[FFmpeg-devel] Realmedia RTSP (RDT) support

[Ronald S. Bultje]

Hi,

On 6/13/07, Benjamin Larsson <banan at ludd.ltu.se> wrote:
>
> Dump a stream from a real server with mplayer -dumpstream. Then run
> ffmpeg -i file -f image2 -acodec copy. This will dump all the frames
> passed to the decoder to files. Then do the same but on the rtsp server.
> Then you should see if the decoder gets the same data from file compared
> to the rtsp server.


Beautiful, it works now. Patch attached.

Caveats, for now:
- I changed default from tcp/udp, I should probably make that dynamically
selectable (right now it always uses udp, even if the sdp says tcp)
- ap->initial_pause is not working for real, since you need to actually
parse frames for av_find_stream_info() to complete, and if you pause the
stream, data never comes in (don't forget that ffplay calls
av_find_stream_info before starting the rtsp stream), so I removed this for
now (if (0)...)
- not sure about the licensing of the checksum calculation code (it's from
librtsp == gpl)
- bitrate selection isn't working yet
- I only have AAC working so far. Cook doesn't work yet. Frames are
reshuffled (and I think it happens correctly), also things like frame sizes
are correct, but the final byte output (as compared through diff -q) is not
the same - still working on this.

If people like this, I'd like help with the checksum calculation code
relicensing, it's only a few lines I'll try to make bitrate selection
working as suggested for mms before, and I'll see if I can get cook-radio
working, and the udp/tcp selection issue. If anyone wants to help with any
of this, please do.

Ronald
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