Hello,
I'm trying to work out how to control latency with pulseaudio CLI scripts.
We're finding that latency varies between a few seconds to about 80 seconds.
We have a system which uses a dedicated embedded board for many channels
of audio I/O. Workstations
connect with the I/O board using RTP over a network.
Pulseaudio 8.0 is used on both I/O board and workstation platforms, both
using pulseaudio CLI scripts.
Modules explicitly loaded include instances of module-rtp-send,
module-rtp-recv, module-null-sink,
module-remap-source, module-remap-sink and module-loopback.
This all works as far as it goes, but with VoIP (using 1 channel in each
direction), we're finding
that the latencies make it pretty much unusable. Ideally, I need to be
able to put a reasonable
upper limit on total latency.
The link
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/Clients/LatencyControl/
provides instructions for use with the API, but I can't find much about
controlling latency with CLI.
A few modules appear to have latency-related parameters I can tweak, but
this seems to be pointless
because other modules are adding latency that I haven't worked out how
to control.
Is there any way to do this?
Steve
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