From 22e28216336da876e1fd17f380ce42eaf1446769 Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Mon, 17 Feb 2014 13:36:38 +0000 Subject: chasing indexing error --- ffmpeg/libavformat/rtsp.h | 632 ---------------------------------------------- 1 file changed, 632 deletions(-) delete mode 100644 ffmpeg/libavformat/rtsp.h (limited to 'ffmpeg/libavformat/rtsp.h') diff --git a/ffmpeg/libavformat/rtsp.h b/ffmpeg/libavformat/rtsp.h deleted file mode 100644 index 76c7f18..0000000 --- a/ffmpeg/libavformat/rtsp.h +++ /dev/null @@ -1,632 +0,0 @@ -/* - * 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 -#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 */ - 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 1 -#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 -#define RTSP_RTP_PORT_MIN 5000 -#define RTSP_RTP_PORT_MAX 65000 - -/** - * 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; - - /** 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; - -/** - * 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: [;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 -}; - -/** - * 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; - - /** - * timeout of socket i/o operations. - */ - int stimeout; - - /** - * Size of RTP packet reordering queue. - */ - int reordering_queue_size; - - /** - * User-Agent string - */ - char *user_agent; -} 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. */ -#define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source - address of received packets. */ - -typedef struct RTSPSource { - char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */ -} RTSPSource; - -/** - * 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 nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */ - struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */ - int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */ - struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (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); - -/** - * Send buffered packets over TCP. - */ -int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st); - -/** - * 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, int send_packets); - -/** - * Open RTSP transport context. - */ -int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st); - -extern const AVOption ff_rtsp_options[]; - -#endif /* AVFORMAT_RTSP_H */ -- cgit v1.2.3