[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 Dec 7, 2010, at 7:53 AM, Martin Storsj? <martin at martin.st> wrote:
> 
> > 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.
> 
> Agreed, hence "OK for now".

Applied, thanks.

// Martin