/*
  * 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"

 #include "libavutil/log.h"

 #include "libavutil/opt.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,
     RTSP_LOWER_TRANSPORT_HTTP = 8,          /**< HTTP tunneled - not a proper
                                                  transport mode as such,
                                                  only for use via AVOptions */

     RTSP_LOWER_TRANSPORT_CUSTOM = 16,       /**< Custom IO - not a public
                                                  option for lower_transport_mask,
                                                  but set in the SDP demuxer based
                                                  on a flag. */

 };
 

 /**
  * 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_RAW, /**< Raw data (over UDP) */

     RTSP_TRANSPORT_NB

 };
 

 /**
  * Transport mode for the RTSP data. This may be plain, or
  * tunneled, which is done over HTTP.
  */
 enum RTSPControlTransport {
     RTSP_MODE_PLAIN,   /**< Normal RTSP */
     RTSP_MODE_TUNNEL   /**< RTSP over HTTP (tunneling) */
 };
 

 #define RTSP_DEFAULT_PORT   554

 #define RTSPS_DEFAULT_PORT  322

 #define RTSP_MAX_TRANSPORTS 8

 #define RTSP_TCP_MAX_PACKET_SIZE 1472

 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1

 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
 #define RTSP_RTP_PORT_MIN 5000

 #define RTSP_RTP_PORT_MAX 65000

 /**
  * 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;
 

     /** transport set to record data */
     int mode_record;
 

     struct sockaddr_storage destination; /**< destination IP address */

     char source[INET6_ADDRSTRLEN + 1]; /**< source 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;

 
     /** The "reason" is meant to specify better the meaning of the error code
      * returned
      */
     char reason[256];

 
     /**
      * Content type header
      */
     char content_type[64];

 } 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 */

 };
 

 /**

  * Identify 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 AVIOContext instead of URLContext

  */

 typedef struct RTSPState {

     const AVClass *class;             /**< Class for private options. */

     URLContext *rtsp_hd; /* RTSP TCP connection 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;
 

     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;

     /** the "RealChallenge1:" field from the server */
     char real_challenge[64];
 

     /** 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;
 
     /** current stream setup. This is a temporary buffer used to compare
      * current setup to previous frame setup. */
     enum AVDiscard *real_setup;

 
     /** 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 following are used for parsing raw mpegts in udp */
     //@{
     struct MpegTSContext *ts;
     int recvbuf_pos;
     int recvbuf_len;
     //@}
 

     /** Additional output handle, used when input and output are done
      * separately, eg for HTTP tunneling. */
     URLContext *rtsp_hd_out;

 
     /** RTSP transport mode, such as plain or tunneled. */
     enum RTSPControlTransport control_transport;

 
     /* Number of RTCP BYE packets the RTSP session has received.
      * An EOF is propagated back if nb_byes == nb_streams.
      * This is reset after a seek. */
     int nb_byes;

 
     /** Reusable buffer for receiving packets */
     uint8_t* recvbuf;

     /**
      * A mask with all requested transport methods
      */
     int lower_transport_mask;
 
     /**
      * The number of returned packets
      */
     uint64_t packets;

 
     /**
      * Polling array for udp
      */
     struct pollfd *p;

     int max_p;

 
     /**
      * Whether the server supports the GET_PARAMETER method.
      */
     int get_parameter_supported;

 
     /**
      * Do not begin to play the stream immediately.
      */
     int initial_pause;

 
     /**
      * Option flags for the chained RTP muxer.
      */
     int rtp_muxer_flags;

 
     /** Whether the server accepts the x-Dynamic-Rate header */
     int accept_dynamic_rate;

 
     /**
      * Various option flags for the RTSP muxer/demuxer.
      */
     int rtsp_flags;

 
     /**
      * Mask of all requested media types
      */
     int media_type_mask;

 
     /**
      * Minimum and maximum local UDP ports.
      */
     int rtp_port_min, rtp_port_max;

 
     /**
      * Timeout to wait for incoming connections.
      */
     int initial_timeout;

 
     /**

      * timeout of socket i/o operations.
      */
     int stimeout;
 
     /**

      * Size of RTP packet reordering queue.
      */
     int reordering_queue_size;

 
     /**
      * User-Agent string
      */
     char *user_agent;

 
     char default_lang[4];

     int buffer_size;

 } RTSPState;
 

 #define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
                                           receive packets only from the right
                                           source address and port. */

 #define RTSP_FLAG_LISTEN      0x2    /**< Wait for incoming connections. */

 #define RTSP_FLAG_CUSTOM_IO   0x4    /**< Do all IO via the AVIOContext. */

 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
                                           address of received packets. */

 #define RTSP_FLAG_PREFER_TCP  0x10   /**< Try RTP via TCP first if possible. */

 typedef struct RTSPSource {
     char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
 } RTSPSource;
 

 /**

  * Describe 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 sockaddr_storage sdp_ip; /**< IP address (from SDP content) */

     int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
     struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
     int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
     struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */

     int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
     int sdp_payload_type;     /**< payload type */
     //@}

     /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */

     //@{
     /** handler structure */
     RTPDynamicProtocolHandler *dynamic_handler;

     /** private data associated with the dynamic protocol */
     PayloadContext *dynamic_protocol_context;
     //@}

 
     /** Enable sending RTCP feedback messages according to RFC 4585 */
     int feedback;

     /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
     uint32_t ssrc;
 

     char crypto_suite[40];
     char crypto_params[100];

 } RTSPStream;
 

 void ff_rtsp_parse_line(AVFormatContext *s,
                         RTSPMessageHeader *reply, const char *buf,

                         RTSPState *rt, const char *method);

 /**
  * Send a command to the RTSP server without waiting for the reply.
  *
  * @see rtsp_send_cmd_with_content_async
  */

 int 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

  *
  * @return zero if success, nonzero otherwise

  */

 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.
  *
  * @see rtsp_send_cmd_with_content
  */

 int 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.

  * @param method the RTSP method this is a reply to. This affects how
  *               some response headers are acted upon. May be NULL.

  *

  * @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, const char *method);

 
 /**

  * 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);

 /**
  * Close all connection handles within the RTSP (de)muxer
  *

  * @param s RTSP (de)muxer context

  */

 void ff_rtsp_close_connections(AVFormatContext *s);

 /**

  * Get the description of the stream and set up the RTSPStream child
  * objects.
  */
 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
 
 /**

  * Announce the stream to the server and set up the RTSPStream child
  * objects for each media stream.
  */
 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
 

 /**

  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
  * listen mode.
  */
 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
 
 /**

  * Parse an SDP description of streams by populating an RTSPState struct
  * within the AVFormatContext; also allocate the RTP streams and the
  * pollfd array used for UDP streams.

  */
 int ff_sdp_parse(AVFormatContext *s, const char *content);
 
 /**
  * Receive one RTP packet from an TCP interleaved RTSP stream.
  */
 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
                             uint8_t *buf, int buf_size);
 
 /**

  * Send buffered packets over TCP.
  */
 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
 
 /**

  * Receive one packet from the RTSPStreams set up in the AVFormatContext
  * (which should contain a RTSPState struct as priv_data).
  */
 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
 

 /**
  * Do the SETUP requests for each stream for the chosen
  * lower transport mode.

  * @return 0 on success, <0 on error, 1 if protocol is unavailable

  */
 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
                                int lower_transport, const char *real_challenge);
 

 /**
  * Undo the effect of ff_rtsp_make_setup_request, close the
  * transport_priv and rtp_handle fields.
  */

 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);

 /**
  * Open RTSP transport context.
  */
 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
 

 extern const AVOption ff_rtsp_options[];
 

 #endif /* AVFORMAT_RTSP_H */