[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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