[Speex-dev] Who is using the jitter buffer?
lis at 1234567890qwertzuiopasdfghjklyxcvbnm.de
Mon Mar 20 16:11:00 PST 2006
> That's basically my question: the timestamps at the
> source and
> destination are not related. Just incrementing by
number of samples
> doesn't really convey the real time, does it? How would
> buffer know that a packet is late/early?
> Simple, I know what packet I just played. That gives me
> the "time". The jitter buffer actually makes no
> difference (and doesn't attempt to) between an increase
> in the network delay and a drift in the soundcard clocks.
> It just increments or decrements the buffer size to make
> sure the packets spend the least amount of time in the
> buffer (before being player), while not having too many
> lost packets.
> So, basically, the sound card play out requests serve as
> a 'reference clock', right?
> Do you have plans to use silence packets to
> grow/shrinking the jitter buffer?
how about tcp?
in tcp you write a packet that got a possible length.
you send one packet after another, whitch stamp is incrementet by one
and if your incoming packet is gone in other steps than 1, the client has
to resend it.
Let me think some days about it and i will get another system.
Time is relative.
Hm, you send a packet that needs to be in a possible chain of n.
Your jitterbuffer needs to be as long as you need to make your
transmition, and you would to send only one packet that contains
all lost packets... ?!? whats the matter ?!?
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