/*
* RTSP definitions
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef RTSP_H
-#define RTSP_H
+#ifndef FFMPEG_RTSP_H
+#define FFMPEG_RTSP_H
-/* RTSP handling */
-enum RTSPStatusCode {
-#define DEF(n, c, s) c = n,
+#include <stdint.h>
+#include "avformat.h"
#include "rtspcodes.h"
-#undef DEF
+#include "rtpdec.h"
+#include "network.h"
+
+/**
+ * Network layer over which RTP/etc packet data will be transported.
+ */
+enum RTSPLowerTransport {
+ 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
};
-enum RTSPProtocol {
- RTSP_PROTOCOL_RTP_UDP = 0,
- RTSP_PROTOCOL_RTP_TCP = 1,
- RTSP_PROTOCOL_RTP_UDP_MULTICAST = 2,
+/**
+ * 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, /**< Standards-compliant RTP */
+ RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
+ RTSP_TRANSPORT_NB
};
#define RTSP_DEFAULT_PORT 554
#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 */
- uint32_t destination; /* destination IP address */
- enum RTSPProtocol protocol;
+ /** 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;
-typedef struct RTSPHeader {
+/**
+ * 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 */
+
+ 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];
-} RTSPHeader;
-
-/* the callback can be used to extend the connection setup/teardown step */
-enum RTSPCallbackAction {
- RTSP_ACTION_SERVER_SETUP,
- RTSP_ACTION_SERVER_TEARDOWN,
- RTSP_ACTION_CLIENT_SETUP,
- RTSP_ACTION_CLIENT_TEARDOWN,
+
+ /** 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];
+
+ /** The "timeout" comes as part of the server response to the "SETUP"
+ * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
+ * time, in seconds, that the server will go without traffic over the
+ * RTSP/TCP connection before it closes the connection. To prevent
+ * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
+ * than this value. */
+ int timeout;
+} 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, /**< 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 */
+ RTSP_SERVER_WMS, /**< Windows Media server */
+ RTSP_SERVER_NB
};
-typedef struct RTSPActionServerSetup {
- uint32_t ipaddr;
- char transport_option[512];
-} RTSPActionServerSetup;
+/**
+ * Private data for the RTSP demuxer.
+ *
+ * @todo Use ByteIOContext instead of URLContext
+ */
+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; /**< 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 */
+
+ /** 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];
+
+ /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
+ * the server will go without traffic on the RTSP/TCP line before it
+ * closes the connection. */
+ int timeout;
+
+ /** timestamp of the last RTSP command that we sent to the RTSP server.
+ * This is used to calculate when to send dummy commands to keep the
+ * connection alive, in conjunction with \p timeout. */
+ int64_t last_cmd_time;
+
+ /** 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];
+ //@}
+
+ /** The following are used for RTP/ASF streams */
+ //@{
+ /** ASF demuxer context for the embedded ASF stream from WMS servers */
+ AVFormatContext *asf_ctx;
+ //@}
+} 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 (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 */
+ //@}
+
+ /** 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;
-typedef int FFRTSPCallback(enum RTSPCallbackAction action,
- const char *session_id,
- char *buf, int buf_size,
- void *arg);
+ /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
+ //@{
+ /** handler structure */
+ RTPDynamicProtocolHandler *dynamic_handler;
-void rtsp_set_callback(FFRTSPCallback *rtsp_cb);
+ /** private data associated with the dynamic protocol */
+ PayloadContext *dynamic_protocol_context;
+ //@}
+} RTSPStream;
int rtsp_init(void);
-void rtsp_parse_line(RTSPHeader *reply, const char *buf);
+void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf);
+#if LIBAVFORMAT_VERSION_INT < (53 << 16)
extern int rtsp_default_protocols;
+#endif
extern int rtsp_rtp_port_min;
extern int rtsp_rtp_port_max;
-extern FFRTSPCallback *ff_rtsp_callback;
-extern AVInputFormat rtsp_demuxer;
int rtsp_pause(AVFormatContext *s);
int rtsp_resume(AVFormatContext *s);
-#endif /* RTSP_H */
+#endif /* FFMPEG_RTSP_H */
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