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Re: [Linphone-developers] question about rtp packets

From: Kosta Welke
Subject: Re: [Linphone-developers] question about rtp packets
Date: Wed, 22 Mar 2006 13:20:57 +0100
User-agent: Thunderbird 1.5 (Windows/20051201)

meng hongling wrote:

>> Try ignoring the buffer size.
>> In coreapi/linphonecore.c, line 1352, there is a check that calculates
>> the jitter delay:
>> - jitt_comp=MAX(lc->sound_conf.latency,lc->rtp_conf.audio_jitt_comp);
>> + jitt_comt = lc->rtp_conf.audio_jitt_comp;

> Yes,I have done this,but the chopped sound is still here.
> I think this may because I discard one channel audio samples in 
> oss_card_read(my sound card only support stereo).this consumes the time.

Hmmm, maybe there is also a problem somewhere else. Is the sound
scrambled on both sides of the call?
Also, do you mean by 'chopped' that the sound has 'holes' in it? Like
when you say "aaaaaaaaaa" into the phone, the other end plays "a-a-a-a-a"?
Also, are you making stereo sound out of your incoming mono packets?

>> Maybe your system is too slow. How big are the "Must catchup" warnings?
> en ,yes,maybe,since my developing board is embedded system
> the Must catchup is about 100 millisecond.

If it is constantly around 100 milliseconds, your system is fast enough.
If it was too slow, the value would increase during the phone call.
This seems to be a scheduling problem. Take a look at
ms_timer_synchronize() in mediastreamer/mstimer.c. Maybe do some debug
output to see how often he's in the loop. If he uses the loop at all,
select() or gettimeofday() could be inaccurate on your platform.
A quick hack could be to simply sleep for a shorter amount of time :)


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