[Speex-dev] How does the jitter buffer "catch up"?
baldvin at rogg.is
Sun Sep 18 07:43:09 PDT 2005
Is is possible to give a short hint about how the jitter buffer would
"catch up" when network condition have been bad and then get better?
I'm using the jitter buffer with success now, but sometimes I have a
long delay that's caused by bad network conditions and then later when
the conditions get better, I would think we would want the audio to
gradually catch up with real-time to minimize the latency in the voice?
Is it not realistic to expect the jitter buffer to do this sort of
"catching up" (of course doing so by "skipping" some of the older
received audio I guess)?
I understand the basic idea of the jitter.c code but am aparently not
bright enough to get the whole point of the short- and long-term margin
values etc. Just wonder if it's possible to get a short description of
each of these variables, their purpose and how they apply to the whole
jitter buffer functionality?
Thank you very much.
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