/*
* 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"
/**
* Network layer over which RTP/etc packet data will be transported.
*/
enum RTSPLowerTransport {
RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
RTSP_LOWER_TRANSPORT_NB
};
/**
* Packet profile of the data that we will be receiving. Real servers
* commonly send RDT (although they can sometimes send RTP as well),
* whereas most others will send RTP.
*/
enum RTSPTransport {
RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
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
/**
* This describes a single item in the "Transport:" line of one stream as
* negotiated by the SETUP RTSP command. Multiple transports are comma-
* separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
* client_port=1000-1001;server_port=1800-1801") and described in separate
* RTSPTransportFields.
*/
typedef struct RTSPTransportField {
/** interleave ids, if TCP transport; each TCP/RTSP data packet starts
* with a '$', stream length and stream ID. If the stream ID is within
* the range of this interleaved_min-max, then the packet belongs to
* this stream. */
int interleaved_min, interleaved_max;
/** UDP multicast port range; the ports to which we should connect to
* receive multicast UDP data. */
int port_min, port_max;
/** UDP client ports; these should be the local ports of the UDP RTP
* (and RTCP) sockets over which we receive RTP/RTCP data. */
int client_port_min, client_port_max;
/** UDP unicast server port range; the ports to which we should connect
* to receive unicast UDP RTP/RTCP data. */
int server_port_min, server_port_max;
/** time-to-live value (required for multicast); the amount of HOPs that
* packets will be allowed to make before being discarded. */
int ttl;
uint32_t destination; /**< destination IP address */
/** data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
enum RTSPLowerTransport lower_transport;
} RTSPTransportField;
/**
* This describes the server response to each RTSP command.
*/
typedef struct RTSPMessageHeader {
/** length of the data following this header */
int content_length;
enum RTSPStatusCode status_code; /**< response code from server */
/** number of items in the 'transports' variable below */
int nb_transports;
/** Time range of the streams that the server will stream. In
* AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
int64_t range_start, range_end;
/** describes the complete "Transport:" line of the server in response
* to a SETUP RTSP command by the client */
RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
int seq; /**< sequence number */
/** the "Session:" field. This value is initially set by the server and
* should be re-transmitted by the client in every RTSP command. */
char session_id[512];
/** the "RealChallenge1:" field from the server */
char real_challenge[64];
/** the "Server: field, which can be used to identify some special-case
* servers that are not 100% standards-compliant. We use this to identify
* Windows Media Server, which has a value "WMServer/v.e.r.sion", where
* version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
* use something like "Helix [..] Server Version v.e.r.sion (platform)
* (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
* where platform is the output of $uname -msr | sed 's/ /-/g'. */
char server[64];
} RTSPMessageHeader;
/**
* Client state, i.e. whether we are currently receiving data (PLAYING) or
* setup-but-not-receiving (PAUSED). State can be changed in applications
* by calling av_read_play/pause().
*/
enum RTSPClientState {
RTSP_STATE_IDLE, /**< not initialized */
RTSP_STATE_PLAYING, /**< initialized and receiving data */
RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
};
/**
* Identifies particular servers that require special handling, such as
* standards-incompliant "Transport:" lines in the SETUP request.
*/
enum RTSPServerType {
RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
RTSP_SERVER_REAL, /**< Realmedia-style server */
RTSP_SERVER_WMS, /**< Windows Media server */
RTSP_SERVER_NB
};
/**
* Private data for the RTSP demuxer.
*/
typedef struct RTSPState {
URLContext *rtsp_hd; /* RTSP TCP connexion handle */
/** number of items in the 'rtsp_streams' variable */
int nb_rtsp_streams;
struct RTSPStream **rtsp_streams; /**< streams in this session */
/** indicator of whether we are currently receiving data from the
* server. Basically this isn't more than a simple cache of the
* last PLAY/PAUSE command sent to the server, to make sure we don't
* send 2x the same unexpectedly or commands in the wrong state. */
enum RTSPClientState state;
/** the seek value requested when calling av_seek_frame(). This value
* is subsequently used as part of the "Range" parameter when emitting
* the RTSP PLAY command. If we are currently playing, this command is
* called instantly. If we are currently paused, this command is called
* whenever we resume playback. Either way, the value is only used once,
* see rtsp_read_play() and rtsp_read_seek(). */
int64_t seek_timestamp;
/* XXX: currently we use unbuffered input */
// ByteIOContext rtsp_gb;
int seq; /**< RTSP command sequence number */
/** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
* identifier that the client should re-transmit in each RTSP command */
char session_id[512];
/** the negotiated data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** the negotiated network layer transport protocol; e.g. TCP or UDP
* uni-/multicast */
enum RTSPLowerTransport lower_transport;
/** brand of server that we're talking to; e.g. WMS, REAL or other.
* Detected based on the value of RTSPMessageHeader->server or the presence
* of RTSPMessageHeader->real_challenge */
enum RTSPServerType server_type;
/** The last reply of the server to a RTSP command */
char last_reply[2048]; /* XXX: allocate ? */
/** RTSPStream->transport_priv of the last stream that we read a
* packet from */
void *cur_transport_priv;
/** The following are used for Real stream selection */
//@{
/** whether we need to send a "SET_PARAMETER Subscribe:" command */
int need_subscription;
/** stream setup during the last frame read. This is used to detect if
* we need to subscribe or unsubscribe to any new streams. */
enum AVDiscard real_setup_cache[MAX_STREAMS];
/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
* this is used to send the same "Unsubscribe:" if stream setup changed,
* before sending a new "Subscribe:" command. */
char last_subscription[1024];
//@}
} RTSPState;
/**
* Describes a single stream, as identified by a single m= line block in the
* SDP content. In the case of RDT, one RTSPStream can represent multiple
* AVStreams. In this case, each AVStream in this set has similar content
* (but different codec/bitrate).
*/
typedef struct RTSPStream {
URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
void *transport_priv; /**< RTP/RDT parse context */
/** corresponding stream index, if any. -1 if none (MPEG2TS case) */
int stream_index;
/** interleave IDs; copies of RTSPTransportField->interleaved_min/max
* for the selected transport. Only used for TCP. */
int interleaved_min, interleaved_max;
char control_url[1024]; /**< url for this stream (from SDP) */
/** The following are used only in SDP, not RTSP */
//@{
int sdp_port; /**< port (from SDP content) */
struct in_addr sdp_ip; /**< IP address (from SDP content) */
int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
int sdp_payload_type; /**< payload type */
//@}
/** rtp payload parsing infos from SDP (i.e. mapping between private
* payload IDs and media-types (string), so that we can derive what
* type of payload we're dealing with (and how to parse it). */
RTPPayloadData rtp_payload_data;
/** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
//@{
/** handler structure */
RTPDynamicProtocolHandler *dynamic_handler;
/** private data associated with the dynamic protocol */
PayloadContext *dynamic_protocol_context;
//@}
} 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 */