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/*
* 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 "rtp.h"
#include "network.h"
enum RTSPLowerTransport {
RTSP_LOWER_TRANSPORT_UDP = 0,
RTSP_LOWER_TRANSPORT_TCP = 1,
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
/**
* This is not part of public API and shouldn't be used outside of ffmpeg.
*/
RTSP_LOWER_TRANSPORT_LAST
};
enum RTSPTransport {
RTSP_TRANSPORT_RTP,
RTSP_TRANSPORT_RDT,
RTSP_TRANSPORT_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 RTSPTransport transport;
enum RTSPLowerTransport lower_transport;
} 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];
char real_challenge[64]; /**< the RealChallenge1 field from the server */
char server[64];
} RTSPHeader;
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_LAST
};
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(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 */
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