FFmpeg  4.3.8
rtsp.h
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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 #include "libavutil/log.h"
32 #include "libavutil/opt.h"
33 
34 /**
35  * Network layer over which RTP/etc packet data will be transported.
36  */
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 */
42  RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
43  transport mode as such,
44  only for use via AVOptions */
45  RTSP_LOWER_TRANSPORT_HTTPS, /**< HTTPS tunneled */
46  RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
47  option for lower_transport_mask,
48  but set in the SDP demuxer based
49  on a flag. */
50 };
51 
52 /**
53  * Packet profile of the data that we will be receiving. Real servers
54  * commonly send RDT (although they can sometimes send RTP as well),
55  * whereas most others will send RTP.
56  */
58  RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
59  RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
60  RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
62 };
63 
64 /**
65  * Transport mode for the RTSP data. This may be plain, or
66  * tunneled, which is done over HTTP.
67  */
69  RTSP_MODE_PLAIN, /**< Normal RTSP */
70  RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
71 };
72 
73 #define RTSP_DEFAULT_PORT 554
74 #define RTSPS_DEFAULT_PORT 322
75 #define RTSP_MAX_TRANSPORTS 8
76 #define RTSP_TCP_MAX_PACKET_SIZE 1472
77 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
78 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
79 #define RTSP_RTP_PORT_MIN 5000
80 #define RTSP_RTP_PORT_MAX 65000
81 
82 /**
83  * This describes a single item in the "Transport:" line of one stream as
84  * negotiated by the SETUP RTSP command. Multiple transports are comma-
85  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
86  * client_port=1000-1001;server_port=1800-1801") and described in separate
87  * RTSPTransportFields.
88  */
89 typedef struct RTSPTransportField {
90  /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
91  * with a '$', stream length and stream ID. If the stream ID is within
92  * the range of this interleaved_min-max, then the packet belongs to
93  * this stream. */
95 
96  /** UDP multicast port range; the ports to which we should connect to
97  * receive multicast UDP data. */
99 
100  /** UDP client ports; these should be the local ports of the UDP RTP
101  * (and RTCP) sockets over which we receive RTP/RTCP data. */
103 
104  /** UDP unicast server port range; the ports to which we should connect
105  * to receive unicast UDP RTP/RTCP data. */
107 
108  /** time-to-live value (required for multicast); the amount of HOPs that
109  * packets will be allowed to make before being discarded. */
110  int ttl;
111 
112  /** transport set to record data */
114 
115  struct sockaddr_storage destination; /**< destination IP address */
116  char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
117 
118  /** data/packet transport protocol; e.g. RTP or RDT */
120 
121  /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
124 
125 /**
126  * This describes the server response to each RTSP command.
127  */
128 typedef struct RTSPMessageHeader {
129  /** length of the data following this header */
131 
132  enum RTSPStatusCode status_code; /**< response code from server */
133 
134  /** number of items in the 'transports' variable below */
136 
137  /** Time range of the streams that the server will stream. In
138  * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
139  int64_t range_start, range_end;
140 
141  /** describes the complete "Transport:" line of the server in response
142  * to a SETUP RTSP command by the client */
144 
145  int seq; /**< sequence number */
146 
147  /** the "Session:" field. This value is initially set by the server and
148  * should be re-transmitted by the client in every RTSP command. */
149  char session_id[512];
150 
151  /** the "Location:" field. This value is used to handle redirection.
152  */
153  char location[4096];
154 
155  /** the "RealChallenge1:" field from the server */
156  char real_challenge[64];
157 
158  /** the "Server: field, which can be used to identify some special-case
159  * servers that are not 100% standards-compliant. We use this to identify
160  * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
161  * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
162  * use something like "Helix [..] Server Version v.e.r.sion (platform)
163  * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
164  * where platform is the output of $uname -msr | sed 's/ /-/g'. */
165  char server[64];
166 
167  /** The "timeout" comes as part of the server response to the "SETUP"
168  * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
169  * time, in seconds, that the server will go without traffic over the
170  * RTSP/TCP connection before it closes the connection. To prevent
171  * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
172  * than this value. */
173  int timeout;
174 
175  /** The "Notice" or "X-Notice" field value. See
176  * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
177  * for a complete list of supported values. */
178  int notice;
179 
180  /** The "reason" is meant to specify better the meaning of the error code
181  * returned
182  */
183  char reason[256];
184 
185  /**
186  * Content type header
187  */
188  char content_type[64];
190 
191 /**
192  * Client state, i.e. whether we are currently receiving data (PLAYING) or
193  * setup-but-not-receiving (PAUSED). State can be changed in applications
194  * by calling av_read_play/pause().
195  */
197  RTSP_STATE_IDLE, /**< not initialized */
198  RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
199  RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
200  RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
201 };
202 
203 /**
204  * Identify particular servers that require special handling, such as
205  * standards-incompliant "Transport:" lines in the SETUP request.
206  */
208  RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
209  RTSP_SERVER_REAL, /**< Realmedia-style server */
210  RTSP_SERVER_WMS, /**< Windows Media server */
212 };
213 
214 /**
215  * Private data for the RTSP demuxer.
216  *
217  * @todo Use AVIOContext instead of URLContext
218  */
219 typedef struct RTSPState {
220  const AVClass *class; /**< Class for private options. */
221  URLContext *rtsp_hd; /* RTSP TCP connection handle */
222 
223  /** number of items in the 'rtsp_streams' variable */
225 
226  struct RTSPStream **rtsp_streams; /**< streams in this session */
227 
228  /** indicator of whether we are currently receiving data from the
229  * server. Basically this isn't more than a simple cache of the
230  * last PLAY/PAUSE command sent to the server, to make sure we don't
231  * send 2x the same unexpectedly or commands in the wrong state. */
233 
234  /** the seek value requested when calling av_seek_frame(). This value
235  * is subsequently used as part of the "Range" parameter when emitting
236  * the RTSP PLAY command. If we are currently playing, this command is
237  * called instantly. If we are currently paused, this command is called
238  * whenever we resume playback. Either way, the value is only used once,
239  * see rtsp_read_play() and rtsp_read_seek(). */
241 
242  int seq; /**< RTSP command sequence number */
243 
244  /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
245  * identifier that the client should re-transmit in each RTSP command */
246  char session_id[512];
247 
248  /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
249  * the server will go without traffic on the RTSP/TCP line before it
250  * closes the connection. */
251  int timeout;
252 
253  /** timestamp of the last RTSP command that we sent to the RTSP server.
254  * This is used to calculate when to send dummy commands to keep the
255  * connection alive, in conjunction with timeout. */
257 
258  /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
260 
261  /** the negotiated network layer transport protocol; e.g. TCP or UDP
262  * uni-/multicast */
264 
265  /** brand of server that we're talking to; e.g. WMS, REAL or other.
266  * Detected based on the value of RTSPMessageHeader->server or the presence
267  * of RTSPMessageHeader->real_challenge */
268  enum RTSPServerType server_type;
269 
270  /** the "RealChallenge1:" field from the server */
271  char real_challenge[64];
272 
273  /** plaintext authorization line (username:password) */
274  char auth[128];
275 
276  /** authentication state */
278 
279  /** The last reply of the server to a RTSP command */
280  char last_reply[2048]; /* XXX: allocate ? */
281 
282  /** RTSPStream->transport_priv of the last stream that we read a
283  * packet from */
285 
286  /** The following are used for Real stream selection */
287  //@{
288  /** whether we need to send a "SET_PARAMETER Subscribe:" command */
290 
291  /** stream setup during the last frame read. This is used to detect if
292  * we need to subscribe or unsubscribe to any new streams. */
294 
295  /** current stream setup. This is a temporary buffer used to compare
296  * current setup to previous frame setup. */
298 
299  /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
300  * this is used to send the same "Unsubscribe:" if stream setup changed,
301  * before sending a new "Subscribe:" command. */
302  char last_subscription[1024];
303  //@}
304 
305  /** The following are used for RTP/ASF streams */
306  //@{
307  /** ASF demuxer context for the embedded ASF stream from WMS servers */
309 
310  /** cache for position of the asf demuxer, since we load a new
311  * data packet in the bytecontext for each incoming RTSP packet. */
312  uint64_t asf_pb_pos;
313  //@}
314 
315  /** some MS RTSP streams contain a URL in the SDP that we need to use
316  * for all subsequent RTSP requests, rather than the input URI; in
317  * other cases, this is a copy of AVFormatContext->filename. */
318  char control_uri[1024];
319 
320  /** The following are used for parsing raw mpegts in udp */
321  //@{
322  struct MpegTSContext *ts;
325  //@}
326 
327  /** Additional output handle, used when input and output are done
328  * separately, eg for HTTP tunneling. */
330 
331  /** RTSP transport mode, such as plain or tunneled. */
332  enum RTSPControlTransport control_transport;
333 
334  /* Number of RTCP BYE packets the RTSP session has received.
335  * An EOF is propagated back if nb_byes == nb_streams.
336  * This is reset after a seek. */
337  int nb_byes;
338 
339  /** Reusable buffer for receiving packets */
341 
342  /**
343  * A mask with all requested transport methods
344  */
346 
347  /**
348  * The number of returned packets
349  */
350  uint64_t packets;
351 
352  /**
353  * Polling array for udp
354  */
355  struct pollfd *p;
356  int max_p;
357 
358  /**
359  * Whether the server supports the GET_PARAMETER method.
360  */
362 
363  /**
364  * Do not begin to play the stream immediately.
365  */
367 
368  /**
369  * Option flags for the chained RTP muxer.
370  */
372 
373  /** Whether the server accepts the x-Dynamic-Rate header */
375 
376  /**
377  * Various option flags for the RTSP muxer/demuxer.
378  */
380 
381  /**
382  * Mask of all requested media types
383  */
385 
386  /**
387  * Minimum and maximum local UDP ports.
388  */
389  int rtp_port_min, rtp_port_max;
390 
391  /**
392  * Timeout to wait for incoming connections.
393  */
395 
396  /**
397  * timeout of socket i/o operations.
398  */
399  int stimeout;
400 
401  /**
402  * Size of RTP packet reordering queue.
403  */
405 
406  /**
407  * User-Agent string
408  */
409  char *user_agent;
410 
411  char default_lang[4];
413  int pkt_size;
414 } RTSPState;
415 
416 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
417  receive packets only from the right
418  source address and port. */
419 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
420 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
421 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
422  address of received packets. */
423 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
424 
425 typedef struct RTSPSource {
426  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
427 } RTSPSource;
429 /**
430  * Describe a single stream, as identified by a single m= line block in the
431  * SDP content. In the case of RDT, one RTSPStream can represent multiple
432  * AVStreams. In this case, each AVStream in this set has similar content
433  * (but different codec/bitrate).
434  */
435 typedef struct RTSPStream {
436  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
437  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
439  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
440  int stream_index;
441 
442  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
443  * for the selected transport. Only used for TCP. */
445 
446  char control_url[1024]; /**< url for this stream (from SDP) */
448  /** The following are used only in SDP, not RTSP */
449  //@{
450  int sdp_port; /**< port (from SDP content) */
451  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
452  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
453  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
454  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
455  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
456  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
457  int sdp_payload_type; /**< payload type */
458  //@}
460  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
461  //@{
462  /** handler structure */
463  const RTPDynamicProtocolHandler *dynamic_handler;
464 
465  /** private data associated with the dynamic protocol */
466  PayloadContext *dynamic_protocol_context;
467  //@}
468 
469  /** Enable sending RTCP feedback messages according to RFC 4585 */
470  int feedback;
471 
472  /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
473  uint32_t ssrc;
474 
475  char crypto_suite[40];
476  char crypto_params[100];
477 } RTSPStream;
480  RTSPMessageHeader *reply, const char *buf,
481  RTSPState *rt, const char *method);
482 
483 /**
484  * Send a command to the RTSP server without waiting for the reply.
485  *
486  * @see rtsp_send_cmd_with_content_async
487  */
488 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
489  const char *url, const char *headers);
490 
491 /**
492  * Send a command to the RTSP server and wait for the reply.
493  *
494  * @param s RTSP (de)muxer context
495  * @param method the method for the request
496  * @param url the target url for the request
497  * @param headers extra header lines to include in the request
498  * @param reply pointer where the RTSP message header will be stored
499  * @param content_ptr pointer where the RTSP message body, if any, will
500  * be stored (length is in reply)
501  * @param send_content if non-null, the data to send as request body content
502  * @param send_content_length the length of the send_content data, or 0 if
503  * send_content is null
504  *
505  * @return zero if success, nonzero otherwise
506  */
508  const char *method, const char *url,
509  const char *headers,
510  RTSPMessageHeader *reply,
511  unsigned char **content_ptr,
512  const unsigned char *send_content,
513  int send_content_length);
514 
515 /**
516  * Send a command to the RTSP server and wait for the reply.
517  *
518  * @see rtsp_send_cmd_with_content
519  */
520 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
521  const char *url, const char *headers,
522  RTSPMessageHeader *reply, unsigned char **content_ptr);
523 
524 /**
525  * Read a RTSP message from the server, or prepare to read data
526  * packets if we're reading data interleaved over the TCP/RTSP
527  * connection as well.
528  *
529  * @param s RTSP (de)muxer context
530  * @param reply pointer where the RTSP message header will be stored
531  * @param content_ptr pointer where the RTSP message body, if any, will
532  * be stored (length is in reply)
533  * @param return_on_interleaved_data whether the function may return if we
534  * encounter a data marker ('$'), which precedes data
535  * packets over interleaved TCP/RTSP connections. If this
536  * is set, this function will return 1 after encountering
537  * a '$'. If it is not set, the function will skip any
538  * data packets (if they are encountered), until a reply
539  * has been fully parsed. If no more data is available
540  * without parsing a reply, it will return an error.
541  * @param method the RTSP method this is a reply to. This affects how
542  * some response headers are acted upon. May be NULL.
543  *
544  * @return 1 if a data packets is ready to be received, -1 on error,
545  * and 0 on success.
546  */
548  unsigned char **content_ptr,
549  int return_on_interleaved_data, const char *method);
550 
551 /**
552  * Skip a RTP/TCP interleaved packet.
553  */
555 
556 /**
557  * Connect to the RTSP server and set up the individual media streams.
558  * This can be used for both muxers and demuxers.
559  *
560  * @param s RTSP (de)muxer context
561  *
562  * @return 0 on success, < 0 on error. Cleans up all allocations done
563  * within the function on error.
564  */
566 
567 /**
568  * Close and free all streams within the RTSP (de)muxer
569  *
570  * @param s RTSP (de)muxer context
571  */
573 
574 /**
575  * Close all connection handles within the RTSP (de)muxer
576  *
577  * @param s RTSP (de)muxer context
578  */
580 
581 /**
582  * Get the description of the stream and set up the RTSPStream child
583  * objects.
584  */
586 
587 /**
588  * Announce the stream to the server and set up the RTSPStream child
589  * objects for each media stream.
590  */
592 
593 /**
594  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
595  * listen mode.
596  */
598 
599 /**
600  * Parse an SDP description of streams by populating an RTSPState struct
601  * within the AVFormatContext; also allocate the RTP streams and the
602  * pollfd array used for UDP streams.
603  */
604 int ff_sdp_parse(AVFormatContext *s, const char *content);
605 
606 /**
607  * Receive one RTP packet from an TCP interleaved RTSP stream.
608  */
610  uint8_t *buf, int buf_size);
611 
612 /**
613  * Send buffered packets over TCP.
614  */
616 
617 /**
618  * Receive one packet from the RTSPStreams set up in the AVFormatContext
619  * (which should contain a RTSPState struct as priv_data).
620  */
622 
623 /**
624  * Do the SETUP requests for each stream for the chosen
625  * lower transport mode.
626  * @return 0 on success, <0 on error, 1 if protocol is unavailable
627  */
628 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
629  int lower_transport, const char *real_challenge);
630 
631 /**
632  * Undo the effect of ff_rtsp_make_setup_request, close the
633  * transport_priv and rtp_handle fields.
634  */
635 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
636 
637 /**
638  * Open RTSP transport context.
639  */
641 
642 extern const AVOption ff_rtsp_options[];
643 
644 #endif /* AVFORMAT_RTSP_H */
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a &#39;$&#39;, stream length and stre...
Definition: rtsp.h:94
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:389
Realmedia Data Transport.
Definition: rtsp.h:59
static struct @314 state
RTSPLowerTransport
Network layer over which RTP/etc packet data will be transported.
Definition: rtsp.h:37
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:807
AVOption.
Definition: opt.h:246
HTTPS tunneled.
Definition: rtsp.h:45
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:116
Windows Media server.
Definition: rtsp.h:210
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:355
RTP/JPEG specific private data.
Definition: rdt.c:83
enum AVDiscard * real_setup
current stream setup.
Definition: rtsp.h:297
char * user_agent
User-Agent string.
Definition: rtsp.h:409
enum AVDiscard * real_setup_cache
stream setup during the last frame read.
Definition: rtsp.h:293
int mode_record
transport set to record data
Definition: rtsp.h:113
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:145
initialized and sending/receiving data
Definition: rtsp.h:198
static AVPacket pkt
RTSPClientState
Client state, i.e.
Definition: rtsp.h:196
HTTP Authentication state structure.
Definition: httpauth.h:55
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:240
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:122
This describes the server response to each RTSP command.
Definition: rtsp.h:128
RTSPTransport
Packet profile of the data that we will be receiving.
Definition: rtsp.h:57
Format I/O context.
Definition: avformat.h:1351
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:208
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:404
int recvbuf_len
Definition: rtsp.h:324
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:361
Standards-compliant RTP.
Definition: rtsp.h:58
uint8_t
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:75
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:110
AVOptions.
int initial_timeout
Timeout to wait for incoming connections.
Definition: rtsp.h:394
int rtp_muxer_flags
Option flags for the chained RTP muxer.
Definition: rtsp.h:371
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:374
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:329
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:438
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:46
RTSPServerType
Identify particular servers that require special handling, such as standards-incompliant "Transport:"...
Definition: rtsp.h:207
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:463
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:69
int nb_transports
number of items in the &#39;transports&#39; variable below
Definition: rtsp.h:135
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:178
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
Private data for the RTSP demuxer.
Definition: rtsp.h:219
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:256
const AVOption ff_rtsp_options[]
Definition: rtsp.c:84
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:251
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:739
URLContext * rtsp_hd
Definition: rtsp.h:221
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:226
uint64_t asf_pb_pos
cache for position of the asf demuxer, since we load a new data packet in the bytecontext for each in...
Definition: rtsp.h:312
int seq
RTSP command sequence number.
Definition: rtsp.h:242
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:340
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:308
int recvbuf_pos
Definition: rtsp.h:323
int nb_rtsp_streams
number of items in the &#39;rtsp_streams&#39; variable
Definition: rtsp.h:224
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:284
int content_length
length of the data following this header
Definition: rtsp.h:130
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:173
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:89
RTSP over HTTP (tunneling)
Definition: rtsp.h:70
#define s(width, name)
Definition: cbs_vp9.c:257
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
Raw data (over UDP)
Definition: rtsp.h:60
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:322
int nb_byes
Definition: rtsp.h:337
RTSPControlTransport
Transport mode for the RTSP data.
Definition: rtsp.h:68
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:429
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:384
int server_port_max
Definition: rtsp.h:106
long long int64_t
Definition: coverity.c:34
Definition: url.h:38
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:379
int client_port_max
Definition: rtsp.h:102
Describe the class of an AVClass context structure.
Definition: log.h:67
not initialized
Definition: rtsp.h:197
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:119
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:748
int max_p
Definition: rtsp.h:356
int buffer_size
Definition: rtsp.h:412
int interleaved_max
Definition: rtsp.h:94
RTSPStatusCode
RTSP handling.
Definition: rtspcodes.h:31
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:139
#define INET6_ADDRSTRLEN
Definition: network.h:237
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
Main libavformat public API header.
initialized, requesting a seek
Definition: rtsp.h:200
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:289
initialized, but not receiving data
Definition: rtsp.h:199
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we&#39;re reading data interleave...
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:771
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:399
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:277
uint64_t packets
The number of returned packets.
Definition: rtsp.h:350
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:591
AVDiscard
Definition: avcodec.h:227
Realmedia-style server.
Definition: rtsp.h:209
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:345
int pkt_size
Definition: rtsp.h:413
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:115
This structure stores compressed data.
Definition: packet.h:332
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:106
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:98
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:102
int initial_pause
Do not begin to play the stream immediately.
Definition: rtsp.h:366