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

[Ronald S. Bultje]

Hi,

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).

Ronald