[FFmpeg-devel] rtp streaming x264+audio issues (and some ideas to fix them)
Tue Feb 9 14:14:10 CET 2010
Timo Ter?s wrote:
>> I had a look at the patch and at the current code. After thinking a
>> bit about it, I think the patch is good and should be committed. After
>> requiring that rtp_write_header() calls for the various streams are
>> in a short time seems reasonable. And should (IMHO) work in 99% of the
> Seems like I got a nice thread started :) ... But yes, it was intended
> to show what is broke. And be just a quick fix for the common case; well
> at least make it a whole lot better than what it was.
> Sending the rtp streams 1-2 secs apart can be problematic for some
> clients, but that's another issue.
Yes, with this solution the buffering is on client side (and some
clients do not like it).
If you want to avoid large buffering in the client, the buffering
must be in the streamer. This can be done at muxer level, by using
something like Martin's RTSP muxer (it does not implement buffering
right now, but this feature can be implemented later), or at
application level (by using avstreamer instead of ffmpeg).
>> 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(),
> I guess it should look something like:
Yes, this is what I queued up locally. It will be in svn soon.
More information about the ffmpeg-devel