/* * RTSP definitions * Copyright (c) 2002 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg 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, * 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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #ifndef AVFORMAT_RTSP_H #define AVFORMAT_RTSP_H #include <stdint.h> #include "avformat.h" #include "rtspcodes.h" #include "rtpdec.h" #include "network.h" #include "httpauth.h" #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 */ }; /** * 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 }; /** * 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 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 { /** 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; 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; /** * 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; /** 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 */ /** 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]; /** 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]; } 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 }; /** * 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]; /** 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; } RTSPState; #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets - receive packets only from the right source address and port. */ /** * 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 (rtp_*.c/rdt.c) */ //@{ /** handler structure */ RTPDynamicProtocolHandler *dynamic_handler; /** private data associated with the dynamic protocol */ PayloadContext *dynamic_protocol_context; //@} } RTSPStream; void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method); extern int rtsp_rtp_port_min; extern int rtsp_rtp_port_max; /** * 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 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); /** * 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); /** * 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); /** * Undo the effect of ff_rtsp_make_setup_request, close the * transport_priv and rtp_handle fields. */ void ff_rtsp_undo_setup(AVFormatContext *s); extern const AVOption ff_rtsp_options[]; #endif /* AVFORMAT_RTSP_H */