Hello, I need help with this issue, any hint what to try or where to look. Can someone please try to reproduce this problem? When I use module-loopback to connect audio input (line-in) to a sink provided by module-tunnel-sink, I get chopped sound. It seems to me that buffer / latency isn't handled correctly and/or module-tunnel-sink does not provide necessary features (?) The sound output is missing parts and that gets worse over time. I've made following observations: - typical debug output (sending side) that mostly comes with this problem is ... D: [pulseaudio] module-loopback.c: Should buffer 0 bytes, buffered at minimum 0 bytes D: [pulseaudio] module-loopback.c: [usbol] Updated sampling rate to 44100 Hz. D: [pulseaudio] module-loopback.c: Loopback overall latency is 149.26 ms + 10.50 ms + 1.01 ms = 160.77 ms ... this is repeated in the output and the chopping gets worse with the time - debug output on the receiving side during output ... D: [alsa-sink-USB Audio] memblock.c: Pool full D: [pulseaudio] memblock.c: Pool full D: [pulseaudio] memblock.c: Pool full D: [alsa-sink-USB Audio] memblock.c: Pool full D: [alsa-sink-USB Audio] memblock.c: Pool full ... repeated ... D: [alsa-sink-USB Audio] memblock.c: Pool full D: [alsa-sink-USB Audio] ratelimit.c: 25 events suppressed D: [alsa-sink-USB Audio] alsa-sink.c: Wakeup from ALSA! D: [alsa-sink-USB Audio] alsa-sink.c: Wakeup from ALSA! D: [alsa-sink-USB Audio] flist.c: pulsecore/memblockq.c: list_items flist is full (don't worry) D: [alsa-sink-USB Audio] flist.c: pulsecore/memblockq.c: list_items flist is full (don't worry) ... repeated ... D: [alsa-sink-USB Audio] flist.c: pulsecore/memblockq.c: list_items flist is full (don't worry) D: [alsa-sink-USB Audio] ratelimit.c: 73 events suppressed D: [alsa-sink-USB Audio] flist.c: pulsecore/memblockq.c: list_items flist is full (don't worry) ... - chopping gets worse over time, and within few minutes there is no (hearable) sound output any more - the chopping depends on latency_msec passed to the module-loopback, but it is always there and gets worse over time. (Lower latency gives less (hearable) chopping at the start.) - at the same time, I can paplay or play-sample in parallel to the output from module-loopback to the same sink without problems - If I suspend the sink on the receiving side and the resume it again, then there is huge latency (the sending side seems to be unaware?), but for some time the sound output is without (AFAIC hear) distortions. module-loopback outputs (debug log) messages about sampling rate adjustment. After some time, the latency gets reduced, but then D: [pulseaudio] module-loopback.c: Should buffer 0 bytes, buffered at minimum 0 bytes appears again and the sound gets chopped. - local loopback from audio-in to audio-out is ok The setup consists of two almost identical systems with a minimal Debian testing install. I've compiled PA 4.0, sources are from Debian unstable. There is no "desktop environment" - no Gnome or similar. I've tried to run PA both as system instance and user's instance - both with root privileges - there is no difference. Examples of full debug log: Client side (sending): http://pastebin.com/xwP4Qhwa Server side (receiving): http://pastebin.com/U3h55jmN Local loopback: http://pastebin.com/S7VT8UZ7 Config (both sides): log-level=debug exit-idle-time = -1 Init (both sides): load-module module-udev-detect load-module module-native-protocol-unix auth-anonymous=1 load-module module-native-protocol-tcp auth-ip-acl=192.168.0.0/24 .ifexists module-dbus-protocol.so load-module module-dbus-protocol access=local .endif load-module module-always-sink load-module module-suspend-on-idle load-module module-cli-protocol-unix Thank you, Zrin
