Dante/AES67-RTP-Stream to SBC: Latency

Nicolas Dufresne nicolas at ndufresne.ca
Tue Feb 18 22:45:54 UTC 2020

Le mar. 18 févr. 2020 11 h 30, cvxdev <cvxdev at gmail.com> a écrit :

> Nicolas Dufresne-5 wrote
> > Alsasink has 200ms latency as default configuration. See buffer-time
> > property.
> Awesome - thanks.
> That improves things a lot. By testing it seems like 20000 microseconds is
> the lower limit, if i put 19000 no sound is played anymore.
> I let it run with 20000 microseconds for a while, after ca. 10min I get
> some
> noise in the audio. I guess the only way to improve that is to increase the
> buffer - or is there any "experience" value for stable playback? (e.g.
> 40000
> microseconds)?

Ideally you want to tweak all together latency-time (size of writes to the
audio card) buffer-time (the buffer accumulation before starting audio, the
added pipeline latency) but also the thresholds. This all play together,
with 20ms, you only have 2 writes buffering to the audio card. It makes it
a little fragile. Ideally I try to maintain 3 or 4 units of buffer.

The lower the latency in GStreamer, the higher the CPU, be aware. You may
also move the ring buffer thread onto deadline scheduler, that will also
help. If that still not enough, best is to move to jack or pipewire audio

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