[FFmpeg-devel] rtp streaming x264+audio issues (and some ideas to fix them)

[Luca Abeni]

Martin Storsj? wrote:
[...]
>> +    s->first_rtcp_ntp_time = ntp_time();
>>
>>     max_packet_size = url_fget_max_packet_size(s1->pb);
>>     if (max_packet_size <= 12)
>>
>> What would be the proper way to fix the synchronization issue?
>> So that the RTP packets sent out are approximately (<1s difference)
>> from the original stream.
> 
> I've also been looking at this same issue, and this is IMO the first step 
> towards getting it right. This way, the sync error between the two streams 
> is reduced to the duration between the calls to rtp_write_header

I had a look at the patch and at the current code. After thinking a little
bit about it, I think the patch is good and should be committed. After all,
requiring that rtp_write_header() calls for the various streams are happen
in a short time seems reasonable. And should (IMHO) work in 99% of the
cases.

So, I'll commit this patch (well, not really this patch... But something
similar: if first_rtcp_ntp_time is initialised in rtp_write_header(),
then rtcp_send_sr() can be simplified) soon, unless anyone disagrees.
Martin, can you see problems with requiring that the calls to
rtp_write_header() happen in a short time? Does this requirement create
problems to your RTSP muxer?



				Luca