Document rtsp.h, see "[PATCH] document rtsp.h" thread.

Originally committed as revision 17614 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files 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);