rtsp: Support receiving plain data over UDP without any RTP encapsulation
[libav.git] / libavformat / rtsp.h
CommitLineData
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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};
46
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47/**
48 * Packet profile of the data that we will be receiving. Real servers
49 * commonly send RDT (although they can sometimes send RTP as well),
50 * whereas most others will send RTP.
51 */
1262d638 52enum RTSPTransport {
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53 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
54 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
df8cf076 55 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
2a1d51c5 56 RTSP_TRANSPORT_NB
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57};
58
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59/**
60 * Transport mode for the RTSP data. This may be plain, or
61 * tunneled, which is done over HTTP.
62 */
63enum RTSPControlTransport {
64 RTSP_MODE_PLAIN, /**< Normal RTSP */
65 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
66};
67
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68#define RTSP_DEFAULT_PORT 554
69#define RTSP_MAX_TRANSPORTS 8
b7b8fc34 70#define RTSP_TCP_MAX_PACKET_SIZE 1472
03a3fcee 71#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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72#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
73#define RTSP_RTP_PORT_MIN 5000
74#define RTSP_RTP_PORT_MAX 10000
1617ad97 75
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76/**
77 * This describes a single item in the "Transport:" line of one stream as
78 * negotiated by the SETUP RTSP command. Multiple transports are comma-
79 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
80 * client_port=1000-1001;server_port=1800-1801") and described in separate
81 * RTSPTransportFields.
82 */
1617ad97 83typedef struct RTSPTransportField {
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84 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
85 * with a '$', stream length and stream ID. If the stream ID is within
86 * the range of this interleaved_min-max, then the packet belongs to
87 * this stream. */
88 int interleaved_min, interleaved_max;
89
90 /** UDP multicast port range; the ports to which we should connect to
91 * receive multicast UDP data. */
92 int port_min, port_max;
93
94 /** UDP client ports; these should be the local ports of the UDP RTP
95 * (and RTCP) sockets over which we receive RTP/RTCP data. */
96 int client_port_min, client_port_max;
97
98 /** UDP unicast server port range; the ports to which we should connect
99 * to receive unicast UDP RTP/RTCP data. */
100 int server_port_min, server_port_max;
101
102 /** time-to-live value (required for multicast); the amount of HOPs that
103 * packets will be allowed to make before being discarded. */
104 int ttl;
105
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106 /** transport set to record data */
107 int mode_record;
108
7934b15d 109 struct sockaddr_storage destination; /**< destination IP address */
619298a8 110 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
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111
112 /** data/packet transport protocol; e.g. RTP or RDT */
1262d638 113 enum RTSPTransport transport;
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114
115 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
90abbdba 116 enum RTSPLowerTransport lower_transport;
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117} RTSPTransportField;
118
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119/**
120 * This describes the server response to each RTSP command.
121 */
a9e534d5 122typedef struct RTSPMessageHeader {
26d6b3e2 123 /** length of the data following this header */
1617ad97 124 int content_length;
26d6b3e2 125
37d2210a 126 enum RTSPStatusCode status_code; /**< response code from server */
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127
128 /** number of items in the 'transports' variable below */
1617ad97 129 int nb_transports;
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130
131 /** Time range of the streams that the server will stream. In
132 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
115329f1 133 int64_t range_start, range_end;
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134
135 /** describes the complete "Transport:" line of the server in response
136 * to a SETUP RTSP command by the client */
1617ad97 137 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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138
139 int seq; /**< sequence number */
140
141 /** the "Session:" field. This value is initially set by the server and
142 * should be re-transmitted by the client in every RTSP command. */
1617ad97 143 char session_id[512];
26d6b3e2 144
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145 /** the "Location:" field. This value is used to handle redirection.
146 */
147 char location[4096];
148
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149 /** the "RealChallenge1:" field from the server */
150 char real_challenge[64];
151
152 /** the "Server: field, which can be used to identify some special-case
153 * servers that are not 100% standards-compliant. We use this to identify
154 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
155 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
156 * use something like "Helix [..] Server Version v.e.r.sion (platform)
157 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
158 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
7a86bafa 159 char server[64];
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160
161 /** The "timeout" comes as part of the server response to the "SETUP"
162 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
163 * time, in seconds, that the server will go without traffic over the
164 * RTSP/TCP connection before it closes the connection. To prevent
165 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
166 * than this value. */
167 int timeout;
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168
169 /** The "Notice" or "X-Notice" field value. See
170 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
171 * for a complete list of supported values. */
172 int notice;
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173
174 /** The "reason" is meant to specify better the meaning of the error code
175 * returned
176 */
177 char reason[256];
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178
179 /**
180 * Content type header
181 */
182 char content_type[64];
a9e534d5 183} RTSPMessageHeader;
1617ad97 184
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185/**
186 * Client state, i.e. whether we are currently receiving data (PLAYING) or
187 * setup-but-not-receiving (PAUSED). State can be changed in applications
188 * by calling av_read_play/pause().
189 */
74272b1c 190enum RTSPClientState {
26d6b3e2 191 RTSP_STATE_IDLE, /**< not initialized */
c02fd3d2 192 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
26d6b3e2 193 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
ec606b36 194 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
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195};
196
26d6b3e2 197/**
58c42af7 198 * Identify particular servers that require special handling, such as
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199 * standards-incompliant "Transport:" lines in the SETUP request.
200 */
74272b1c 201enum RTSPServerType {
6e5f27ca 202 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
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203 RTSP_SERVER_REAL, /**< Realmedia-style server */
204 RTSP_SERVER_WMS, /**< Windows Media server */
2a1d51c5 205 RTSP_SERVER_NB
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206};
207
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208/**
209 * Private data for the RTSP demuxer.
9c610b76 210 *
ae628ec1 211 * @todo Use AVIOContext instead of URLContext
26d6b3e2 212 */
74272b1c 213typedef struct RTSPState {
4779f593 214 const AVClass *class; /**< Class for private options. */
48e77473 215 URLContext *rtsp_hd; /* RTSP TCP connection handle */
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216
217 /** number of items in the 'rtsp_streams' variable */
74272b1c 218 int nb_rtsp_streams;
74272b1c 219
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220 struct RTSPStream **rtsp_streams; /**< streams in this session */
221
222 /** indicator of whether we are currently receiving data from the
223 * server. Basically this isn't more than a simple cache of the
224 * last PLAY/PAUSE command sent to the server, to make sure we don't
225 * send 2x the same unexpectedly or commands in the wrong state. */
74272b1c 226 enum RTSPClientState state;
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227
228 /** the seek value requested when calling av_seek_frame(). This value
229 * is subsequently used as part of the "Range" parameter when emitting
230 * the RTSP PLAY command. If we are currently playing, this command is
231 * called instantly. If we are currently paused, this command is called
232 * whenever we resume playback. Either way, the value is only used once,
233 * see rtsp_read_play() and rtsp_read_seek(). */
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234 int64_t seek_timestamp;
235
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236 int seq; /**< RTSP command sequence number */
237
238 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
239 * identifier that the client should re-transmit in each RTSP command */
74272b1c 240 char session_id[512];
26d6b3e2 241
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242 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
243 * the server will go without traffic on the RTSP/TCP line before it
244 * closes the connection. */
245 int timeout;
246
247 /** timestamp of the last RTSP command that we sent to the RTSP server.
248 * This is used to calculate when to send dummy commands to keep the
bf7e799c 249 * connection alive, in conjunction with timeout. */
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250 int64_t last_cmd_time;
251
26d6b3e2 252 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
74272b1c 253 enum RTSPTransport transport;
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254
255 /** the negotiated network layer transport protocol; e.g. TCP or UDP
256 * uni-/multicast */
74272b1c 257 enum RTSPLowerTransport lower_transport;
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258
259 /** brand of server that we're talking to; e.g. WMS, REAL or other.
260 * Detected based on the value of RTSPMessageHeader->server or the presence
261 * of RTSPMessageHeader->real_challenge */
74272b1c 262 enum RTSPServerType server_type;
26d6b3e2 263
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264 /** the "RealChallenge1:" field from the server */
265 char real_challenge[64];
266
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267 /** plaintext authorization line (username:password) */
268 char auth[128];
269
270 /** authentication state */
271 HTTPAuthState auth_state;
f9337897 272
26d6b3e2 273 /** The last reply of the server to a RTSP command */
74272b1c 274 char last_reply[2048]; /* XXX: allocate ? */
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275
276 /** RTSPStream->transport_priv of the last stream that we read a
277 * packet from */
0a861b6f 278 void *cur_transport_priv;
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279
280 /** The following are used for Real stream selection */
281 //@{
282 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
74272b1c 283 int need_subscription;
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284
285 /** stream setup during the last frame read. This is used to detect if
286 * we need to subscribe or unsubscribe to any new streams. */
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287 enum AVDiscard *real_setup_cache;
288
289 /** current stream setup. This is a temporary buffer used to compare
290 * current setup to previous frame setup. */
291 enum AVDiscard *real_setup;
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292
293 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
294 * this is used to send the same "Unsubscribe:" if stream setup changed,
295 * before sending a new "Subscribe:" command. */
74272b1c 296 char last_subscription[1024];
26d6b3e2 297 //@}
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298
299 /** The following are used for RTP/ASF streams */
300 //@{
301 /** ASF demuxer context for the embedded ASF stream from WMS servers */
302 AVFormatContext *asf_ctx;
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303
304 /** cache for position of the asf demuxer, since we load a new
305 * data packet in the bytecontext for each incoming RTSP packet. */
306 uint64_t asf_pb_pos;
1a30d541 307 //@}
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308
309 /** some MS RTSP streams contain a URL in the SDP that we need to use
310 * for all subsequent RTSP requests, rather than the input URI; in
311 * other cases, this is a copy of AVFormatContext->filename. */
312 char control_uri[1024];
c07c6f81 313
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314 /** Additional output handle, used when input and output are done
315 * separately, eg for HTTP tunneling. */
316 URLContext *rtsp_hd_out;
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317
318 /** RTSP transport mode, such as plain or tunneled. */
319 enum RTSPControlTransport control_transport;
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320
321 /* Number of RTCP BYE packets the RTSP session has received.
322 * An EOF is propagated back if nb_byes == nb_streams.
323 * This is reset after a seek. */
324 int nb_byes;
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325
326 /** Reusable buffer for receiving packets */
327 uint8_t* recvbuf;
a92c30d7 328
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329 /**
330 * A mask with all requested transport methods
331 */
332 int lower_transport_mask;
333
334 /**
335 * The number of returned packets
336 */
337 uint64_t packets;
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338
339 /**
340 * Polling array for udp
341 */
342 struct pollfd *p;
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343
344 /**
345 * Whether the server supports the GET_PARAMETER method.
346 */
347 int get_parameter_supported;
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348
349 /**
350 * Do not begin to play the stream immediately.
351 */
352 int initial_pause;
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353
354 /**
355 * Option flags for the chained RTP muxer.
356 */
357 int rtp_muxer_flags;
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358
359 /** Whether the server accepts the x-Dynamic-Rate header */
360 int accept_dynamic_rate;
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361
362 /**
363 * Various option flags for the RTSP muxer/demuxer.
364 */
365 int rtsp_flags;
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366
367 /**
368 * Mask of all requested media types
369 */
370 int media_type_mask;
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371
372 /**
373 * Minimum and maximum local UDP ports.
374 */
375 int rtp_port_min, rtp_port_max;
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376
377 /**
378 * Timeout to wait for incoming connections.
379 */
380 int initial_timeout;
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381} RTSPState;
382
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383#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
384 receive packets only from the right
385 source address and port. */
a8ad6ffa 386#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
eca4850c 387
26d6b3e2 388/**
58c42af7 389 * Describe a single stream, as identified by a single m= line block in the
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390 * SDP content. In the case of RDT, one RTSPStream can represent multiple
391 * AVStreams. In this case, each AVStream in this set has similar content
392 * (but different codec/bitrate).
393 */
74272b1c 394typedef struct RTSPStream {
26d6b3e2 395 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
fd450a51 396 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
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397
398 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
399 int stream_index;
400
401 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
402 * for the selected transport. Only used for TCP. */
403 int interleaved_min, interleaved_max;
404
405 char control_url[1024]; /**< url for this stream (from SDP) */
406
407 /** The following are used only in SDP, not RTSP */
408 //@{
409 int sdp_port; /**< port (from SDP content) */
3fbd12d1 410 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
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411 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
412 int sdp_payload_type; /**< payload type */
413 //@}
74272b1c 414
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415 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
416 //@{
417 /** handler structure */
418 RTPDynamicProtocolHandler *dynamic_handler;
74272b1c 419
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420 /** private data associated with the dynamic protocol */
421 PayloadContext *dynamic_protocol_context;
422 //@}
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423} RTSPStream;
424
2626308a 425void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
77223c53 426 RTSPState *rt, const char *method);
1617ad97 427
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428/**
429 * Send a command to the RTSP server without waiting for the reply.
430 *
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431 * @see rtsp_send_cmd_with_content_async
432 */
d0382374 433int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
fc490fcf 434 const char *url, const char *headers);
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435
436/**
437 * Send a command to the RTSP server and wait for the reply.
438 *
439 * @param s RTSP (de)muxer context
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440 * @param method the method for the request
441 * @param url the target url for the request
442 * @param headers extra header lines to include in the request
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443 * @param reply pointer where the RTSP message header will be stored
444 * @param content_ptr pointer where the RTSP message body, if any, will
445 * be stored (length is in reply)
446 * @param send_content if non-null, the data to send as request body content
447 * @param send_content_length the length of the send_content data, or 0 if
448 * send_content is null
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449 *
450 * @return zero if success, nonzero otherwise
15ba2315 451 */
d0382374 452int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
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453 const char *method, const char *url,
454 const char *headers,
455 RTSPMessageHeader *reply,
456 unsigned char **content_ptr,
457 const unsigned char *send_content,
458 int send_content_length);
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459
460/**
461 * Send a command to the RTSP server and wait for the reply.
462 *
463 * @see rtsp_send_cmd_with_content
464 */
d0382374 465int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
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466 const char *url, const char *headers,
467 RTSPMessageHeader *reply, unsigned char **content_ptr);
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468
469/**
470 * Read a RTSP message from the server, or prepare to read data
471 * packets if we're reading data interleaved over the TCP/RTSP
472 * connection as well.
473 *
474 * @param s RTSP (de)muxer context
475 * @param reply pointer where the RTSP message header will be stored
476 * @param content_ptr pointer where the RTSP message body, if any, will
477 * be stored (length is in reply)
478 * @param return_on_interleaved_data whether the function may return if we
479 * encounter a data marker ('$'), which precedes data
480 * packets over interleaved TCP/RTSP connections. If this
481 * is set, this function will return 1 after encountering
482 * a '$'. If it is not set, the function will skip any
483 * data packets (if they are encountered), until a reply
484 * has been fully parsed. If no more data is available
485 * without parsing a reply, it will return an error.
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486 * @param method the RTSP method this is a reply to. This affects how
487 * some response headers are acted upon. May be NULL.
15ba2315 488 *
32e543f8 489 * @return 1 if a data packets is ready to be received, -1 on error,
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490 * and 0 on success.
491 */
3307e6ea 492int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
93993933 493 unsigned char **content_ptr,
3df54c6b 494 int return_on_interleaved_data, const char *method);
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495
496/**
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497 * Skip a RTP/TCP interleaved packet.
498 */
499void ff_rtsp_skip_packet(AVFormatContext *s);
500
501/**
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502 * Connect to the RTSP server and set up the individual media streams.
503 * This can be used for both muxers and demuxers.
504 *
505 * @param s RTSP (de)muxer context
506 *
32e543f8 507 * @return 0 on success, < 0 on error. Cleans up all allocations done
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508 * within the function on error.
509 */
3307e6ea 510int ff_rtsp_connect(AVFormatContext *s);
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511
512/**
513 * Close and free all streams within the RTSP (de)muxer
514 *
515 * @param s RTSP (de)muxer context
516 */
3307e6ea 517void ff_rtsp_close_streams(AVFormatContext *s);
15ba2315 518
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519/**
520 * Close all connection handles within the RTSP (de)muxer
521 *
96c1e6d4 522 * @param s RTSP (de)muxer context
b8c2c41d 523 */
96c1e6d4 524void ff_rtsp_close_connections(AVFormatContext *s);
b8c2c41d 525
c2688f3a 526/**
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527 * Get the description of the stream and set up the RTSPStream child
528 * objects.
529 */
530int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
531
532/**
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533 * Announce the stream to the server and set up the RTSPStream child
534 * objects for each media stream.
535 */
536int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
537
0526c6f7 538/**
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539 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
540 * listen mode.
541 */
542int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
543
544/**
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545 * Parse an SDP description of streams by populating an RTSPState struct
546 * within the AVFormatContext; also allocate the RTP streams and the
547 * pollfd array used for UDP streams.
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548 */
549int ff_sdp_parse(AVFormatContext *s, const char *content);
550
551/**
552 * Receive one RTP packet from an TCP interleaved RTSP stream.
553 */
554int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
555 uint8_t *buf, int buf_size);
556
557/**
558 * Receive one packet from the RTSPStreams set up in the AVFormatContext
559 * (which should contain a RTSPState struct as priv_data).
560 */
561int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
562
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563/**
564 * Do the SETUP requests for each stream for the chosen
565 * lower transport mode.
f75e3da5 566 * @return 0 on success, <0 on error, 1 if protocol is unavailable
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567 */
568int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
569 int lower_transport, const char *real_challenge);
570
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571/**
572 * Undo the effect of ff_rtsp_make_setup_request, close the
573 * transport_priv and rtp_handle fields.
574 */
575void ff_rtsp_undo_setup(AVFormatContext *s);
576
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577/**
578 * Open RTSP transport context.
579 */
580int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
581
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582extern const AVOption ff_rtsp_options[];
583
58ad770f 584#endif /* AVFORMAT_RTSP_H */