[FFmpeg-devel] [PATCH] document rtsp.h
Ronald S. Bultje
rsbultje
Thu Feb 5 15:24:21 CET 2009
Hi,
new thread, otherwise the subject line doesn't make any sense.
Attached patch fully (I think) documents rtsp.h. The changes are sort
of, ehm, well, everywhere, so I basically just didn't bother too much
to distinguish "move comment to new line" or "align comment /
whitespace" or "add new comment" in separate patches, because in the
end I pretty much touch every single line out there anyway. I didn't
add any newlines between commented items:
/** bla */
int var1;
<- there is no newline here, although AVCodecContext does have that
/** bla2 */
char var2;
I can add more newlines if you think it's unreadable this way, but
regardless, I think this is better than what it was.
Ronald
-------------- next part --------------
Index: ffmpeg-svn/libavformat/rtsp.h
===================================================================
--- ffmpeg-svn.orig/libavformat/rtsp.h 2009-02-05 09:10:30.000000000 -0500
+++ ffmpeg-svn/libavformat/rtsp.h 2009-02-05 09:18:10.000000000 -0500
@@ -27,19 +27,30 @@
#include "rtp.h"
#include "network.h"
+/**
+ * Network layer over which RTP 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 */
/**
* This is not part of public API and shouldn't be used outside of ffmpeg.
*/
RTSP_LOWER_TRANSPORT_LAST
};
+/**
+ * Packet protocol 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 */
+ /**
+ * This is not part of the public API and shouldn't be used outside ffmpeg.
+ */
RTSP_TRANSPORT_LAST
};
@@ -51,81 +62,181 @@
#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 RTSPHeader {
+ /** 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];
} RTSPHeader;
+/**
+ * Client state, i.e. whether we are currently streaming data (PLAYING) or
+ * setup-but-not-streaming (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 streaming data */
+ RTSP_STATE_PAUSED, /**< initialized, but not streaming 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 */
RTSP_SERVER_WMS, /**< Windows Media server */
+ /**
+ * This is not part of the public API and shouldn't be used outside ffmpeg.
+ */
RTSP_SERVER_LAST
};
+/**
+ * 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 */
+ /** whether we are currently receiving data from the server */
enum RTSPClientState state;
+ /** the seek value requested when calling av_seek_frame(). This way,
+ * the seek value is saved if we are currently paused and will be
+ * transmitted at the next PLAY RTSP command. See rtsp_read_play(). */
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 RTSPHeader->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 RTSPHeader->server or the presence
+ * of RTSPHeader->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_ctx of the last stream that we read a
+ * packet from */
void *cur_transport_priv_ctx;
+
+ /** 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_ctx; /* RTP/RDT parse context */
-
- 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) */
-
- 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 */
+ URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
+ void *transport_priv_ctx; /**< RTP/RDT parse context */
- 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)
+ /** 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/private protocols (payloads) */
+ //@{
+ /** handler structure */
+ RTPDynamicProtocolHandler *dynamic_handler;
+ /** private data associated with the dynamic protocol */
+ PayloadContext *dynamic_protocol_context;
+ //@}
} RTSPStream;
int rtsp_init(void);
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