[Speex-dev] speex voice seems to be bit breaking over long distance.

Tay YueWeng yueweng at yahoo.com
Mon Apr 18 08:11:36 PDT 2005


Hi,

Ok, what you suggest sound logical to me. Currently, I
have done a small trick to prevent this problem. What
I did is that whenever windows request a voice packet
from me and if I do not have the voice packet, I
repeat the previous packet. Hence, all the breaking
portion is filled with previous packet. This trick
seems to work so far. I am not sure what is the side
effect.

I think jitter buffering is more correct way to solve
this problem but I think this is not so easy. Is there
got any source code to reference from?

Regards,
YueWeng 

--- "Kleijn, J.S. de" <J.S.d.Kleijn at student.tue.nl>
wrote:
> Hi,
>  
> You should implement a jitter buffer, this buffers
> the audio to prevent the 'breaking'.
>  
> My VoIP application called VoIPerized has a jitter
> buffer. You can try it to see if you then don't
> experience the breaking of the voice.
>  
> Greetings Jeroen de Kleijn
> 
> 	-----Oorspronkelijk bericht----- 
> 	Van: speex-dev-bounces at xiph.org namens Tay YueWeng 
> 	Verzonden: ma 18-4-2005 11:28 
> 	Aan: speex-dev at xiph.org 
> 	CC: 
> 	Onderwerp: [Speex-dev] speex voice seems to be bit
> breaking over long distance.
> 	
> 	
> 
> 	Dear all,
> 	I have implemented speex. Under LAN environment,
> 	everything is working fine. However, when the
> source
> 	and destination is about 20 hrs away, with ping
> 	response time of about 800ms, the voice is
> breaking.
> 	Breaking means you can not hear a smooth voice.
> Like
> 	the voice is being "chopped" into many pieces.
> 	
> 	The amount of packet lost is less than 10%. I have
> 	tried 8KHz, 16KHz, 32KHz. The bigger it gets, the
> more
> 	lousy the quality. I believe I have enough
> bandwidth
> 	since the packet lost is less than 10%.  The only
> 	factor that could affect the quality is may be
> 	delayed.
> 	
> 	Any one have any idea of what is wrong?
> 	
> 	Regards,
> 	YueWeng
> 	
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> 

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