aarch64: Add assembly support for -fsanitize=hwaddress tagged globals.
[libav.git] / libavformat / rtsp.h
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1/*
2 * RTSP definitions
406792e7 3 * Copyright (c) 2002 Fabrice Bellard
1617ad97 4 *
2912e87a 5 * This file is part of Libav.
b78e7197 6 *
2912e87a 7 * Libav is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
b78e7197 10 * version 2.1 of the License, or (at your option) any later version.
1617ad97 11 *
2912e87a 12 * Libav is distributed in the hope that it will be useful,
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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
2912e87a 18 * License along with Libav; if not, write to the Free Software
5509bffa 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
1617ad97 20 */
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21#ifndef AVFORMAT_RTSP_H
22#define AVFORMAT_RTSP_H
1617ad97 23
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24#include <stdint.h>
25#include "avformat.h"
1617ad97 26#include "rtspcodes.h"
302879cb 27#include "rtpdec.h"
74272b1c 28#include "network.h"
aa8bf2fb 29#include "httpauth.h"
1617ad97 30
4779f593 31#include "libavutil/log.h"
17fff881 32#include "libavutil/opt.h"
4779f593 33
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34/**
35 * Network layer over which RTP/etc packet data will be transported.
36 */
90abbdba 37enum RTSPLowerTransport {
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38 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
39 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
40 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
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41 RTSP_LOWER_TRANSPORT_NB,
42 RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
43 transport mode as such,
44 only for use via AVOptions */
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45 RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
46 option for lower_transport_mask,
47 but set in the SDP demuxer based
48 on a flag. */
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49};
50
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51/**
52 * Packet profile of the data that we will be receiving. Real servers
53 * commonly send RDT (although they can sometimes send RTP as well),
54 * whereas most others will send RTP.
55 */
1262d638 56enum RTSPTransport {
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57 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
58 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
df8cf076 59 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
2a1d51c5 60 RTSP_TRANSPORT_NB
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61};
62
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63/**
64 * Transport mode for the RTSP data. This may be plain, or
65 * tunneled, which is done over HTTP.
66 */
67enum RTSPControlTransport {
68 RTSP_MODE_PLAIN, /**< Normal RTSP */
69 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
70};
71
1617ad97 72#define RTSP_DEFAULT_PORT 554
8b2e9636 73#define RTSPS_DEFAULT_PORT 322
1617ad97 74#define RTSP_MAX_TRANSPORTS 8
b7b8fc34 75#define RTSP_TCP_MAX_PACKET_SIZE 1472
03a3fcee 76#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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77#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
78#define RTSP_RTP_PORT_MIN 5000
79#define RTSP_RTP_PORT_MAX 10000
1617ad97 80
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81/**
82 * This describes a single item in the "Transport:" line of one stream as
83 * negotiated by the SETUP RTSP command. Multiple transports are comma-
84 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
85 * client_port=1000-1001;server_port=1800-1801") and described in separate
86 * RTSPTransportFields.
87 */
1617ad97 88typedef struct RTSPTransportField {
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89 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
90 * with a '$', stream length and stream ID. If the stream ID is within
91 * the range of this interleaved_min-max, then the packet belongs to
92 * this stream. */
93 int interleaved_min, interleaved_max;
94
95 /** UDP multicast port range; the ports to which we should connect to
96 * receive multicast UDP data. */
97 int port_min, port_max;
98
99 /** UDP client ports; these should be the local ports of the UDP RTP
100 * (and RTCP) sockets over which we receive RTP/RTCP data. */
101 int client_port_min, client_port_max;
102
103 /** UDP unicast server port range; the ports to which we should connect
104 * to receive unicast UDP RTP/RTCP data. */
105 int server_port_min, server_port_max;
106
107 /** time-to-live value (required for multicast); the amount of HOPs that
108 * packets will be allowed to make before being discarded. */
109 int ttl;
110
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111 /** transport set to record data */
112 int mode_record;
113
7934b15d 114 struct sockaddr_storage destination; /**< destination IP address */
619298a8 115 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
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116
117 /** data/packet transport protocol; e.g. RTP or RDT */
1262d638 118 enum RTSPTransport transport;
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119
120 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
90abbdba 121 enum RTSPLowerTransport lower_transport;
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122} RTSPTransportField;
123
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124/**
125 * This describes the server response to each RTSP command.
126 */
a9e534d5 127typedef struct RTSPMessageHeader {
26d6b3e2 128 /** length of the data following this header */
1617ad97 129 int content_length;
26d6b3e2 130
37d2210a 131 enum RTSPStatusCode status_code; /**< response code from server */
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132
133 /** number of items in the 'transports' variable below */
1617ad97 134 int nb_transports;
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135
136 /** Time range of the streams that the server will stream. In
137 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
115329f1 138 int64_t range_start, range_end;
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139
140 /** describes the complete "Transport:" line of the server in response
141 * to a SETUP RTSP command by the client */
1617ad97 142 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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143
144 int seq; /**< sequence number */
145
146 /** the "Session:" field. This value is initially set by the server and
147 * should be re-transmitted by the client in every RTSP command. */
1617ad97 148 char session_id[512];
26d6b3e2 149
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150 /** the "Location:" field. This value is used to handle redirection.
151 */
152 char location[4096];
153
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154 /** the "RealChallenge1:" field from the server */
155 char real_challenge[64];
156
157 /** the "Server: field, which can be used to identify some special-case
158 * servers that are not 100% standards-compliant. We use this to identify
159 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
160 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
161 * use something like "Helix [..] Server Version v.e.r.sion (platform)
162 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
163 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
7a86bafa 164 char server[64];
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165
166 /** The "timeout" comes as part of the server response to the "SETUP"
167 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
168 * time, in seconds, that the server will go without traffic over the
169 * RTSP/TCP connection before it closes the connection. To prevent
170 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
171 * than this value. */
172 int timeout;
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173
174 /** The "Notice" or "X-Notice" field value. See
175 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
176 * for a complete list of supported values. */
177 int notice;
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178
179 /** The "reason" is meant to specify better the meaning of the error code
180 * returned
181 */
182 char reason[256];
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183
184 /**
185 * Content type header
186 */
187 char content_type[64];
a9e534d5 188} RTSPMessageHeader;
1617ad97 189
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190/**
191 * Client state, i.e. whether we are currently receiving data (PLAYING) or
192 * setup-but-not-receiving (PAUSED). State can be changed in applications
193 * by calling av_read_play/pause().
194 */
74272b1c 195enum RTSPClientState {
26d6b3e2 196 RTSP_STATE_IDLE, /**< not initialized */
c02fd3d2 197 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
26d6b3e2 198 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
ec606b36 199 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
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200};
201
26d6b3e2 202/**
58c42af7 203 * Identify particular servers that require special handling, such as
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204 * standards-incompliant "Transport:" lines in the SETUP request.
205 */
74272b1c 206enum RTSPServerType {
6e5f27ca 207 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
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208 RTSP_SERVER_REAL, /**< Realmedia-style server */
209 RTSP_SERVER_WMS, /**< Windows Media server */
2a1d51c5 210 RTSP_SERVER_NB
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211};
212
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213/**
214 * Private data for the RTSP demuxer.
9c610b76 215 *
ae628ec1 216 * @todo Use AVIOContext instead of URLContext
26d6b3e2 217 */
74272b1c 218typedef struct RTSPState {
4779f593 219 const AVClass *class; /**< Class for private options. */
48e77473 220 URLContext *rtsp_hd; /* RTSP TCP connection handle */
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221
222 /** number of items in the 'rtsp_streams' variable */
74272b1c 223 int nb_rtsp_streams;
74272b1c 224
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225 struct RTSPStream **rtsp_streams; /**< streams in this session */
226
227 /** indicator of whether we are currently receiving data from the
228 * server. Basically this isn't more than a simple cache of the
229 * last PLAY/PAUSE command sent to the server, to make sure we don't
230 * send 2x the same unexpectedly or commands in the wrong state. */
74272b1c 231 enum RTSPClientState state;
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232
233 /** the seek value requested when calling av_seek_frame(). This value
234 * is subsequently used as part of the "Range" parameter when emitting
235 * the RTSP PLAY command. If we are currently playing, this command is
236 * called instantly. If we are currently paused, this command is called
237 * whenever we resume playback. Either way, the value is only used once,
238 * see rtsp_read_play() and rtsp_read_seek(). */
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239 int64_t seek_timestamp;
240
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241 int seq; /**< RTSP command sequence number */
242
243 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
244 * identifier that the client should re-transmit in each RTSP command */
74272b1c 245 char session_id[512];
26d6b3e2 246
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247 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
248 * the server will go without traffic on the RTSP/TCP line before it
249 * closes the connection. */
250 int timeout;
251
252 /** timestamp of the last RTSP command that we sent to the RTSP server.
253 * This is used to calculate when to send dummy commands to keep the
bf7e799c 254 * connection alive, in conjunction with timeout. */
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255 int64_t last_cmd_time;
256
26d6b3e2 257 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
74272b1c 258 enum RTSPTransport transport;
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259
260 /** the negotiated network layer transport protocol; e.g. TCP or UDP
261 * uni-/multicast */
74272b1c 262 enum RTSPLowerTransport lower_transport;
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263
264 /** brand of server that we're talking to; e.g. WMS, REAL or other.
265 * Detected based on the value of RTSPMessageHeader->server or the presence
266 * of RTSPMessageHeader->real_challenge */
74272b1c 267 enum RTSPServerType server_type;
26d6b3e2 268
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269 /** the "RealChallenge1:" field from the server */
270 char real_challenge[64];
271
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272 /** plaintext authorization line (username:password) */
273 char auth[128];
274
275 /** authentication state */
276 HTTPAuthState auth_state;
f9337897 277
26d6b3e2 278 /** The last reply of the server to a RTSP command */
74272b1c 279 char last_reply[2048]; /* XXX: allocate ? */
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280
281 /** RTSPStream->transport_priv of the last stream that we read a
282 * packet from */
0a861b6f 283 void *cur_transport_priv;
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284
285 /** The following are used for Real stream selection */
286 //@{
287 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
74272b1c 288 int need_subscription;
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289
290 /** stream setup during the last frame read. This is used to detect if
291 * we need to subscribe or unsubscribe to any new streams. */
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292 enum AVDiscard *real_setup_cache;
293
294 /** current stream setup. This is a temporary buffer used to compare
295 * current setup to previous frame setup. */
296 enum AVDiscard *real_setup;
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297
298 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
299 * this is used to send the same "Unsubscribe:" if stream setup changed,
300 * before sending a new "Subscribe:" command. */
74272b1c 301 char last_subscription[1024];
26d6b3e2 302 //@}
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303
304 /** The following are used for RTP/ASF streams */
305 //@{
306 /** ASF demuxer context for the embedded ASF stream from WMS servers */
307 AVFormatContext *asf_ctx;
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308
309 /** cache for position of the asf demuxer, since we load a new
310 * data packet in the bytecontext for each incoming RTSP packet. */
311 uint64_t asf_pb_pos;
1a30d541 312 //@}
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313
314 /** some MS RTSP streams contain a URL in the SDP that we need to use
315 * for all subsequent RTSP requests, rather than the input URI; in
316 * other cases, this is a copy of AVFormatContext->filename. */
317 char control_uri[1024];
c07c6f81 318
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319 /** The following are used for parsing raw mpegts in udp */
320 //@{
321 struct MpegTSContext *ts;
322 int recvbuf_pos;
323 int recvbuf_len;
324 //@}
325
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326 /** Additional output handle, used when input and output are done
327 * separately, eg for HTTP tunneling. */
328 URLContext *rtsp_hd_out;
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329
330 /** RTSP transport mode, such as plain or tunneled. */
331 enum RTSPControlTransport control_transport;
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332
333 /* Number of RTCP BYE packets the RTSP session has received.
334 * An EOF is propagated back if nb_byes == nb_streams.
335 * This is reset after a seek. */
336 int nb_byes;
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337
338 /** Reusable buffer for receiving packets */
339 uint8_t* recvbuf;
a92c30d7 340
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341 /**
342 * A mask with all requested transport methods
343 */
344 int lower_transport_mask;
345
346 /**
347 * The number of returned packets
348 */
349 uint64_t packets;
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350
351 /**
352 * Polling array for udp
353 */
354 struct pollfd *p;
79331df3 355 int max_p;
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356
357 /**
358 * Whether the server supports the GET_PARAMETER method.
359 */
360 int get_parameter_supported;
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361
362 /**
363 * Do not begin to play the stream immediately.
364 */
365 int initial_pause;
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366
367 /**
368 * Option flags for the chained RTP muxer.
369 */
370 int rtp_muxer_flags;
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371
372 /** Whether the server accepts the x-Dynamic-Rate header */
373 int accept_dynamic_rate;
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374
375 /**
376 * Various option flags for the RTSP muxer/demuxer.
377 */
378 int rtsp_flags;
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379
380 /**
381 * Mask of all requested media types
382 */
383 int media_type_mask;
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384
385 /**
386 * Minimum and maximum local UDP ports.
387 */
388 int rtp_port_min, rtp_port_max;
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389
390 /**
391 * Timeout to wait for incoming connections.
392 */
393 int initial_timeout;
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394
395 /**
396 * Size of RTP packet reordering queue.
397 */
398 int reordering_queue_size;
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399
400 char default_lang[4];
e3ec6fe7 401 int buffer_size;
1e561735 402 int pkt_size;
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403
404 const URLProtocol **protocols;
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405} RTSPState;
406
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407#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
408 receive packets only from the right
409 source address and port. */
1c377449 410#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
e96406ed 411#define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
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412#define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
413 address of received packets. */
eca4850c 414
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415typedef struct RTSPSource {
416 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
417} RTSPSource;
418
26d6b3e2 419/**
58c42af7 420 * Describe a single stream, as identified by a single m= line block in the
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421 * SDP content. In the case of RDT, one RTSPStream can represent multiple
422 * AVStreams. In this case, each AVStream in this set has similar content
423 * (but different codec/bitrate).
424 */
74272b1c 425typedef struct RTSPStream {
26d6b3e2 426 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
fd450a51 427 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
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428
429 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
430 int stream_index;
431
432 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
433 * for the selected transport. Only used for TCP. */
434 int interleaved_min, interleaved_max;
435
436 char control_url[1024]; /**< url for this stream (from SDP) */
437
438 /** The following are used only in SDP, not RTSP */
439 //@{
440 int sdp_port; /**< port (from SDP content) */
3fbd12d1 441 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
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442 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
443 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
444 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
445 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
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446 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
447 int sdp_payload_type; /**< payload type */
448 //@}
74272b1c 449
e0d5ac6a 450 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
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451 //@{
452 /** handler structure */
453 RTPDynamicProtocolHandler *dynamic_handler;
74272b1c 454
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455 /** private data associated with the dynamic protocol */
456 PayloadContext *dynamic_protocol_context;
457 //@}
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458
459 /** Enable sending RTCP feedback messages according to RFC 4585 */
460 int feedback;
424da308 461
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462 /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
463 uint32_t ssrc;
464
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465 char crypto_suite[40];
466 char crypto_params[100];
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467} RTSPStream;
468
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469void ff_rtsp_parse_line(AVFormatContext *s,
470 RTSPMessageHeader *reply, const char *buf,
77223c53 471 RTSPState *rt, const char *method);
1617ad97 472
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473/**
474 * Send a command to the RTSP server without waiting for the reply.
475 *
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476 * @see rtsp_send_cmd_with_content_async
477 */
d0382374 478int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
fc490fcf 479 const char *url, const char *headers);
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480
481/**
482 * Send a command to the RTSP server and wait for the reply.
483 *
484 * @param s RTSP (de)muxer context
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485 * @param method the method for the request
486 * @param url the target url for the request
487 * @param headers extra header lines to include in the request
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488 * @param reply pointer where the RTSP message header will be stored
489 * @param content_ptr pointer where the RTSP message body, if any, will
490 * be stored (length is in reply)
491 * @param send_content if non-null, the data to send as request body content
492 * @param send_content_length the length of the send_content data, or 0 if
493 * send_content is null
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494 *
495 * @return zero if success, nonzero otherwise
15ba2315 496 */
d0382374 497int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
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498 const char *method, const char *url,
499 const char *headers,
500 RTSPMessageHeader *reply,
501 unsigned char **content_ptr,
502 const unsigned char *send_content,
503 int send_content_length);
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504
505/**
506 * Send a command to the RTSP server and wait for the reply.
507 *
508 * @see rtsp_send_cmd_with_content
509 */
d0382374 510int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
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511 const char *url, const char *headers,
512 RTSPMessageHeader *reply, unsigned char **content_ptr);
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513
514/**
515 * Read a RTSP message from the server, or prepare to read data
516 * packets if we're reading data interleaved over the TCP/RTSP
517 * connection as well.
518 *
519 * @param s RTSP (de)muxer context
520 * @param reply pointer where the RTSP message header will be stored
521 * @param content_ptr pointer where the RTSP message body, if any, will
522 * be stored (length is in reply)
523 * @param return_on_interleaved_data whether the function may return if we
524 * encounter a data marker ('$'), which precedes data
525 * packets over interleaved TCP/RTSP connections. If this
526 * is set, this function will return 1 after encountering
527 * a '$'. If it is not set, the function will skip any
528 * data packets (if they are encountered), until a reply
529 * has been fully parsed. If no more data is available
530 * without parsing a reply, it will return an error.
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531 * @param method the RTSP method this is a reply to. This affects how
532 * some response headers are acted upon. May be NULL.
15ba2315 533 *
32e543f8 534 * @return 1 if a data packets is ready to be received, -1 on error,
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535 * and 0 on success.
536 */
3307e6ea 537int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
93993933 538 unsigned char **content_ptr,
3df54c6b 539 int return_on_interleaved_data, const char *method);
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540
541/**
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542 * Skip a RTP/TCP interleaved packet.
543 */
544void ff_rtsp_skip_packet(AVFormatContext *s);
545
546/**
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547 * Connect to the RTSP server and set up the individual media streams.
548 * This can be used for both muxers and demuxers.
549 *
550 * @param s RTSP (de)muxer context
551 *
32e543f8 552 * @return 0 on success, < 0 on error. Cleans up all allocations done
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553 * within the function on error.
554 */
3307e6ea 555int ff_rtsp_connect(AVFormatContext *s);
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556
557/**
558 * Close and free all streams within the RTSP (de)muxer
559 *
560 * @param s RTSP (de)muxer context
561 */
3307e6ea 562void ff_rtsp_close_streams(AVFormatContext *s);
15ba2315 563
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564/**
565 * Close all connection handles within the RTSP (de)muxer
566 *
96c1e6d4 567 * @param s RTSP (de)muxer context
b8c2c41d 568 */
96c1e6d4 569void ff_rtsp_close_connections(AVFormatContext *s);
b8c2c41d 570
c2688f3a 571/**
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572 * Get the description of the stream and set up the RTSPStream child
573 * objects.
574 */
575int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
576
577/**
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578 * Announce the stream to the server and set up the RTSPStream child
579 * objects for each media stream.
580 */
581int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
582
0526c6f7 583/**
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584 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
585 * listen mode.
586 */
587int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
588
589/**
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590 * Parse an SDP description of streams by populating an RTSPState struct
591 * within the AVFormatContext; also allocate the RTP streams and the
592 * pollfd array used for UDP streams.
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593 */
594int ff_sdp_parse(AVFormatContext *s, const char *content);
595
596/**
597 * Receive one RTP packet from an TCP interleaved RTSP stream.
598 */
599int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
600 uint8_t *buf, int buf_size);
601
602/**
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603 * Send buffered packets over TCP.
604 */
605int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
606
607/**
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608 * Receive one packet from the RTSPStreams set up in the AVFormatContext
609 * (which should contain a RTSPState struct as priv_data).
610 */
611int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
612
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613/**
614 * Do the SETUP requests for each stream for the chosen
615 * lower transport mode.
f75e3da5 616 * @return 0 on success, <0 on error, 1 if protocol is unavailable
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617 */
618int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
619 int lower_transport, const char *real_challenge);
620
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621/**
622 * Undo the effect of ff_rtsp_make_setup_request, close the
623 * transport_priv and rtp_handle fields.
624 */
50aef03b 625void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
93e7490e 626
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627/**
628 * Open RTSP transport context.
629 */
630int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
631
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632extern const AVOption ff_rtsp_options[];
633
58ad770f 634#endif /* AVFORMAT_RTSP_H */