2011-02-12 06:43:43     Linphone glitches

Document created by Aaronwu Employee on Aug 26, 2013
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2011-02-12 06:43:43     Linphone glitches


Message: 98113   


Hey guys,


I'm getting a problem with linphone on my blackfin board. I'm using a custom board now with blackfin and ad73311. in the middle of a few of my calls I get these pauses where it seems like playback has stopped and then started again. Seems that it might be some sort of underrun issue but i was wondering if anyone else has seen similar problems and was hoping someone could push me in the right direction.


I've disabled MMAP (under AC97 in the menu). I haven't made any changes to the distribution yet.


Arecord | aplay work fine for days.


I've run the netperf tests and my network interface is fine. I've also got the board connected directly to my desktop so network congestion shouldn't be a problem.


I ran linphone in verbose mode and paid more attention to the stats at the end of the call. I'm getting a lot of packets that have "arrived late" but would that cause this problem.


I started looking through the discussion of the following bug:


[#3597] linphone sound cutting out and in after several minutes


Can anyone help?


I'm using 2010RC5.









2011-02-13 20:46:17     Re: Linphone glitches

Aaron Wu (CHINA)

Message: 98123   


If you have time, you may try to run Linphone on two PCs In your network environment, just to narrow down the problem, then we can have idea where the problem is. Network, Linphone on blackfin or Linphone itself.




2011-02-13 22:38:13     Re: Linphone glitches

Sonic Zhang (CHINA)

Message: 98124   


Which version of Linphone do you run on your host PC? You'd better run the same version as that in blackfin Linux 2010R1.




2011-02-14 11:38:42     Re: Linphone glitches

Steve Strobel (UNITED STATES)

Message: 98158   


We have seen sample rate differences cause audio glitches.  In other words, when you ask for 8KHz you don't always get exactly 8KHz.  On some PCs, starting audio playback at 44.1KHz such as by playing an MP3 or starting Pandora before starting the VoIP software (softphone, etc) seems to help;  apparently the software sample-rate adjustment more accurately hits the 8KHz sampe rate than if you just let the VoIP software try to set the soundcard directly to 8KHz.


A quick way to check for sample rate differences is to listen to the output of a tone generator directly while also feeding it to your Blackfin to be sent via VoIP to the PC.  Turn up the audio on the PC and compare the tone pitch.  If the frequencies are close, you may be able to count the beats.  If they are a long way off, they will sound seriously out of tune.