1/* 2 * RTSP definitions 3 * Copyright (c) 2002 Fabrice Bellard 4 * 5 * This file is part of FFmpeg. 6 * 7 * FFmpeg is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Lesser General Public 9 * License as published by the Free Software Foundation; either 10 * version 2.1 of the License, or (at your option) any later version. 11 * 12 * FFmpeg is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Lesser General Public License for more details. 16 * 17 * You should have received a copy of the GNU Lesser General Public 18 * License along with FFmpeg; if not, write to the Free Software 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 20 */ 21#ifndef AVFORMAT_RTSP_H 22#define AVFORMAT_RTSP_H 23 24#include <stdint.h> 25#include "avformat.h" 26#include "rtspcodes.h" 27#include "rtpdec.h" 28#include "network.h" 29#include "httpauth.h" 30 31/** 32 * Network layer over which RTP/etc packet data will be transported. 33 */ 34enum RTSPLowerTransport { 35 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */ 36 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */ 37 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ 38 RTSP_LOWER_TRANSPORT_NB 39}; 40 41/** 42 * Packet profile of the data that we will be receiving. Real servers 43 * commonly send RDT (although they can sometimes send RTP as well), 44 * whereas most others will send RTP. 45 */ 46enum RTSPTransport { 47 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ 48 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ 49 RTSP_TRANSPORT_NB 50}; 51 52#define RTSP_DEFAULT_PORT 554 53#define RTSP_MAX_TRANSPORTS 8 54#define RTSP_TCP_MAX_PACKET_SIZE 1472 55#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2 56#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 57#define RTSP_RTP_PORT_MIN 5000 58#define RTSP_RTP_PORT_MAX 10000 59 60/** 61 * This describes a single item in the "Transport:" line of one stream as 62 * negotiated by the SETUP RTSP command. Multiple transports are comma- 63 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; 64 * client_port=1000-1001;server_port=1800-1801") and described in separate 65 * RTSPTransportFields. 66 */ 67typedef struct RTSPTransportField { 68 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts 69 * with a '$', stream length and stream ID. If the stream ID is within 70 * the range of this interleaved_min-max, then the packet belongs to 71 * this stream. */ 72 int interleaved_min, interleaved_max; 73 74 /** UDP multicast port range; the ports to which we should connect to 75 * receive multicast UDP data. */ 76 int port_min, port_max; 77 78 /** UDP client ports; these should be the local ports of the UDP RTP 79 * (and RTCP) sockets over which we receive RTP/RTCP data. */ 80 int client_port_min, client_port_max; 81 82 /** UDP unicast server port range; the ports to which we should connect 83 * to receive unicast UDP RTP/RTCP data. */ 84 int server_port_min, server_port_max; 85 86 /** time-to-live value (required for multicast); the amount of HOPs that 87 * packets will be allowed to make before being discarded. */ 88 int ttl; 89 90 uint32_t destination; /**< destination IP address */ 91 92 /** data/packet transport protocol; e.g. RTP or RDT */ 93 enum RTSPTransport transport; 94 95 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ 96 enum RTSPLowerTransport lower_transport; 97} RTSPTransportField; 98 99/** 100 * This describes the server response to each RTSP command. 101 */ 102typedef struct RTSPMessageHeader { 103 /** length of the data following this header */ 104 int content_length; 105 106 enum RTSPStatusCode status_code; /**< response code from server */ 107 108 /** number of items in the 'transports' variable below */ 109 int nb_transports; 110 111 /** Time range of the streams that the server will stream. In 112 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ 113 int64_t range_start, range_end; 114 115 /** describes the complete "Transport:" line of the server in response 116 * to a SETUP RTSP command by the client */ 117 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; 118 119 int seq; /**< sequence number */ 120 121 /** the "Session:" field. This value is initially set by the server and 122 * should be re-transmitted by the client in every RTSP command. */ 123 char session_id[512]; 124 125 /** the "Location:" field. This value is used to handle redirection. 126 */ 127 char location[4096]; 128 129 /** the "RealChallenge1:" field from the server */ 130 char real_challenge[64]; 131 132 /** the "Server: field, which can be used to identify some special-case 133 * servers that are not 100% standards-compliant. We use this to identify 134 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where 135 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers 136 * use something like "Helix [..] Server Version v.e.r.sion (platform) 137 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", 138 * where platform is the output of $uname -msr | sed 's/ /-/g'. */ 139 char server[64]; 140 141 /** The "timeout" comes as part of the server response to the "SETUP" 142 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the 143 * time, in seconds, that the server will go without traffic over the 144 * RTSP/TCP connection before it closes the connection. To prevent 145 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller 146 * than this value. */ 147 int timeout; 148 149 /** The "Notice" or "X-Notice" field value. See 150 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 151 * for a complete list of supported values. */ 152 int notice; 153} RTSPMessageHeader; 154 155/** 156 * Client state, i.e. whether we are currently receiving data (PLAYING) or 157 * setup-but-not-receiving (PAUSED). State can be changed in applications 158 * by calling av_read_play/pause(). 159 */ 160enum RTSPClientState { 161 RTSP_STATE_IDLE, /**< not initialized */ 162 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ 163 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ 164 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ 165}; 166 167/** 168 * Identifies particular servers that require special handling, such as 169 * standards-incompliant "Transport:" lines in the SETUP request. 170 */ 171enum RTSPServerType { 172 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ 173 RTSP_SERVER_REAL, /**< Realmedia-style server */ 174 RTSP_SERVER_WMS, /**< Windows Media server */ 175 RTSP_SERVER_NB 176}; 177 178/** 179 * Private data for the RTSP demuxer. 180 * 181 * @todo Use ByteIOContext instead of URLContext 182 */ 183typedef struct RTSPState { 184 URLContext *rtsp_hd; /* RTSP TCP connexion handle */ 185 186 /** number of items in the 'rtsp_streams' variable */ 187 int nb_rtsp_streams; 188 189 struct RTSPStream **rtsp_streams; /**< streams in this session */ 190 191 /** indicator of whether we are currently receiving data from the 192 * server. Basically this isn't more than a simple cache of the 193 * last PLAY/PAUSE command sent to the server, to make sure we don't 194 * send 2x the same unexpectedly or commands in the wrong state. */ 195 enum RTSPClientState state; 196 197 /** the seek value requested when calling av_seek_frame(). This value 198 * is subsequently used as part of the "Range" parameter when emitting 199 * the RTSP PLAY command. If we are currently playing, this command is 200 * called instantly. If we are currently paused, this command is called 201 * whenever we resume playback. Either way, the value is only used once, 202 * see rtsp_read_play() and rtsp_read_seek(). */ 203 int64_t seek_timestamp; 204 205 /* XXX: currently we use unbuffered input */ 206 // ByteIOContext rtsp_gb; 207 208 int seq; /**< RTSP command sequence number */ 209 210 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session 211 * identifier that the client should re-transmit in each RTSP command */ 212 char session_id[512]; 213 214 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that 215 * the server will go without traffic on the RTSP/TCP line before it 216 * closes the connection. */ 217 int timeout; 218 219 /** timestamp of the last RTSP command that we sent to the RTSP server. 220 * This is used to calculate when to send dummy commands to keep the 221 * connection alive, in conjunction with timeout. */ 222 int64_t last_cmd_time; 223 224 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ 225 enum RTSPTransport transport; 226 227 /** the negotiated network layer transport protocol; e.g. TCP or UDP 228 * uni-/multicast */ 229 enum RTSPLowerTransport lower_transport; 230 231 /** brand of server that we're talking to; e.g. WMS, REAL or other. 232 * Detected based on the value of RTSPMessageHeader->server or the presence 233 * of RTSPMessageHeader->real_challenge */ 234 enum RTSPServerType server_type; 235 236 /** plaintext authorization line (username:password) */ 237 char auth[128]; 238 239 /** authentication state */ 240 HTTPAuthState auth_state; 241 242 /** The last reply of the server to a RTSP command */ 243 char last_reply[2048]; /* XXX: allocate ? */ 244 245 /** RTSPStream->transport_priv of the last stream that we read a 246 * packet from */ 247 void *cur_transport_priv; 248 249 /** The following are used for Real stream selection */ 250 //@{ 251 /** whether we need to send a "SET_PARAMETER Subscribe:" command */ 252 int need_subscription; 253 254 /** stream setup during the last frame read. This is used to detect if 255 * we need to subscribe or unsubscribe to any new streams. */ 256 enum AVDiscard real_setup_cache[MAX_STREAMS]; 257 258 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. 259 * this is used to send the same "Unsubscribe:" if stream setup changed, 260 * before sending a new "Subscribe:" command. */ 261 char last_subscription[1024]; 262 //@} 263 264 /** The following are used for RTP/ASF streams */ 265 //@{ 266 /** ASF demuxer context for the embedded ASF stream from WMS servers */ 267 AVFormatContext *asf_ctx; 268 269 /** cache for position of the asf demuxer, since we load a new 270 * data packet in the bytecontext for each incoming RTSP packet. */ 271 uint64_t asf_pb_pos; 272 //@} 273 274 /** some MS RTSP streams contain a URL in the SDP that we need to use 275 * for all subsequent RTSP requests, rather than the input URI; in 276 * other cases, this is a copy of AVFormatContext->filename. */ 277 char control_uri[1024]; 278 279 /** The synchronized start time of the output streams. */ 280 int64_t start_time; 281} RTSPState; 282 283/** 284 * Describes a single stream, as identified by a single m= line block in the 285 * SDP content. In the case of RDT, one RTSPStream can represent multiple 286 * AVStreams. In this case, each AVStream in this set has similar content 287 * (but different codec/bitrate). 288 */ 289typedef struct RTSPStream { 290 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ 291 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */ 292 293 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ 294 int stream_index; 295 296 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max 297 * for the selected transport. Only used for TCP. */ 298 int interleaved_min, interleaved_max; 299 300 char control_url[1024]; /**< url for this stream (from SDP) */ 301 302 /** The following are used only in SDP, not RTSP */ 303 //@{ 304 int sdp_port; /**< port (from SDP content) */ 305 struct in_addr sdp_ip; /**< IP address (from SDP content) */ 306 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ 307 int sdp_payload_type; /**< payload type */ 308 //@} 309 310 /** rtp payload parsing infos from SDP (i.e. mapping between private 311 * payload IDs and media-types (string), so that we can derive what 312 * type of payload we're dealing with (and how to parse it). */ 313 RTPPayloadData rtp_payload_data; 314 315 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */ 316 //@{ 317 /** handler structure */ 318 RTPDynamicProtocolHandler *dynamic_handler; 319 320 /** private data associated with the dynamic protocol */ 321 PayloadContext *dynamic_protocol_context; 322 //@} 323} RTSPStream; 324 325void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, 326 HTTPAuthState *auth_state); 327 328#if LIBAVFORMAT_VERSION_INT < (53 << 16) 329extern int rtsp_default_protocols; 330#endif 331extern int rtsp_rtp_port_min; 332extern int rtsp_rtp_port_max; 333 334/** 335 * Send a command to the RTSP server without waiting for the reply. 336 * 337 * @param s RTSP (de)muxer context 338 * @param method the method for the request 339 * @param url the target url for the request 340 * @param headers extra header lines to include in the request 341 * @param send_content if non-null, the data to send as request body content 342 * @param send_content_length the length of the send_content data, or 0 if 343 * send_content is null 344 */ 345void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s, 346 const char *method, const char *url, 347 const char *headers, 348 const unsigned char *send_content, 349 int send_content_length); 350/** 351 * Send a command to the RTSP server without waiting for the reply. 352 * 353 * @see rtsp_send_cmd_with_content_async 354 */ 355void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, 356 const char *url, const char *headers); 357 358/** 359 * Send a command to the RTSP server and wait for the reply. 360 * 361 * @param s RTSP (de)muxer context 362 * @param method the method for the request 363 * @param url the target url for the request 364 * @param headers extra header lines to include in the request 365 * @param reply pointer where the RTSP message header will be stored 366 * @param content_ptr pointer where the RTSP message body, if any, will 367 * be stored (length is in reply) 368 * @param send_content if non-null, the data to send as request body content 369 * @param send_content_length the length of the send_content data, or 0 if 370 * send_content is null 371 */ 372void ff_rtsp_send_cmd_with_content(AVFormatContext *s, 373 const char *method, const char *url, 374 const char *headers, 375 RTSPMessageHeader *reply, 376 unsigned char **content_ptr, 377 const unsigned char *send_content, 378 int send_content_length); 379 380/** 381 * Send a command to the RTSP server and wait for the reply. 382 * 383 * @see rtsp_send_cmd_with_content 384 */ 385void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, 386 const char *url, const char *headers, 387 RTSPMessageHeader *reply, unsigned char **content_ptr); 388 389/** 390 * Read a RTSP message from the server, or prepare to read data 391 * packets if we're reading data interleaved over the TCP/RTSP 392 * connection as well. 393 * 394 * @param s RTSP (de)muxer context 395 * @param reply pointer where the RTSP message header will be stored 396 * @param content_ptr pointer where the RTSP message body, if any, will 397 * be stored (length is in reply) 398 * @param return_on_interleaved_data whether the function may return if we 399 * encounter a data marker ('$'), which precedes data 400 * packets over interleaved TCP/RTSP connections. If this 401 * is set, this function will return 1 after encountering 402 * a '$'. If it is not set, the function will skip any 403 * data packets (if they are encountered), until a reply 404 * has been fully parsed. If no more data is available 405 * without parsing a reply, it will return an error. 406 * 407 * @return 1 if a data packets is ready to be received, -1 on error, 408 * and 0 on success. 409 */ 410int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, 411 unsigned char **content_ptr, 412 int return_on_interleaved_data); 413 414/** 415 * Skip a RTP/TCP interleaved packet. 416 */ 417void ff_rtsp_skip_packet(AVFormatContext *s); 418 419/** 420 * Connect to the RTSP server and set up the individual media streams. 421 * This can be used for both muxers and demuxers. 422 * 423 * @param s RTSP (de)muxer context 424 * 425 * @return 0 on success, < 0 on error. Cleans up all allocations done 426 * within the function on error. 427 */ 428int ff_rtsp_connect(AVFormatContext *s); 429 430/** 431 * Close and free all streams within the RTSP (de)muxer 432 * 433 * @param s RTSP (de)muxer context 434 */ 435void ff_rtsp_close_streams(AVFormatContext *s); 436 437#endif /* AVFORMAT_RTSP_H */ 438