[FFmpeg-devel] WunderRadio changes

[Michael Niedermayer]

Hi martin

the patch below (on ffmpeg-dev from you) still applies to rtspdec.c)
is this patch still correct (in which case id like to apply it to
ffmpeg)
?


On Mon, Aug 23, 2010 at 04:50:27PM +0300, Martin Storsjö 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?
> 
> // Martin
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-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Asymptotically faster algorithms should always be preferred if you have
asymptotical amounts of data
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