I've got a very weird issue with PulseAudio when trying to route audio from one application (Firefox 64.0b7 (64-bit)) to another one (WSJT-X v1.9.1). I'm experiencing the same issue with different browsers (Chrome and Chromium too). The browser is receiving audio from a radio transceiver through a WebRTC connection and I'm feeding it to WSJT-X to decode the data in the audio signal. I'm using two module-null-sink modules to transfer the audio between the browser and WSJT-X. I use pavucontrol to make the browser play audio to null sink called "radio-output" and then let WSJT-X listen to the audio via "radio-output.monitor". The kind of setup exists for transmitted audio where WSJT-X feeds audio to the browser through a null sink called "radio-input". However, the issue her is that WSJT-X is mostly not able to decode the data when it's using "radio-output.monitor" as audio input. Sometimes it works and sometimes it does not. As a technical detail, I'm trying to decode FT8 digital mode traffic and I've confirmed that the reason is not related to bad time sync (which FT8 requires), because I can even play the browser audio through laptop speakers and let WSJT-X use the laptop microphone as audio input and it decodes the data just fine -- the audio sounds clean and strong with no audible artifacts. By looking at the waterfall display of WSJT-X when using "radio-output.monitor" as audio input, I can see some artifacts appearing randomly in the frequency domain data -- I would call this distortion, but it's hard to know as I cannot hear it. I can set up a loopback module to feed "radio-output.monitor" to the speakers and the audio sounds fine. Lowering the incoming volume radically, e.g. to 10-20% of the full volume, does not help. And occasionally, couple of times a minute -- and sometimes for a longer period of time -- these artifacts disappear and WSJT-X is able to decode all data flawlessly. Please see this screenshot for details: https://ibb.co/eiEo0A I'm using a powerful PC (a quad-core Xeon with 48GB RAM) running Fedora 28 with latest updates applied and CPU usage is quite low (20-30%) during decoding. PulseAudio version is 12.2-rebootstrapped packaged by Fedora 28. Pulseaudio daemon.conf has default settings, except for the sample rate: default-sample-format = s16le default-sample-rate = 48000 Also, I have the following null sink / loopback setup in default.pa file: # Loopback devices load-module module-null-sink sink_name=radio-output sink_properties=device.description="radio-output" load-module module-null-sink sink_name=radio-input sink_properties=device.description="radio-input" load-module module-loopback source=radio-output.monitor latency_msec=100 adjust_time=0 rate=48000 load-module module-loopback source=radio-input.monitor latency_msec=100 adjust_time=0 rate=48000 The latter loopback modules are there so that I can hear what is being fed into the null sinks, but I've confirmed the same behavior even *WITHOUT* the loopback modules. I've also set up a fixed latency range for the hardware sound cards, but I believe this should not have any effect on the null sinks or loopback modules: load-module module-udev-detect tsched=yes tsched_buffer_size=65536 fixed_latency_range=yes I am not sure if these audio artifacts are caused by PulseAudio, WSJT-X or some specific configuration issue in my system. Any ideas? _______________________________________________ pulseaudio-discuss mailing list pulseaudio-discuss@xxxxxxxxxxxxxxxxxxxxx https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
