/*
* RTSP definitions
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFORMAT_RTSP_H
#include <stdint.h>
#include "avformat.h"
#include "rtspcodes.h"
+#include "rtpdec.h"
+#include "network.h"
+#include "httpauth.h"
-enum RTSPProtocol {
- RTSP_PROTOCOL_RTP_UDP = 0,
- RTSP_PROTOCOL_RTP_TCP = 1,
- RTSP_PROTOCOL_RTP_UDP_MULTICAST = 2,
- /**
- * This is not part of public API and shouldn't be used outside of ffmpeg.
- */
- RTSP_PROTOCOL_RTP_LAST
+#include "libavutil/log.h"
+#include "libavutil/opt.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,
+ RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
+ transport mode as such,
+ only for use via AVOptions */
+ RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
+ option for lower_transport_mask,
+ but set in the SDP demuxer based
+ on a flag. */
+};
+
+/**
+ * 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_RAW, /**< Raw data (over UDP) */
+ RTSP_TRANSPORT_NB
+};
+
+/**
+ * Transport mode for the RTSP data. This may be plain, or
+ * tunneled, which is done over HTTP.
+ */
+enum RTSPControlTransport {
+ RTSP_MODE_PLAIN, /**< Normal RTSP */
+ RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
};
#define RTSP_DEFAULT_PORT 554
#define RTSP_MAX_TRANSPORTS 8
#define RTSP_TCP_MAX_PACKET_SIZE 1472
-#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
+#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
#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;
+
+ /** transport set to record data */
+ int mode_record;
+
+ struct sockaddr_storage destination; /**< destination IP address */
+ char source[INET6_ADDRSTRLEN + 1]; /**< source 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 */
+
+ /** 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 */
-} 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 "Location:" field. This value is used to handle redirection.
+ */
+ char location[4096];
+
+ /** 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;
+
+ /** The "Notice" or "X-Notice" field value. See
+ * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
+ * for a complete list of supported values. */
+ int notice;
+
+ /** The "reason" is meant to specify better the meaning of the error code
+ * returned
+ */
+ char reason[256];
+
+ /**
+ * Content type header
+ */
+ char content_type[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, /**< not initialized */
+ RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
+ RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
+ RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
+};
+
+/**
+ * Identify 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 AVIOContext instead of URLContext
+ */
+typedef struct RTSPState {
+ const AVClass *class; /**< Class for private options. */
+ URLContext *rtsp_hd; /* RTSP TCP connection 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;
+
+ 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 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 "RealChallenge1:" field from the server */
+ char real_challenge[64];
+
+ /** plaintext authorization line (username:password) */
+ char auth[128];
+
+ /** authentication state */
+ HTTPAuthState auth_state;
+
+ /** 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;
+
+ /** current stream setup. This is a temporary buffer used to compare
+ * current setup to previous frame setup. */
+ enum AVDiscard *real_setup;
+
+ /** 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;
+
+ /** cache for position of the asf demuxer, since we load a new
+ * data packet in the bytecontext for each incoming RTSP packet. */
+ uint64_t asf_pb_pos;
+ //@}
+
+ /** some MS RTSP streams contain a URL in the SDP that we need to use
+ * for all subsequent RTSP requests, rather than the input URI; in
+ * other cases, this is a copy of AVFormatContext->filename. */
+ char control_uri[1024];
+
+ /** The following are used for parsing raw mpegts in udp */
+ //@{
+ struct MpegTSContext *ts;
+ int recvbuf_pos;
+ int recvbuf_len;
+ //@}
+
+ /** Additional output handle, used when input and output are done
+ * separately, eg for HTTP tunneling. */
+ URLContext *rtsp_hd_out;
+
+ /** RTSP transport mode, such as plain or tunneled. */
+ enum RTSPControlTransport control_transport;
+
+ /* Number of RTCP BYE packets the RTSP session has received.
+ * An EOF is propagated back if nb_byes == nb_streams.
+ * This is reset after a seek. */
+ int nb_byes;
+
+ /** Reusable buffer for receiving packets */
+ uint8_t* recvbuf;
+
+ /**
+ * A mask with all requested transport methods
+ */
+ int lower_transport_mask;
+
+ /**
+ * The number of returned packets
+ */
+ uint64_t packets;
+
+ /**
+ * Polling array for udp
+ */
+ struct pollfd *p;
+
+ /**
+ * Whether the server supports the GET_PARAMETER method.
+ */
+ int get_parameter_supported;
+
+ /**
+ * Do not begin to play the stream immediately.
+ */
+ int initial_pause;
+
+ /**
+ * Option flags for the chained RTP muxer.
+ */
+ int rtp_muxer_flags;
+
+ /** Whether the server accepts the x-Dynamic-Rate header */
+ int accept_dynamic_rate;
+
+ /**
+ * Various option flags for the RTSP muxer/demuxer.
+ */
+ int rtsp_flags;
+
+ /**
+ * Mask of all requested media types
+ */
+ int media_type_mask;
+
+ /**
+ * Minimum and maximum local UDP ports.
+ */
+ int rtp_port_min, rtp_port_max;
+
+ /**
+ * Timeout to wait for incoming connections.
+ */
+ int initial_timeout;
+
+ /**
+ * Size of RTP packet reordering queue.
+ */
+ int reordering_queue_size;
+} RTSPState;
+
+#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
+ receive packets only from the right
+ source address and port. */
+#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
+#define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
+
+/**
+ * Describe 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 if input, RTP AVFormatContext if output */
+
+ /** 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 sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
+ int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
+ int sdp_payload_type; /**< payload type */
+ //@}
+
+ /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
+ //@{
+ /** handler structure */
+ RTPDynamicProtocolHandler *dynamic_handler;
+
+ /** private data associated with the dynamic protocol */
+ PayloadContext *dynamic_protocol_context;
+ //@}
+
+ /** Enable sending RTCP feedback messages according to RFC 4585 */
+ int feedback;
+
+ char crypto_suite[40];
+ char crypto_params[100];
+} RTSPStream;
+
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
+ RTSPState *rt, const char *method);
+
+/**
+ * Send a command to the RTSP server without waiting for the reply.
+ *
+ * @see rtsp_send_cmd_with_content_async
+ */
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers);
+
+/**
+ * Send a command to the RTSP server and wait for the reply.
+ *
+ * @param s RTSP (de)muxer context
+ * @param method the method for the request
+ * @param url the target url for the request
+ * @param headers extra header lines to include in the request
+ * @param reply pointer where the RTSP message header will be stored
+ * @param content_ptr pointer where the RTSP message body, if any, will
+ * be stored (length is in reply)
+ * @param send_content if non-null, the data to send as request body content
+ * @param send_content_length the length of the send_content data, or 0 if
+ * send_content is null
+ *
+ * @return zero if success, nonzero otherwise
+ */
+int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *headers,
+ RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ const unsigned char *send_content,
+ int send_content_length);
+
+/**
+ * Send a command to the RTSP server and wait for the reply.
+ *
+ * @see rtsp_send_cmd_with_content
+ */
+int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
+ const char *url, const char *headers,
+ RTSPMessageHeader *reply, unsigned char **content_ptr);
+
+/**
+ * Read a RTSP message from the server, or prepare to read data
+ * packets if we're reading data interleaved over the TCP/RTSP
+ * connection as well.
+ *
+ * @param s RTSP (de)muxer context
+ * @param reply pointer where the RTSP message header will be stored
+ * @param content_ptr pointer where the RTSP message body, if any, will
+ * be stored (length is in reply)
+ * @param return_on_interleaved_data whether the function may return if we
+ * encounter a data marker ('$'), which precedes data
+ * packets over interleaved TCP/RTSP connections. If this
+ * is set, this function will return 1 after encountering
+ * a '$'. If it is not set, the function will skip any
+ * data packets (if they are encountered), until a reply
+ * has been fully parsed. If no more data is available
+ * without parsing a reply, it will return an error.
+ * @param method the RTSP method this is a reply to. This affects how
+ * some response headers are acted upon. May be NULL.
+ *
+ * @return 1 if a data packets is ready to be received, -1 on error,
+ * and 0 on success.
+ */
+int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ int return_on_interleaved_data, const char *method);
+
+/**
+ * Skip a RTP/TCP interleaved packet.
+ */
+void ff_rtsp_skip_packet(AVFormatContext *s);
+
+/**
+ * Connect to the RTSP server and set up the individual media streams.
+ * This can be used for both muxers and demuxers.
+ *
+ * @param s RTSP (de)muxer context
+ *
+ * @return 0 on success, < 0 on error. Cleans up all allocations done
+ * within the function on error.
+ */
+int ff_rtsp_connect(AVFormatContext *s);
+
+/**
+ * Close and free all streams within the RTSP (de)muxer
+ *
+ * @param s RTSP (de)muxer context
+ */
+void ff_rtsp_close_streams(AVFormatContext *s);
+
+/**
+ * Close all connection handles within the RTSP (de)muxer
+ *
+ * @param s RTSP (de)muxer context
+ */
+void ff_rtsp_close_connections(AVFormatContext *s);
+
+/**
+ * Get the description of the stream and set up the RTSPStream child
+ * objects.
+ */
+int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
+
+/**
+ * Announce the stream to the server and set up the RTSPStream child
+ * objects for each media stream.
+ */
+int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
+
+/**
+ * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
+ * listen mode.
+ */
+int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
+
+/**
+ * Parse an SDP description of streams by populating an RTSPState struct
+ * within the AVFormatContext; also allocate the RTP streams and the
+ * pollfd array used for UDP streams.
+ */
+int ff_sdp_parse(AVFormatContext *s, const char *content);
+
+/**
+ * Receive one RTP packet from an TCP interleaved RTSP stream.
+ */
+int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+ uint8_t *buf, int buf_size);
-typedef int FFRTSPCallback(enum RTSPCallbackAction action,
- const char *session_id,
- char *buf, int buf_size,
- void *arg);
+/**
+ * Receive one packet from the RTSPStreams set up in the AVFormatContext
+ * (which should contain a RTSPState struct as priv_data).
+ */
+int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
-int rtsp_init(void);
-void rtsp_parse_line(RTSPHeader *reply, const char *buf);
+/**
+ * Do the SETUP requests for each stream for the chosen
+ * lower transport mode.
+ * @return 0 on success, <0 on error, 1 if protocol is unavailable
+ */
+int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
+ int lower_transport, const char *real_challenge);
-#if LIBAVFORMAT_VERSION_INT < (53 << 16)
-extern int rtsp_default_protocols;
-#endif
-extern int rtsp_rtp_port_min;
-extern int rtsp_rtp_port_max;
+/**
+ * Undo the effect of ff_rtsp_make_setup_request, close the
+ * transport_priv and rtp_handle fields.
+ */
+void ff_rtsp_undo_setup(AVFormatContext *s);
+
+/**
+ * Open RTSP transport context.
+ */
+int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
-int rtsp_pause(AVFormatContext *s);
-int rtsp_resume(AVFormatContext *s);
+extern const AVOption ff_rtsp_options[];
#endif /* AVFORMAT_RTSP_H */
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