ffmpeg/libavformat/rtsp.h
Romain Degez d1ccf0e0a6 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
2005-05-26 07:47:51 +00:00

97 lines
3.0 KiB
C

/*
* RTSP definitions
* Copyright (c) 2002 Fabrice Bellard.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifndef RTSP_H
#define RTSP_H
/* RTSP handling */
enum RTSPStatusCode {
#define DEF(n, c, s) c = n,
#include "rtspcodes.h"
#undef DEF
};
enum RTSPProtocol {
RTSP_PROTOCOL_RTP_UDP = 0,
RTSP_PROTOCOL_RTP_TCP = 1,
RTSP_PROTOCOL_RTP_UDP_MULTICAST = 2,
};
#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 */
int port_min, port_max; /* RTP ports */
int client_port_min, client_port_max; /* RTP ports */
int server_port_min, server_port_max; /* RTP ports */
int ttl; /* ttl value */
uint32_t destination; /* destination IP address */
enum RTSPProtocol protocol;
} RTSPTransportField;
typedef struct RTSPHeader {
int content_length;
enum RTSPStatusCode status_code; /* response code from server */
int nb_transports;
/* in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
int64_t range_start, range_end;
RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
int seq; /* sequence number */
char session_id[512];
} RTSPHeader;
/* the callback can be used to extend the connection setup/teardown step */
enum RTSPCallbackAction {
RTSP_ACTION_SERVER_SETUP,
RTSP_ACTION_SERVER_TEARDOWN,
RTSP_ACTION_CLIENT_SETUP,
RTSP_ACTION_CLIENT_TEARDOWN,
};
typedef struct RTSPActionServerSetup {
uint32_t ipaddr;
char transport_option[512];
} RTSPActionServerSetup;
typedef int FFRTSPCallback(enum RTSPCallbackAction action,
const char *session_id,
char *buf, int buf_size,
void *arg);
void rtsp_set_callback(FFRTSPCallback *rtsp_cb);
int rtsp_init(void);
void rtsp_parse_line(RTSPHeader *reply, const char *buf);
extern int rtsp_default_protocols;
extern int rtsp_rtp_port_min;
extern int rtsp_rtp_port_max;
extern FFRTSPCallback *ff_rtsp_callback;
extern AVInputFormat rtsp_demux;
int rtsp_pause(AVFormatContext *s);
int rtsp_resume(AVFormatContext *s);
#endif /* RTSP_H */