1. ffmpeg
  2. Commit 26d6b3e230067833edb551795e834679c820d96e
Browse code

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

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

Ronald S. Bultje authored on 2009/02/26 23:15:41
Showing 1 changed files
libavformat/rtsp.h
History View file @ 26d6b3e
... ... 
@@ -27,16 +27,24 @@
27 27 
 #include "rtpdec.h"
28 28 
 #include "network.h"
29 29
30 
+/**
31 
+ * Network layer over which RTP/etc packet data will be transported.
32 
+ */
30 33 
 enum RTSPLowerTransport {
31 
-    RTSP_LOWER_TRANSPORT_UDP = 0,
32 
-    RTSP_LOWER_TRANSPORT_TCP = 1,
33 
-    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
34 
+    RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
35 
+    RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
36 
+    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
34 37 
     RTSP_LOWER_TRANSPORT_NB
35 38 
 };
36 39
40 
+/**
41 
+ * Packet profile of the data that we will be receiving. Real servers
42 
+ * commonly send RDT (although they can sometimes send RTP as well),
43 
+ * whereas most others will send RTP.
44 
+ */
37 45 
 enum RTSPTransport {
38 
-    RTSP_TRANSPORT_RTP,
39 
-    RTSP_TRANSPORT_RDT,
46 
+    RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
47 
+    RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
40 48 
     RTSP_TRANSPORT_NB
41 49 
 };
42 50
... ... 
@@ -48,36 +56,99 @@ enum RTSPTransport {
48 48 
 #define RTSP_RTP_PORT_MIN 5000
49 49 
 #define RTSP_RTP_PORT_MAX 10000
50 50
51 
+/**
52 
+ * This describes a single item in the "Transport:" line of one stream as
53 
+ * negotiated by the SETUP RTSP command. Multiple transports are comma-
54 
+ * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
55 
+ * client_port=1000-1001;server_port=1800-1801") and described in separate
56 
+ * RTSPTransportFields.
57 
+ */
51 58 
 typedef struct RTSPTransportField {
52 
-    int interleaved_min, interleaved_max;  /**< interleave ids, if TCP transport */
53 
-    int port_min, port_max; /**< RTP ports */
54 
-    int client_port_min, client_port_max; /**< RTP ports */
55 
-    int server_port_min, server_port_max; /**< RTP ports */
56 
-    int ttl; /**< ttl value */
59 
+    /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
60 
+     * with a '$', stream length and stream ID. If the stream ID is within
61 
+     * the range of this interleaved_min-max, then the packet belongs to
62 
+     * this stream. */
63 
+    int interleaved_min, interleaved_max;
64 
+
65 
+    /** UDP multicast port range; the ports to which we should connect to
66 
+     * receive multicast UDP data. */
67 
+    int port_min, port_max;
68 
+
69 
+    /** UDP client ports; these should be the local ports of the UDP RTP
70 
+     * (and RTCP) sockets over which we receive RTP/RTCP data. */
71 
+    int client_port_min, client_port_max;
72 
+
73 
+    /** UDP unicast server port range; the ports to which we should connect
74 
+     * to receive unicast UDP RTP/RTCP data. */
75 
+    int server_port_min, server_port_max;
76 
+
77 
+    /** time-to-live value (required for multicast); the amount of HOPs that
78 
+     * packets will be allowed to make before being discarded. */
79 
+    int ttl;
80 
+
57 81 
     uint32_t destination; /**< destination IP address */
82 
+
83 
+    /** data/packet transport protocol; e.g. RTP or RDT */
58 84 
     enum RTSPTransport transport;
85 
+
86 
+    /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
59 87 
     enum RTSPLowerTransport lower_transport;
60 88 
 } RTSPTransportField;
61 89
90 
+/**
91 
+ * This describes the server response to each RTSP command.
92 
+ */
62 93 
 typedef struct RTSPMessageHeader {
94 
+    /** length of the data following this header */
63 95 
     int content_length;
96 
+
64 97 
     enum RTSPStatusCode status_code; /**< response code from server */
98 
+
99 
+    /** number of items in the 'transports' variable below */
65 100 
     int nb_transports;
66 
-    /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
101 
+
102 
+    /** Time range of the streams that the server will stream. In
103 
+     * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
67 104 
     int64_t range_start, range_end;
105 
+
106 
+    /** describes the complete "Transport:" line of the server in response
107 
+     * to a SETUP RTSP command by the client */
68 108 
     RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
69 
-    int seq; /**< sequence number */
109 
+
110 
+    int seq;                         /**< sequence number */
111 
+
112 
+    /** the "Session:" field. This value is initially set by the server and
113 
+     * should be re-transmitted by the client in every RTSP command. */
70 114 
     char session_id[512];
71 
-    char real_challenge[64]; /**< the RealChallenge1 field from the server */
115 
+
116 
+    /** the "RealChallenge1:" field from the server */
117 
+    char real_challenge[64];
118 
+
119 
+    /** the "Server: field, which can be used to identify some special-case
120 
+     * servers that are not 100% standards-compliant. We use this to identify
121 
+     * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
122 
+     * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
123 
+     * use something like "Helix [..] Server Version v.e.r.sion (platform)
124 
+     * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
125 
+     * where platform is the output of $uname -msr | sed 's/ /-/g'. */
72 126 
     char server[64];
73 127 
 } RTSPMessageHeader;
74 128
129 
+/**
130 
+ * Client state, i.e. whether we are currently receiving data (PLAYING) or
131 
+ * setup-but-not-receiving (PAUSED). State can be changed in applications
132 
+ * by calling av_read_play/pause().
133 
+ */
75 134 
 enum RTSPClientState {
76 
-    RTSP_STATE_IDLE,
77 
-    RTSP_STATE_PLAYING,
78 
-    RTSP_STATE_PAUSED,
135 
+    RTSP_STATE_IDLE,    /**< not initialized */
136 
+    RTSP_STATE_PLAYING, /**< initialized and receiving data */
137 
+    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
79 138 
 };
80 139
140 
+/**
141 
+ * Identifies particular servers that require special handling, such as
142 
+ * standards-incompliant "Transport:" lines in the SETUP request.
143 
+ */
81 144 
 enum RTSPServerType {
82 145 
     RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
83 146 
     RTSP_SERVER_REAL, /**< Realmedia-style server */
... ... 
@@ -85,44 +156,115 @@ enum RTSPServerType {
85 85 
     RTSP_SERVER_NB
86 86 
 };
87 87
88 
+/**
89 
+ * Private data for the RTSP demuxer.
90 
+ */
88 91 
 typedef struct RTSPState {
89 92 
     URLContext *rtsp_hd; /* RTSP TCP connexion handle */
93 
+
94 
+    /** number of items in the 'rtsp_streams' variable */
90 95 
     int nb_rtsp_streams;
91 
-    struct RTSPStream **rtsp_streams;
92 96
97 
+    struct RTSPStream **rtsp_streams; /**< streams in this session */
98 
+
99 
+    /** indicator of whether we are currently receiving data from the
100 
+     * server. Basically this isn't more than a simple cache of the
101 
+     * last PLAY/PAUSE command sent to the server, to make sure we don't
102 
+     * send 2x the same unexpectedly or commands in the wrong state. */
93 103 
     enum RTSPClientState state;
104 
+
105 
+    /** the seek value requested when calling av_seek_frame(). This value
106 
+     * is subsequently used as part of the "Range" parameter when emitting
107 
+     * the RTSP PLAY command. If we are currently playing, this command is
108 
+     * called instantly. If we are currently paused, this command is called
109 
+     * whenever we resume playback. Either way, the value is only used once,
110 
+     * see rtsp_read_play() and rtsp_read_seek(). */
94 111 
     int64_t seek_timestamp;
95 112
96 113 
     /* XXX: currently we use unbuffered input */
97 114 
     //    ByteIOContext rtsp_gb;
98 
-    int seq;        /* RTSP command sequence number */
115 
+
116 
+    int seq;                          /**< RTSP command sequence number */
117 
+
118 
+    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
119 
+     * identifier that the client should re-transmit in each RTSP command */
99 120 
     char session_id[512];
121 
+
122 
+    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
100 123 
     enum RTSPTransport transport;
124 
+
125 
+    /** the negotiated network layer transport protocol; e.g. TCP or UDP
126 
+     * uni-/multicast */
101 127 
     enum RTSPLowerTransport lower_transport;
128 
+
129 
+    /** brand of server that we're talking to; e.g. WMS, REAL or other.
130 
+     * Detected based on the value of RTSPMessageHeader->server or the presence
131 
+     * of RTSPMessageHeader->real_challenge */
102 132 
     enum RTSPServerType server_type;
133 
+
134 
+    /** The last reply of the server to a RTSP command */
103 135 
     char last_reply[2048]; /* XXX: allocate ? */
136 
+
137 
+    /** RTSPStream->transport_priv of the last stream that we read a
138 
+     * packet from */
104 139 
     void *cur_transport_priv;
140 
+
141 
+    /** The following are used for Real stream selection */
142 
+    //@{
143 
+    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
105 144 
     int need_subscription;
145 
+
146 
+    /** stream setup during the last frame read. This is used to detect if
147 
+     * we need to subscribe or unsubscribe to any new streams. */
106 148 
     enum AVDiscard real_setup_cache[MAX_STREAMS];
149 
+
150 
+    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
151 
+     * this is used to send the same "Unsubscribe:" if stream setup changed,
152 
+     * before sending a new "Subscribe:" command. */
107 153 
     char last_subscription[1024];
154 
+    //@}
108 155 
 } RTSPState;
109 156
157 
+/**
158 
+ * Describes a single stream, as identified by a single m= line block in the
159 
+ * SDP content. In the case of RDT, one RTSPStream can represent multiple
160 
+ * AVStreams. In this case, each AVStream in this set has similar content
161 
+ * (but different codec/bitrate).
162 
+ */
110 163 
 typedef struct RTSPStream {
111 
-    URLContext *rtp_handle; /* RTP stream handle */
112 
-    void *transport_priv; /* RTP/RDT parse context */
164 
+    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
165 
+    void *transport_priv; /**< RTP/RDT parse context */
166 
+
167 
+    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
168 
+    int stream_index;
169 
+
170 
+    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
171 
+     * for the selected transport. Only used for TCP. */
172 
+    int interleaved_min, interleaved_max;
173 
+
174 
+    char control_url[1024];   /**< url for this stream (from SDP) */
175 
+
176 
+    /** The following are used only in SDP, not RTSP */
177 
+    //@{
178 
+    int sdp_port;             /**< port (from SDP content) */
179 
+    struct in_addr sdp_ip;    /**< IP address (from SDP content) */
180 
+    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
181 
+    int sdp_payload_type;     /**< payload type */
182 
+    //@}
113 183
114 
-    int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
115 
-    int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */
116 
-    char control_url[1024]; /* url for this stream (from SDP) */
184 
+    /** rtp payload parsing infos from SDP (i.e. mapping between private
185 
+     * payload IDs and media-types (string), so that we can derive what
186 
+     * type of payload we're dealing with (and how to parse it). */
187 
+    RTPPayloadData rtp_payload_data;
117 188
118 
-    int sdp_port; /* port (from SDP content - not used in RTSP) */
119 
-    struct in_addr sdp_ip; /* IP address  (from SDP content - not used in RTSP) */
120 
-    int sdp_ttl;  /* IP TTL (from SDP content - not used in RTSP) */
121 
-    int sdp_payload_type; /* payload type - only used in SDP */
122 
-    RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
189 
+    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
190 
+    //@{
191 
+    /** handler structure */
192 
+    RTPDynamicProtocolHandler *dynamic_handler;
123 193
124 
-    RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
125 
-    PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
194 
+    /** private data associated with the dynamic protocol */
195 
+    PayloadContext *dynamic_protocol_context;
196 
+    //@}
126 197 
 } RTSPStream;
127 198
128 199 
 int rtsp_init(void);