[pulseaudio-discuss] RTP makes high tone pitch

[Lars Stollenwerk]

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 at Opppf.de
> >    O-O-@
> >    / | \    PGP Key  : 0x6E5651C4
> >   FORMICA   Homepage : http://www.Opppf.de/
> 
-- 
       /   ePistulae: formica at Opppf.de
  O-O-@
  / | \    PGP Key  : 0x6E5651C4
 FORMICA   Homepage : http://www.Opppf.de/