<html> <head> <base href="https://bugs.freedesktop.org/" /> </head> <body> <div> <a class="bz_bug_link bz_status_NEW " title="NEW --- - Very bad static following audio input." href="https://bugs.freedesktop.org/show_bug.cgi?id=76328#c3">Comment # 3</a> on <a class="bz_bug_link bz_status_NEW " title="NEW --- - Very bad static following audio input." href="https://bugs.freedesktop.org/show_bug.cgi?id=76328">bug 76328</a> from <a class="email" href="mailto:dhv2712@gmail.com" title="Dhaval <dhv2712@gmail.com>"> Dhaval</a> <pre>(In reply to <a href="show_bug.cgi?id=76328#c2">comment #2</a>) > if you want low latency for real-time voice/video chat, there is no point to > use a two seconds of capture and playback buffer > > sleeping for 1.98 seconds and wake up to capture > > > : [alsa-source-VT1708S Analog] alsa-util.c: Set buffer size first (to 88192 > samples), period size second (to 44096 samples). > I: [alsa-source-VT1708S Analog] alsa-util.c: ALSA period wakeups disabled > D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0 > D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87650 > D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=0 > D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=87310 > I: [alsa-source-VT1708S Analog] alsa-source.c: Time scheduling watermark is > 20.00ms > I: [alsa-source-VT1708S Analog] alsa-source.c: Resumed successfully... > I: [alsa-source-VT1708S Analog] alsa-source.c: Starting capture. > D: [pulseaudio] module-suspend-on-idle.c: Source > alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds. > D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change > event. > I: [pulseaudio] source-output.c: Rate changed to 44100 Hz > D: [pulseaudio] module-suspend-on-idle.c: Source > alsa_input.pci-0000_00_14.2.analog-stereo becomes busy, resuming. > D: [pulseaudio] resampler.c: Channel matrix: > D: [pulseaudio] resampler.c: I00 I01 > D: [pulseaudio] resampler.c: +------------ > D: [pulseaudio] resampler.c: O00 | 0.500 0.500 > I: [pulseaudio] remap.c: Using generic matrix remapping > I: [pulseaudio] resampler.c: Using resampler 'speex-float-1' > I: [pulseaudio] resampler.c: Using float32le as working format. > I: [pulseaudio] resampler.c: Choosing speex quality setting 1. > D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, > tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 > D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, > tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 > I: [pulseaudio] source-output.c: Created output 11 "ALSA Capture" on > alsa_input.pci-0000_00_14.2.analog-stereo with sample spec float32le 1ch > 88200Hz and channel map mono > I: [pulseaudio] source-output.c: media.name = "ALSA Capture" > I: [pulseaudio] source-output.c: application.name = "ALSA plug-in > [audacity]" > I: [pulseaudio] source-output.c: native-protocol.peer = "UNIX socket > client" > I: [pulseaudio] source-output.c: native-protocol.version = "29" > I: [pulseaudio] source-output.c: application.process.id = "18069" > I: [pulseaudio] source-output.c: application.process.user = "dhaval" > I: [pulseaudio] source-output.c: application.process.host = "dhaval" > I: [pulseaudio] source-output.c: application.process.binary = "audacity" > I: [pulseaudio] source-output.c: application.language = "en_US.UTF-8" > I: [pulseaudio] source-output.c: window.x11.display = ":0" > I: [pulseaudio] source-output.c: application.process.machine_id = > "fbb0c4896a9c0f680475c0a07edd2c5a" > I: [pulseaudio] source-output.c: application.process.session_id = "1" > I: [pulseaudio] source-output.c: application.icon_name = "audacity" > I: [pulseaudio] source-output.c: module-stream-restore.id = > "source-output-by-application-name:ALSA plug-in [audacity]" > D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, > tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0 > D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, > tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0 > I: [pulseaudio] protocol-native.c: Final latency 23.22 ms = 11.61 ms + 11.61 > ms > D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms > D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724 > D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257 > D: [alsa-source-VT1708S Analog] alsa-source.c: latency set to 11.61ms > D: [alsa-source-VT1708S Analog] alsa-source.c: hwbuf_unused=350724 > D: [alsa-source-VT1708S Analog] alsa-source.c: setting avail_min=257 So how exactly do I change it? I mean is there a configuration file or do I have to recompile pulseaudio with certain options? I'm sorry but I'm not as familiar with pulseaudio as I should be.</pre> </div> <hr> You are receiving this mail because: <ul> <li>You are the QA Contact for the bug.</li> <li>You are the assignee for the bug.</li> </ul> </body> </html>