mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-24 00:02:52 +00:00
a9e534d561
structure is meant to represent. See "[PATCH] rtsp.[ch]: RTSPHeader -> RTSPServerResponse" and "[PATCH] document rtsp.h" threads on ML. Originally committed as revision 17504 to svn://svn.ffmpeg.org/ffmpeg/trunk
141 lines
4.7 KiB
C
141 lines
4.7 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"
|
|
#include "rtpdec.h"
|
|
#include "network.h"
|
|
|
|
enum RTSPLowerTransport {
|
|
RTSP_LOWER_TRANSPORT_UDP = 0,
|
|
RTSP_LOWER_TRANSPORT_TCP = 1,
|
|
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
|
|
RTSP_LOWER_TRANSPORT_NB
|
|
};
|
|
|
|
enum RTSPTransport {
|
|
RTSP_TRANSPORT_RTP,
|
|
RTSP_TRANSPORT_RDT,
|
|
RTSP_TRANSPORT_NB
|
|
};
|
|
|
|
#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 RTSPTransport transport;
|
|
enum RTSPLowerTransport lower_transport;
|
|
} RTSPTransportField;
|
|
|
|
typedef struct RTSPMessageHeader {
|
|
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];
|
|
char real_challenge[64]; /**< the RealChallenge1 field from the server */
|
|
char server[64];
|
|
} RTSPMessageHeader;
|
|
|
|
enum RTSPClientState {
|
|
RTSP_STATE_IDLE,
|
|
RTSP_STATE_PLAYING,
|
|
RTSP_STATE_PAUSED,
|
|
};
|
|
|
|
enum RTSPServerType {
|
|
RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
|
|
RTSP_SERVER_REAL, /**< Realmedia-style server */
|
|
RTSP_SERVER_WMS, /**< Windows Media server */
|
|
RTSP_SERVER_NB
|
|
};
|
|
|
|
typedef struct RTSPState {
|
|
URLContext *rtsp_hd; /* RTSP TCP connexion handle */
|
|
int nb_rtsp_streams;
|
|
struct RTSPStream **rtsp_streams;
|
|
|
|
enum RTSPClientState state;
|
|
int64_t seek_timestamp;
|
|
|
|
/* XXX: currently we use unbuffered input */
|
|
// ByteIOContext rtsp_gb;
|
|
int seq; /* RTSP command sequence number */
|
|
char session_id[512];
|
|
enum RTSPTransport transport;
|
|
enum RTSPLowerTransport lower_transport;
|
|
enum RTSPServerType server_type;
|
|
char last_reply[2048]; /* XXX: allocate ? */
|
|
void *cur_transport_priv;
|
|
int need_subscription;
|
|
enum AVDiscard real_setup_cache[MAX_STREAMS];
|
|
char last_subscription[1024];
|
|
} RTSPState;
|
|
|
|
typedef struct RTSPStream {
|
|
URLContext *rtp_handle; /* RTP stream handle */
|
|
void *transport_priv; /* RTP/RDT parse context */
|
|
|
|
int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
|
|
int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
|
|
char control_url[1024]; /* url for this stream (from SDP) */
|
|
|
|
int sdp_port; /* port (from SDP content - not used in RTSP) */
|
|
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
|
|
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
|
|
int sdp_payload_type; /* payload type - only used in SDP */
|
|
RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
|
|
|
|
RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
|
|
PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
|
|
} RTSPStream;
|
|
|
|
int rtsp_init(void);
|
|
void rtsp_parse_line(RTSPMessageHeader *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 */
|