Hi,
I am working with low-latency audio streaming and found a curious behavior
using the alsasink. When I, for instance, request 40ms buffer-time and 20ms
latency-time, I do not get the expected minimum latency for the sink of 40
ms but instead one latency-time extra reported minimum latency(i.e. 60ms).
I have traced it down to this line in gstaudiosink.c (line 402) which was
added by a patch in 2008:
/* set latency to one more segment as we need some headroom */
spec->seglatency = spec->segtotal + 1;
https://github.com/GStreamer/gst-plugins-base/commit/fc523e047ceefcba5db19b94bcfd66289a409374If I remove this line I get the expected minimum latency of 40ms and audio
seems to play fine.
What is the reason for the additional latency?
Best Regards,
Danny
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