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

[Ronald S. Bultje]

Hi,

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".

Ronald