| /* |
| * 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 */ |