yelo3 | Owner: lennart Type: defect | Status: closed Milestone: | Component: module-alsa-* Resolution: elsewhere |Keywords: +--- Comment(by coling): Yes I think this is a valid closer Tanu. For reference, I recently helped to solve this problem on my own h/w. The error was a combination of three problems: one in the h/w itself, one in the alsa driver and one in the alsa userspace library. * The hardware did not follow the HDA spec and muted the device when it was the slider was set to 0. * The driver did not have a quirk defined for this. * The library did not pay attention to the quirk when it was added. So there are a few commits needed to fix it: * kernel: [http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=de8c85f7840e5e29629de95f5af24297fb325e0b de8c85] and [http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=a74ccea51d4314632a81d568d59bf885e5b09d93 a74cce] (but obviously this is specific to my h/w - yours may need similar fixes if it doesn't have them already) * alsa-lib: [http://git.alsa-project.org/?p=alsa- lib.git;a=commit;h=2f6206da0c1ff88235e6eca0077343f22a4b43ee 2f6206] -- Ticket URL: http://pulseaudio.org/ticket/821#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #848: Loud noise for pa_stream_flush()
#848: Loud noise for pa_stream_flush() +--- Reporter: mschwendt | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: very loud short sound on stream flushing +--- Changes (by adi): * cc: a...@drcomp.erfurt.thur.de (added) -- Ticket URL: http://pulseaudio.org/ticket/848#comment:12 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #821: volume mute at 15%
#821: volume mute at 15% -+-- Reporter: yelo3 | Owner: lennart Type: defect | Status: new Milestone: | Component: module-alsa-* Resolution: |Keywords: -+-- Comment(by starT_T): {{{ set-sink-volume 0 9175 D: alsa-sink.c: Requested volume: 0: 14% 1: 14% D: alsa-sink.c: Got hardware volume: 0: 14% 1: 14% D: alsa-sink.c: Calculated software volume: 0: 99% 1: 99% (accurate- enough=yes) list-sinks 1 sink(s) available. * index: 0 name: alsa_output.0.analog-stereo driver: module-alsa-sink.c flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME state: RUNNING suspend cause: priority: 9009 volume: 0: 14% 1: 14% 0: -51.23 dB 1: -51.23 dB balance 0.00 base volume: 10% -60.00 dB volume steps: 65537 }}} I tested, set the volume to 14%, -51.23 dB, I am sure that I can hear the sound. -48 dB isn't mute. But which value is good for mute? -- Ticket URL: http://pulseaudio.org/ticket/821#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use
#789: 100% CPU use -+-- Reporter: porton | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Comment(by Ford_Prefect): If you are familiar with gdb, when CPU usage hits 100%, could you attach to the process and provide a backtrace? Roughly, this would required: gdb path-to-pulseaudio pid-of-pulseaudio thread apply all bt Also would help if you're using the latest version of PA available on your distro. -- Ticket URL: http://pulseaudio.org/ticket/789#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #822: Closing the last stream on an ALSA sink causes distortion
#822: Closing the last stream on an ALSA sink causes distortion +--- Reporter: adi| Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: duplicate |Keywords: +--- Changes (by Ford_Prefect): * status: new = closed * resolution: = duplicate Comment: Marking as a duplicate as pointed out. -- Ticket URL: http://pulseaudio.org/ticket/822#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #782: loud woof when closing a player
#782: loud woof when closing a player -+-- Reporter: patra...@gmail.com | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: duplicate |Keywords: -+-- Changes (by Ford_Prefect): * cc: a...@accosted.net (added) * status: new = closed * resolution: = duplicate Comment: Marking as duplicate of #848. -- Ticket URL: http://pulseaudio.org/ticket/782#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #822: Closing the last stream on an ALSA sink causes distortion
#822: Closing the last stream on an ALSA sink causes distortion +--- Reporter: adi| Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: duplicate |Keywords: +--- Changes (by Ford_Prefect): * cc: a...@accosted.net (added) -- Ticket URL: http://pulseaudio.org/ticket/822#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #789: 100% CPU use
#789: 100% CPU use -+-- Reporter: porton | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: -+-- Changes (by Ford_Prefect): * cc: a...@accosted.net (added) -- Ticket URL: http://pulseaudio.org/ticket/789#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #871: PA compile fails on MAC OS X due to bonjour error
#871: PA compile fails on MAC OS X due to bonjour error --+- Reporter: spitfire | Owner: lennart Type: defect| Status: new Milestone:| Component: module-zeroconf-* Keywords:| --+- I tried to compile pulseaudio from git HEAD, and got errors about bonjour (of course I don't have avahi installed): In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:25:33: error: avahi-client/client.h: No such file or directory In file included from modules/module-bonjour-publish.c:47: ./pulsecore/avahi-wrap.h:29: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ./pulsecore/avahi-wrap.h:30: error: expected ')' before '*' token lipo: can't open input file: /var/tmp//cc4jQLPi.out (No such file or directory) make[3]: *** [module-bonjour-publish.lo] Error 1 -- Ticket URL: http://pulseaudio.org/ticket/871 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #868: No digital out with Audiotrak Prodigy HD2
#868: No digital out with Audiotrak Prodigy HD2 --+- Reporter: anbello | Owner: lennart Type: defect| Status: new Milestone:| Component: daemon Keywords: Audiotrak Prodigy HD2 ICE1724 Envy24 | --+- I have an Audiotrak Prodigy HD2 sound card on a PC with ubuntu 10.10 (i have tried both 32bit and 64bit) and i am not able to use the digital spdif out with pulseaudio. In Sound Preferences - Hardware tab - Profile: combobox i have only Analog Stereo Input, Analog Stereo Output and Analog Stereo Duplex and no Digital Stereo * of any sort so i cannot use pulseaudio with spdif output. With another Envy24HT based sound card (Terratec Aureon 7.1 Space) i had no problems and in Profile: combobox i had Digital Stereo Output (IEC958) (or something similar). With alsa i have no problems: aplay -l[[BR]] List of PLAYBACK Hardware Devices [[BR]] card 0: HD2 [Audiotrak Prodigy HD2], device 0: ICE1724 [ICE1724][[BR]] Subdevices: 1/1[[BR]] Subdevice #0: subdevice #0[[BR]] card 0: HD2 [Audiotrak Prodigy HD2], device 1: ICE1724 IEC958 [ICE1724 IEC958][[BR]] Subdevices: 1/1[[BR]] Subdevice #0: subdevice #0[[BR]] Thanks[[BR]] Andrea -- Ticket URL: http://pulseaudio.org/ticket/868 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #869: 100% cpu use
#869: 100% cpu use +--- Reporter: mccann | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | +--- Noticed my battery drain very very quickly. Looked at top: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 17959 mccann 9 -11 107m 2944 2320 R 99.7 0.1 850:13.90 pulseaudio Yikes. strace showed: read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device) ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37, events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10, events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20, events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36, events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN}, {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN}, {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN}, {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15, events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL, NULL, 8) = 1 ([{fd=32, revents=POLLERR}]) read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device) ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37, events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10, events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20, events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36, events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN}, {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN}, {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN}, {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15, events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL, NULL, 8) = 1 ([{fd=32, revents=POLLERR}]) read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device) ppoll([{fd=4, events=POLLIN}, {fd=38, events=POLLIN}, {fd=37, events=POLLIN}, {fd=27, events=POLLIN}, {fd=33, events=POLLIN}, {fd=10, events=POLLIN}, {fd=21, events=POLLIN}, {fd=7, events=POLLIN}, {fd=20, events=POLLIN}, {fd=26, events=POLLIN}, {fd=32, events=POLLIN}, {fd=36, events=POLLIN|POLLERR|POLLHUP}, {fd=36, events=0}, {fd=35, events=POLLIN}, {fd=34, events=POLLIN}, {fd=14, events=POLLIN}, {fd=28, events=POLLIN}, {fd=31, events=POLLIN}, {fd=22, events=POLLIN}, {fd=25, events=POLLIN}, {fd=19, events=POLLIN|POLLERR|POLLHUP}, {fd=19, events=0}, {fd=15, events=POLLIN}, {fd=18, events=POLLIN}, {fd=8, events=POLLIN}], 25, NULL, NULL, 8) = 1 ([{fd=32, revents=POLLERR}]) read(32, 0xbfeba8c4, 72)= -1 ENODEV (No such device) And gdb showed: Thread 4 (Thread 0xb753bb70 (LWP 17960)): #0 0x00efc416 in __kernel_vsyscall () #1 0x006bdec5 in ppoll (fds=0x847aa58, nfds=2, timeout=value optimized out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58 #2 0x0059dea0 in pa_rtpoll_run (p=0x8477dc0, wait_op=true) at pulsecore/rtpoll.c:304 #3 0x001d061c in thread_func (userdata=0x847a890) at modules/alsa/alsa- sink.c:1430 #4 0x003bd442 in internal_thread_func (userdata=0x84c6410) at pulsecore /thread-posix.c:72 #5 0x00510919 in start_thread (arg=0xb753bb70) at pthread_create.c:301 #6 0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133 Thread 3 (Thread 0xb294eb70 (LWP 17961)): #0 0x00efc416 in __kernel_vsyscall () #1 0x006bdec5 in ppoll (fds=0x847f958, nfds=2, timeout=value optimized out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58 #2 0x0059dea0 in pa_rtpoll_run (p=0x84bf188, wait_op=true) at pulsecore/rtpoll.c:304 #3 0x001d6d86 in thread_func (userdata=0x84a0a10) at modules/alsa/alsa- source.c:1274 #4 0x003bd442 in internal_thread_func (userdata=0x84d1290) at pulsecore /thread-posix.c:72 #5 0x00510919 in start_thread (arg=0xb294eb70) at pthread_create.c:301 #6 0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133 Thread 2 (Thread 0xb1dffb70 (LWP 17962)): #0 0x00efc416 in __kernel_vsyscall () #1 0x006bdec5 in ppoll (fds=0x847e7b8, nfds=2, timeout=value optimized out, sigmask=0x0) at ../sysdeps/unix/sysv/linux/ppoll.c:58 #2 0x0059dea0 in pa_rtpoll_run (p=0x847e790, wait_op=true) at pulsecore/rtpoll.c:304 #3 0x001d6d86 in thread_func (userdata=0x847e5e0) at modules/alsa/alsa- source.c:1274 #4 0x003bd442 in internal_thread_func (userdata=0x84d8ba8) at pulsecore /thread-posix.c:72 #5 0x00510919 in start_thread (arg=0xb1dffb70) at pthread_create.c:301 #6 0x006c8cbe in clone () at ../sysdeps/unix/sysv/linux/i386/clone.S:133 Thread 1 (Thread 0xb7743700 (LWP 17959)): #0 0x00efc416 in __kernel_vsyscall () #1 0x006bdec5 in ppoll (fds=0x84700b8, nfds=25, timeout=value optimized out
Re: [pulseaudio-tickets] [PulseAudio] #858: Pulseaudio uses all CPU on ALSA problems
#858: Pulseaudio uses all CPU on ALSA problems +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by LCID Fire): After upgrading to 2.6.36 I don't get the dmesgs anymore - but it still locks the whole machine. Even worse - apps that use Pulseaudio like Thunderbird and Empathy do the same. -- Ticket URL: http://pulseaudio.org/ticket/858#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing
#851: pulsesink not playing +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: gst-pulse Resolution: |Keywords: gst-launch pulseaudio pulsesink +--- Comment(by LCID Fire): Using the released versions in Ubuntu 10.10 it works for me again, too. -- Ticket URL: http://pulseaudio.org/ticket/851#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing
#851: pulsesink not playing +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: closed Milestone: | Component: gst-pulse Resolution: fixed |Keywords: gst-launch pulseaudio pulsesink +--- Changes (by LCID Fire): * status: new = closed * resolution: = fixed -- Ticket URL: http://pulseaudio.org/ticket/851#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #867: Please update POTFILES.in
#867: Please update POTFILES.in --+- Reporter: kelemeng | Owner: lennart Type: defect| Status: new Milestone:| Component: build-system Keywords:| --+- I just added Pulseaudio to Gnome's Damned Lies (so that Gnome translators can see the stats where they are looking for it: http://l10n.gnome.org/module/Pulseaudio/ ), and intltool is complaining, please fix this: There are some missing files from POTFILES.in: * src/daemon/pulseaudio-kde.desktop.in * src/modules/module-equalizer-sink.c * src/modules/module-virtual-sink.c * src/modules/module-virtual-source.c Also, it would be great to sync the master-tx and master branches :). -- Ticket URL: http://pulseaudio.org/ticket/867 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults ---+ Reporter: rabbit64 | Owner: lennart Type: defect| Status: new Milestone:| Component: module-combine-* Resolution:|Keywords: ---+ Comment(by rabbit64): So you've patched alsa-sink.c to define DEBUG_TIMING. Why? These segfaults are not segfaults, instead the daemon gets the debug trap signal. To get some more information about timing in log files. Raymond from alsa has advised me to do so. Here is link to the original thread: [https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5146]. I think you need an account in alsa bugtracker. So what to do? Should this ticket be closed? Should we remove the PA_DEBUG_TRAP line from check_left_to_play()? No, it is probably intentional. There may be a bug in emu10k1 driver and coredump should provide more information if I understand it correctly. If DEBUG_TIMING is disabled, pulseaudio recovers from this state by spawning new process. But after some respawns audacious audio player freezes and has to be SIGKILLed. -- Ticket URL: http://pulseaudio.org/ticket/865#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #851: pulsesink not playing
#851: pulsesink not playing +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: gst-pulse Resolution: |Keywords: gst-launch pulseaudio pulsesink +--- Comment(by starT_T): {{{ s...@ubuntu:~$ gst-launch-0.10 -v pulsesrc ! pulsesink Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-buffer-time = 2376 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-latency-time = 8 /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps = audio/x -raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstPulseSink:pulsesink0.GstPad:sink: caps = audio/x -raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstPulseSink:pulsesink0: volume = 1.00 /GstPipeline:pipeline0/GstPulseSink:pulsesink0: mute = FALSE ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 56762866587 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstPulseSink:pulsesink0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... s...@ubuntu:~$ pulseaudio --version pulseaudio 0.9.21-63-gd3efa-dirty s...@ubuntu:~$ gst-launch-0.10 --version gst-launch-0.10 version 0.10.28 GStreamer 0.10.28 https://launchpad.net/distros/ubuntu/+source/gstreamer0.10 }}} works fine for me too -- Ticket URL: http://pulseaudio.org/ticket/851#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..
#855: Illegal instruction.. ---+ Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: new Milestone: 0.9.22| Component: gst-pulse Resolution:|Keywords: ---+ Comment(by starT_T): {{{ s...@ubuntu:~$gst-launch-0.10 filesrc location=/home/star/Desktop/bbb.wav ! wavparse ! audioconvert ! audioresample ! pulsesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstPulseSinkClock Got EOS from element pipeline0. Execution ended after 3720897522 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... }}} my GStreamer version is 0.10.28 I played very well -- Ticket URL: http://pulseaudio.org/ticket/855#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults ---+ Reporter: rabbit64 | Owner: lennart Type: defect| Status: new Milestone:| Component: module-combine-* Resolution:|Keywords: ---+ Comment(by tanuk): So you've patched alsa-sink.c to define DEBUG_TIMING. Why? These segfaults are not segfaults, instead the daemon gets the debug trap signal. The check_left_to_play() function has this piece of code: {{{ #ifdef DEBUG_TIMING PA_DEBUG_TRAP; #endif }}} Are you arguing that it's a bug to trigger the signal when DEBUG_TIMING is defined? I don't have an opinion myself, since it's not documented what DEBUG_TIMING means, so I don't know what's the intended behaviour - is triggering the signal a good thing to do or not. Anyway, this message is the reason why this code path is executed: {{{ pulseaudio[14730]: ( 3.543| 0.000) alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 319392 bytes (1810 ms). pulseaudio[14730]: ( 3.543| 0.000) alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_emu10k1'. Please report this issue to the ALSA developers. pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: snd_pcm_dump(): pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: Hardware PCM card 1 'SB Live! 5.1 [SB0220]' device 0 subdevice 0 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: Its setup is: pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: stream : CAPTURE pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: access : MMAP_INTERLEAVED pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: format : S16_LE pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: subformat: STD pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: channels : 2 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: rate : 44100 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: exact rate : 44100 (44100/1) pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: msbits : 16 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: buffer_size : 4096 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: period_size : 2048 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: period_time : 46439 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: tstamp_mode : ENABLE pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: period_step : 1 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: avail_min: 2048 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: period_event : 1 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: start_threshold : -1 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: stop_threshold : 4611686018427387904 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: silence_threshold: 0 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: silence_size : 0 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: boundary : 4611686018427387904 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: appl_ptr : 49256 pulseaudio[14730]: ( 3.544| 0.000) alsa-util.c: hw_ptr : 129104 }}} So what to do? Should this ticket be closed? Should we remove the PA_DEBUG_TRAP line from check_left_to_play()? -- Ticket URL: http://pulseaudio.org/ticket/865#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
of silence already written to the hardware buffer, which means that it's possible to rewind 0.1 seconds. Your stream, however, does not know that there's only 0.1 seconds in the buffer. It should be informed about that, but the current implementation doesn't. Since it doesn't know that, it has to play safe, and assume that the whole hardware buffer can be rewound, i.e. 2 seconds. So, when the sink asks for more data, the stream notices that draining has been requested, but it checks the rewind buffer and sees that it's not empty - the rewind buffer contains all data from the last 2 seconds, and since the clip started 1.9 seconds ago and finished 1.4 seconds ago, it means that the stream must still wait 1.4 seconds until it can be sure that it's safe to send the drain acknowledgement. And so the sink fills the buffer again with silence and sleeps 1.9 seconds. After that wait is over, the stream sees that the most recent data is 1.4 + 1.9 = 3.3 seconds old, which is more than the maximum rewind amount, so now it's safe to send the drain acknowledgement - 3.3 seconds late. That's how it works in principle - the calculations used a bunch of assumptions that aren't entirely correct, so don't take the numbers as exact, but as your experiments show, the extra delay is always measured in seconds, if the hw buffer is two seconds. So, who wants to implement the needed fixes? -- Ticket URL: http://pulseaudio.org/ticket/866#comment:7 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete -+-- Reporter: th | Owner: lennart Type: defect | Status: new Milestone: | Component: libpulse Resolution: |Keywords: -+-- Comment(by th): I ran the test program with pulse running with verbose logging, if you look at the Launchpad bug entry you can see it: [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567] (also pasted below). It seems that pulse gets a buffer underrun at the time that I would expect pa_simple_drain() to return, but it doesn't, and waits for 2s after that. In simplistic terms, I guess pa_simple_drain() could be thought of as wait for buffer underrun, but for some reason it doesn't signal the client when the buffer runs out. {{{ ... ( 0.034| 0.000) D: source.c: Processing rewind... [0.344269] play done [0.344328] not calling pa_simple_drain() [0.358380] playing 680 hz tone volume 50 for 500 ms [0.829451] play done [0.829489] not calling pa_simple_drain() [0.840411] playing 440 hz tone volume 50 for 500 ms [1.303775] play done ( 1.471| 1.437) D: protocol-native.c: Underrun on 'beep', 0 bytes in queue. [3.505551] pa_simple_drain() done [3.519491] playing 680 hz tone volume 50 for 500 ms ( 3.532| 2.060) D: protocol-native.c: Requesting rewind due to end of underrun. ( 3.532| 0.000) D: alsa-sink.c: Requested to rewind 10156 bytes. ( 3.532| 0.000) D: alsa-sink.c: Limited to 10156 bytes. ( 3.532| 0.000) D: alsa-sink.c: before: 2539 ( 3.532| 0.000) D: alsa-sink.c: after: 2539 ( 3.532| 0.000) D: alsa-sink.c: Rewound 10156 bytes. ( 3.532| 0.000) D: sink.c: Processing rewind... ( 3.532| 0.000) D: sink-input.c: Have to rewind 10156 bytes on render memblockq. ( 3.533| 0.000) D: source.c: Processing rewind... [3.848197] play done ( 4.016| 0.483) D: protocol-native.c: Underrun on 'beep', 0 bytes in queue. [6.050165] pa_simple_drain() done [6.062775] playing 440 hz tone volume 50 for 500 ms ... }}} -- Ticket URL: http://pulseaudio.org/ticket/866#comment:6 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te
#790: pulsecore/svolume_arm.c fails to compile for armv5te -+-- Reporter: ao2 | Owner: wtay Type: defect | Status: new Milestone: 0.9.22 | Component: core Resolution: |Keywords: -+-- Changes (by dromi): * cc: alexandre.rela...@gmail.com (added) Comment: You should be able to compile for a generic ARM architecture. It will compile this ARMv6 assembly code but it should not be called on an ARMv5 processor. But since you want to test some patches, here are they ;) The two patches do the same, ie they remove this ARMv6 code if you compile for an ARMv5 specific architecture. In the first one, I use a macro that was created elsewhere but whose name is not explicit at all. The second one is a bit cleaner since the macro is renamed. I have tested the first one, not the second one. Any kind of feedback is welcome ! -- Ticket URL: http://pulseaudio.org/ticket/790#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te
#790: pulsecore/svolume_arm.c fails to compile for armv5te -+-- Reporter: ao2 | Owner: wtay Type: defect | Status: new Milestone: 0.9.22 | Component: core Resolution: |Keywords: -+-- Comment(by flokli): Thanks for your patches! I applied the second one to the source code, and it compiled well on this device (see [http://pastebin.com/aj7VrDDm]). However, there are some ugly autoconf warnings... Will now test a build of pulseaudio with network/avahi support and post results... -- Ticket URL: http://pulseaudio.org/ticket/790#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te
#790: pulsecore/svolume_arm.c fails to compile for armv5te -+-- Reporter: ao2 | Owner: wtay Type: defect | Status: new Milestone: 0.9.22 | Component: core Resolution: |Keywords: -+-- Comment(by flokli): Ok I tested it, it works! I will need to experiment a bit with resampling methods and such because sound sometimes is stuttering (cpu a bit weak), but all in all everything works like it should :-) Can you commit this patch so that it will be in the next release? -- Ticket URL: http://pulseaudio.org/ticket/790#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults --+- Reporter: rabbit64 | Owner: lennart Type: defect| Status: new Milestone:| Component: module-combine-* Keywords:| --+- PA segfaults when combining analog stereo and digital output on my CT4780 SBLive card. I have uploaded coredumps and corresponding messages to http://fi.muni.cz/~xsakalik/ . I have in default.pa {{{ load-module module-combine slaves=alsa_output.pci-_05_00.0.iec958-stereo,alsa_output.pci-_05_00.0 .analog-stereo resample_method=trivial sink_name=combined-trivial }}} daemon.conf: {{{ daemonize = no allow-exit = no high-priority = yes nice-level = -11 realtime-scheduling = yes realtime-priority = 5 log-target = syslog log-level = debug log-time = yes }}} I have compiled pulseaudio from git (commit 3de129f3ac8dd6cf51178b266837db4d5e4a1215) with this patch: {{{ diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c index 1108a79..6de95b6 100644 --- a/src/modules/alsa/alsa-sink.c +++ b/src/modules/alsa/alsa-sink.c @@ -59,7 +59,7 @@ #include alsa-util.h #include alsa-sink.h -/* #define DEBUG_TIMING */ +#define DEBUG_TIMING #define DEFAULT_DEVICE default diff --git a/src/modules/alsa/alsa-source.c b/src/modules/alsa/alsa- source.c index 5f12675..47f914b 100644 --- a/src/modules/alsa/alsa-source.c +++ b/src/modules/alsa/alsa-source.c @@ -56,7 +56,7 @@ #include alsa-util.h #include alsa-source.h -/* #define DEBUG_TIMING */ +#define DEBUG_TIMING #define DEFAULT_DEVICE default }}} -- Ticket URL: http://pulseaudio.org/ticket/865 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults ---+ Reporter: rabbit64 | Owner: lennart Type: defect| Status: new Milestone:| Component: module-combine-* Resolution:|Keywords: ---+ Comment(by coling): Hi there. Thanks for the detailed report. Would you be able to get a backtrace from those coredumps? You should be able to load them in gdb and get the necessary backtraces from that. As you've compiled your own, the core dumps on their own will not really help remotes, but a nice detailed backtrace could help a lot. See the wiki:Community page for some bactrace tips. -- Ticket URL: http://pulseaudio.org/ticket/865#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #865: Pulseaudio segfaults
#865: Pulseaudio segfaults ---+ Reporter: rabbit64 | Owner: lennart Type: defect| Status: new Milestone:| Component: module-combine-* Resolution:|Keywords: ---+ Comment(by rabbit64): Hi, I've compiled it with -O2 so some debugging information are not there. I should have also said that two instances of pulseaudio (under different users) were running (this was under root). But I think it isn't a problem because it has segfaulted even when only one instance was running. I have uploaded backtraces on my webpage. -- Ticket URL: http://pulseaudio.org/ticket/865#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete -+-- Reporter: th | Owner: lennart Type: defect | Status: new Milestone: | Component: libpulse Resolution: |Keywords: -+-- Changes (by th): * cc: tho...@horsten.com (added) * component: daemon = libpulse -- Ticket URL: http://pulseaudio.org/ticket/866#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete -+-- Reporter: th | Owner: lennart Type: defect | Status: new Milestone: | Component: libpulse Resolution: |Keywords: -+-- Comment(by th): Here's the test program output with automatic linewrap removed...: {{{ $ ./beep [0.039147] playing 440 hz tone volume 50 for 500 ms [0.291107] play done [0.291116] not calling pa_simple_drain() [0.293467] playing 680 hz tone volume 50 for 500 ms [0.855440] play done [0.855459] not calling pa_simple_drain() [0.857808] playing 440 hz tone volume 50 for 500 ms [1.312107] play done [3.544309] pa_simple_drain() done [3.546687] playing 680 hz tone volume 50 for 500 ms [3.798973] play done [6.132183] pa_simple_drain() done [6.134535] playing 440 hz tone volume 50 for 500 ms [6.386808] play done [6.386824] not calling pa_simple_drain() [6.386851] playing 440 hz tone volume 0 for 500 ms [6.951772] play done [6.951789] not calling pa_simple_drain() [6.956659] playing 880 hz tone volume 50 for 1000 ms [7.912965] play done [10.144917] pa_simple_drain() done }}} And the gcc command was supposed to be: {{{ gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c }}} -- Ticket URL: http://pulseaudio.org/ticket/866#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete -+-- Reporter: th | Owner: lennart Type: defect | Status: new Milestone: | Component: libpulse Resolution: |Keywords: -+-- Description changed by coling: Old description: The libpulse-simple API has a function, pa_simple_drain() which is supposed to [w]ait until all data already written is played by the daemon. However with the version that ships with Ubuntu 10.10, it waits a *fair* bit more than that, approximately 2 seconds (see below). This is also the case for Fedora 14 according to a user on IRC who confirmed this issue, hence I am reporting it here. This means, that when I want to synchronize audio in my program ie. ensure that previously streamed audio has finished playing, e.g. before I start playing a new sound or just prior to exiting the program, there is a typically 2.2 second extra delay. Without this call, at the end of playback before exiting, the final sound will be clipped off at the end, since there is still unbuffered audio waiting to be sent, so it is needed to call this at the end of the program, but now causes an extra delay of several seconds before the program exits. I have attached a program that demonstrates this behaviour. Compile with: $ gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c Here is the output from the program: $ ./beep [0.039147] playing 440 hz tone volume 50 for 500 ms [0.291107] play done [0.291116] not calling pa_simple_drain() [0.293467] playing 680 hz tone volume 50 for 500 ms [0.855440] play done [0.855459] not calling pa_simple_drain() [0.857808] playing 440 hz tone volume 50 for 500 ms [1.312107] play done [3.544309] pa_simple_drain() done [3.546687] playing 680 hz tone volume 50 for 500 ms [3.798973] play done [6.132183] pa_simple_drain() done [6.134535] playing 440 hz tone volume 50 for 500 ms [6.386808] play done [6.386824] not calling pa_simple_drain() [6.386851] playing 440 hz tone volume 0 for 500 ms [6.951772] play done [6.951789] not calling pa_simple_drain() [6.956659] playing 880 hz tone volume 50 for 1000 ms [7.912965] play done [10.144917] pa_simple_drain() done I have also reported this on Ubuntu's bug tracker: [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567] New description: The libpulse-simple API has a function, pa_simple_drain() which is supposed to [w]ait until all data already written is played by the daemon. However with the version that ships with Ubuntu 10.10, it waits a *fair* bit more than that, approximately 2 seconds (see below). This is also the case for Fedora 14 according to a user on IRC who confirmed this issue, hence I am reporting it here. This means, that when I want to synchronize audio in my program ie. ensure that previously streamed audio has finished playing, e.g. before I start playing a new sound or just prior to exiting the program, there is a typically 2.2 second extra delay. Without this call, at the end of playback before exiting, the final sound will be clipped off at the end, since there is still unbuffered audio waiting to be sent, so it is needed to call this at the end of the program, but now causes an extra delay of several seconds before the program exits. I have attached a program that demonstrates this behaviour. Compile with: $ gcc `pkg-config --cflags --libs libpulse-simple` -o beep beep.c Here is the output from the program: {{{ $ ./beep [0.039147] playing 440 hz tone volume 50 for 500 ms [0.291107] play done [0.291116] not calling pa_simple_drain() [0.293467] playing 680 hz tone volume 50 for 500 ms [0.855440] play done [0.855459] not calling pa_simple_drain() [0.857808] playing 440 hz tone volume 50 for 500 ms [1.312107] play done [3.544309] pa_simple_drain() done [3.546687] playing 680 hz tone volume 50 for 500 ms [3.798973] play done [6.132183] pa_simple_drain() done [6.134535] playing 440 hz tone volume 50 for 500 ms [6.386808] play done [6.386824] not calling pa_simple_drain() [6.386851] playing 440 hz tone volume 0 for 500 ms [6.951772] play done [6.951789] not calling pa_simple_drain() [6.956659] playing 880 hz tone volume 50 for 1000 ms [7.912965] play done [10.144917] pa_simple_drain() done }}} I have also reported this on Ubuntu's bug tracker: [https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/660567] -- -- Ticket URL: http://pulseaudio.org/ticket/866#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete
#866: pa_simple_drain() takes much longer (about 2000ms) than expected to complete -+-- Reporter: th | Owner: lennart Type: defect | Status: new Milestone: | Component: libpulse Resolution: |Keywords: -+-- Comment(by coling): Thanks for the test case. Replicated here also. Not sure what cases it, but hopefully I or someone else will find time to investigate soon. -- Ticket URL: http://pulseaudio.org/ticket/866#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #790: pulsecore/svolume_arm.c fails to compile for armv5te
#790: pulsecore/svolume_arm.c fails to compile for armv5te -+-- Reporter: ao2 | Owner: wtay Type: defect | Status: new Milestone: 0.9.22 | Component: core Resolution: |Keywords: -+-- Changes (by flokli): * cc: flo...@flokli.de (added) Comment: I also see build failure on armv5tel-softfloat-linux-gnueabi gcc-4.4.4, glibc-2.11.2-r0, 2.6.36-rc7-00125-gd2a63db armv5tel in file pulsecore/svolume_arm.c with pulseaudio pulseaudio-0.9.21.2-r2. {{{ CC libpulsecore_0.9.21_la-svolume_arm.lo {standard input}: Assembler messages: {standard input}:34: Error: selected processor does not support `ssat r0,#16,r0' {standard input}:48: Error: selected processor does not support `ssat r2,#16,r2' {standard input}:49: Error: selected processor does not support `ssat r3,#16,r3' {standard input}:50: Error: selected processor does not support `pkhbt r0,r3,r2,LSL#16' {standard input}:66: Error: selected processor does not support `ssat r2,#16,r2' {standard input}:67: Error: selected processor does not support `ssat r3,#16,r3' {standard input}:68: Error: selected processor does not support `ssat r4,#16,r4' {standard input}:69: Error: selected processor does not support `ssat r5,#16,r5' {standard input}:70: Error: selected processor does not support `pkhbt r0,r3,r2,LSL#16' {standard input}:71: Error: selected processor does not support `pkhbt r1,r5,r4,LSL#16' make[3]: *** [libpulsecore_0.9.21_la-svolume_arm.lo] Error 1 }}} pasted build logs (link from ao2 doesn't work anymore) for -O2 [http://pastebin.com/PTs1we1s] and -Os [http://pastebin.com/zSau03SJ] Having pulseaudio running on this device would be very nice (I want to use it as a music jukebox with network sound) Will test patches and stuff if needed :-) -- Ticket URL: http://pulseaudio.org/ticket/790#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink
#863: assign different device.description when more than one sink +--- Reporter: skierpage | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by coling): Yeah in an ideal world I'd have done the renaming via that UI but it would require some mechanism (i.e. API) in Phonon which in turn I could proxy on to PA, so not as trivial as it may first seem. Chances are that something made PA crash (likely in the module-device- manager code that I wrote!) and it then autospawned again (as it's meant to do) but would lack the nice modules that are loaded at login via start- pulseaudio-x11 and start-pulseaudio-kde. KDE support is certainly a whole lot better than it was. I've still got more features to add and obviously fix up any bugs etc. Aside from the name of the devices do you experience any other major issues? I have tried the Kubuntu live CD and while IMO it would be better if the Ubuntu Pulseaudio had even more of the stable-queue fixes (I'm not 100% sure what they do and don't have but I'm pretty sure it's not the full compliment), it did seem to work fine for me, even with some exotic devices and bluetooth headsets etc. which is nice. Personally I find that Xine sucks with regards to it's PA output and still far prefer GStreamer backend for that, and hopefully VLC backend will prove more reliable in the future than the Xine one. -- Ticket URL: http://pulseaudio.org/ticket/863#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink
#863: assign different device.description when more than one sink +--- Reporter: skierpage | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by coling): Oh incidentally, I noticed something you mentioned on my blog about the mute status being odd (PA apparently muting one of the two devices but kmix showing it as unmuted - to be honest this could be a bug in kmix - I did use names as an index at one point IIRC - will have to double check). The “deciding to mute” scenario is interesting. Not quite sure what the reason for that would be but I suspect it’s something non-obvious. I’d be very interested to get the “pacmd list” output for this muted device – and a screenshot from kmix at the same time when this misbehaviour is observed. -- Ticket URL: http://pulseaudio.org/ticket/863#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink
#863: assign different device.description when more than one sink +--- Reporter: skierpage | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by skierpage): Thanks for responding! pavucontrol is not on the Kubuntu CD. Ideally System Settings Multimedia Phonon preferred output dialog would let me right-click to rename outputs, which is another Launchpad bug to file. I updated bug #723 with my `lspci -vvv` info, hope it helps. I don't know why my fiddling in pacmd made Phonon and/or PulseAudio lose track of names and/or unload module-device-manager, if I can reproduce I'll file a Launchpad bug. You must be the Tolstoy who penned KDE + PulseAudio != Sucks; it sounds like PulseAudio and KDE work better in other distributions than Kubuntu, even though 10.10 made Phonon Xine PulseAudio the default. I don't know whether to upgrade to 10.10 or switch distros. -- Ticket URL: http://pulseaudio.org/ticket/863#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink
#863: assign different device.description when more than one sink +--- Reporter: skierpage | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by coling): It's actually easier than that to change the description. If you run pavucontrol (assuming it has support for it which is not guaranteed), then you can just right click on a sink and select rename. The problem you get is really that both cards are being detected as being part of the the motherboard (i.e. built in). There is possibly something you can do with udev to stop this happening for one card, but I'm not sure and would have to read over the code which I can't do right this minute. Ultimately I think this is a duplicate of http://pulseaudio.org/ticket/723 It doesn't seem to affect too many people overall (I've not hear anyone mention it for ages) but it's obviously still a problem. When you just see PulseAudio Sound Server in Phonon (note that it's NOT a backend, it's listed under devices) then it means that PA itself does not have the module-device-manager loaded. Simply running start- pulseaudio-kde will reload this module. It should be run automatically on login. There is a lot more information on the http://pulseaudio.org/wiki/KDE page. -- Ticket URL: http://pulseaudio.org/ticket/863#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #863: assign different device.description when more than one sink
#863: assign different device.description when more than one sink ---+ Reporter: skierpage | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | ---+ I'm trying to get sound working on Kubuntu 10.10 (KDE Phonon Xine PulseAudio 0.9.21-63-gd3efa-dirty). My computer has two sound outputs, a built-in VT8233/A/8235/8237 AC97 Audio Controller and a Creative Labs SB Audigy PCI card; the names in quotes are their device.product_names from `pacmd list-sinks` The problem is PA gives these sinks identical device.description strings, Internal Audio Analog Stereo. These names appear in the UI of KDE's System Settings Multimedia Phonon Backend , so it's impossible to tell which is which! It would be friendlier if PA disambiguated the devices by assigning different device.descriptions. This sounds like bug #708 (unable to differentiate multiple identical soundcards), but in my case the sound cards aren't even the same. A workaround according to [wiki:Modules#Sinks] is to run `update-sink- proplist SINKNAME device.description=NEWNAME`. That works for PA but it messed up KDE System Settings which now shows only PulseAudio Sound Server as a backend -- I probably have to restart KDE. -- Ticket URL: http://pulseaudio.org/ticket/863 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #864: use GNU readline in pacmd
#864: use GNU readline in pacmd -+-- Reporter: skierpage| Owner: lennart Type: enhancement | Status: new Milestone: | Component: daemon Keywords: | -+-- I'm using Kubuntu 10.10 RC (KDE Phonon Xine PulseAudio 0.9.21-63 -gd3efa-dirty). `pacmd` is nice, but it's strange that up arrow and editing keys don't work in it, unlike most other command-line environments in Linux (ftp, gdb, python, etc.). I think the fix is to integrate GNU readline. I dunno if the separation of pacmd from the pulseaudio server and module-cli-protocol-unix makes this harder or easier. -- Ticket URL: http://pulseaudio.org/ticket/864 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #828: Flash causes PA to get into broken state; affects other apps
#828: Flash causes PA to get into broken state; affects other apps -+-- Reporter: lukehutch | Owner: lennart Type: defect | Status: closed Milestone: | Component: daemon Resolution: lackofresponse |Keywords: -+-- Changes (by lukehutch): * status: new = closed * resolution: = lackofresponse -- Ticket URL: http://pulseaudio.org/ticket/828#comment:2 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values
#862: snd_pcm_avail_delay() returned strange values +--- Reporter: oniram | Owner: lennart Type: defect | Status: new Milestone: | Component: module-alsa-* Keywords: | +--- I have a Dell Latitude E6410 and sound is working fine with pulseaudio. However, I get some error messages when I start pulseaudio: $ pulseaudio E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 24. E: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: alsa-util.c: snd_pcm_dump(): E: alsa-util.c: Soft volume PCM E: alsa-util.c: Control: PCM Playback Volume E: alsa-util.c: min_dB: -51 E: alsa-util.c: max_dB: 0 E: alsa-util.c: resolution: 256 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat: STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 16384 E: alsa-util.c: period_size : 8192 E: alsa-util.c: period_time : 185759 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min: 15502 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 4611686018427387904 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 4611686018427387904 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat: STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 16384 E: alsa-util.c: period_size : 8192 E: alsa-util.c: period_time : 185759 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min: 15502 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 4611686018427387904 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 4611686018427387904 E: alsa-util.c: appl_ptr : 15528 E: alsa-util.c: hw_ptr : 15528 As suggested by the messages I reported the bug to ALSA. Here's the bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5154 The ALSA developers seemed to suggest that I get pulseaudio developers involved ... so here we are. -- Ticket URL: http://pulseaudio.org/ticket/862 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #862: snd_pcm_avail_delay() returned strange values
#862: snd_pcm_avail_delay() returned strange values -+-- Reporter: oniram | Owner: lennart Type: defect | Status: new Milestone: | Component: module-alsa-* Resolution: |Keywords: -+-- Comment(by oniram): Let me try to fix that formatting mess: {{{ E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 24. E: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: alsa-util.c: snd_pcm_dump(): E: alsa-util.c: Soft volume PCM E: alsa-util.c: Control: PCM Playback Volume E: alsa-util.c: min_dB: -51 E: alsa-util.c: max_dB: 0 E: alsa-util.c: resolution: 256 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat: STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 16384 E: alsa-util.c: period_size : 8192 E: alsa-util.c: period_time : 185759 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min: 15502 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 4611686018427387904 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 4611686018427387904 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat: STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 16384 E: alsa-util.c: period_size : 8192 E: alsa-util.c: period_time : 185759 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min: 15502 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 4611686018427387904 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 4611686018427387904 E: alsa-util.c: appl_ptr : 15528 E: alsa-util.c: hw_ptr : 15528 }}} -- Ticket URL: http://pulseaudio.org/ticket/862#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #861: audio stops working with multiple users
#861: audio stops working with multiple users ---+ Reporter: sergei | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: stops working, multiple users | ---+ I have discovered that when I switch to another user in Mint 9 (gnome edition) the audio stops working until I reboot the computer. -- Ticket URL: http://pulseaudio.org/ticket/861 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #860: No audio through speakers on Dell M1330 Laptop
#860: No audio through speakers on Dell M1330 Laptop -+-- Reporter: jjardon | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | -+-- I have no sound in my laptop. It only works if I choose Analog Surround 4.0 in sound preferences. -- Ticket URL: http://pulseaudio.org/ticket/860 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices
#706: [PATCH] Automatically switch to new devices --+- Reporter: mterry | Owner: lennart Type: enhancement | Status: new Milestone: | Component: daemon Resolution: |Keywords: --+- Comment(by mgedmin): Two questions: * If a new network sink appears, will this module switch to it? That would be undesirable. (Use case: I'm listening to music on my laptop via headphones when someone reboots the shared office jukebox machine.) * If I switch an audio stream to a different device (say, the above- mentioned network sink), and then I switch it back to my laptop's current default device (internal audio) -- will that keep the save_device flag set and prevent it from moving to a USB headset? That would be inconvenient. -- Ticket URL: http://pulseaudio.org/ticket/706#comment:9 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #859: racy crashes
#859: racy crashes ---+ Reporter: jankratochvil | Owner: lennart Type: defect | Status: new Milestone: | Component: clients Keywords: | ---+ Goal: Five rows in Tetris: http://js1k.com.nyud.net:8080/demo/730 firefox-3.6.10-1.fc13.x86_64 pulseaudio-0.9.21-6.fc13.x86_64 It crashes in 1 to 5 seconds. You must run no debugging tools as they reduce threading raciness. Happenning on i7-920 (4 cores + multithreading = 8 cores). Finally making so many rows in Tetris now! -- Ticket URL: http://pulseaudio.org/ticket/859 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #843: Some PA strings are not translated in the Gnome Sound Properties window
#843: Some PA strings are not translated in the Gnome Sound Properties window ---+ Reporter: kelemeng | Owner: lennart Type: defect| Status: new Milestone:| Component: module-alsa-* Resolution:|Keywords: i18n ---+ Comment(by kelemeng): Ping? -- Ticket URL: http://pulseaudio.org/ticket/843#comment:7 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices
#706: [PATCH] Automatically switch to new devices --+- Reporter: mterry | Owner: lennart Type: enhancement | Status: new Milestone: | Component: daemon Resolution: |Keywords: --+- Comment(by mterry): I just updated a more recent version of the patch (against 0.9.22). This one drops the intended-roles logic, as it didn't work right (BT devices had roles set, as I commented above). It also adds logic to not switch to ISA or PCI devices in case they show up for some reason after the module loads. -- Ticket URL: http://pulseaudio.org/ticket/706#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices
#706: [PATCH] Automatically switch to new devices --+- Reporter: mterry | Owner: lennart Type: enhancement | Status: new Milestone: | Component: daemon Resolution: |Keywords: --+- Comment(by coling): Hi there, On first glance the module looks fine. It may ultimately go through some changes once I've finished some other work but as I've been particularly tardy about that I see no reason to wait for that before this goes in! Couple points re the patch itself: 1. It's (c) Canonical. Do you work for them or have you just copied it from elsewhere and forgot to update it? 2. If you want to retain authorship can you post your full name and email address (for the git commit feel free to obfuscate email address here but it'll go into the git history so will likely be listed on the web somewhere). If prefer to remain anonymous, that's fine too :) Cheers -- Ticket URL: http://pulseaudio.org/ticket/706#comment:6 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #706: [PATCH] Automatically switch to new devices
#706: [PATCH] Automatically switch to new devices --+- Reporter: mterry | Owner: lennart Type: enhancement | Status: new Milestone: | Component: daemon Resolution: |Keywords: --+- Comment(by mterry): Uh, that was at canonical.com. Trac tried to help me out there. -- Ticket URL: http://pulseaudio.org/ticket/706#comment:8 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #858: Pulseaudio uses all CPU on ALSA problems
#858: Pulseaudio uses all CPU on ALSA problems ---+ Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | ---+ Using Pulseaudio with a USB device I frequently get dmesg like: cannot submit datapipe for urb 0, error -28: not enough bandwidth after which Pulseaudio goes to 100% CPU utilization and stays there. The way to get the process to behave again is to kill it. -- Ticket URL: http://pulseaudio.org/ticket/858 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #799: libpulse use of Xlib is not thread-safe
#799: libpulse use of Xlib is not thread-safe +--- Reporter: courmisch | Owner: coling Type: defect | Status: new Milestone: | Component: module-x11-* Resolution: |Keywords: +--- Changes (by coling): * owner: lennart = coling -- Ticket URL: http://pulseaudio.org/ticket/799#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #857: Gstreamer audio glitches when playback unpaused
#857: Gstreamer audio glitches when playback unpaused +--- Reporter: ettlz | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | +--- The audio from Gstreamer-based applications (tested with Rhythmbox and Totem) does not unpause cleanly. Typically there is a glitch a fraction of a second into the unpaused audio. This is on Intel HDA audio with a ALC-883 codec. pulseaudio-0.9.21-6.fc13.x86_64 rhythmbox-0.12.8-4.fc13.x86_64 totem-2.30.2-1.fc13.x86_64 gstreamer-0.10.30-1.fc13.x86_64 gstreamer-0.10.30-1.fc13.i686 gstreamer-plugins-good-0.10.25-1.fc13.x86_64 Attached is the output of pulseaudio -. -- Ticket URL: http://pulseaudio.org/ticket/857 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free -+-- Reporter: erich | Owner: erich Type: defect | Status: new Milestone: | Component: module-rtp-* Resolution: |Keywords: latency,glitch-free -+-- Changes (by jw): * cc: devnull.pulseau...@molb.org (added) -- Ticket URL: http://pulseaudio.org/ticket/477#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free -+-- Reporter: erich | Owner: erich Type: defect | Status: new Milestone: | Component: module-rtp-* Resolution: |Keywords: latency,glitch-free -+-- Comment(by coling): jw, are you using all the patches on stable-queue git branch (which is 0.9.21 + lots of fixes)? There are some rtp related fixes there that fix header parsing that could have previously broken things. Not certainly they'll fix this issue, but all the same it's worth trying. -- Ticket URL: http://pulseaudio.org/ticket/477#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free -+-- Reporter: erich | Owner: erich Type: defect | Status: new Milestone: | Component: module-rtp-* Resolution: |Keywords: latency,glitch-free -+-- Comment(by jw): Replying to [comment:4 coling]: You mean commit 678f12d? I will check that out and report afterwards! -- Ticket URL: http://pulseaudio.org/ticket/477#comment:5 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free
#477: module-rtp-recv - rtp sink latency is random/playback distorts randomly without glitch-free -+-- Reporter: erich | Owner: erich Type: defect | Status: new Milestone: | Component: module-rtp-* Resolution: |Keywords: latency,glitch-free -+-- Comment(by jw): I updated pulseaudio (both sender and receiver), but the speed still changes pretty often and randomly (LAN connection). With vlc on the receiver side it plays normal... -- Ticket URL: http://pulseaudio.org/ticket/477#comment:6 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization) --+- Reporter: peter| Owner: lennart Type: enhancement | Status: new Milestone: | Component: clients Resolution: |Keywords: network virtual --+- Comment(by coling): @gauss: Just don't talk directly to the remote side. Always create a local PA daemon even if there is no physical h/w. Pulseaudio automatically loads a Null sink when no real hardware is present (I could easily make it do the same with sources if there were a null source module!). By always talking to the local PA daemon there is a much reduced likelihood of any problems. You can then load a tunnel sink to the remote machine and output to the right place as you desire. If there are any network issues, the sink will unload (TCP connection broken) and the auto- null sink will kick in (i.e. exactly what you suggest). -- Ticket URL: http://pulseaudio.org/ticket/194#comment:4 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization) --+- Reporter: peter| Owner: lennart Type: enhancement | Status: new Milestone: | Component: clients Resolution: |Keywords: network virtual --+- Comment(by coling): So you are suggesting that we essentially just accept that PA crashes very often and create some kind of wrapper system to deal with this? This seems fundamentally wrong: why not just fix the crashes? And I really don't think that PA itself crashes very often anyway. I've not seen any crash related tickets for a long time. If you have any back traces available, I'll be happy to take a look, but without some kind of information about the crashes you refer to, there is little we can do to help fix them. Now some kind of auto-reconnect logic is not necessarily a bad idea (I would find it useful when debugging) but this is really something up to the individual clients to implement. I've implemented such features in the various clients I've written, so really other clients need to do the same. -- Ticket URL: http://pulseaudio.org/ticket/194#comment:6 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization) --+- Reporter: peter| Owner: lennart Type: enhancement | Status: new Milestone: | Component: clients Resolution: |Keywords: network virtual --+- Comment(by gauss-gs): We just mean that pulseaudio CAN crash (of course crashes need to fix) but it need to give smoother user experience by default(if pulseaudio stated as a consumer, not а professional sound server), with allow (but not force) client developers to deal with reconnections. -- Ticket URL: http://pulseaudio.org/ticket/194#comment:7 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #856: found existing unlinked compatible pad pulsesink0:sink
#856: found existing unlinked compatible pad pulsesink0:sink --+- Reporter: ganeshpetkar | Owner: lennart Type: defect| Status: new Milestone:| Component: gst-pulse Keywords:| --+- Hi, i m using below gst pipeline gst-launch filesrc location=/home/Test.wav ! wavparse ! audioconvert ! audioresample ! pulsesink find the log message after execution 0:00:21.157652439 996 0x416050 DEBUG GST_ELEMENT_PADS gstutils.c:1671:gst_element_link_pads: looping through allowed src and dest pads 0:00:21.158231708 996 0x416050 DEBUG GST_ELEMENT_PADS gstutils.c:1674:gst_element_link_pads: trying src pad audioresample0:src 0:00:21.158841464 996 0x416050 DEBUG GST_ELEMENT_PADS gstutils.c::gst_element_get_compatible_pad: finding pad in pulsesink0 compatible with audioresample0:src 0:00:21.159481708 996 0x416050 LOG GST_ELEMENT_PADS gstutils.c:1130:gst_element_get_compatible_pad: examining pad pulsesink0:sink 0:00:21.178048781 996 0x416050 DEBUG GST_ELEMENT_PADS gstutils.c:1155:gst_element_get_compatible_pad: found existing unlinked compatible pad pulsesink0:sink 0:00:21.196920732 996 0x416050 DEBUG GST_ELEMENT_PADS gstutils.c:1688:gst_element_link_pads: linked pad audioresample0:src to pad pulsesink0:sink Setting pipeline to PAUSED ... Illegal instruction I m using pulseaudio-0.9.21 gst-plugins-good_0.10.23 gst-plugin-pulse_0.9.7 Please do the needful. Regards, ganesh -- Ticket URL: http://pulseaudio.org/ticket/856 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization)
#194: $PULSE_SERVER: permanent loss of audio in client when facing a transient network problem (networking / virtualization) --+- Reporter: peter| Owner: lennart Type: enhancement | Status: new Milestone: | Component: clients Resolution: |Keywords: network virtual --+- Changes (by gauss-gs): * cc: gauss...@gmail.com (added) Comment: I think small sound gap is VERY MUCH better than -KILL-ing and restarting freezed apps (damn skype won the 1st prize for this, second is padsp/alsa- pulse :-!) while pulseaudio crash or network disconnect happens. Maybe just make sound go /dev/null until connection is established by default and let app override this behavior if needed? This is definitely need to fix. -- Ticket URL: http://pulseaudio.org/ticket/194#comment:3 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs
#852: Pulseaudio doesn't map multiple inputs/outputs ---+ Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Keywords: | ---+ I have a soundcard with multiple inputs and outputs. {{{ cat /proc/asound/devices }}} shows {{{ 13: [ 2- 0]: digital audio playback 14: [ 2- 0]: digital audio capture 15: [ 2] : control 16: [ 2- 1]: digital audio playback 17: [ 2- 1]: digital audio capture 18: [ 2- 2]: digital audio capture }}} But in Pulseaudio only the first capture and playback get mapped. The second is not accessible via Pulseaudio. -- Ticket URL: http://pulseaudio.org/ticket/852 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #853: Illegal instruction..
#853: Illegal instruction.. --+- Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: new Milestone: 0.9.22| Component: gst-pulse Keywords:| --+- Hi, i m trying to play wav file through gstreamer and pulseaudio as a sink but it is giving error as '''Illegal instruction''' But i m not able to weather my pipe line is correct or not. Find the pipe line and log messages gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert ! audioresample ! pulsesink 0:00:20.388993903 1008 0x416050 LOG GST_ELEMENT_FACTORY gstelementfactory.c:441:gst_element_factory_make:elementfactory37 found factory 0x434d10 0:00:20.488414635 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434d10 (pulsesink) 0:00:20.490823171 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434da0 (pulsesrc) 0:00:20.492621952 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434e30 (pulsemixer) 0:00:20.493628050 1008 0x416050 INFO GST_ELEMENT_FACTORY gstelementfactory.c:363:gst_element_factory_create: creating element pulsesink 0:00:20.504664635 1008 0x416050 INFOGST_ELEMENT_PADS gstelement.c:698:gst_element_add_pad:gstbases...@0x472078 adding pad 'sink' Illegal instruction -- Ticket URL: http://pulseaudio.org/ticket/853 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #854: Illegal instruction..
#854: Illegal instruction.. --+- Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: new Milestone: 0.9.22| Component: gst-pulse Keywords:| --+- Hi, i m trying to play wav file through gstreamer and pulseaudio as a sink but it is giving error as '''Illegal instruction''' But i m not able to weather my pipe line is correct or not. Find the pipe line and log messages gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert ! audioresample ! pulsesink 0:00:20.388993903 1008 0x416050 LOG GST_ELEMENT_FACTORY gstelementfactory.c:441:gst_element_factory_make:elementfactory37 found factory 0x434d10 0:00:20.488414635 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434d10 (pulsesink) 0:00:20.490823171 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434da0 (pulsesrc) 0:00:20.492621952 1008 0x416050 DEBUGGST_ELEMENT_FACTORY gstelementfactory.c:233:gst_element_register:registry0 update existing feature 0x434e30 (pulsemixer) 0:00:20.493628050 1008 0x416050 INFO GST_ELEMENT_FACTORY gstelementfactory.c:363:gst_element_factory_create: creating element pulsesink 0:00:20.504664635 1008 0x416050 INFOGST_ELEMENT_PADS gstelement.c:698:gst_element_add_pad:gstbases...@0x472078 adding pad 'sink' Illegal instruction Please do the needful. Regards, Ganesh -- Ticket URL: http://pulseaudio.org/ticket/854 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
[pulseaudio-tickets] [PulseAudio] #855: Illegal instruction..
#855: Illegal instruction.. --+- Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: new Milestone: 0.9.22| Component: gst-pulse Keywords:| --+- Hi, i m trying to play wav file through gstreamer and pulseaudio as a sink but it is giving error as '''Illegal instruction''' But i m not able to weather my pipe line is correct or not. But if i use alsa as sink it is working fine. Find the pipe line and log messages gst-launch filesrc location=/home/test.wav ! wavparse ! audioconvert ! audioresample ! pulsesink Please do the needful. Regards, Ganesh -- Ticket URL: http://pulseaudio.org/ticket/855 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #852: Pulseaudio doesn't map multiple inputs/outputs
#852: Pulseaudio doesn't map multiple inputs/outputs +--- Reporter: LCID Fire | Owner: lennart Type: defect | Status: new Milestone: | Component: daemon Resolution: |Keywords: +--- Comment(by coling): Please attach output from {{{pacmd list}}} -- Ticket URL: http://pulseaudio.org/ticket/852#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets
Re: [pulseaudio-tickets] [PulseAudio] #854: Illegal instruction..
#854: Illegal instruction.. ---+ Reporter: ganeshpetkar | Owner: lennart Type: tracking | Status: closed Milestone: 0.9.22| Component: gst-pulse Resolution: duplicate |Keywords: ---+ Changes (by coling): * status: new = closed * resolution: = duplicate Comment: Dupe of #853 -- Ticket URL: http://pulseaudio.org/ticket/854#comment:1 PulseAudio http://pulseaudio.org/ The PulseAudio Sound Server ___ pulseaudio-tickets mailing list pulseaudio-tickets@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-tickets