[FFmpeg-devel] [PATCH] Speex RTP support

[Dmitry Samonenko]

Hello, Michael.

> > +static int speex_parse_sdp_line(AVFormatContext *s, int st_index,
> > +                               PayloadContext *data, const char *line)
> > +{
> > +    av_log(NULL, AV_LOG_WARNING, "fmtp line parsing is not implemented yet\n");
> 
> something else than NULL should be used as context

I am kinda new to ffmpeg internals, have scratched only the surface and
missed logging subsystem.
I hope this will do:

-    av_log(NULL, AV_LOG_WARNING, "fmtp line parsing is not implemented yet\n");
+    av_log(s, AV_LOG_WARNING, "fmtp line parsing is not implemented yet\n");

> and do you have (a) testcase(s) for these patches ?

I've tested it using 2 ffmpeg processes: 
* the first one was generating RTP stream;
* the other one was receiving and dumping it to raw audio

Like this:
* ffmpeg -re -i ./audio.ogg -acodec libspeex -ar 16K -ac 1 -f rtp rtp://127.0.0.1:12345
* ffmpeg -i ./speex.sdp -acodec pcm_s16le -f s16le - | aplay - -t raw -f S16_LE -r 16000

According to the test it works flawlessly.

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