<div dir="auto"><div><br><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">Le mar. 18 févr. 2020 11 h 30, cvxdev <<a href="mailto:cvxdev@gmail.com">cvxdev@gmail.com</a>> a écrit :<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Nicolas Dufresne-5 wrote<br>
> Alsasink has 200ms latency as default configuration. See buffer-time<br>
> property.<br>
<br>
Awesome - thanks. <br>
That improves things a lot. By testing it seems like 20000 microseconds is<br>
the lower limit, if i put 19000 no sound is played anymore. <br>
I let it run with 20000 microseconds for a while, after ca. 10min I get some<br>
noise in the audio. I guess the only way to improve that is to increase the<br>
buffer - or is there any "experience" value for stable playback? (e.g. 40000<br>
microseconds)?<br></blockquote></div></div><div dir="auto"><br></div><div dir="auto">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.</div><div dir="auto"><br></div><div dir="auto">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 daemons.</div><div dir="auto"><br></div><div dir="auto"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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