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645 lines
22 KiB
C
Executable File
645 lines
22 KiB
C
Executable File
/*
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* RTSP definitions
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVFORMAT_RTSP_H
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#define AVFORMAT_RTSP_H
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#include <stdint.h>
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#include "avformat.h"
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#include "rtspcodes.h"
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#include "rtpdec.h"
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#include "network.h"
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#include "httpauth.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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/**
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* Network layer over which RTP/etc packet data will be transported.
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*/
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enum RTSPLowerTransport {
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RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
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RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
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RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
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RTSP_LOWER_TRANSPORT_NB,
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RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
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transport mode as such,
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only for use via AVOptions */
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RTSP_LOWER_TRANSPORT_HTTPS, /**< HTTPS tunneled */
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RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
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option for lower_transport_mask,
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but set in the SDP demuxer based
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on a flag. */
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};
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/**
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* Packet profile of the data that we will be receiving. Real servers
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* commonly send RDT (although they can sometimes send RTP as well),
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* whereas most others will send RTP.
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*/
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enum RTSPTransport {
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RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
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RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
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RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
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RTSP_TRANSPORT_NB
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};
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/**
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* Transport mode for the RTSP data. This may be plain, or
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* tunneled, which is done over HTTP.
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*/
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enum RTSPControlTransport {
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RTSP_MODE_PLAIN, /**< Normal RTSP */
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RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
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};
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#define RTSP_DEFAULT_PORT 554
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#define RTSPS_DEFAULT_PORT 322
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#define RTSP_MAX_TRANSPORTS 8
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#define RTSP_TCP_MAX_PACKET_SIZE 1472
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#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
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#define RTSP_RTP_PORT_MIN 5000
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#define RTSP_RTP_PORT_MAX 65000
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/**
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* This describes a single item in the "Transport:" line of one stream as
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* negotiated by the SETUP RTSP command. Multiple transports are comma-
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* separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
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* client_port=1000-1001;server_port=1800-1801") and described in separate
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* RTSPTransportFields.
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*/
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typedef struct RTSPTransportField {
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/** interleave ids, if TCP transport; each TCP/RTSP data packet starts
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* with a '$', stream length and stream ID. If the stream ID is within
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* the range of this interleaved_min-max, then the packet belongs to
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* this stream. */
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int interleaved_min, interleaved_max;
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/** UDP multicast port range; the ports to which we should connect to
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* receive multicast UDP data. */
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int port_min, port_max;
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/** UDP client ports; these should be the local ports of the UDP RTP
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* (and RTCP) sockets over which we receive RTP/RTCP data. */
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int client_port_min, client_port_max;
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/** UDP unicast server port range; the ports to which we should connect
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* to receive unicast UDP RTP/RTCP data. */
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int server_port_min, server_port_max;
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/** time-to-live value (required for multicast); the amount of HOPs that
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* packets will be allowed to make before being discarded. */
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int ttl;
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/** transport set to record data */
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int mode_record;
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struct sockaddr_storage destination; /**< destination IP address */
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char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
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/** data/packet transport protocol; e.g. RTP or RDT */
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enum RTSPTransport transport;
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/** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
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enum RTSPLowerTransport lower_transport;
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} RTSPTransportField;
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/**
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* This describes the server response to each RTSP command.
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*/
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typedef struct RTSPMessageHeader {
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/** length of the data following this header */
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int content_length;
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enum RTSPStatusCode status_code; /**< response code from server */
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/** number of items in the 'transports' variable below */
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int nb_transports;
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/** Time range of the streams that the server will stream. In
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* AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
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int64_t range_start, range_end;
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/** describes the complete "Transport:" line of the server in response
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* to a SETUP RTSP command by the client */
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RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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int seq; /**< sequence number */
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/** the "Session:" field. This value is initially set by the server and
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* should be re-transmitted by the client in every RTSP command. */
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char session_id[512];
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/** the "Location:" field. This value is used to handle redirection.
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*/
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char location[4096];
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/** the "RealChallenge1:" field from the server */
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char real_challenge[64];
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/** the "Server: field, which can be used to identify some special-case
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* servers that are not 100% standards-compliant. We use this to identify
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* Windows Media Server, which has a value "WMServer/v.e.r.sion", where
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* version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
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* use something like "Helix [..] Server Version v.e.r.sion (platform)
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* (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
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* where platform is the output of $uname -msr | sed 's/ /-/g'. */
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char server[64];
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/** The "timeout" comes as part of the server response to the "SETUP"
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* command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
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* time, in seconds, that the server will go without traffic over the
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* RTSP/TCP connection before it closes the connection. To prevent
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* this, sent dummy requests (e.g. OPTIONS) with intervals smaller
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* than this value. */
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int timeout;
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/** The "Notice" or "X-Notice" field value. See
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* http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
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* for a complete list of supported values. */
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int notice;
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/** The "reason" is meant to specify better the meaning of the error code
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* returned
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*/
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char reason[256];
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/**
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* Content type header
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*/
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char content_type[64];
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} RTSPMessageHeader;
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/**
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* Client state, i.e. whether we are currently receiving data (PLAYING) or
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* setup-but-not-receiving (PAUSED). State can be changed in applications
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* by calling av_read_play/pause().
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*/
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enum RTSPClientState {
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RTSP_STATE_IDLE, /**< not initialized */
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RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
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RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
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RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
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};
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/**
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* Identify particular servers that require special handling, such as
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* standards-incompliant "Transport:" lines in the SETUP request.
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*/
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enum RTSPServerType {
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RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
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RTSP_SERVER_REAL, /**< Realmedia-style server */
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RTSP_SERVER_WMS, /**< Windows Media server */
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RTSP_SERVER_NB
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};
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/**
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* Private data for the RTSP demuxer.
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*
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* @todo Use AVIOContext instead of URLContext
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*/
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typedef struct RTSPState {
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const AVClass *class; /**< Class for private options. */
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URLContext *rtsp_hd; /* RTSP TCP connection handle */
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/** number of items in the 'rtsp_streams' variable */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams; /**< streams in this session */
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/** indicator of whether we are currently receiving data from the
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* server. Basically this isn't more than a simple cache of the
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* last PLAY/PAUSE command sent to the server, to make sure we don't
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* send 2x the same unexpectedly or commands in the wrong state. */
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enum RTSPClientState state;
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/** the seek value requested when calling av_seek_frame(). This value
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* is subsequently used as part of the "Range" parameter when emitting
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* the RTSP PLAY command. If we are currently playing, this command is
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* called instantly. If we are currently paused, this command is called
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* whenever we resume playback. Either way, the value is only used once,
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* see rtsp_read_play() and rtsp_read_seek(). */
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int64_t seek_timestamp;
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int seq; /**< RTSP command sequence number */
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/** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
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* identifier that the client should re-transmit in each RTSP command */
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char session_id[512];
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/** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
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* the server will go without traffic on the RTSP/TCP line before it
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* closes the connection. */
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int timeout;
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/** timestamp of the last RTSP command that we sent to the RTSP server.
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* This is used to calculate when to send dummy commands to keep the
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* connection alive, in conjunction with timeout. */
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int64_t last_cmd_time;
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/** the negotiated data/packet transport protocol; e.g. RTP or RDT */
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enum RTSPTransport transport;
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/** the negotiated network layer transport protocol; e.g. TCP or UDP
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* uni-/multicast */
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enum RTSPLowerTransport lower_transport;
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/** brand of server that we're talking to; e.g. WMS, REAL or other.
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* Detected based on the value of RTSPMessageHeader->server or the presence
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* of RTSPMessageHeader->real_challenge */
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enum RTSPServerType server_type;
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/** the "RealChallenge1:" field from the server */
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char real_challenge[64];
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/** plaintext authorization line (username:password) */
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char auth[128];
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/** authentication state */
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HTTPAuthState auth_state;
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/** The last reply of the server to a RTSP command */
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char last_reply[2048]; /* XXX: allocate ? */
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/** RTSPStream->transport_priv of the last stream that we read a
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* packet from */
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void *cur_transport_priv;
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/** The following are used for Real stream selection */
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//@{
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/** whether we need to send a "SET_PARAMETER Subscribe:" command */
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int need_subscription;
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/** stream setup during the last frame read. This is used to detect if
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* we need to subscribe or unsubscribe to any new streams. */
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enum AVDiscard *real_setup_cache;
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/** current stream setup. This is a temporary buffer used to compare
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* current setup to previous frame setup. */
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enum AVDiscard *real_setup;
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/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
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* this is used to send the same "Unsubscribe:" if stream setup changed,
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* before sending a new "Subscribe:" command. */
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char last_subscription[1024];
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//@}
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/** The following are used for RTP/ASF streams */
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//@{
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/** ASF demuxer context for the embedded ASF stream from WMS servers */
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AVFormatContext *asf_ctx;
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/** cache for position of the asf demuxer, since we load a new
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* data packet in the bytecontext for each incoming RTSP packet. */
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uint64_t asf_pb_pos;
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//@}
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/** some MS RTSP streams contain a URL in the SDP that we need to use
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* for all subsequent RTSP requests, rather than the input URI; in
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* other cases, this is a copy of AVFormatContext->filename. */
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char control_uri[1024];
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/** The following are used for parsing raw mpegts in udp */
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//@{
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struct MpegTSContext *ts;
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int recvbuf_pos;
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int recvbuf_len;
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//@}
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/** Additional output handle, used when input and output are done
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* separately, eg for HTTP tunneling. */
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URLContext *rtsp_hd_out;
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/** RTSP transport mode, such as plain or tunneled. */
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enum RTSPControlTransport control_transport;
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/* Number of RTCP BYE packets the RTSP session has received.
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* An EOF is propagated back if nb_byes == nb_streams.
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* This is reset after a seek. */
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int nb_byes;
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/** Reusable buffer for receiving packets */
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uint8_t* recvbuf;
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/**
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* A mask with all requested transport methods
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*/
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int lower_transport_mask;
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/**
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* The number of returned packets
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*/
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uint64_t packets;
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/**
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* Polling array for udp
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*/
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struct pollfd *p;
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int max_p;
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/**
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* Whether the server supports the GET_PARAMETER method.
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*/
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int get_parameter_supported;
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/**
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* Do not begin to play the stream immediately.
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*/
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int initial_pause;
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/**
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* Option flags for the chained RTP muxer.
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*/
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int rtp_muxer_flags;
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/** Whether the server accepts the x-Dynamic-Rate header */
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int accept_dynamic_rate;
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/**
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* Various option flags for the RTSP muxer/demuxer.
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*/
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int rtsp_flags;
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/**
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* Mask of all requested media types
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*/
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int media_type_mask;
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/**
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* Minimum and maximum local UDP ports.
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*/
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int rtp_port_min, rtp_port_max;
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/**
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* Timeout to wait for incoming connections.
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*/
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int initial_timeout;
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/**
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* timeout of socket i/o operations.
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*/
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int stimeout;
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/**
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* Size of RTP packet reordering queue.
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*/
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int reordering_queue_size;
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/**
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* User-Agent string
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*/
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char *user_agent;
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char default_lang[4];
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int buffer_size;
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int pkt_size;
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} RTSPState;
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#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
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receive packets only from the right
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source address and port. */
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#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
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#define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
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#define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
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address of received packets. */
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#define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
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typedef struct RTSPSource {
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char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
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} RTSPSource;
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/**
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* Describe a single stream, as identified by a single m= line block in the
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* SDP content. In the case of RDT, one RTSPStream can represent multiple
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* AVStreams. In this case, each AVStream in this set has similar content
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* (but different codec/bitrate).
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*/
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typedef struct RTSPStream {
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URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
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void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
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/** corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int stream_index;
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/** interleave IDs; copies of RTSPTransportField->interleaved_min/max
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* for the selected transport. Only used for TCP. */
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int interleaved_min, interleaved_max;
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char control_url[1024]; /**< url for this stream (from SDP) */
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/** The following are used only in SDP, not RTSP */
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//@{
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int sdp_port; /**< port (from SDP content) */
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struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
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int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
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struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
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int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
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struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
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int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
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int sdp_payload_type; /**< payload type */
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//@}
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/** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
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//@{
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/** handler structure */
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const RTPDynamicProtocolHandler *dynamic_handler;
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/** private data associated with the dynamic protocol */
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PayloadContext *dynamic_protocol_context;
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//@}
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/** Enable sending RTCP feedback messages according to RFC 4585 */
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int feedback;
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/** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
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uint32_t ssrc;
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char crypto_suite[40];
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char crypto_params[100];
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} RTSPStream;
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void ff_rtsp_parse_line(AVFormatContext *s,
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RTSPMessageHeader *reply, const char *buf,
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RTSPState *rt, const char *method);
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/**
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* Send a command to the RTSP server without waiting for the reply.
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*
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* @see rtsp_send_cmd_with_content_async
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*/
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int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
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const char *url, const char *headers);
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/**
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* Send a command to the RTSP server and wait for the reply.
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*
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* @param s RTSP (de)muxer context
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* @param method the method for the request
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* @param url the target url for the request
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* @param headers extra header lines to include in the request
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* @param reply pointer where the RTSP message header will be stored
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* @param content_ptr pointer where the RTSP message body, if any, will
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* be stored (length is in reply)
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* @param send_content if non-null, the data to send as request body content
|
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* @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 */
|