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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.
*
* 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
* 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
* 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 |
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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
#define AVFORMAT_RTSP_H |
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|
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#include <stdint.h>
#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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/**
* Network layer over which RTP/etc packet data will be transported.
*/ |
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enum RTSPLowerTransport { |
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RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */ |
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RTSP_LOWER_TRANSPORT_NB |
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};
|
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/**
* Packet profile of the data that we will be receiving. Real servers
* commonly send RDT (although they can sometimes send RTP as well),
* whereas most others will send RTP.
*/ |
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enum RTSPTransport { |
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RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ |
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RTSP_TRANSPORT_NB |
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};
|
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/**
* 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) */
};
|
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#define RTSP_DEFAULT_PORT 554
#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
#define RTSP_RTP_PORT_MIN 5000
#define RTSP_RTP_PORT_MAX 10000 |
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|
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/**
* This describes a single item in the "Transport:" line of one stream as
* negotiated by the SETUP RTSP command. Multiple transports are comma-
* separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
* client_port=1000-1001;server_port=1800-1801") and described in separate
* RTSPTransportFields.
*/ |
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typedef struct RTSPTransportField { |
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/** interleave ids, if TCP transport; each TCP/RTSP data packet starts
* with a '$', stream length and stream ID. If the stream ID is within
* the range of this interleaved_min-max, then the packet belongs to
* this stream. */
int interleaved_min, interleaved_max;
/** UDP multicast port range; the ports to which we should connect to
* receive multicast UDP data. */
int port_min, port_max;
/** UDP client ports; these should be the local ports of the UDP RTP
* (and RTCP) sockets over which we receive RTP/RTCP data. */
int client_port_min, client_port_max;
/** UDP unicast server port range; the ports to which we should connect
* to receive unicast UDP RTP/RTCP data. */
int server_port_min, server_port_max;
/** time-to-live value (required for multicast); the amount of HOPs that
* packets will be allowed to make before being discarded. */
int ttl;
|
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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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/**
* This describes the server response to each RTSP command.
*/ |
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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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|
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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
* 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
* 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 */
/** the "Session:" field. This value is initially set by the server and
* 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.
*/
char location[4096];
|
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/** the "RealChallenge1:" field from the server */
char real_challenge[64];
/** the "Server: field, which can be used to identify some special-case
* servers that are not 100% standards-compliant. We use this to identify
* Windows Media Server, which has a value "WMServer/v.e.r.sion", where
* version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
* use something like "Helix [..] Server Version v.e.r.sion (platform)
* (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
* 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"
* command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
* time, in seconds, that the server will go without traffic over the
* RTSP/TCP connection before it closes the connection. To prevent
* this, sent dummy requests (e.g. OPTIONS) with intervals smaller
* than this value. */
int timeout; |
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/** The "Notice" or "X-Notice" field value. See
* http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
* for a complete list of supported values. */
int notice; |
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/** The "reason" is meant to specify better the meaning of the error code
* returned
*/
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
* setup-but-not-receiving (PAUSED). State can be changed in applications
* by calling av_read_play/pause().
*/ |
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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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/**
* Identifies particular servers that require special handling, such as
* standards-incompliant "Transport:" lines in the SETUP request.
*/ |
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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 */
RTSP_SERVER_WMS, /**< Windows Media server */ |
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RTSP_SERVER_NB |
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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 */
/** indicator of whether we are currently receiving data from the
* server. Basically this isn't more than a simple cache of the
* last PLAY/PAUSE command sent to the server, to make sure we don't
* 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
* is subsequently used as part of the "Range" parameter when emitting
* the RTSP PLAY command. If we are currently playing, this command is
* called instantly. If we are currently paused, this command is called
* whenever we resume playback. Either way, the value is only used once,
* see rtsp_read_play() and rtsp_read_seek(). */ |
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int64_t seek_timestamp;
/* XXX: currently we use unbuffered input */ |
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// AVIOContext rtsp_gb; |
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int seq; /**< RTSP command sequence number */
/** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
* identifier that the client should re-transmit in each RTSP command */ |
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char session_id[512]; |
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|
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/** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
* the server will go without traffic on the RTSP/TCP line before it
* closes the connection. */
int timeout;
/** timestamp of the last RTSP command that we sent to the RTSP server.
* 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
* 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.
* Detected based on the value of RTSPMessageHeader->server or the presence
* of RTSPMessageHeader->real_challenge */ |
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enum RTSPServerType server_type; |
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|
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/** the "RealChallenge1:" field from the server */
char real_challenge[64];
|
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/** plaintext authorization line (username:password) */
char auth[128];
/** authentication state */
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
* packet from */ |
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void *cur_transport_priv; |
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/** The following are used for Real stream selection */
//@{
/** 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
* we need to subscribe or unsubscribe to any new streams. */ |
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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; |
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/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
* this is used to send the same "Unsubscribe:" if stream setup changed,
* 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 */
//@{
/** ASF demuxer context for the embedded ASF stream from WMS servers */
AVFormatContext *asf_ctx; |
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/** cache for position of the asf demuxer, since we load a new
* data packet in the bytecontext for each incoming RTSP packet. */
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
* for all subsequent RTSP requests, rather than the input URI; in
* other cases, this is a copy of AVFormatContext->filename. */
char control_uri[1024]; |
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|
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/** Additional output handle, used when input and output are done
* separately, eg for HTTP tunneling. */
URLContext *rtsp_hd_out; |
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/** RTSP transport mode, such as plain or tunneled. */
enum RTSPControlTransport control_transport; |
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/* 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; |
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/** Reusable buffer for receiving packets */
uint8_t* recvbuf; |
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/** Filter incoming UDP packets - receive packets only from the right
* source address and port. */
int filter_source; |
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/**
* A mask with all requested transport methods
*/
int lower_transport_mask;
/**
* The number of returned packets
*/
uint64_t packets; |
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/**
* Polling array for udp
*/
struct pollfd *p; |
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/**
* Whether the server supports the GET_PARAMETER method.
*/
int get_parameter_supported; |
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/**
* Do not begin to play the stream immediately.
*/
int initial_pause; |
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/**
* Option flags for the chained RTP muxer.
*/
int rtp_muxer_flags; |
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} RTSPState;
|
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/**
* Describes 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).
*/ |
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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) */
int stream_index;
/** interleave IDs; copies of RTSPTransportField->interleaved_min/max
* for the selected transport. Only used for TCP. */
int interleaved_min, interleaved_max;
char control_url[1024]; /**< url for this stream (from SDP) */
/** The following are used only in SDP, not RTSP */
//@{
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) */
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) */
//@{
/** handler structure */
RTPDynamicProtocolHandler *dynamic_handler; |
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|
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/** private data associated with the dynamic protocol */
PayloadContext *dynamic_protocol_context;
//@} |
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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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extern int rtsp_rtp_port_min;
extern int rtsp_rtp_port_max; |
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|
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/**
* Send a command to the RTSP server without waiting for the reply.
*
* @see rtsp_send_cmd_with_content_async
*/ |
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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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/**
* Send a command to the RTSP server and wait for the reply.
*
* @param s RTSP (de)muxer context |
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* @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 |
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* @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 |
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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,
const char *headers,
RTSPMessageHeader *reply,
unsigned char **content_ptr,
const unsigned char *send_content,
int send_content_length); |
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/**
* Send a command to the RTSP server and wait for the reply.
*
* @see rtsp_send_cmd_with_content
*/ |
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int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, |
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const char *url, const char *headers,
RTSPMessageHeader *reply, unsigned char **content_ptr); |
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/**
* 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. |
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* @param method the RTSP method this is a reply to. This affects how
* 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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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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* Skip a RTP/TCP interleaved packet.
*/
void ff_rtsp_skip_packet(AVFormatContext *s);
/** |
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* 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
* |
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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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int ff_rtsp_connect(AVFormatContext *s); |
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/**
* Close and free all streams within the RTSP (de)muxer
*
* @param s RTSP (de)muxer context
*/ |
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void ff_rtsp_close_streams(AVFormatContext *s); |
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|
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/**
* Close all connection handles within the RTSP (de)muxer
*
* @param rt RTSP (de)muxer context
*/
void ff_rtsp_close_connections(AVFormatContext *rt);
|
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/** |
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* Get the description of the stream and set up the RTSPStream child
* objects.
*/
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
/** |
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* 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);
|
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/** |
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* 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. |
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*/
int ff_sdp_parse(AVFormatContext *s, const char *content);
/**
* Receive one RTP packet from an TCP interleaved RTSP stream.
*/
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size);
/**
* Receive one packet from the RTSPStreams set up in the AVFormatContext
* (which should contain a RTSPState struct as priv_data).
*/
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
|
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/**
* Do the SETUP requests for each stream for the chosen
* lower transport mode. |
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* @return 0 on success, <0 on error, 1 if protocol is unavailable |
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*/
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge);
|
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/**
* Undo the effect of ff_rtsp_make_setup_request, close the
* transport_priv and rtp_handle fields.
*/
void ff_rtsp_undo_setup(AVFormatContext *s);
|
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#endif /* AVFORMAT_RTSP_H */ |