X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtsp.h;h=44240c1d0fa851c9ab36c69b53a270a830eb205a;hb=3d035d5a6a9116a334fc82dcb37704da2d074958;hp=537d4bf333e35c875574d22e5d524bada6d462e0;hpb=987903826b0dba2e134be200ac94be66b4a3acf1;p=ffmpeg diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 537d4bf333e..44240c1d0fa 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -1,21 +1,21 @@ /* * 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 @@ -24,75 +24,584 @@ #include #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: [;timeout=]" 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 */