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553 lines
20 KiB
553 lines
20 KiB
/* |
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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 Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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|
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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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|
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#include "libavutil/log.h" |
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#include "libavutil/opt.h" |
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|
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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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}; |
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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_NB |
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}; |
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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 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 10000 |
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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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/** data/packet transport protocol; e.g. RTP or RDT */ |
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enum RTSPTransport transport; |
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|
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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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|
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/** number of items in the 'transports' variable below */ |
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int nb_transports; |
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|
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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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|
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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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|
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int seq; /**< sequence number */ |
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|
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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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|
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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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|
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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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|
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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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} 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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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/** 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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/** |
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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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} 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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/** |
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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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|
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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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|
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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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|
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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 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 (rtp_*.c/rdt.c) */ |
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//@{ |
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/** handler structure */ |
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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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} RTSPStream; |
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void ff_rtsp_parse_line(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 |
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* send_content is null |
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* |
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* @return zero if success, nonzero otherwise |
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*/ |
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int ff_rtsp_send_cmd_with_content(AVFormatContext *s, |
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const char *method, const char *url, |
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const char *headers, |
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RTSPMessageHeader *reply, |
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unsigned char **content_ptr, |
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const unsigned char *send_content, |
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int send_content_length); |
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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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* @see rtsp_send_cmd_with_content |
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*/ |
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int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, |
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const char *url, const char *headers, |
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RTSPMessageHeader *reply, unsigned char **content_ptr); |
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|
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/** |
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* Read a RTSP message from the server, or prepare to read data |
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* packets if we're reading data interleaved over the TCP/RTSP |
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* connection as well. |
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* |
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* @param s RTSP (de)muxer context |
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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 return_on_interleaved_data whether the function may return if we |
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* encounter a data marker ('$'), which precedes data |
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* packets over interleaved TCP/RTSP connections. If this |
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* is set, this function will return 1 after encountering |
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* a '$'. If it is not set, the function will skip any |
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* data packets (if they are encountered), until a reply |
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* has been fully parsed. If no more data is available |
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* without parsing a reply, it will return an error. |
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* @param method the RTSP method this is a reply to. This affects how |
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* some response headers are acted upon. May be NULL. |
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* |
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* @return 1 if a data packets is ready to be received, -1 on error, |
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* and 0 on success. |
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*/ |
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int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, |
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unsigned char **content_ptr, |
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int return_on_interleaved_data, const char *method); |
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|
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/** |
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* Skip a RTP/TCP interleaved packet. |
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*/ |
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void ff_rtsp_skip_packet(AVFormatContext *s); |
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|
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/** |
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* Connect to the RTSP server and set up the individual media streams. |
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* This can be used for both muxers and demuxers. |
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* |
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* @param s RTSP (de)muxer context |
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* |
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* @return 0 on success, < 0 on error. Cleans up all allocations done |
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* within the function on error. |
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*/ |
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int ff_rtsp_connect(AVFormatContext *s); |
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|
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/** |
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* Close and free all streams within the RTSP (de)muxer |
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* |
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* @param s RTSP (de)muxer context |
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*/ |
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void ff_rtsp_close_streams(AVFormatContext *s); |
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|
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/** |
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* Close all connection handles within the RTSP (de)muxer |
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* |
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* @param s RTSP (de)muxer context |
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*/ |
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void ff_rtsp_close_connections(AVFormatContext *s); |
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|
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/** |
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* Get the description of the stream and set up the RTSPStream child |
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* objects. |
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*/ |
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int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply); |
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|
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/** |
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* Announce the stream to the server and set up the RTSPStream child |
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* objects for each media stream. |
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*/ |
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int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr); |
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|
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/** |
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* Parse an SDP description of streams by populating an RTSPState struct |
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* within the AVFormatContext; also allocate the RTP streams and the |
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* pollfd array used for UDP streams. |
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*/ |
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int ff_sdp_parse(AVFormatContext *s, const char *content); |
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|
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/** |
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* Receive one RTP packet from an TCP interleaved RTSP stream. |
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*/ |
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int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
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uint8_t *buf, int buf_size); |
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|
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/** |
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* Receive one packet from the RTSPStreams set up in the AVFormatContext |
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* (which should contain a RTSPState struct as priv_data). |
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*/ |
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int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt); |
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|
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/** |
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* Do the SETUP requests for each stream for the chosen |
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* lower transport mode. |
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* @return 0 on success, <0 on error, 1 if protocol is unavailable |
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*/ |
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int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, |
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int lower_transport, const char *real_challenge); |
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|
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/** |
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* Undo the effect of ff_rtsp_make_setup_request, close the |
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* transport_priv and rtp_handle fields. |
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*/ |
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void ff_rtsp_undo_setup(AVFormatContext *s); |
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|
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extern const AVOption ff_rtsp_options[]; |
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#endif /* AVFORMAT_RTSP_H */
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