libavformat/rtsp.h

   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
  30 /**
  31  * Network layer over which RTP/etc packet data will be transported.
  32  */
  33 enum RTSPLowerTransport {
  34     RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
  35     RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
  36     RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
  37     RTSP_LOWER_TRANSPORT_NB
  38 };
  39
  40 /**
  41  * Packet profile of the data that we will be receiving. Real servers
  42  * commonly send RDT (although they can sometimes send RTP as well),
  43  * whereas most others will send RTP.
  44  */
  45 enum RTSPTransport {
  46     RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
  47     RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
  48     RTSP_TRANSPORT_NB
  49 };
  50
  51 #define RTSP_DEFAULT_PORT   554
  52 #define RTSP_MAX_TRANSPORTS 8
  53 #define RTSP_TCP_MAX_PACKET_SIZE 1472
  54 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
  55 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
  56 #define RTSP_RTP_PORT_MIN 5000
  57 #define RTSP_RTP_PORT_MAX 10000
  58
  59 /**
  60  * This describes a single item in the "Transport:" line of one stream as
  61  * negotiated by the SETUP RTSP command. Multiple transports are comma-
  62  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
  63  * client_port=1000-1001;server_port=1800-1801") and described in separate
  64  * RTSPTransportFields.
  65  */
  66 typedef struct RTSPTransportField {
  67     /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
  68      * with a '$', stream length and stream ID. If the stream ID is within
  69      * the range of this interleaved_min-max, then the packet belongs to
  70      * this stream. */
  71     int interleaved_min, interleaved_max;
  72
  73     /** UDP multicast port range; the ports to which we should connect to
  74      * receive multicast UDP data. */
  75     int port_min, port_max;
  76
  77     /** UDP client ports; these should be the local ports of the UDP RTP
  78      * (and RTCP) sockets over which we receive RTP/RTCP data. */
  79     int client_port_min, client_port_max;
  80
  81     /** UDP unicast server port range; the ports to which we should connect
  82      * to receive unicast UDP RTP/RTCP data. */
  83     int server_port_min, server_port_max;
  84
  85     /** time-to-live value (required for multicast); the amount of HOPs that
  86      * packets will be allowed to make before being discarded. */
  87     int ttl;
  88
  89     uint32_t destination; /**< destination IP address */
  90
  91     /** data/packet transport protocol; e.g. RTP or RDT */
  92     enum RTSPTransport transport;
  93
  94     /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
  95     enum RTSPLowerTransport lower_transport;
  96 } RTSPTransportField;
  97
  98 /**
  99  * This describes the server response to each RTSP command.
 100  */
 101 typedef struct RTSPMessageHeader {
 102     /** length of the data following this header */
 103     int content_length;
 104
 105     enum RTSPStatusCode status_code; /**< response code from server */
 106
 107     /** number of items in the 'transports' variable below */
 108     int nb_transports;
 109
 110     /** Time range of the streams that the server will stream. In
 111      * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
 112     int64_t range_start, range_end;
 113
 114     /** describes the complete "Transport:" line of the server in response
 115      * to a SETUP RTSP command by the client */
 116     RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
 117
 118     int seq;                         /**< sequence number */
 119
 120     /** the "Session:" field. This value is initially set by the server and
 121      * should be re-transmitted by the client in every RTSP command. */
 122     char session_id[512];
 123
 124     /** the "RealChallenge1:" field from the server */
 125     char real_challenge[64];
 126
 127     /** the "Server: field, which can be used to identify some special-case
 128      * servers that are not 100% standards-compliant. We use this to identify
 129      * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
 130      * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
 131      * use something like "Helix [..] Server Version v.e.r.sion (platform)
 132      * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
 133      * where platform is the output of $uname -msr | sed 's/ /-/g'. */
 134     char server[64];
 135
 136     /** The "timeout" comes as part of the server response to the "SETUP"
 137      * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
 138      * time, in seconds, that the server will go without traffic over the
 139      * RTSP/TCP connection before it closes the connection. To prevent
 140      * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
 141      * than this value. */
 142     int timeout;
 143
 144     /** The "Notice" or "X-Notice" field value. See
 145      * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
 146      * for a complete list of supported values. */
 147     int notice;
 148 } RTSPMessageHeader;
 149
 150 /**
 151  * Client state, i.e. whether we are currently receiving data (PLAYING) or
 152  * setup-but-not-receiving (PAUSED). State can be changed in applications
 153  * by calling av_read_play/pause().
 154  */
 155 enum RTSPClientState {
 156     RTSP_STATE_IDLE,    /**< not initialized */
 157     RTSP_STATE_PLAYING, /**< initialized and receiving data */
 158     RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
 159     RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
 160 };
 161
 162 /**
 163  * Identifies particular servers that require special handling, such as
 164  * standards-incompliant "Transport:" lines in the SETUP request.
 165  */
 166 enum RTSPServerType {
 167     RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
 168     RTSP_SERVER_REAL, /**< Realmedia-style server */
 169     RTSP_SERVER_WMS,  /**< Windows Media server */
 170     RTSP_SERVER_NB
 171 };
 172
 173 /**
 174  * Private data for the RTSP demuxer.
 175  *
 176  * @todo Use ByteIOContext instead of URLContext
 177  */
 178 typedef struct RTSPState {
 179     URLContext *rtsp_hd; /* RTSP TCP connexion handle */
 180
 181     /** number of items in the 'rtsp_streams' variable */
 182     int nb_rtsp_streams;
 183
 184     struct RTSPStream **rtsp_streams; /**< streams in this session */
 185
 186     /** indicator of whether we are currently receiving data from the
 187      * server. Basically this isn't more than a simple cache of the
 188      * last PLAY/PAUSE command sent to the server, to make sure we don't
 189      * send 2x the same unexpectedly or commands in the wrong state. */
 190     enum RTSPClientState state;
 191
 192     /** the seek value requested when calling av_seek_frame(). This value
 193      * is subsequently used as part of the "Range" parameter when emitting
 194      * the RTSP PLAY command. If we are currently playing, this command is
 195      * called instantly. If we are currently paused, this command is called
 196      * whenever we resume playback. Either way, the value is only used once,
 197      * see rtsp_read_play() and rtsp_read_seek(). */
 198     int64_t seek_timestamp;
 199
 200     /* XXX: currently we use unbuffered input */
 201     //    ByteIOContext rtsp_gb;
 202
 203     int seq;                          /**< RTSP command sequence number */
 204
 205     /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
 206      * identifier that the client should re-transmit in each RTSP command */
 207     char session_id[512];
 208
 209     /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
 210      * the server will go without traffic on the RTSP/TCP line before it
 211      * closes the connection. */
 212     int timeout;
 213
 214     /** timestamp of the last RTSP command that we sent to the RTSP server.
 215      * This is used to calculate when to send dummy commands to keep the
 216      * connection alive, in conjunction with timeout. */
 217     int64_t last_cmd_time;
 218
 219     /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
 220     enum RTSPTransport transport;
 221
 222     /** the negotiated network layer transport protocol; e.g. TCP or UDP
 223      * uni-/multicast */
 224     enum RTSPLowerTransport lower_transport;
 225
 226     /** brand of server that we're talking to; e.g. WMS, REAL or other.
 227      * Detected based on the value of RTSPMessageHeader->server or the presence
 228      * of RTSPMessageHeader->real_challenge */
 229     enum RTSPServerType server_type;
 230
 231     /** base64-encoded authorization lines (username:password) */
 232     char *auth_b64;
 233
 234     /** The last reply of the server to a RTSP command */
 235     char last_reply[2048]; /* XXX: allocate ? */
 236
 237     /** RTSPStream->transport_priv of the last stream that we read a
 238      * packet from */
 239     void *cur_transport_priv;
 240
 241     /** The following are used for Real stream selection */
 242     //@{
 243     /** whether we need to send a "SET_PARAMETER Subscribe:" command */
 244     int need_subscription;
 245
 246     /** stream setup during the last frame read. This is used to detect if
 247      * we need to subscribe or unsubscribe to any new streams. */
 248     enum AVDiscard real_setup_cache[MAX_STREAMS];
 249
 250     /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
 251      * this is used to send the same "Unsubscribe:" if stream setup changed,
 252      * before sending a new "Subscribe:" command. */
 253     char last_subscription[1024];
 254     //@}
 255
 256     /** The following are used for RTP/ASF streams */
 257     //@{
 258     /** ASF demuxer context for the embedded ASF stream from WMS servers */
 259     AVFormatContext *asf_ctx;
 260
 261     /** cache for position of the asf demuxer, since we load a new
 262      * data packet in the bytecontext for each incoming RTSP packet. */
 263     uint64_t asf_pb_pos;
 264     //@}
 265 } RTSPState;
 266
 267 /**
 268  * Describes a single stream, as identified by a single m= line block in the
 269  * SDP content. In the case of RDT, one RTSPStream can represent multiple
 270  * AVStreams. In this case, each AVStream in this set has similar content
 271  * (but different codec/bitrate).
 272  */
 273 typedef struct RTSPStream {
 274     URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
 275     void *transport_priv; /**< RTP/RDT parse context */
 276
 277     /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
 278     int stream_index;
 279
 280     /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
 281      * for the selected transport. Only used for TCP. */
 282     int interleaved_min, interleaved_max;
 283
 284     char control_url[1024];   /**< url for this stream (from SDP) */
 285
 286     /** The following are used only in SDP, not RTSP */
 287     //@{
 288     int sdp_port;             /**< port (from SDP content) */
 289     struct in_addr sdp_ip;    /**< IP address (from SDP content) */
 290     int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
 291     int sdp_payload_type;     /**< payload type */
 292     //@}
 293
 294     /** rtp payload parsing infos from SDP (i.e. mapping between private
 295      * payload IDs and media-types (string), so that we can derive what
 296      * type of payload we're dealing with (and how to parse it). */
 297     RTPPayloadData rtp_payload_data;
 298
 299     /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
 300     //@{
 301     /** handler structure */
 302     RTPDynamicProtocolHandler *dynamic_handler;
 303
 304     /** private data associated with the dynamic protocol */
 305     PayloadContext *dynamic_protocol_context;
 306     //@}
 307 } RTSPStream;
 308
 309 int rtsp_init(void);
 310 void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf);
 311
 312 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
 313 extern int rtsp_default_protocols;
 314 #endif
 315 extern int rtsp_rtp_port_min;
 316 extern int rtsp_rtp_port_max;
 317
 318 int rtsp_pause(AVFormatContext *s);
 319 int rtsp_resume(AVFormatContext *s);
 320
 321 #endif /* AVFORMAT_RTSP_H */