text: /*
   * RTSP definitions
   * Copyright (c) 2002 Fabrice Bellard
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  #ifndef AVFORMAT_RTSP_H
  #define AVFORMAT_RTSP_H
  
  #include <stdint.h>
  #include "avformat.h"
  #include "rtspcodes.h"
  #include "rtpdec.h"
  #include "network.h"
  #include "httpauth.h"
  
  /**
   * Network layer over which RTP/etc packet data will be transported.
   */
  enum RTSPLowerTransport {
      RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
      RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
      RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
      RTSP_LOWER_TRANSPORT_NB
  };
  
  /**
   * Packet profile of the data that we will be receiving. Real servers
   * commonly send RDT (although they can sometimes send RTP as well),
   * whereas most others will send RTP.
   */
  enum RTSPTransport {
      RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
      RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
      RTSP_TRANSPORT_NB
  };
  
  #define RTSP_DEFAULT_PORT   554
  #define RTSP_MAX_TRANSPORTS 8
  #define RTSP_TCP_MAX_PACKET_SIZE 1472
  #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
  #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
  #define RTSP_RTP_PORT_MIN 5000
  #define RTSP_RTP_PORT_MAX 10000
  
  /**
   * This describes a single item in the "Transport:" line of one stream as
   * negotiated by the SETUP RTSP command. Multiple transports are comma-
   * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
   * client_port=1000-1001;server_port=1800-1801") and described in separate
   * RTSPTransportFields.
   */
  typedef struct RTSPTransportField {
      /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
       * with a '$', stream length and stream ID. If the stream ID is within
       * the range of this interleaved_min-max, then the packet belongs to
       * this stream. */
      int interleaved_min, interleaved_max;
  
      /** UDP multicast port range; the ports to which we should connect to
       * receive multicast UDP data. */
      int port_min, port_max;
  
      /** UDP client ports; these should be the local ports of the UDP RTP
       * (and RTCP) sockets over which we receive RTP/RTCP data. */
      int client_port_min, client_port_max;
  
      /** UDP unicast server port range; the ports to which we should connect
       * to receive unicast UDP RTP/RTCP data. */
      int server_port_min, server_port_max;
  
      /** time-to-live value (required for multicast); the amount of HOPs that
       * packets will be allowed to make before being discarded. */
      int ttl;
  
      uint32_t destination; /**< destination IP address */
  
      /** data/packet transport protocol; e.g. RTP or RDT */
      enum RTSPTransport transport;
  
      /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
      enum RTSPLowerTransport lower_transport;
  } RTSPTransportField;
  
  /**
   * This describes the server response to each RTSP command.
   */
  typedef struct RTSPMessageHeader {
      /** length of the data following this header */
      int content_length;
  
      enum RTSPStatusCode status_code; /**< response code from server */
  
      /** number of items in the 'transports' variable below */
      int nb_transports;
  
      /** Time range of the streams that the server will stream. In
       * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
      int64_t range_start, range_end;
  
      /** describes the complete "Transport:" line of the server in response
       * to a SETUP RTSP command by the client */
      RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
  
      int seq;                         /**< sequence number */
  
      /** the "Session:" field. This value is initially set by the server and
       * should be re-transmitted by the client in every RTSP command. */
      char session_id[512];
  
      /** the "Location:" field. This value is used to handle redirection.
       */
      char location[4096];
  
      /** the "RealChallenge1:" field from the server */
      char real_challenge[64];
  
      /** the "Server: field, which can be used to identify some special-case
       * servers that are not 100% standards-compliant. We use this to identify
       * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
       * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
       * use something like "Helix [..] Server Version v.e.r.sion (platform)
       * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
       * where platform is the output of $uname -msr | sed 's/ /-/g'. */
      char server[64];
  
      /** The "timeout" comes as part of the server response to the "SETUP"
       * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
       * time, in seconds, that the server will go without traffic over the
       * RTSP/TCP connection before it closes the connection. To prevent
       * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
       * than this value. */
      int timeout;
  
      /** The "Notice" or "X-Notice" field value. See
       * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
       * for a complete list of supported values. */
      int notice;
  } RTSPMessageHeader;
  
  /**
   * Client state, i.e. whether we are currently receiving data (PLAYING) or
   * setup-but-not-receiving (PAUSED). State can be changed in applications
   * by calling av_read_play/pause().
   */
  enum RTSPClientState {
      RTSP_STATE_IDLE,    /**< not initialized */
      RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
      RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
      RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
  };
  
  /**
   * Identifies particular servers that require special handling, such as
   * standards-incompliant "Transport:" lines in the SETUP request.
   */
  enum RTSPServerType {
      RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
      RTSP_SERVER_REAL, /**< Realmedia-style server */
      RTSP_SERVER_WMS,  /**< Windows Media server */
      RTSP_SERVER_NB
  };
  
  /**
   * Private data for the RTSP demuxer.
   *
   * @todo Use ByteIOContext instead of URLContext
   */
  typedef struct RTSPState {
      URLContext *rtsp_hd; /* RTSP TCP connexion handle */
  
      /** number of items in the 'rtsp_streams' variable */
      int nb_rtsp_streams;
  
      struct RTSPStream **rtsp_streams; /**< streams in this session */
  
      /** indicator of whether we are currently receiving data from the
       * server. Basically this isn't more than a simple cache of the
       * last PLAY/PAUSE command sent to the server, to make sure we don't
       * send 2x the same unexpectedly or commands in the wrong state. */
      enum RTSPClientState state;
  
      /** the seek value requested when calling av_seek_frame(). This value
       * is subsequently used as part of the "Range" parameter when emitting
       * the RTSP PLAY command. If we are currently playing, this command is
       * called instantly. If we are currently paused, this command is called
       * whenever we resume playback. Either way, the value is only used once,
       * see rtsp_read_play() and rtsp_read_seek(). */
      int64_t seek_timestamp;
  
      /* XXX: currently we use unbuffered input */
      //    ByteIOContext rtsp_gb;
  
      int seq;                          /**< RTSP command sequence number */
  
      /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
       * identifier that the client should re-transmit in each RTSP command */
      char session_id[512];
  
      /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
       * the server will go without traffic on the RTSP/TCP line before it
       * closes the connection. */
      int timeout;
  
      /** timestamp of the last RTSP command that we sent to the RTSP server.
       * This is used to calculate when to send dummy commands to keep the
       * connection alive, in conjunction with timeout. */
      int64_t last_cmd_time;
  
      /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
      enum RTSPTransport transport;
  
      /** the negotiated network layer transport protocol; e.g. TCP or UDP
       * uni-/multicast */
      enum RTSPLowerTransport lower_transport;
  
      /** brand of server that we're talking to; e.g. WMS, REAL or other.
       * Detected based on the value of RTSPMessageHeader->server or the presence
       * of RTSPMessageHeader->real_challenge */
      enum RTSPServerType server_type;
  
      /** plaintext authorization line (username:password) */
      char auth[128];
  
      /** authentication state */
      HTTPAuthState auth_state;
  
      /** The last reply of the server to a RTSP command */
      char last_reply[2048]; /* XXX: allocate ? */
  
      /** RTSPStream->transport_priv of the last stream that we read a
       * packet from */
      void *cur_transport_priv;
  
      /** The following are used for Real stream selection */
      //@{
      /** whether we need to send a "SET_PARAMETER Subscribe:" command */
      int need_subscription;
  
      /** stream setup during the last frame read. This is used to detect if
       * we need to subscribe or unsubscribe to any new streams. */
      enum AVDiscard real_setup_cache[MAX_STREAMS];
  
      /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
       * this is used to send the same "Unsubscribe:" if stream setup changed,
       * before sending a new "Subscribe:" command. */
      char last_subscription[1024];
      //@}
  
      /** The following are used for RTP/ASF streams */
      //@{
      /** ASF demuxer context for the embedded ASF stream from WMS servers */
      AVFormatContext *asf_ctx;
  
      /** cache for position of the asf demuxer, since we load a new
       * data packet in the bytecontext for each incoming RTSP packet. */
      uint64_t asf_pb_pos;
      //@}
  
      /** some MS RTSP streams contain a URL in the SDP that we need to use
       * for all subsequent RTSP requests, rather than the input URI; in
       * other cases, this is a copy of AVFormatContext->filename. */
      char control_uri[1024];
  
      /** The synchronized start time of the output streams. */
      int64_t start_time;
  } RTSPState;
  
  /**
   * Describes a single stream, as identified by a single m= line block in the
   * SDP content. In the case of RDT, one RTSPStream can represent multiple
   * AVStreams. In this case, each AVStream in this set has similar content
   * (but different codec/bitrate).
   */
  typedef struct RTSPStream {
      URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
      void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
  
      /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
      int stream_index;
  
      /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
       * for the selected transport. Only used for TCP. */
      int interleaved_min, interleaved_max;
  
      char control_url[1024];   /**< url for this stream (from SDP) */
  
      /** The following are used only in SDP, not RTSP */
      //@{
      int sdp_port;             /**< port (from SDP content) */
      struct in_addr sdp_ip;    /**< IP address (from SDP content) */
      int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
      int sdp_payload_type;     /**< payload type */
      //@}
  
      /** rtp payload parsing infos from SDP (i.e. mapping between private
       * payload IDs and media-types (string), so that we can derive what
       * type of payload we're dealing with (and how to parse it). */
      RTPPayloadData rtp_payload_data;
  
      /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
      //@{
      /** handler structure */
      RTPDynamicProtocolHandler *dynamic_handler;
  
      /** private data associated with the dynamic protocol */
      PayloadContext *dynamic_protocol_context;
      //@}
  } RTSPStream;
  
  void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
                          HTTPAuthState *auth_state);
  
  #if LIBAVFORMAT_VERSION_INT < (53 << 16)
  extern int rtsp_default_protocols;
  #endif
  extern int rtsp_rtp_port_min;
  extern int rtsp_rtp_port_max;
  
  /**
   * Send a command to the RTSP server without waiting for the reply.
   *
   * @param s RTSP (de)muxer context
   * @param method the method for the request
   * @param url the target url for the request
   * @param headers extra header lines to include in the request
   * @param send_content if non-null, the data to send as request body content
   * @param send_content_length the length of the send_content data, or 0 if
   *                            send_content is null
   */
  void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
                                           const char *method, const char *url,
                                           const char *headers,
                                           const unsigned char *send_content,
                                           int send_content_length);
  /**
   * Send a command to the RTSP server without waiting for the reply.
   *
   * @see rtsp_send_cmd_with_content_async
   */
  void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
                              const char *url, const char *headers);
  
  /**
   * Send a command to the RTSP server and wait for the reply.
   *
   * @param s RTSP (de)muxer context
   * @param method the method for the request
   * @param url the target url for the request
   * @param headers extra header lines to include in the request
   * @param reply pointer where the RTSP message header will be stored
   * @param content_ptr pointer where the RTSP message body, if any, will
   *                    be stored (length is in reply)
   * @param send_content if non-null, the data to send as request body content
   * @param send_content_length the length of the send_content data, or 0 if
   *                            send_content is null
   */
  void 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.
   *
   * @see rtsp_send_cmd_with_content
   */
  void ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
                        const char *url, const char *headers,
                        RTSPMessageHeader *reply, unsigned char **content_ptr);
  
  /**
   * Read a RTSP message from the server, or prepare to read data
   * packets if we're reading data interleaved over the TCP/RTSP
   * connection as well.
   *
   * @param s RTSP (de)muxer context
   * @param reply pointer where the RTSP message header will be stored
   * @param content_ptr pointer where the RTSP message body, if any, will
   *                    be stored (length is in reply)
   * @param return_on_interleaved_data whether the function may return if we
   *                   encounter a data marker ('$'), which precedes data
   *                   packets over interleaved TCP/RTSP connections. If this
   *                   is set, this function will return 1 after encountering
   *                   a '$'. If it is not set, the function will skip any
   *                   data packets (if they are encountered), until a reply
   *                   has been fully parsed. If no more data is available
   *                   without parsing a reply, it will return an error.
   *
   * @return 1 if a data packets is ready to be received, -1 on error,
   *          and 0 on success.
   */
  int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
                         unsigned char **content_ptr,
                         int return_on_interleaved_data);
  
  /**
   * Skip a RTP/TCP interleaved packet.
   */
  void ff_rtsp_skip_packet(AVFormatContext *s);
  
  /**
   * Connect to the RTSP server and set up the individual media streams.
   * This can be used for both muxers and demuxers.
   *
   * @param s RTSP (de)muxer context
   *
   * @return 0 on success, < 0 on error. Cleans up all allocations done
   *          within the function on error.
   */
  int ff_rtsp_connect(AVFormatContext *s);
  
  /**
   * Close and free all streams within the RTSP (de)muxer
   *
   * @param s RTSP (de)muxer context
   */
  void ff_rtsp_close_streams(AVFormatContext *s);
  
  #endif /* AVFORMAT_RTSP_H */
  