[FFmpeg-devel] WunderRadio changes

[Martin Storsjö]

On Tue, 20 Sep 2011, Michael Niedermayer wrote:

> the patch below (on ffmpeg-dev from you) still applies to rtspdec.c)
> is this patch still correct (in which case id like to apply it to
> ffmpeg)
> ?
>
>
> On Mon, Aug 23, 2010 at 04:50:27PM +0300, Martin Storsjö wrote:
>> On Mon, 23 Aug 2010, Diego Biurrun wrote:
>>
>>> I just had a 5 minute look, so I don't know any details, but there
>>> are some changes in there that we might wish to pick up: RTSP stuff
>>> and some ARM libswscale improvements.
>>
>> A quick check of their RTSP changes:
>> - Hardcoded to use TCP as lower transport
>> - Send keep-alive OPTIONS regularly (which we added in trunk a few weeks
>> ago)
>> - Require TCP-interleaved RTP packets to be at least 11 bytes, instead of
>> 12. (A minimal RTP packet is 12 bytes, but a minimal RTCP packet can be
>> much smaller, at least as small as 8 bytes.) This could be adjusted with
>> something like this:
>>
>> diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
>> index 36fe753..8c4c29b 100644
>> --- a/libavformat/rtsp.c
>> +++ b/libavformat/rtsp.c
>> @@ -1735,7 +1735,7 @@ redo:
>>  #ifdef DEBUG_RTP_TCP
>>      dprintf(s, "id=%d len=%d\n", id, len);
>>  #endif
>> -    if (len > buf_size || len < 12)
>> +    if (len > buf_size || len < 8)
>>          goto redo;
>>      /* get the data */
>>      ret = url_read_complete(rt->rtsp_hd, buf, len);
>>

It's as usable now as it was then - that is, it might fix some odd corner 
case - I haven't heard about any such concrete case, but I don't think it 
will hurt anything in practice to do this change.

// Martin