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/*
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 * RTSP definitions
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 * Copyright (c) 2002 Fabrice Bellard
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#ifndef AVFORMAT_RTSP_H
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#define AVFORMAT_RTSP_H
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#include <stdint.h>
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#include "avformat.h"
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#include "rtspcodes.h"
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#include "rtpdec.h"
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#include "network.h"
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#include "httpauth.h"
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/**
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 * Network layer over which RTP/etc packet data will be transported.
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 */
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enum RTSPLowerTransport {
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    RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
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    RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
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    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
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    RTSP_LOWER_TRANSPORT_NB
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};
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/**
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 * Packet profile of the data that we will be receiving. Real servers
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 * commonly send RDT (although they can sometimes send RTP as well),
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 * whereas most others will send RTP.
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 */
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enum RTSPTransport {
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    RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
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    RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
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    RTSP_TRANSPORT_NB
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};
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/**
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 * Transport mode for the RTSP data. This may be plain, or
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 * tunneled, which is done over HTTP.
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 */
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enum RTSPControlTransport {
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    RTSP_MODE_PLAIN,   /**< Normal RTSP */
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    RTSP_MODE_TUNNEL   /**< RTSP over HTTP (tunneling) */
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};
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#define RTSP_DEFAULT_PORT   554
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#define RTSP_MAX_TRANSPORTS 8
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#define RTSP_TCP_MAX_PACKET_SIZE 1472
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#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
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#define RTSP_RTP_PORT_MIN 5000
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#define RTSP_RTP_PORT_MAX 10000
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/**
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 * This describes a single item in the "Transport:" line of one stream as
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 * negotiated by the SETUP RTSP command. Multiple transports are comma-
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 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
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 * client_port=1000-1001;server_port=1800-1801") and described in separate
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 * RTSPTransportFields.
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 */
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typedef struct RTSPTransportField {
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    /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
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     * with a '$', stream length and stream ID. If the stream ID is within
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     * the range of this interleaved_min-max, then the packet belongs to
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     * this stream. */
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    int interleaved_min, interleaved_max;
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    /** UDP multicast port range; the ports to which we should connect to
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     * receive multicast UDP data. */
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    int port_min, port_max;
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    /** UDP client ports; these should be the local ports of the UDP RTP
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     * (and RTCP) sockets over which we receive RTP/RTCP data. */
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    int client_port_min, client_port_max;
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    /** UDP unicast server port range; the ports to which we should connect
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     * to receive unicast UDP RTP/RTCP data. */
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    int server_port_min, server_port_max;
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    /** time-to-live value (required for multicast); the amount of HOPs that
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     * packets will be allowed to make before being discarded. */
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    int ttl;
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    struct sockaddr_storage destination; /**< destination IP address */
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    /** data/packet transport protocol; e.g. RTP or RDT */
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    enum RTSPTransport transport;
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    /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
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    enum RTSPLowerTransport lower_transport;
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} RTSPTransportField;
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/**
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 * This describes the server response to each RTSP command.
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 */
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typedef struct RTSPMessageHeader {
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    /** length of the data following this header */
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    int content_length;
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    enum RTSPStatusCode status_code; /**< response code from server */
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    /** number of items in the 'transports' variable below */
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    int nb_transports;
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    /** Time range of the streams that the server will stream. In
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     * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
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    int64_t range_start, range_end;
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    /** describes the complete "Transport:" line of the server in response
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     * to a SETUP RTSP command by the client */
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    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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    int seq;                         /**< sequence number */
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    /** the "Session:" field. This value is initially set by the server and
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     * should be re-transmitted by the client in every RTSP command. */
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    char session_id[512];
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    /** the "Location:" field. This value is used to handle redirection.
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     */
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    char location[4096];
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    /** the "RealChallenge1:" field from the server */
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    char real_challenge[64];
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    /** the "Server: field, which can be used to identify some special-case
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     * servers that are not 100% standards-compliant. We use this to identify
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     * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
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     * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
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     * use something like "Helix [..] Server Version v.e.r.sion (platform)
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     * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
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     * where platform is the output of $uname -msr | sed 's/ /-/g'. */
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    char server[64];
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    /** The "timeout" comes as part of the server response to the "SETUP"
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     * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
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     * time, in seconds, that the server will go without traffic over the
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     * RTSP/TCP connection before it closes the connection. To prevent
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     * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
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     * than this value. */
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    int timeout;
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    /** The "Notice" or "X-Notice" field value. See
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     * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
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     * for a complete list of supported values. */
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    int notice;
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    /** The "reason" is meant to specify better the meaning of the error code
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     * returned
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     */
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    char reason[256];
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} RTSPMessageHeader;
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/**
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 * Client state, i.e. whether we are currently receiving data (PLAYING) or
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 * setup-but-not-receiving (PAUSED). State can be changed in applications
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 * by calling av_read_play/pause().
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 */
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enum RTSPClientState {
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    RTSP_STATE_IDLE,    /**< not initialized */
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    RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
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    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
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    RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
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};
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/**
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 * Identifies particular servers that require special handling, such as
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 * standards-incompliant "Transport:" lines in the SETUP request.
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 */
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enum RTSPServerType {
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    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
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    RTSP_SERVER_REAL, /**< Realmedia-style server */
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    RTSP_SERVER_WMS,  /**< Windows Media server */
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    RTSP_SERVER_NB
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};
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/**
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 * Private data for the RTSP demuxer.
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 *
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 * @todo Use ByteIOContext instead of URLContext
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 */
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typedef struct RTSPState {
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    URLContext *rtsp_hd; /* RTSP TCP connection handle */
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    /** number of items in the 'rtsp_streams' variable */
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    int nb_rtsp_streams;
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    struct RTSPStream **rtsp_streams; /**< streams in this session */
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    /** indicator of whether we are currently receiving data from the
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     * server. Basically this isn't more than a simple cache of the
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     * last PLAY/PAUSE command sent to the server, to make sure we don't
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     * send 2x the same unexpectedly or commands in the wrong state. */
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    enum RTSPClientState state;
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    /** the seek value requested when calling av_seek_frame(). This value
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     * is subsequently used as part of the "Range" parameter when emitting
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     * the RTSP PLAY command. If we are currently playing, this command is
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     * called instantly. If we are currently paused, this command is called
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     * whenever we resume playback. Either way, the value is only used once,
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     * see rtsp_read_play() and rtsp_read_seek(). */
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    int64_t seek_timestamp;
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    /* XXX: currently we use unbuffered input */
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    //    ByteIOContext rtsp_gb;
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    int seq;                          /**< RTSP command sequence number */
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    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
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     * identifier that the client should re-transmit in each RTSP command */
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    char session_id[512];
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    /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
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     * the server will go without traffic on the RTSP/TCP line before it
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     * closes the connection. */
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    int timeout;
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    /** timestamp of the last RTSP command that we sent to the RTSP server.
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     * This is used to calculate when to send dummy commands to keep the
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     * connection alive, in conjunction with timeout. */
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    int64_t last_cmd_time;
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    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
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    enum RTSPTransport transport;
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    /** the negotiated network layer transport protocol; e.g. TCP or UDP
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     * uni-/multicast */
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    enum RTSPLowerTransport lower_transport;
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    /** brand of server that we're talking to; e.g. WMS, REAL or other.
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     * Detected based on the value of RTSPMessageHeader->server or the presence
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     * of RTSPMessageHeader->real_challenge */
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    enum RTSPServerType server_type;
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    /** plaintext authorization line (username:password) */
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    char auth[128];
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    /** authentication state */
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    HTTPAuthState auth_state;
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    /** The last reply of the server to a RTSP command */
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    char last_reply[2048]; /* XXX: allocate ? */
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    /** RTSPStream->transport_priv of the last stream that we read a
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     * packet from */
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    void *cur_transport_priv;
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    /** The following are used for Real stream selection */
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    //@{
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    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
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    int need_subscription;
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    /** stream setup during the last frame read. This is used to detect if
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     * we need to subscribe or unsubscribe to any new streams. */
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    enum AVDiscard *real_setup_cache;
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    /** current stream setup. This is a temporary buffer used to compare
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     * current setup to previous frame setup. */
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    enum AVDiscard *real_setup;
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    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
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     * this is used to send the same "Unsubscribe:" if stream setup changed,
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     * before sending a new "Subscribe:" command. */
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    char last_subscription[1024];
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    //@}
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    /** The following are used for RTP/ASF streams */
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    //@{
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    /** ASF demuxer context for the embedded ASF stream from WMS servers */
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    AVFormatContext *asf_ctx;
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    /** cache for position of the asf demuxer, since we load a new
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     * data packet in the bytecontext for each incoming RTSP packet. */
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    uint64_t asf_pb_pos;
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    //@}
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    /** some MS RTSP streams contain a URL in the SDP that we need to use
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     * for all subsequent RTSP requests, rather than the input URI; in
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     * other cases, this is a copy of AVFormatContext->filename. */
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    char control_uri[1024];
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    /** The synchronized start time of the output streams. */
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    int64_t start_time;
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    /** Additional output handle, used when input and output are done
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     * separately, eg for HTTP tunneling. */
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    URLContext *rtsp_hd_out;
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    /** RTSP transport mode, such as plain or tunneled. */
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    enum RTSPControlTransport control_transport;
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} RTSPState;
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/**
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 * Describes a single stream, as identified by a single m= line block in the
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 * SDP content. In the case of RDT, one RTSPStream can represent multiple
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 * AVStreams. In this case, each AVStream in this set has similar content
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 * (but different codec/bitrate).
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 */
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typedef struct RTSPStream {
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    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
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    void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
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    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
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    int stream_index;
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    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
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     * for the selected transport. Only used for TCP. */
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    int interleaved_min, interleaved_max;
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    char control_url[1024];   /**< url for this stream (from SDP) */
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    /** The following are used only in SDP, not RTSP */
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    //@{
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    int sdp_port;             /**< port (from SDP content) */
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    struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
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    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
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    int sdp_payload_type;     /**< payload type */
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    //@}
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    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
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    //@{
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    /** handler structure */
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    RTPDynamicProtocolHandler *dynamic_handler;
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    /** private data associated with the dynamic protocol */
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    PayloadContext *dynamic_protocol_context;
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    //@}
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} RTSPStream;
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void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
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                        HTTPAuthState *auth_state);
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#if LIBAVFORMAT_VERSION_INT < (53 << 16)
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extern int rtsp_default_protocols;
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#endif
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extern int rtsp_rtp_port_min;
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extern int rtsp_rtp_port_max;
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/**
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 * Send a command to the RTSP server without waiting for the reply.
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 *
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 * @param s RTSP (de)muxer context
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 * @param method the method for the request
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 * @param url the target url for the request
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 * @param headers extra header lines to include in the request
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 * @param send_content if non-null, the data to send as request body content
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 * @param send_content_length the length of the send_content data, or 0 if
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 *                            send_content is null
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 *
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 * @return zero if success, nonzero otherwise
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 */
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int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
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                                        const char *method, const char *url,
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                                        const char *headers,
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                                        const unsigned char *send_content,
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                                        int send_content_length);
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/**
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 * Send a command to the RTSP server without waiting for the reply.
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 *
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 * @see rtsp_send_cmd_with_content_async
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 */
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int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
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                           const char *url, const char *headers);
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/**
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 * Send a command to the RTSP server and wait for the reply.
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 *
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 * @param s RTSP (de)muxer context
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 * @param method the method for the request
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 * @param url the target url for the request
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 * @param headers extra header lines to include in the request
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 * @param reply pointer where the RTSP message header will be stored
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 * @param content_ptr pointer where the RTSP message body, if any, will
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 *                    be stored (length is in reply)
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 * @param send_content if non-null, the data to send as request body content
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 * @param send_content_length the length of the send_content data, or 0 if
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 *                            send_content is null
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 *
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 * @return zero if success, nonzero otherwise
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 */
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int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
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                                  const char *method, const char *url,
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                                  const char *headers,
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                                  RTSPMessageHeader *reply,
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                                  unsigned char **content_ptr,
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                                  const unsigned char *send_content,
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                                  int send_content_length);
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/**
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 * Send a command to the RTSP server and wait for the reply.
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 *
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 * @see rtsp_send_cmd_with_content
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 */
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int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
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                     const char *url, const char *headers,
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                     RTSPMessageHeader *reply, unsigned char **content_ptr);
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/**
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 * Read a RTSP message from the server, or prepare to read data
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 * packets if we're reading data interleaved over the TCP/RTSP
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 * connection as well.
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 *
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 * @param s RTSP (de)muxer context
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 * @param reply pointer where the RTSP message header will be stored
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 * @param content_ptr pointer where the RTSP message body, if any, will
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 *                    be stored (length is in reply)
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 * @param return_on_interleaved_data whether the function may return if we
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 *                   encounter a data marker ('$'), which precedes data
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 *                   packets over interleaved TCP/RTSP connections. If this
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 *                   is set, this function will return 1 after encountering
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 *                   a '$'. If it is not set, the function will skip any
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 *                   data packets (if they are encountered), until a reply
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 *                   has been fully parsed. If no more data is available
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 *                   without parsing a reply, it will return an error.
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 *
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 * @return 1 if a data packets is ready to be received, -1 on error,
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 *          and 0 on success.
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 */
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int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
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                       unsigned char **content_ptr,
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                       int return_on_interleaved_data);
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/**
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 * Skip a RTP/TCP interleaved packet.
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 */
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void ff_rtsp_skip_packet(AVFormatContext *s);
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/**
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 * Connect to the RTSP server and set up the individual media streams.
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 * This can be used for both muxers and demuxers.
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 *
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 * @param s RTSP (de)muxer context
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 *
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 * @return 0 on success, < 0 on error. Cleans up all allocations done
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 *          within the function on error.
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 */
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int ff_rtsp_connect(AVFormatContext *s);
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/**
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 * Close and free all streams within the RTSP (de)muxer
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 *
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 * @param s RTSP (de)muxer context
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 */
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void ff_rtsp_close_streams(AVFormatContext *s);
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/**
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 * Close all connection handles within the RTSP (de)muxer
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 *
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 * @param rt RTSP (de)muxer context
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 */
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void ff_rtsp_close_connections(AVFormatContext *rt);
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#endif /* AVFORMAT_RTSP_H */