Hi Georg, thank you for your detailed answer! Especially the hint regarding the number of samples in the latency buffer was very enlightening. Now at the weekend I had time to test your suggestions. I'm using pulseaudio version 14.2 as part of the current raspberry pi os. The parameter log_interval to the module-loopback seems not to be supported; the corresponding call of 'pactl load-module ...' returns with non-zero return code. Setting log-level to 4 did not produce any relevant log lines. Nevertheless, I increased the latency step by step to 70 ms for the module-loopback with max_latency_msec of 100. For module-rtp-recv I set the latency to 100 ms. Now we are testing the intercom with these parameters. Right now, it works since several hours without helium-voice. I am optimistic... By the way, the documentation of the modules at [1] seems not to contain all module parameters (e. g. log_interval is not mentioned). Furthermore, it is not clear to me which version the documentation belongs to. Is there another resource with the full documentation of a given version? Many thanks and best regards Lars [1] https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/ > Hi, > > you say below, that a few hundred milliseconds would be an acceptable > latency. > Why then do you set the values so low? Setting the loopback latency to > 1ms and > max_latency to 10ms is not a good idea and will probably not work as > expected. > The latency you specify for the loopback includes source and sink latency > (= source latency + sink latency + loopback buffer latency), so a > reasonable value > for you would be around 50 ms or larger when you are using an USB device. > The max_latency parameter is only intended to be used when the latency grows > too much and should not be specified unless the latency increases > significantly > over time. > > Something similar applies to module-rtp-recv. Specifying a latency of 1ms at > 8000 Hz means that you have just 8 audio frames of latency. So here I would > go for something like 100ms. > > Both, module-loopback and module-rtp-recv try to adjust to the requested > latency by changing the sample rate. This is probably what you hear. In > particular, while module-loopback stops when the sample rate is more than > 1% away from the base rate, there is no such limit for module-rtp-recv. > > To debug the situation I would run PA with debug logging and also add > the log-interval=1 parameter to module-loopback. This produces quite a > lot of output, but then you should be able to see what happens and can > adjust your parameters for optimum performance. To enable debug logging > and to send the output to a file you can use > > pacmd set-log-level 4 > pacmd set-log-target file://tmp/pulse.log > > The log settings will revert to the default values when PA is restarted. > Hope this helps. > > Regards > Georg > > On 19.03.23 16:14, Lars Stollenwerk wrote: > > Hi all, > > > > I am using PulseAudio to implement a kind of intercom: In our ham radio club > > house we want to install in every room a raspberry pi with PulseAudio running; > > this is what I call a 'client'. Each client is connecet to a microphone and a > > speaker using a USB sound card. To send and receive the audio signal over > > network, on each client a set of PulseAudio modules is started using 'pactl > > load-module ...'. The follwing modules are started: > > > > Sending path > > ------------ > > Signal path: mic-source -> loopback -> null-sink -> rtp-send > > > > module-null-sink > > sink_name=rtp-tx-sink > > rate=8000 > > channels=1 > > module-loopback > > source=<mic-source> > > sink=rtp-tx-sink > > latency_msec=1 > > max_latency_msec=10 > > fast_adjust_threshold_msec=100 > > adjust_time=1 > > rate=8000 > > channels=1 > > module-rtp-send > > rate=8000 > > channels=1 > > loop=false > > source=rtp-tx-sink.monitor > > destination=224.0.0.47 > > > > Receiving path > > -------------- > > Signal path: rtp-recv -> null-sink -> loopback -> speaker-sink > > > > module-null-sink > > sink_name=rtp-rx-sink > > rate=8000 > > channels=1 > > module-loopback > > source=rtp-rx-sink.monitor > > sink=<speaker-sink> > > latency_msec=1 > > max_latency_msec=10 > > fast_adjust_threshold_msec=100 > > adjust_time=1 > > rate=8000 > > channels=1 > > module-rtp-recv > > latency_msec=1 > > sink=rtp-rx-sink > > sap_address=224.0.0.47 > > > > The null-sinks are used to mute the stream when necessary. > > > > Generally, the set-up works. However, often the pitch of the voice is much too > > high. I. e., the voice can be well understood but sounds more or less like a > > Helium voice. > > > > The values for latency are all chosen to be very small. The reason is that we > > can accept short drop-outs, but not a big latency (few 100 ms is the maxium > > acceptable). I have tryed to increase max_latency_msec of the loopback > > devices to 60 ms - without success. > > > > My question to the community is: Does anyone know where the high pitch voice > > comes from? And how to avoid this? > > > > I am thankful for any help! > > > > Best ragards > > Lars > > -- > > / ePistulae: formica@xxxxxxxx > > O-O-@ > > / | \ PGP Key : 0x6E5651C4 > > FORMICA Homepage : http://www.Opppf.de/ > -- / ePistulae: formica@xxxxxxxx O-O-@ / | \ PGP Key : 0x6E5651C4 FORMICA Homepage : http://www.Opppf.de/
