Mercurial > libavformat.hg
view rtsp.h @ 3383:2e9f19e0493b libavformat
Tell the user if a AVI is non interleaved.
author | michael |
---|---|
date | Wed, 28 May 2008 00:07:41 +0000 |
parents | 5a7a7406ab1f |
children | e43947d04029 |
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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 FFMPEG_RTSP_H #define FFMPEG_RTSP_H #include <stdint.h> #include "avformat.h" #include "rtspcodes.h" enum RTSPProtocol { RTSP_PROTOCOL_RTP_UDP = 0, RTSP_PROTOCOL_RTP_TCP = 1, RTSP_PROTOCOL_RTP_UDP_MULTICAST = 2, /** * This is not part of public API and shouldn't be used outside of ffmpeg. */ RTSP_PROTOCOL_RTP_LAST }; #define RTSP_DEFAULT_PORT 554 #define RTSP_MAX_TRANSPORTS 8 #define RTSP_TCP_MAX_PACKET_SIZE 1472 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 #define RTSP_RTP_PORT_MIN 5000 #define RTSP_RTP_PORT_MAX 10000 typedef struct RTSPTransportField { int interleaved_min, interleaved_max; /**< interleave ids, if TCP transport */ int port_min, port_max; /**< RTP ports */ int client_port_min, client_port_max; /**< RTP ports */ int server_port_min, server_port_max; /**< RTP ports */ int ttl; /**< ttl value */ uint32_t destination; /**< destination IP address */ enum RTSPProtocol protocol; } RTSPTransportField; typedef struct RTSPHeader { int content_length; enum RTSPStatusCode status_code; /**< response code from server */ int nb_transports; /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ int64_t range_start, range_end; RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; int seq; /**< sequence number */ char session_id[512]; } RTSPHeader; /** the callback can be used to extend the connection setup/teardown step */ enum RTSPCallbackAction { RTSP_ACTION_SERVER_SETUP, RTSP_ACTION_SERVER_TEARDOWN, RTSP_ACTION_CLIENT_SETUP, RTSP_ACTION_CLIENT_TEARDOWN, }; typedef struct RTSPActionServerSetup { uint32_t ipaddr; char transport_option[512]; } RTSPActionServerSetup; typedef int FFRTSPCallback(enum RTSPCallbackAction action, const char *session_id, char *buf, int buf_size, void *arg); int rtsp_init(void); void rtsp_parse_line(RTSPHeader *reply, const char *buf); #if LIBAVFORMAT_VERSION_INT < (53 << 16) extern int rtsp_default_protocols; #endif extern int rtsp_rtp_port_min; extern int rtsp_rtp_port_max; int rtsp_pause(AVFormatContext *s); int rtsp_resume(AVFormatContext *s); #endif /* FFMPEG_RTSP_H */