mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-11 17:55:21 +00:00
8a8754d80f
one doesn't work, we can try the next one (i.e. trial-error protocol auto- probing). Discussed and approved in "[PATCH] RTSP alternate protocol 2-3/3". Originally committed as revision 12504 to svn://svn.ffmpeg.org/ffmpeg/trunk
98 lines
3.0 KiB
C
98 lines
3.0 KiB
C
/*
|
|
* RTSP definitions
|
|
* Copyright (c) 2002 Fabrice Bellard.
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg 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 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg 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 FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
#ifndef FFMPEG_RTSP_H
|
|
#define FFMPEG_RTSP_H
|
|
|
|
#include <stdint.h>
|
|
#include "avformat.h"
|
|
#include "rtspcodes.h"
|
|
|
|
enum RTSPProtocol {
|
|
RTSP_PROTOCOL_RTP_UDP = 0,
|
|
RTSP_PROTOCOL_RTP_TCP = 1,
|
|
RTSP_PROTOCOL_RTP_UDP_MULTICAST = 2,
|
|
/**
|
|
* This is not part of public API and shouldn't be used outside of ffmpeg.
|
|
*/
|
|
RTSP_PROTOCOL_RTP_LAST
|
|
};
|
|
|
|
#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);
|
|
|
|
int rtsp_init(void);
|
|
void rtsp_parse_line(RTSPHeader *reply, const char *buf);
|
|
|
|
#if LIBAVFORMAT_VERSION_INT < (53 << 16)
|
|
extern int rtsp_default_protocols;
|
|
#endif
|
|
extern int rtsp_rtp_port_min;
|
|
extern int rtsp_rtp_port_max;
|
|
|
|
int rtsp_pause(AVFormatContext *s);
|
|
int rtsp_resume(AVFormatContext *s);
|
|
|
|
#endif /* FFMPEG_RTSP_H */
|