[FFmpeg-devel] [PATCH] Handle MP3ADU in RealRTSP, restructure rtpdec/rtsp handling of AVStream time_base

[Martin Storsjö]

On Tue, 7 Dec 2010, Ronald S. Bultje wrote:

> On Tue, Dec 7, 2010 at 5:29 AM, Martin Storsj? <martin at martin.st> wrote:
> > On Mon, 6 Dec 2010, Martin Storsj? wrote:
> >> On Mon, 6 Dec 2010, Luca Barbato wrote:
> >> > On 12/5/10 12:59 PM, Martin Storsj? wrote:
> >> > > I've tested this change with quite a few different streams, and didn't see
> >> > > any regression anywhere, but please do check if you know of any weird
> >> > > stream that might break.
> >> >
> >> > It looks safe, from what I read the default if anything is present is missing
> >> > (I just woke up so I can be wrong), but that shouldn't be an issue.
> >>
> >> Yes, I don't set the default explicitly any longer, but it's set to the
> >> same, 90 kHz, in av_new_stream anyway. If it would make things better, I
> >> could add it to be explicitly set within rtsp.c after creating the
> >> streams, so that we don't rely on defaults set anywhere else.
> >
> > Ok to commit?
> [..]
> > --- a/libavformat/rtpdec.c
> > +++ b/libavformat/rtpdec.c
> > @@ -43,6 +43,12 @@
> >           'url_open_dyn_packet_buf')
> >  */
> >
> > +RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
> > +    .enc_name           = "X-MP3-draft-00",
> > +    .codec_type         = AVMEDIA_TYPE_AUDIO,
> > +    .codec_id           = CODEC_ID_MP3ADU,
> > +};
> 
> Does this have to go in rtpdec.c? I guess it's OK for now but at some
> point this needs to go in a new file (with all dynamic-but-standard
> rtp formats).

Well, creating a new file only containing this and nothing else would feel 
like overkill at the moment.

// Martin