Hi Lars,
14.2 is pretty old, current version is 16.1 and we are shortly before
releasing 17.0.
So if possible, I would recommend to update to a newer version, at least
if your
problems return.
Concerning documentation - you are right, the module page has not been
updated
for quite some time and unfortunately there is no newer documentation.
You can
use "pacmd describe-module <module-name>" to get a list and a short
description
of supported module arguments. The arguments mentioned on the module page
should still all work, we try not to break existing configurations, but
meanwhile
there may be quite a few new arguments that have been added over time.
If you set the log level to 4 you should see quite a bit of output, for
example
module-loopback should tell you every adjust_time seconds the current
source,
sink and buffer latency. Also module-rtp-recv and module-loopback should
both
log the current sample rate they are using. If you did not see any of
that, something
was wrong with the logging. In 14.2, the log_interval parameter to
module-loopback
should not be necessary.
Regards
Georg
On 26.03.23 19:43, Lars Stollenwerk wrote:
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
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