mirror of
https://gitee.com/openharmony/third_party_ffmpeg
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131 lines
3.6 KiB
C
131 lines
3.6 KiB
C
/*
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* RTP 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 RTP_H
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#define RTP_H
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#include "avcodec.h"
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#include "avformat.h"
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#define RTP_MIN_PACKET_LENGTH 12
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#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
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int rtp_init(void);
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int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
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/** return < 0 if unknown payload type */
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int rtp_get_payload_type(AVCodecContext *codec);
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typedef struct RTPDemuxContext RTPDemuxContext;
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typedef struct rtp_payload_data_s rtp_payload_data_s;
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len);
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void rtp_parse_close(RTPDemuxContext *s);
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extern AVOutputFormat rtp_muxer;
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extern AVInputFormat rtp_demuxer;
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int rtp_get_local_port(URLContext *h);
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int rtp_set_remote_url(URLContext *h, const char *uri);
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void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
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/**
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* some rtp servers assume client is dead if they don't hear from them...
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* so we send a Receiver Report to the provided ByteIO context
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* (we don't have access to the rtcp handle from here)
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*/
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
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extern URLProtocol rtp_protocol;
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#define RTP_PT_PRIVATE 96
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#define RTP_VERSION 2
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#define RTP_MAX_SDES 256 /**< maximum text length for SDES */
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/* RTCP paquets use 0.5 % of the bandwidth */
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#define RTCP_TX_RATIO_NUM 5
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#define RTCP_TX_RATIO_DEN 1000
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/** Structure listing useful vars to parse RTP packet payload*/
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typedef struct rtp_payload_data_s
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{
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int sizelength;
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int indexlength;
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int indexdeltalength;
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int profile_level_id;
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int streamtype;
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int objecttype;
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char *mode;
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/** mpeg 4 AU headers */
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struct AUHeaders {
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int size;
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int index;
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int cts_flag;
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int cts;
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int dts_flag;
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int dts;
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int rap_flag;
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int streamstate;
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} *au_headers;
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int nb_au_headers;
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int au_headers_length_bytes;
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int cur_au_index;
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} rtp_payload_data_t;
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typedef struct AVRtpPayloadType_s
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{
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int pt;
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const char enc_name[50]; /* XXX: why 50 ? */
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enum CodecType codec_type;
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enum CodecID codec_id;
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int clock_rate;
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int audio_channels;
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} AVRtpPayloadType_t;
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#if 0
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typedef enum {
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RTCP_SR = 200,
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RTCP_RR = 201,
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RTCP_SDES = 202,
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RTCP_BYE = 203,
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RTCP_APP = 204
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} rtcp_type_t;
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typedef enum {
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RTCP_SDES_END = 0,
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RTCP_SDES_CNAME = 1,
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RTCP_SDES_NAME = 2,
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RTCP_SDES_EMAIL = 3,
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RTCP_SDES_PHONE = 4,
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RTCP_SDES_LOC = 5,
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RTCP_SDES_TOOL = 6,
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RTCP_SDES_NOTE = 7,
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RTCP_SDES_PRIV = 8,
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RTCP_SDES_IMG = 9,
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RTCP_SDES_DOOR = 10,
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RTCP_SDES_SOURCE = 11
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} rtcp_sdes_type_t;
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#endif
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extern AVRtpPayloadType_t AVRtpPayloadTypes[];
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#endif /* RTP_H */
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