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119b466811
types and their non-standard extensions, and the data they serve. Practically, this patch allows Real servers to serve normal non-RDT (standard RTP) data. See discussion on ML in "Realmedia patch" thread. Originally committed as revision 15484 to svn://svn.ffmpeg.org/ffmpeg/trunk
100 lines
3.2 KiB
C
100 lines
3.2 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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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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/**
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* This is not part of public API and shouldn't be used outside of ffmpeg.
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*/
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RTSP_LOWER_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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int 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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} RTSPHeader;
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/** the callback can be used to extend the connection setup/teardown step */
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enum RTSPCallbackAction {
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RTSP_ACTION_SERVER_SETUP,
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RTSP_ACTION_SERVER_TEARDOWN,
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RTSP_ACTION_CLIENT_SETUP,
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RTSP_ACTION_CLIENT_TEARDOWN,
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};
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typedef struct RTSPActionServerSetup {
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uint32_t ipaddr;
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char transport_option[512];
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} RTSPActionServerSetup;
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typedef int FFRTSPCallback(enum RTSPCallbackAction action,
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const char *session_id,
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char *buf, int buf_size,
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void *arg);
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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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