回复： RE: [Speex-dev] Bitrate problem

[wu xudong]

We use speex-vad to solve this problem.That's means we
only send voice packets at send part.Though it can't
really solve the sampling rate distortion.I have
tested different sample frequency and the result data
rate is different.


--- Shantanu Gramsci <shantanu_khan at yahoo.com>写道:

> 
> Thanks all for your prompt reply.
> 
> I agree with Alex. I think jitter buffer will have
> to
> drop packets if its input bitrate is higher than it
> is
> configured for.
> 
> I never tried speex jitter buffer. What does it do
> when input bitrate is higher or lower ?
> 
> As an alternate solution, I was thinking to play
> with
> directsound playrate [using SetFrequency()]. Has
> anyone used it to solve such problem?
> 
> Any comment is appreciated.
> 
> -- Shantanu
> 
> 
> --- Alex Bakaev <Alex.Bakaev at citrix.com> wrote:
> 
> > But that's what a jitter buffer will do,
> eventually
> > -
> > drop a packet. And it's better to drop a silence
> > packet.
> > 
> > Cheers,
> > .a 
> > 
> > -----Original Message-----
> > From: Jean-Marc Valin
> > [mailto:jean-marc.valin at usherbrooke.ca] 
> > Sent: Wednesday, December 21, 2005 3:39 PM
> > To: Alex Bakaev
> > Cc: Shantanu Gramsci; speex-dev at xiph.org
> > Subject: RE: [Speex-dev] Bitrate problem
> > 
> > On Wed, 2005-12-21 at 15:29 -0800, Alex Bakaev
> > wrote:
> > > This is a common problem. Sound cards run at
> > different
> > clock speeds 
> > > (but your difference seems kinda high).
> > > What I'd suggest is dropping silence packets
> once
> > in a
> > while.
> > 
> > Or just use a decent jitter buffer!
> > 
> > 	Jean-Marc
> > 
> > > .a
> > > 
> > > -----Original Message-----
> > > From: speex-dev-bounces at xiph.org
> > > [mailto:speex-dev-bounces at xiph.org] On Behalf Of
> > Shantanu Gramsci
> > > Sent: Wednesday, December 21, 2005 2:05 PM
> > > To: speex-dev at xiph.org
> > > Subject: [Speex-dev] Bitrate problem
> > > 
> > > Hello all,
> > > 
> > > This is a bit off topic but I am sure many of
> you
> > can
> > answer it.
> > > 
> > > I use speex and speex-wb in my VOIP application.
> > With
> > speex [at 8 KHz] 
> > > bitrate is arround 30KBPS, including UDP,RTP
> > headers.
> > This is OK for 
> > > me and I get good quality audio.
> > > 
> > > Problem is, if call is made from a HIGH END
> > machine
> > [like
> > > P4 3.2 GHz with good soundcard] to a LOW END
> > machine
> > [like P3 550 
> > > MHz], high end pc sends at more bitrate than as
> > usual
> > [arround 
> > > 31Kbps].And apparently low end PC receives this
> > HIGH
> > bitrate.
> > > 
> > > So, slow end PC receives audio packets at
> arround
> > 8100
> > Hz but it is 
> > > confiured for 8000 Hz. So there are overflows in
> > the
> > directsound 
> > > buffer at receiver end at periodic interval [in
> my
> > case
> > after every 30 
> > > sec], than I flush the buffer which makes some
> > distortion.
> > > 
> > > What should be approapriate solution for this
> > resynch
> > problem?
> > > 
> > > 
> > > Thanks
> > > 
> > > Shantanu
> > > 
> > > 
> > >
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> > 
> 
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