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/*
* RTSP definitions
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFORMAT_RTSP_H
#define AVFORMAT_RTSP_H
#include <stdint.h>
#include "avformat.h"
#include "rtspcodes.h"
#include "rtpdec.h"
#include "network.h"
#include "httpauth.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
/**
* Network layer over which RTP/etc packet data will be transported.
*/
enum RTSPLowerTransport {
RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
RTSP_LOWER_TRANSPORT_NB,
RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
transport mode as such,
only for use via AVOptions */
};
/**
* 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.
*/
enum RTSPTransport {
RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
RTSP_TRANSPORT_NB
};
/**
* Transport mode for the RTSP data. This may be plain, or
* tunneled, which is done over HTTP.
*/
enum RTSPControlTransport {
RTSP_MODE_PLAIN, /**< Normal RTSP */
RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
};
#define RTSP_DEFAULT_PORT 554
#define RTSP_MAX_TRANSPORTS 8
#define RTSP_TCP_MAX_PACKET_SIZE 1472
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
#define RTSP_RTP_PORT_MIN 5000
#define RTSP_RTP_PORT_MAX 10000
/**
* 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.
*/
typedef struct RTSPTransportField {
/** 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;
struct sockaddr_storage destination; /**< destination IP address */
char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
/** data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
enum RTSPLowerTransport lower_transport;
} RTSPTransportField;
/**
* This describes the server response to each RTSP command.
*/
typedef struct RTSPMessageHeader {
/** length of the data following this header */
int content_length;
enum RTSPStatusCode status_code; /**< response code from server */
/** number of items in the 'transports' variable below */
int nb_transports;
/** Time range of the streams that the server will stream. In
* AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
int64_t range_start, range_end;
/** describes the complete "Transport:" line of the server in response
* to a SETUP RTSP command by the client */
RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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. */
char session_id[512];
/** the "Location:" field. This value is used to handle redirection.
*/
char location[4096];
/** 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'. */
char server[64];
/** 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;
/** 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;
/** The "reason" is meant to specify better the meaning of the error code
* returned
*/
char reason[256];
} RTSPMessageHeader;
/**
* 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().
*/
enum RTSPClientState {
RTSP_STATE_IDLE, /**< not initialized */
RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
};
/**
* Identify particular servers that require special handling, such as
* standards-incompliant "Transport:" lines in the SETUP request.
*/
enum RTSPServerType {
RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
RTSP_SERVER_REAL, /**< Realmedia-style server */
RTSP_SERVER_WMS, /**< Windows Media server */
RTSP_SERVER_NB
};
/**
* Private data for the RTSP demuxer.
*
* @todo Use AVIOContext instead of URLContext
*/
typedef struct RTSPState {
const AVClass *class; /**< Class for private options. */
URLContext *rtsp_hd; /* RTSP TCP connection handle */
/** number of items in the 'rtsp_streams' variable */
int nb_rtsp_streams;
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. */
enum RTSPClientState state;
/** 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(). */
int64_t seek_timestamp;
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 */
char session_id[512];
/** 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
* connection alive, in conjunction with timeout. */
int64_t last_cmd_time;
/** the negotiated data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** the negotiated network layer transport protocol; e.g. TCP or UDP
* uni-/multicast */
enum RTSPLowerTransport lower_transport;
/** 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 */
enum RTSPServerType server_type;
/** the "RealChallenge1:" field from the server */
char real_challenge[64];
/** plaintext authorization line (username:password) */
char auth[128];
/** authentication state */
HTTPAuthState auth_state;
/** The last reply of the server to a RTSP command */
char last_reply[2048]; /* XXX: allocate ? */
/** RTSPStream->transport_priv of the last stream that we read a
* packet from */
void *cur_transport_priv;
/** The following are used for Real stream selection */
//@{
/** whether we need to send a "SET_PARAMETER Subscribe:" command */
int need_subscription;
/** stream setup during the last frame read. This is used to detect if
* we need to subscribe or unsubscribe to any new streams. */
enum AVDiscard *real_setup_cache;
/** current stream setup. This is a temporary buffer used to compare
* current setup to previous frame setup. */
enum AVDiscard *real_setup;
/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
* this is used to send the same "Unsubscribe:" if stream setup changed,
* before sending a new "Subscribe:" command. */
char last_subscription[1024];
//@}
/** The following are used for RTP/ASF streams */
//@{
/** ASF demuxer context for the embedded ASF stream from WMS servers */
AVFormatContext *asf_ctx;
/** 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;
//@}
/** some MS RTSP streams contain a URL in the SDP that we need to use
* for all subsequent RTSP requests, rather than the input URI; in
* other cases, this is a copy of AVFormatContext->filename. */
char control_uri[1024];
/** Additional output handle, used when input and output are done
* separately, eg for HTTP tunneling. */
URLContext *rtsp_hd_out;
/** RTSP transport mode, such as plain or tunneled. */
enum RTSPControlTransport control_transport;
/* Number of RTCP BYE packets the RTSP session has received.
* An EOF is propagated back if nb_byes == nb_streams.
* This is reset after a seek. */
int nb_byes;
/** Reusable buffer for receiving packets */
uint8_t* recvbuf;
/**
* A mask with all requested transport methods
*/
int lower_transport_mask;
/**
* The number of returned packets
*/
uint64_t packets;
/**
* Polling array for udp
*/
struct pollfd *p;
/**
* Whether the server supports the GET_PARAMETER method.
*/
int get_parameter_supported;
/**
* Do not begin to play the stream immediately.
*/
int initial_pause;
/**
* Option flags for the chained RTP muxer.
*/
int rtp_muxer_flags;
/** Whether the server accepts the x-Dynamic-Rate header */
int accept_dynamic_rate;
/**
* Various option flags for the RTSP muxer/demuxer.
*/
int rtsp_flags;
/**
* Mask of all requested media types
*/
int media_type_mask;
} RTSPState;
#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
receive packets only from the right
source address and port. */
/**
* Describe a single stream, as identified by a single m= line block in the
* SDP content. In the case of RDT, one RTSPStream can represent multiple
* AVStreams. In this case, each AVStream in this set has similar content
* (but different codec/bitrate).
*/
typedef struct RTSPStream {
URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
/** 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) */
struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
int sdp_payload_type; /**< payload type */
//@}
/** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
//@{
/** handler structure */
RTPDynamicProtocolHandler *dynamic_handler;
/** private data associated with the dynamic protocol */
PayloadContext *dynamic_protocol_context;
//@}
} RTSPStream;
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
RTSPState *rt, const char *method);
extern int rtsp_rtp_port_min;
extern int rtsp_rtp_port_max;
/**
* Send a command to the RTSP server without waiting for the reply.
*
* @see rtsp_send_cmd_with_content_async
*/
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
const char *url, const char *headers);
/**
* Send a command to the RTSP server and wait for the reply.
*
* @param s RTSP (de)muxer context
* @param method the method for the request
* @param url the target url for the request
* @param headers extra header lines to include in the request
* @param reply pointer where the RTSP message header will be stored
* @param content_ptr pointer where the RTSP message body, if any, will
* be stored (length is in reply)
* @param send_content if non-null, the data to send as request body content
* @param send_content_length the length of the send_content data, or 0 if
* send_content is null
*
* @return zero if success, nonzero otherwise
*/
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
const char *method, const char *url,
const char *headers,
RTSPMessageHeader *reply,
unsigned char **content_ptr,
const unsigned char *send_content,
int send_content_length);
/**
* Send a command to the RTSP server and wait for the reply.
*
* @see rtsp_send_cmd_with_content
*/
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
const char *url, const char *headers,
RTSPMessageHeader *reply, unsigned char **content_ptr);
/**
* Read a RTSP message from the server, or prepare to read data
* packets if we're reading data interleaved over the TCP/RTSP
* connection as well.
*
* @param s RTSP (de)muxer context
* @param reply pointer where the RTSP message header will be stored
* @param content_ptr pointer where the RTSP message body, if any, will
* be stored (length is in reply)
* @param return_on_interleaved_data whether the function may return if we
* encounter a data marker ('$'), which precedes data
* packets over interleaved TCP/RTSP connections. If this
* is set, this function will return 1 after encountering
* a '$'. If it is not set, the function will skip any
* data packets (if they are encountered), until a reply
* has been fully parsed. If no more data is available
* without parsing a reply, it will return an error.
* @param method the RTSP method this is a reply to. This affects how
* some response headers are acted upon. May be NULL.
*
* @return 1 if a data packets is ready to be received, -1 on error,
* and 0 on success.
*/
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
int return_on_interleaved_data, const char *method);
/**
* Skip a RTP/TCP interleaved packet.
*/
void ff_rtsp_skip_packet(AVFormatContext *s);
/**
* Connect to the RTSP server and set up the individual media streams.
* This can be used for both muxers and demuxers.
*
* @param s RTSP (de)muxer context
*
* @return 0 on success, < 0 on error. Cleans up all allocations done
* within the function on error.
*/
int ff_rtsp_connect(AVFormatContext *s);
/**
* Close and free all streams within the RTSP (de)muxer
*
* @param s RTSP (de)muxer context
*/
void ff_rtsp_close_streams(AVFormatContext *s);
/**
* Close all connection handles within the RTSP (de)muxer
*
* @param s RTSP (de)muxer context
*/
void ff_rtsp_close_connections(AVFormatContext *s);
/**
* Get the description of the stream and set up the RTSPStream child
* objects.
*/
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
/**
* Announce the stream to the server and set up the RTSPStream child
* objects for each media stream.
*/
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
/**
* Parse 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);
/**
* 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);
extern const AVOption ff_rtsp_options[];
#endif /* AVFORMAT_RTSP_H */
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