Mercurial > libavformat.hg
view rtp.h @ 831:8e1b338096a0 libavformat
changes to ignore command media embedded in MS WMV files patch by ("Brown, Mike": mikeb, vibephone com)
author | michael |
---|---|
date | Wed, 10 Aug 2005 23:41:20 +0000 |
parents | 021c8bf015c2 |
children | da1d5db0ce5c |
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/* * RTP definitions * Copyright (c) 2002 Fabrice Bellard. * * This library 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 of the License, or (at your option) any later version. * * This library 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 this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #ifndef RTP_H #define RTP_H #define RTP_MIN_PACKET_LENGTH 12 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */ int rtp_init(void); int rtp_get_codec_info(AVCodecContext *codec, int payload_type); int rtp_get_payload_type(AVCodecContext *codec); typedef struct RTPDemuxContext RTPDemuxContext; typedef struct rtp_payload_data_s rtp_payload_data_s; RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data); int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len); void rtp_parse_close(RTPDemuxContext *s); extern AVOutputFormat rtp_mux; extern AVInputFormat rtp_demux; int rtp_get_local_port(URLContext *h); int rtp_set_remote_url(URLContext *h, const char *uri); void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd); extern URLProtocol rtp_protocol; #define RTP_PT_PRIVATE 96 #define RTP_VERSION 2 #define RTP_MAX_SDES 256 /* maximum text length for SDES */ /* RTCP paquets use 0.5 % of the bandwidth */ #define RTCP_TX_RATIO_NUM 5 #define RTCP_TX_RATIO_DEN 1000 /* Structure listing usefull vars to parse RTP packet payload*/ typedef struct rtp_payload_data_s { int sizelength; int indexlength; int indexdeltalength; int profile_level_id; int streamtype; int objecttype; char *mode; /* mpeg 4 AU headers */ struct AUHeaders { int size; int index; int cts_flag; int cts; int dts_flag; int dts; int rap_flag; int streamstate; } *au_headers; int nb_au_headers; int au_headers_length_bytes; int cur_au_index; } rtp_payload_data_t; typedef struct AVRtpPayloadType_s { int pt; const char enc_name[50]; /* XXX: why 50 ? */ enum CodecType codec_type; enum CodecID codec_id; int clock_rate; int audio_channels; } AVRtpPayloadType_t; typedef struct AVRtpDynamicPayloadType_s /* payload type >= 96 */ { const char enc_name[50]; /* XXX: still why 50 ? ;-) */ enum CodecType codec_type; enum CodecID codec_id; } AVRtpDynamicPayloadType_t; #if 0 typedef enum { RTCP_SR = 200, RTCP_RR = 201, RTCP_SDES = 202, RTCP_BYE = 203, RTCP_APP = 204 } rtcp_type_t; typedef enum { RTCP_SDES_END = 0, RTCP_SDES_CNAME = 1, RTCP_SDES_NAME = 2, RTCP_SDES_EMAIL = 3, RTCP_SDES_PHONE = 4, RTCP_SDES_LOC = 5, RTCP_SDES_TOOL = 6, RTCP_SDES_NOTE = 7, RTCP_SDES_PRIV = 8, RTCP_SDES_IMG = 9, RTCP_SDES_DOOR = 10, RTCP_SDES_SOURCE = 11 } rtcp_sdes_type_t; #endif extern AVRtpPayloadType_t AVRtpPayloadTypes[]; extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[]; #endif /* RTP_H */