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| author | Romain Degez <romain.degez@smartjog.com> | 2005-05-26 07:47:51 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2005-05-26 07:47:51 +0000 |
| commit | d1ccf0e0a66bf9b09d86d799ca3b3616a14d2427 (patch) | |
| tree | b7163163fc4fff788efc969c1bce3811345c0037 /libavformat/rtsp.h | |
| parent | 3072f0cb2ede227b803eee3b3e7d325991192858 (diff) | |
| download | ffmpeg-d1ccf0e0a66bf9b09d86d799ca3b3616a14d2427.tar.gz | |
RTP/RTSP and MPEG4-AAC audio
- preliminary support for mpeg4-aac rtp payload (no interleaving support)
- use udp transport as default (makes more sense with rtp, doesn't it ?)
- some code factorization, so adding support for new rtp payload will be easier
(I hope ;-)
patch by (Romain DEGEZ: romain degez, smartjog com)
Originally committed as revision 4306 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtsp.h')
| -rw-r--r-- | libavformat/rtsp.h | 4 |
1 files changed, 4 insertions, 0 deletions
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 3a2713f655..6c2c5efd52 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -35,6 +35,10 @@ enum RTSPProtocol { #define RTSP_DEFAULT_PORT 554 #define RTSP_MAX_TRANSPORTS 8 #define RTSP_TCP_MAX_PACKET_SIZE 1472 +#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2 +#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 +#define RTSP_RTP_PORT_MIN 5000 +#define RTSP_RTP_PORT_MAX 10000 typedef struct RTSPTransportField { int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ |