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127 lines
3.5 KiB
127 lines
3.5 KiB
/* |
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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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#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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