From 26d6b3e230067833edb551795e834679c820d96e Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Thu, 26 Feb 2009 14:15:41 +0000 Subject: [PATCH] Document rtsp.h, see "[PATCH] document rtsp.h" thread. Originally committed as revision 17614 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavformat/rtsp.h | 202 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 172 insertions(+), 30 deletions(-) diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 94ee719519..d44926ac3a 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -27,16 +27,24 @@ #include "rtpdec.h" #include "network.h" +/** + * Network layer over which RTP/etc packet data will be transported. + */ enum RTSPLowerTransport { - RTSP_LOWER_TRANSPORT_UDP = 0, - RTSP_LOWER_TRANSPORT_TCP = 1, - RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, + 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, - RTSP_TRANSPORT_RDT, + RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ + RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ RTSP_TRANSPORT_NB }; @@ -48,36 +56,99 @@ enum RTSPTransport { #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 { - int interleaved_min, interleaved_max; /**< interleave ids, if TCP transport */ - int port_min, port_max; /**< RTP ports */ - int client_port_min, client_port_max; /**< RTP ports */ - int server_port_min, server_port_max; /**< RTP ports */ - int ttl; /**< ttl value */ + /** 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; - /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ + + /** 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 */ + + 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]; - char real_challenge[64]; /**< the RealChallenge1 field from the server */ + + /** 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]; } 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, - RTSP_STATE_PLAYING, - RTSP_STATE_PAUSED, + RTSP_STATE_IDLE, /**< not initialized */ + RTSP_STATE_PLAYING, /**< initialized and receiving data */ + RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ }; +/** + * 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 */ @@ -85,44 +156,115 @@ enum RTSPServerType { RTSP_SERVER_NB }; +/** + * Private data for the RTSP demuxer. + */ 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; + 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 */ + + 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]; + + /** 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 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]; + //@} } 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 */ - void *transport_priv; /* RTP/RDT parse context */ + URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ + void *transport_priv; /**< RTP/RDT parse context */ + + /** 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 */ + //@} - int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ - int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ - char control_url[1024]; /* url for this stream (from SDP) */ + /** 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; - int sdp_port; /* port (from SDP content - not used in RTSP) */ - struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ - int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ - int sdp_payload_type; /* payload type - only used in SDP */ - RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */ + /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */ + //@{ + /** handler structure */ + RTPDynamicProtocolHandler *dynamic_handler; - RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) - PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) + /** private data associated with the dynamic protocol */ + PayloadContext *dynamic_protocol_context; + //@} } RTSPStream; int rtsp_init(void);