RTP AAC timestamps issue

Question

I'm trying to stream AAC audio to Wowza server via RTP. I managed to make it work but I hear alternately the sound very fast and then a 1s blank. Sample rate is 22050 and frames per packet is 1024.

Currently my timestamps are generated like that :

long _timestamp;

//--- called each time I have to send a new AAC packet ---//
-(void) updateTimestamp{
    if(! _timestamp)
        _timestamp = random();
    else
        _timestamp += 1024;
    rtptime = big_endian(_timestamp);
}

is it correct ? I don't know why the output is that weird ans any help would be apreciated :)

Answer 1

是的，逻辑是正确的，但是使用unsigned long代替了long，因此可以在unsigned long达到最大值时舍入。