X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtsp.h;h=662407f2993ade8d9d0c32e6b329d2ccf11a7ff2;hb=0b8b3387a977dcdb6fb9e53bcc9966d34b2e4cec;hp=814d8d8e8dcc6041ed04b7ec0621c9bf91d8cd4e;hpb=be73ba2fa4890b857d987b79958e46af8c5e545b;p=ffmpeg diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 814d8d8e8dc..662407f2993 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -2,20 +2,20 @@ * RTSP definitions * 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 @@ -28,6 +28,9 @@ #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. */ @@ -35,7 +38,10 @@ 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_NB, + RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper + transport mode as such, + only for use via AVOptions */ }; /** @@ -96,7 +102,8 @@ typedef struct RTSPTransportField { * packets will be allowed to make before being discarded. */ int ttl; - uint32_t destination; /**< destination IP address */ + 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; @@ -179,7 +186,7 @@ enum RTSPClientState { }; /** - * Identifies particular servers that require special handling, such as + * Identify particular servers that require special handling, such as * standards-incompliant "Transport:" lines in the SETUP request. */ enum RTSPServerType { @@ -192,9 +199,10 @@ enum RTSPServerType { /** * Private data for the RTSP demuxer. * - * @todo Use ByteIOContext instead of URLContext + * @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 */ @@ -216,9 +224,6 @@ typedef struct RTSPState { * see rtsp_read_play() and rtsp_read_seek(). */ int64_t seek_timestamp; - /* XXX: currently we use unbuffered input */ - // ByteIOContext rtsp_gb; - int seq; /**< RTSP command sequence number */ /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session @@ -247,6 +252,9 @@ typedef struct RTSPState { * 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]; @@ -294,19 +302,71 @@ typedef struct RTSPState { * other cases, this is a copy of AVFormatContext->filename. */ char control_uri[1024]; - /** The synchronized start time of the output streams. */ - int64_t start_time; - /** 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. */ + /** - * Describes a single stream, as identified by a single m= line block in the + * 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). @@ -327,7 +387,7 @@ typedef struct RTSPStream { /** The following are used only in SDP, not RTSP */ //@{ int sdp_port; /**< port (from SDP content) */ - struct in_addr sdp_ip; /**< IP address (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 */ //@} @@ -343,32 +403,11 @@ typedef struct RTSPStream { } RTSPStream; void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, - HTTPAuthState *auth_state); + RTSPState *rt, const char *method); -#if LIBAVFORMAT_VERSION_INT < (53 << 16) -extern int rtsp_default_protocols; -#endif extern int rtsp_rtp_port_min; extern int rtsp_rtp_port_max; -/** - * Send a command to the RTSP server without waiting 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 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_async(AVFormatContext *s, - const char *method, const char *url, - const char *headers, - const unsigned char *send_content, - int send_content_length); /** * Send a command to the RTSP server without waiting for the reply. * @@ -427,13 +466,15 @@ int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, * 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); + int return_on_interleaved_data, const char *method); /** * Skip a RTP/TCP interleaved packet. @@ -461,8 +502,55 @@ void ff_rtsp_close_streams(AVFormatContext *s); /** * Close all connection handles within the RTSP (de)muxer * - * @param rt RTSP (de)muxer context + * @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. */ -void ff_rtsp_close_connections(AVFormatContext *rt); +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 */