Thanks for the comments. I do know that the network connection will be a factor in unrealiability of FT8 decoding, however, I've measured the latency to be very low, around 100-200ms all the time. The audio codec being used is Opus with 64 kbps wide-band encoding, which is definitely enough for all narrow-band digital modes.
An additional reason to why I'm being optimistic is that this setup works very reliably in Windows with VB Audio Virtual Cable and I also usually get 100% successful decoding when playing through speakers and using microphone as audio input -> and I assume going through sound hardware introduces even more artifacts and higher latency than loopback internally in PulseAudio. So there is definitely something weird going on with loopback audio and I can't figure out what. I've tried to disable sample rate conversion and remixing in PulseAudio daemon.conf (and restart PulseAudio after that, of course), but neither would help. So considering these points, I would say that this is _certainly_ not about the time offsets or about the quality of the audio received by the browser. -Mikael On Thu, Nov 8, 2018, at 13:17, Bill Somerville wrote: > On 08/11/2018 09:01, Mikael Nousiainen wrote: > > 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. > > Hi Mikael, > > you are being very optimistic to expect audio delivered from a remote > web SDR receiver will decode reliably in WSJT-X. There are several > factors that are likely to contribute to unreliability. Firstly as you > imply latency in the connection may exceed the allowable time offset for > the mode being decoded. The web link likely uses aggressive and variable > audio compression that will leave artefacts after expansion. Glitches > will occur when latency changes and the stream is re-aligned. It may > also be as simple as rate conversions being applied through the > pulseaudio loopback hookup (the null sinks) that do not respond well in > the presence of the above distortions. > > 73 > Bill > G4WJS. > > > > _______________________________________________ > wsjt-devel mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/wsjt-devel _______________________________________________ wsjt-devel mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/wsjt-devel
