annotate rtsp.h @ 6085:72c7c3d5c4e9 libavformat

matroskaenc: Mux clusters better Start them on keyframes when reasonable, and delay writing audio packets to help ensure that there's audio samples available for the first frame in clusters. Patch by James Zern <jzern at google>

/*
 * 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 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;

    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 */