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 #include "internal.h" 31 32 #include "libavutil/log.h" 33 #include "libavutil/opt.h" 34 35 /** 36 * Network layer over which RTP/etc packet data will be transported. 37 */ 38 enum RTSPLowerTransport { 39 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */ 40 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */ 41 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ 42 RTSP_LOWER_TRANSPORT_NB, 43 RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper 44 transport mode as such, 45 only for use via AVOptions */ 46 RTSP_LOWER_TRANSPORT_HTTPS, /**< HTTPS tunneled */ 47 RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public 48 option for lower_transport_mask, 49 but set in the SDP demuxer based 50 on a flag. */ 51 }; 52 53 /** 54 * Packet profile of the data that we will be receiving. Real servers 55 * commonly send RDT (although they can sometimes send RTP as well), 56 * whereas most others will send RTP. 57 */ 58 enum RTSPTransport { 59 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ 60 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ 61 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */ 62 RTSP_TRANSPORT_NB 63 }; 64 65 /** 66 * Transport mode for the RTSP data. This may be plain, or 67 * tunneled, which is done over HTTP. 68 */ 69 enum RTSPControlTransport { 70 RTSP_MODE_PLAIN, /**< Normal RTSP */ 71 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */ 72 }; 73 74 #define RTSP_DEFAULT_PORT 554 75 #define RTSPS_DEFAULT_PORT 322 76 #define RTSP_MAX_TRANSPORTS 8 77 #define RTSP_TCP_MAX_PACKET_SIZE 1472 78 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1 79 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 80 #define RTSP_RTP_PORT_MIN 5000 81 #define RTSP_RTP_PORT_MAX 65000 82 #define SDP_MAX_SIZE 16384 83 84 /** 85 * This describes a single item in the "Transport:" line of one stream as 86 * negotiated by the SETUP RTSP command. Multiple transports are comma- 87 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; 88 * client_port=1000-1001;server_port=1800-1801") and described in separate 89 * RTSPTransportFields. 90 */ 91 typedef struct RTSPTransportField { 92 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts 93 * with a '$', stream length and stream ID. If the stream ID is within 94 * the range of this interleaved_min-max, then the packet belongs to 95 * this stream. */ 96 int interleaved_min, interleaved_max; 97 98 /** UDP multicast port range; the ports to which we should connect to 99 * receive multicast UDP data. */ 100 int port_min, port_max; 101 102 /** UDP client ports; these should be the local ports of the UDP RTP 103 * (and RTCP) sockets over which we receive RTP/RTCP data. */ 104 int client_port_min, client_port_max; 105 106 /** UDP unicast server port range; the ports to which we should connect 107 * to receive unicast UDP RTP/RTCP data. */ 108 int server_port_min, server_port_max; 109 110 /** time-to-live value (required for multicast); the amount of HOPs that 111 * packets will be allowed to make before being discarded. */ 112 int ttl; 113 114 /** transport set to record data */ 115 int mode_record; 116 117 struct sockaddr_storage destination; /**< destination IP address */ 118 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */ 119 120 /** data/packet transport protocol; e.g. RTP or RDT */ 121 enum RTSPTransport transport; 122 123 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ 124 enum RTSPLowerTransport lower_transport; 125 } RTSPTransportField; 126 127 /** 128 * This describes the server response to each RTSP command. 129 */ 130 typedef struct RTSPMessageHeader { 131 /** length of the data following this header */ 132 int content_length; 133 134 enum RTSPStatusCode status_code; /**< response code from server */ 135 136 /** number of items in the 'transports' variable below */ 137 int nb_transports; 138 139 /** Time range of the streams that the server will stream. In 140 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ 141 int64_t range_start, range_end; 142 143 /** describes the complete "Transport:" line of the server in response 144 * to a SETUP RTSP command by the client */ 145 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; 146 147 int seq; /**< sequence number */ 148 149 /** the "Session:" field. This value is initially set by the server and 150 * should be re-transmitted by the client in every RTSP command. */ 151 char session_id[512]; 152 153 /** the "Location:" field. This value is used to handle redirection. 154 */ 155 char location[4096]; 156 157 /** the "RealChallenge1:" field from the server */ 158 char real_challenge[64]; 159 160 /** the "Server: field, which can be used to identify some special-case 161 * servers that are not 100% standards-compliant. We use this to identify 162 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where 163 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers 164 * use something like "Helix [..] Server Version v.e.r.sion (platform) 165 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", 166 * where platform is the output of $uname -msr | sed 's/ /-/g'. */ 167 char server[64]; 168 169 /** The "timeout" comes as part of the server response to the "SETUP" 170 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the 171 * time, in seconds, that the server will go without traffic over the 172 * RTSP/TCP connection before it closes the connection. To prevent 173 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller 174 * than this value. */ 175 int timeout; 176 177 /** The "Notice" or "X-Notice" field value. See 178 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 179 * for a complete list of supported values. */ 180 int notice; 181 182 /** The "reason" is meant to specify better the meaning of the error code 183 * returned 184 */ 185 char reason[256]; 186 187 /** 188 * Content type header 189 */ 190 char content_type[64]; 191 192 /** 193 * SAT>IP com.ses.streamID header 194 */ 195 char stream_id[64]; 196 } RTSPMessageHeader; 197 198 /** 199 * Client state, i.e. whether we are currently receiving data (PLAYING) or 200 * setup-but-not-receiving (PAUSED). State can be changed in applications 201 * by calling av_read_play/pause(). 202 */ 203 enum RTSPClientState { 204 RTSP_STATE_IDLE, /**< not initialized */ 205 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ 206 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ 207 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ 208 }; 209 210 /** 211 * Identify particular servers that require special handling, such as 212 * standards-incompliant "Transport:" lines in the SETUP request. 213 */ 214 enum RTSPServerType { 215 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ 216 RTSP_SERVER_REAL, /**< Realmedia-style server */ 217 RTSP_SERVER_WMS, /**< Windows Media server */ 218 RTSP_SERVER_SATIP,/**< SAT>IP server */ 219 RTSP_SERVER_NB 220 }; 221 222 /** 223 * Private data for the RTSP demuxer. 224 * 225 * @todo Use AVIOContext instead of URLContext 226 */ 227 typedef struct RTSPState { 228 const AVClass *class; /**< Class for private options. */ 229 URLContext *rtsp_hd; /* RTSP TCP connection handle */ 230 231 /** number of items in the 'rtsp_streams' variable */ 232 int nb_rtsp_streams; 233 234 struct RTSPStream **rtsp_streams; /**< streams in this session */ 235 236 /** indicator of whether we are currently receiving data from the 237 * server. Basically this isn't more than a simple cache of the 238 * last PLAY/PAUSE command sent to the server, to make sure we don't 239 * send 2x the same unexpectedly or commands in the wrong state. */ 240 enum RTSPClientState state; 241 242 /** the seek value requested when calling av_seek_frame(). This value 243 * is subsequently used as part of the "Range" parameter when emitting 244 * the RTSP PLAY command. If we are currently playing, this command is 245 * called instantly. If we are currently paused, this command is called 246 * whenever we resume playback. Either way, the value is only used once, 247 * see rtsp_read_play() and rtsp_read_seek(). */ 248 int64_t seek_timestamp; 249 250 int seq; /**< RTSP command sequence number */ 251 252 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session 253 * identifier that the client should re-transmit in each RTSP command */ 254 char session_id[512]; 255 256 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that 257 * the server will go without traffic on the RTSP/TCP line before it 258 * closes the connection. */ 259 int timeout; 260 261 /** timestamp of the last RTSP command that we sent to the RTSP server. 262 * This is used to calculate when to send dummy commands to keep the 263 * connection alive, in conjunction with timeout. */ 264 int64_t last_cmd_time; 265 266 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ 267 enum RTSPTransport transport; 268 269 /** the negotiated network layer transport protocol; e.g. TCP or UDP 270 * uni-/multicast */ 271 enum RTSPLowerTransport lower_transport; 272 273 /** brand of server that we're talking to; e.g. WMS, REAL or other. 274 * Detected based on the value of RTSPMessageHeader->server or the presence 275 * of RTSPMessageHeader->real_challenge */ 276 enum RTSPServerType server_type; 277 278 /** the "RealChallenge1:" field from the server */ 279 char real_challenge[64]; 280 281 /** plaintext authorization line (username:password) */ 282 char auth[128]; 283 284 /** authentication state */ 285 HTTPAuthState auth_state; 286 287 /** The last reply of the server to a RTSP command */ 288 char last_reply[2048]; /* XXX: allocate ? */ 289 290 /** RTSPStream->transport_priv of the last stream that we read a 291 * packet from */ 292 void *cur_transport_priv; 293 294 /** The following are used for Real stream selection */ 295 //@{ 296 /** whether we need to send a "SET_PARAMETER Subscribe:" command */ 297 int need_subscription; 298 299 /** stream setup during the last frame read. This is used to detect if 300 * we need to subscribe or unsubscribe to any new streams. */ 301 enum AVDiscard *real_setup_cache; 302 303 /** current stream setup. This is a temporary buffer used to compare 304 * current setup to previous frame setup. */ 305 enum AVDiscard *real_setup; 306 307 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. 308 * this is used to send the same "Unsubscribe:" if stream setup changed, 309 * before sending a new "Subscribe:" command. */ 310 char last_subscription[1024]; 311 //@} 312 313 /** The following are used for RTP/ASF streams */ 314 //@{ 315 /** ASF demuxer context for the embedded ASF stream from WMS servers */ 316 AVFormatContext *asf_ctx; 317 318 /** cache for position of the asf demuxer, since we load a new 319 * data packet in the bytecontext for each incoming RTSP packet. */ 320 uint64_t asf_pb_pos; 321 //@} 322 323 /** some MS RTSP streams contain a URL in the SDP that we need to use 324 * for all subsequent RTSP requests, rather than the input URI; in 325 * other cases, this is a copy of AVFormatContext->filename. */ 326 char control_uri[MAX_URL_SIZE]; 327 328 /** The following are used for parsing raw mpegts in udp */ 329 //@{ 330 struct MpegTSContext *ts; 331 int recvbuf_pos; 332 int recvbuf_len; 333 //@} 334 335 /** Additional output handle, used when input and output are done 336 * separately, eg for HTTP tunneling. */ 337 URLContext *rtsp_hd_out; 338 339 /** RTSP transport mode, such as plain or tunneled. */ 340 enum RTSPControlTransport control_transport; 341 342 /* Number of RTCP BYE packets the RTSP session has received. 343 * An EOF is propagated back if nb_byes == nb_streams. 344 * This is reset after a seek. */ 345 int nb_byes; 346 347 /** Reusable buffer for receiving packets */ 348 uint8_t* recvbuf; 349 350 /** 351 * A mask with all requested transport methods 352 */ 353 int lower_transport_mask; 354 355 /** 356 * The number of returned packets 357 */ 358 uint64_t packets; 359 360 /** 361 * Polling array for udp 362 */ 363 struct pollfd *p; 364 int max_p; 365 366 /** 367 * Whether the server supports the GET_PARAMETER method. 368 */ 369 int get_parameter_supported; 370 371 /** 372 * Do not begin to play the stream immediately. 373 */ 374 int initial_pause; 375 376 /** 377 * Option flags for the chained RTP muxer. 378 */ 379 int rtp_muxer_flags; 380 381 /** Whether the server accepts the x-Dynamic-Rate header */ 382 int accept_dynamic_rate; 383 384 /** 385 * Various option flags for the RTSP muxer/demuxer. 386 */ 387 int rtsp_flags; 388 389 /** 390 * Mask of all requested media types 391 */ 392 int media_type_mask; 393 394 /** 395 * Minimum and maximum local UDP ports. 396 */ 397 int rtp_port_min, rtp_port_max; 398 399 /** 400 * Timeout to wait for incoming connections. 401 */ 402 int initial_timeout; 403 404 /** 405 * timeout of socket i/o operations. 406 */ 407 int stimeout; 408 409 /** 410 * Size of RTP packet reordering queue. 411 */ 412 int reordering_queue_size; 413 414 /** 415 * User-Agent string 416 */ 417 char *user_agent; 418 419 char default_lang[4]; 420 int buffer_size; 421 int pkt_size; 422 } RTSPState; 423 424 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets - 425 receive packets only from the right 426 source address and port. */ 427 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */ 428 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */ 429 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source 430 address of received packets. */ 431 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */ 432 #define RTSP_FLAG_SATIP_RAW 0x20 /**< Export SAT>IP stream as raw MPEG-TS */ 433 434 typedef struct RTSPSource { 435 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */ 436 } RTSPSource; 437 438 /** 439 * Describe a single stream, as identified by a single m= line block in the 440 * SDP content. In the case of RDT, one RTSPStream can represent multiple 441 * AVStreams. In this case, each AVStream in this set has similar content 442 * (but different codec/bitrate). 443 */ 444 typedef struct RTSPStream { 445 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ 446 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */ 447 448 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ 449 int stream_index; 450 451 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max 452 * for the selected transport. Only used for TCP. */ 453 int interleaved_min, interleaved_max; 454 455 char control_url[MAX_URL_SIZE]; /**< url for this stream (from SDP) */ 456 457 /** The following are used only in SDP, not RTSP */ 458 //@{ 459 int sdp_port; /**< port (from SDP content) */ 460 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */ 461 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */ 462 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */ 463 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */ 464 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */ 465 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ 466 int sdp_payload_type; /**< payload type */ 467 //@} 468 469 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */ 470 //@{ 471 /** handler structure */ 472 const RTPDynamicProtocolHandler *dynamic_handler; 473 474 /** private data associated with the dynamic protocol */ 475 PayloadContext *dynamic_protocol_context; 476 //@} 477 478 /** Enable sending RTCP feedback messages according to RFC 4585 */ 479 int feedback; 480 481 /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */ 482 uint32_t ssrc; 483 484 char crypto_suite[40]; 485 char crypto_params[100]; 486 } RTSPStream; 487 488 void ff_rtsp_parse_line(AVFormatContext *s, 489 RTSPMessageHeader *reply, const char *buf, 490 RTSPState *rt, const char *method); 491 492 /** 493 * Send a command to the RTSP server without waiting for the reply. 494 * 495 * @see rtsp_send_cmd_with_content_async 496 */ 497 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, 498 const char *url, const char *headers); 499 500 /** 501 * Send a command to the RTSP server and wait for the reply. 502 * 503 * @param s RTSP (de)muxer context 504 * @param method the method for the request 505 * @param url the target url for the request 506 * @param headers extra header lines to include in the request 507 * @param reply pointer where the RTSP message header will be stored 508 * @param content_ptr pointer where the RTSP message body, if any, will 509 * be stored (length is in reply) 510 * @param send_content if non-null, the data to send as request body content 511 * @param send_content_length the length of the send_content data, or 0 if 512 * send_content is null 513 * 514 * @return zero if success, nonzero otherwise 515 */ 516 int ff_rtsp_send_cmd_with_content(AVFormatContext *s, 517 const char *method, const char *url, 518 const char *headers, 519 RTSPMessageHeader *reply, 520 unsigned char **content_ptr, 521 const unsigned char *send_content, 522 int send_content_length); 523 524 /** 525 * Send a command to the RTSP server and wait for the reply. 526 * 527 * @see rtsp_send_cmd_with_content 528 */ 529 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, 530 const char *url, const char *headers, 531 RTSPMessageHeader *reply, unsigned char **content_ptr); 532 533 /** 534 * Read a RTSP message from the server, or prepare to read data 535 * packets if we're reading data interleaved over the TCP/RTSP 536 * connection as well. 537 * 538 * @param s RTSP (de)muxer context 539 * @param reply pointer where the RTSP message header will be stored 540 * @param content_ptr pointer where the RTSP message body, if any, will 541 * be stored (length is in reply) 542 * @param return_on_interleaved_data whether the function may return if we 543 * encounter a data marker ('$'), which precedes data 544 * packets over interleaved TCP/RTSP connections. If this 545 * is set, this function will return 1 after encountering 546 * a '$'. If it is not set, the function will skip any 547 * data packets (if they are encountered), until a reply 548 * has been fully parsed. If no more data is available 549 * without parsing a reply, it will return an error. 550 * @param method the RTSP method this is a reply to. This affects how 551 * some response headers are acted upon. May be NULL. 552 * 553 * @return 1 if a data packets is ready to be received, -1 on error, 554 * and 0 on success. 555 */ 556 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, 557 unsigned char **content_ptr, 558 int return_on_interleaved_data, const char *method); 559 560 /** 561 * Skip a RTP/TCP interleaved packet. 562 */ 563 void ff_rtsp_skip_packet(AVFormatContext *s); 564 565 /** 566 * Connect to the RTSP server and set up the individual media streams. 567 * This can be used for both muxers and demuxers. 568 * 569 * @param s RTSP (de)muxer context 570 * 571 * @return 0 on success, < 0 on error. Cleans up all allocations done 572 * within the function on error. 573 */ 574 int ff_rtsp_connect(AVFormatContext *s); 575 576 /** 577 * Close and free all streams within the RTSP (de)muxer 578 * 579 * @param s RTSP (de)muxer context 580 */ 581 void ff_rtsp_close_streams(AVFormatContext *s); 582 583 /** 584 * Close all connection handles within the RTSP (de)muxer 585 * 586 * @param s RTSP (de)muxer context 587 */ 588 void ff_rtsp_close_connections(AVFormatContext *s); 589 590 /** 591 * Get the description of the stream and set up the RTSPStream child 592 * objects. 593 */ 594 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply); 595 596 /** 597 * Announce the stream to the server and set up the RTSPStream child 598 * objects for each media stream. 599 */ 600 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr); 601 602 /** 603 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in 604 * listen mode. 605 */ 606 int ff_rtsp_parse_streaming_commands(AVFormatContext *s); 607 608 /** 609 * Parse an SDP description of streams by populating an RTSPState struct 610 * within the AVFormatContext; also allocate the RTP streams and the 611 * pollfd array used for UDP streams. 612 */ 613 int ff_sdp_parse(AVFormatContext *s, const char *content); 614 615 /** 616 * Receive one RTP packet from an TCP interleaved RTSP stream. 617 */ 618 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, 619 uint8_t *buf, int buf_size); 620 621 /** 622 * Send buffered packets over TCP. 623 */ 624 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st); 625 626 /** 627 * Receive one packet from the RTSPStreams set up in the AVFormatContext 628 * (which should contain a RTSPState struct as priv_data). 629 */ 630 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt); 631 632 /** 633 * Do the SETUP requests for each stream for the chosen 634 * lower transport mode. 635 * @return 0 on success, <0 on error, 1 if protocol is unavailable 636 */ 637 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, 638 int lower_transport, const char *real_challenge); 639 640 /** 641 * Undo the effect of ff_rtsp_make_setup_request, close the 642 * transport_priv and rtp_handle fields. 643 */ 644 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets); 645 646 /** 647 * Open RTSP transport context. 648 */ 649 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st); 650 651 extern const AVOption ff_rtsp_options[]; 652 653 #endif /* AVFORMAT_RTSP_H */ 654