[FFmpeg-devel] WunderRadio changes

Ronald S. Bultje rsbultje
Mon Aug 23 16:36:28 CEST 2010


Hi,

On Mon, Aug 23, 2010 at 9:50 AM, Martin Storsj? <martin at martin.st> wrote:
> On Mon, 23 Aug 2010, Diego Biurrun wrote:
>
>> I just had a 5 minute look, so I don't know any details, but there
>> are some changes in there that we might wish to pick up: RTSP stuff
>> and some ARM libswscale improvements.
>
> A quick check of their RTSP changes:
> - Hardcoded to use TCP as lower transport
> - Send keep-alive OPTIONS regularly (which we added in trunk a few weeks
> ago)
> - Require TCP-interleaved RTP packets to be at least 11 bytes, instead of
> 12. (A minimal RTP packet is 12 bytes, but a minimal RTCP packet can be
> much smaller, at least as small as 8 bytes.) This could be adjusted with
> something like this:
>
> diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
> index 36fe753..8c4c29b 100644
> --- a/libavformat/rtsp.c
> +++ b/libavformat/rtsp.c
> @@ -1735,7 +1735,7 @@ redo:
> ?#ifdef DEBUG_RTP_TCP
> ? ? dprintf(s, "id=%d len=%d\n", id, len);
> ?#endif
> - ? ?if (len > buf_size || len < 12)
> + ? ?if (len > buf_size || len < 8)
> ? ? ? ? goto redo;
> ? ? /* get the data */
> ? ? ret = url_read_complete(rt->rtsp_hd, buf, len);
>
>
> Luca B, Ronald, any opinions on adjusting this?

Only if we actually do something with the RTCP packet. Otherwise
ignoring is better.

Ronald



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