Mercurial > libavformat.hg
view rtp.h @ 3534:e04ae2357e66 libavformat
Fix A/V synch for RTP streams that do not contain MPEG1 or 2
(correctly compute the presentation times based on the RTP timestamps
and the RTCP SR packets)
author | lucabe |
---|---|
date | Wed, 02 Jul 2008 10:23:27 +0000 |
parents | 6f61c3b36632 |
children | bf39ed8d3d69 |
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/* * RTP 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_RTP_H #define FFMPEG_RTP_H #include "libavcodec/avcodec.h" #include "avformat.h" #define RTP_MIN_PACKET_LENGTH 12 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */ int rtp_get_codec_info(AVCodecContext *codec, int payload_type); /** return < 0 if unknown 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, URLContext *rtpc, 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); 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); /** * some rtp servers assume client is dead if they don't hear from them... * so we send a Receiver Report to the provided ByteIO context * (we don't have access to the rtcp handle from here) */ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count); #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 useful 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; #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 #endif /* FFMPEG_RTP_H */