/* * 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 #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]; /** The "timeout" comes as part of the server response to the "SETUP" * command, in the "Session: [;timeout=]" line. It is the * time, in seconds, that the server will go without traffic over the * RTSP/TCP connection before it closes the connection. To prevent * this, sent dummy requests (e.g. OPTIONS) with intervals smaller * than this value. */ int timeout; } 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. * * @todo Use ByteIOContext instead of URLContext */ 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]; /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that * the server will go without traffic on the RTSP/TCP line before it * closes the connection. */ int timeout; /** timestamp of the last RTSP command that we sent to the RTSP server. * This is used to calculate when to send dummy commands to keep the * connection alive, in conjunction with timeout. */ int64_t last_cmd_time; /** 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]; //@} /** The following are used for RTP/ASF streams */ //@{ /** ASF demuxer context for the embedded ASF stream from WMS servers */ AVFormatContext *asf_ctx; //@} } 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 */