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