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Document rtsp.h, see "[PATCH] document rtsp.h" thread.
Originally committed as revision 17614 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Ronald S. Bultje
parent
34583e1bbe
commit
26d6b3e230
@ -27,16 +27,24 @@
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#include "rtpdec.h"
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#include "network.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,
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RTSP_LOWER_TRANSPORT_TCP = 1,
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RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
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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,
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RTSP_TRANSPORT_RDT,
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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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@ -48,36 +56,99 @@ enum RTSPTransport {
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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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int interleaved_min, interleaved_max; /**< interleave ids, if TCP transport */
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int port_min, port_max; /**< RTP ports */
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int client_port_min, client_port_max; /**< RTP ports */
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int server_port_min, server_port_max; /**< RTP ports */
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int ttl; /**< ttl value */
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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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uint32_t 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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/** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
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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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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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char real_challenge[64]; /**< the RealChallenge1 field from the server */
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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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} 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,
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RTSP_STATE_PLAYING,
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RTSP_STATE_PAUSED,
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RTSP_STATE_IDLE, /**< not initialized */
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RTSP_STATE_PLAYING, /**< initialized and receiving data */
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RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
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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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@ -85,44 +156,115 @@ enum RTSPServerType {
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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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typedef struct RTSPState {
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams;
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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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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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/** 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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/** 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[MAX_STREAMS];
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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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} 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 */
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void *transport_priv; /* RTP/RDT parse context */
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URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
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void *transport_priv; /**< RTP/RDT parse context */
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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char control_url[1024]; /* url for this stream (from SDP) */
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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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int sdp_port; /* port (from SDP content - not used in RTSP) */
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
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int sdp_payload_type; /* payload type - only used in SDP */
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RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
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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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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
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PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
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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 in_addr 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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/** rtp payload parsing infos from SDP (i.e. mapping between private
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* payload IDs and media-types (string), so that we can derive what
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* type of payload we're dealing with (and how to parse it). */
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RTPPayloadData rtp_payload_data;
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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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int rtsp_init(void);
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