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3b0fc60592
which isn't installed anyway (so it doesn't work). In the process, also remove public/private API comments from rtsp headers because they are unnecessary. Originally committed as revision 17379 to svn://svn.ffmpeg.org/ffmpeg/trunk
141 lines
4.7 KiB
C
141 lines
4.7 KiB
C
/*
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* RTSP definitions
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef FFMPEG_RTSP_H
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#define FFMPEG_RTSP_H
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#include <stdint.h>
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#include "avformat.h"
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#include "rtspcodes.h"
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#include "rtpdec.h"
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#include "network.h"
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enum RTSPLowerTransport {
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RTSP_LOWER_TRANSPORT_UDP = 0,
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RTSP_LOWER_TRANSPORT_TCP = 1,
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RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
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RTSP_LOWER_TRANSPORT_LAST
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};
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enum RTSPTransport {
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RTSP_TRANSPORT_RTP,
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RTSP_TRANSPORT_RDT,
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RTSP_TRANSPORT_LAST
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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 2
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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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typedef struct RTSPTransportField {
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int interleaved_min, interleaved_max; /**< interleave ids, if TCP transport */
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int port_min, port_max; /**< RTP ports */
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int client_port_min, client_port_max; /**< RTP ports */
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int server_port_min, server_port_max; /**< RTP ports */
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int ttl; /**< ttl value */
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uint32_t destination; /**< destination IP address */
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enum RTSPTransport transport;
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enum RTSPLowerTransport lower_transport;
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} RTSPTransportField;
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typedef struct RTSPHeader {
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int content_length;
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enum RTSPStatusCode status_code; /**< response code from server */
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int nb_transports;
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/** 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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RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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int seq; /**< sequence number */
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char session_id[512];
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char real_challenge[64]; /**< the RealChallenge1 field from the server */
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char server[64];
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} RTSPHeader;
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enum RTSPClientState {
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RTSP_STATE_IDLE,
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RTSP_STATE_PLAYING,
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RTSP_STATE_PAUSED,
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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_LAST
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};
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typedef struct RTSPState {
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams;
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enum RTSPClientState state;
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int64_t seek_timestamp;
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/* XXX: currently we use unbuffered input */
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// ByteIOContext rtsp_gb;
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int seq; /* RTSP command sequence number */
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char session_id[512];
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enum RTSPTransport transport;
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enum RTSPLowerTransport lower_transport;
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enum RTSPServerType server_type;
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char last_reply[2048]; /* XXX: allocate ? */
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void *cur_transport_priv;
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int need_subscription;
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enum AVDiscard real_setup_cache[MAX_STREAMS];
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char last_subscription[1024];
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} RTSPState;
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typedef struct RTSPStream {
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URLContext *rtp_handle; /* RTP stream handle */
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void *transport_priv; /* RTP/RDT parse context */
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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char control_url[1024]; /* url for this stream (from SDP) */
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int sdp_port; /* port (from SDP content - not used in RTSP) */
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
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int sdp_payload_type; /* payload type - only used in SDP */
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RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
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PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
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} RTSPStream;
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int rtsp_init(void);
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void rtsp_parse_line(RTSPHeader *reply, const char *buf);
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#if LIBAVFORMAT_VERSION_INT < (53 << 16)
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extern int rtsp_default_protocols;
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#endif
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extern int rtsp_rtp_port_min;
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extern int rtsp_rtp_port_max;
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int rtsp_pause(AVFormatContext *s);
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int rtsp_resume(AVFormatContext *s);
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#endif /* FFMPEG_RTSP_H */
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