X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtsp.h;h=7a910b06db2fd01256a0329a87076f3084f4d675;hb=acaffdca21f63b3b465892e8e9d85fd8ca0022bd;hp=c6c39725764594836f6c320c9c6e51855bd5fbd2;hpb=b20359f51a1c3be5603be9908061b27f883f9467;p=ffmpeg diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index c6c39725764..7a910b06db2 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 +38,14 @@ 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 */ + 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. */ }; /** @@ -46,6 +56,7 @@ enum RTSPLowerTransport { enum RTSPTransport { RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ + RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */ RTSP_TRANSPORT_NB }; @@ -96,7 +107,11 @@ typedef struct RTSPTransportField { * 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; @@ -164,6 +179,11 @@ typedef struct RTSPMessageHeader { * returned */ char reason[256]; + + /** + * Content type header + */ + char content_type[64]; } RTSPMessageHeader; /** @@ -179,7 +199,7 @@ 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 +212,10 @@ 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 +237,6 @@ 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 +265,9 @@ 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]; @@ -294,8 +315,12 @@ 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; + /** 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. */ @@ -308,10 +333,83 @@ typedef struct RTSPState { * 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. */ +#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; + /** - * 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). @@ -333,11 +431,15 @@ typedef struct RTSPStream { //@{ 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 (rtp_*.c/rdt.c) */ + /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */ //@{ /** handler structure */ RTPDynamicProtocolHandler *dynamic_handler; @@ -345,35 +447,17 @@ typedef struct RTSPStream { /** 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, - HTTPAuthState *auth_state); - -#if LIBAVFORMAT_VERSION_INT < (53 << 16) -extern int rtsp_default_protocols; -#endif -extern int rtsp_rtp_port_min; -extern int rtsp_rtp_port_max; + RTSPState *rt, const char *method); -/** - * 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. * @@ -432,13 +516,15 @@ 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. @@ -466,8 +552,71 @@ 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. + */ +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). */ -void ff_rtsp_close_connections(AVFormatContext *rt); +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 */