PeerStreamer

/*

 * RTSP definitions

 * Copyright (c) 2002 Fabrice Bellard

 *

 * This file is part of Libav.

 *

 * Libav 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.

 *

 * Libav 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 Libav; 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

};

/**

 * 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 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 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;

    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];

} 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 AVIOContext instead of URLContext

 */

typedef struct RTSPState {

    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;

    /* XXX: currently we use unbuffered input */

    //    AVIOContext 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;

    /** 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];

    /** 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;

    /** Filter incoming UDP packets - receive packets only from the right

     * source address and port. */

    int filter_source;

    /**

     * 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;

} 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 sockaddr_storage sdp_ip; /**< IP address (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 (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,

                        RTSPState *rt, const char *method);

extern int rtsp_rtp_port_min;

extern int rtsp_rtp_port_max;

/**

 * 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 rt RTSP (de)muxer context

 */

void ff_rtsp_close_connections(AVFormatContext *rt);

/**

 * 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 a SDP description of streams by populating an RTSPState struct

 * within the AVFormatContext.

 */

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);

/**

 * 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.

 */

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);

#endif /* AVFORMAT_RTSP_H */