X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtsp.h;h=a738a3d4349336981e349bb4caa948496d6d19ab;hb=3505d5574e1d87ab8af9ea38337bfa0a1ca6381d;hp=8f79c759ff0997ab26ac0f90b21875e50730f497;hpb=d243ba30b824513e75f5836c6dde789034152976;p=ffmpeg diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 8f79c759ff0..a738a3d4349 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 @@ -26,6 +26,10 @@ #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. @@ -34,7 +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 */ }; /** @@ -48,10 +55,19 @@ enum RTSPTransport { 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 @@ -86,7 +102,11 @@ typedef struct RTSPTransportField { * packets will be allowed to make before being discarded. */ int ttl; - uint32_t destination; /**< destination IP address */ + /** 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; @@ -149,6 +169,16 @@ typedef struct RTSPMessageHeader { * 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; /** @@ -158,13 +188,13 @@ typedef struct RTSPMessageHeader { */ enum RTSPClientState { RTSP_STATE_IDLE, /**< not initialized */ - RTSP_STATE_PLAYING, /**< initialized and receiving data */ + RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ }; /** - * 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 { @@ -177,10 +207,11 @@ enum RTSPServerType { /** * Private data for the RTSP demuxer. * - * @todo Use ByteIOContext instead of URLContext + * @todo Use AVIOContext instead of URLContext */ typedef struct RTSPState { - URLContext *rtsp_hd; /* RTSP TCP connexion handle */ + 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; @@ -201,9 +232,6 @@ 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 @@ -232,8 +260,14 @@ typedef struct RTSPState { * of RTSPMessageHeader->real_challenge */ enum RTSPServerType server_type; - /** base64-encoded authorization lines (username:password) */ - char *auth_b64; + /** 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 ? */ @@ -249,7 +283,11 @@ typedef struct RTSPState { /** 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[MAX_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, @@ -266,17 +304,95 @@ typedef struct RTSPState { * 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]; + + /** 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; } 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. */ + /** - * 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). */ typedef struct RTSPStream { URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ - void *transport_priv; /**< RTP/RDT parse context */ + 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; @@ -290,16 +406,11 @@ 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 */ //@} - /** rtp payload parsing infos from SDP (i.e. mapping between private - * payload IDs and media-types (string), so that we can derive what - * type of payload we're dealing with (and how to parse it). */ - RTPPayloadData rtp_payload_data; - /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */ //@{ /** handler structure */ @@ -310,16 +421,163 @@ typedef struct RTSPStream { //@} } RTSPStream; -int rtsp_init(void); -void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf); +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); -#if LIBAVFORMAT_VERSION_INT < (53 << 16) -extern int rtsp_default_protocols; -#endif -extern int rtsp_rtp_port_min; -extern int rtsp_rtp_port_max; +/** + * 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); + +/** + * 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 */