| /* |
| * 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 "internal.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_HTTPS, /**< HTTPS tunneled */ |
| 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 RTSPS_DEFAULT_PORT 322 |
| #define RTSP_MAX_TRANSPORTS 8 |
| #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 |
| #define RTSP_RTP_PORT_MIN 5000 |
| #define RTSP_RTP_PORT_MAX 65000 |
| #define SDP_MAX_SIZE 16384 |
| |
| /** |
| * 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: <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]; |
| |
| /** |
| * SAT>IP com.ses.streamID header |
| */ |
| char stream_id[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_SATIP,/**< SAT>IP 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[MAX_URL_SIZE]; |
| |
| /** 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; |
| int max_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. |
| */ |
| int64_t stimeout; |
| |
| /** |
| * Size of RTP packet reordering queue. |
| */ |
| int reordering_queue_size; |
| |
| /** |
| * User-Agent string |
| */ |
| char *user_agent; |
| |
| char default_lang[4]; |
| int buffer_size; |
| int pkt_size; |
| char *localaddr; |
| } 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. */ |
| #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */ |
| #define RTSP_FLAG_SATIP_RAW 0x20 /**< Export SAT>IP stream as raw MPEG-TS */ |
| |
| 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[MAX_URL_SIZE]; /**< 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 */ |
| const 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; |
| |
| /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */ |
| uint32_t ssrc; |
| |
| char crypto_suite[40]; |
| char crypto_params[100]; |
| } RTSPStream; |
| |
| void ff_rtsp_parse_line(AVFormatContext *s, |
| 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. |
| * |
| * @return 0 on success, < 0 on failure. |
| */ |
| int 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 */ |
