Mercurial > libavformat.hg
annotate rtp.h @ 3125:bc818669cdbd libavformat
Put network-related code under #ifdef
| author | ramiro |
|---|---|
| date | Sat, 08 Mar 2008 17:43:53 +0000 |
| parents | 0d0ffa1a0478 |
| children | 6f61c3b36632 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 * RTP definitions | |
| 3 * Copyright (c) 2002 Fabrice Bellard. | |
| 4 * | |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
| 0 | 8 * modify it under the terms of the GNU Lesser General Public |
| 9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
| 0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
| 0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
| 15 * Lesser General Public License for more details. | |
| 16 * | |
| 17 * You should have received a copy of the GNU Lesser General Public | |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 0 | 20 */ |
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21 #ifndef FFMPEG_RTP_H |
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22 #define FFMPEG_RTP_H |
| 0 | 23 |
| 2171 | 24 #include "avcodec.h" |
| 25 #include "avformat.h" | |
| 26 | |
| 0 | 27 #define RTP_MIN_PACKET_LENGTH 12 |
| 28 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */ | |
| 29 | |
| 30 int rtp_get_codec_info(AVCodecContext *codec, int payload_type); | |
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Move rtp_get_payload_type()'s unaltered comments to the header file.
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31 |
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32 /** return < 0 if unknown payload type */ |
| 0 | 33 int rtp_get_payload_type(AVCodecContext *codec); |
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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34 |
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35 typedef struct RTPDemuxContext RTPDemuxContext; |
| 774 | 36 typedef struct rtp_payload_data_s rtp_payload_data_s; |
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make ffmpeg able to send back a RTCP receiver report.
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37 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data); |
| 885 | 38 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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39 const uint8_t *buf, int len); |
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40 void rtp_parse_close(RTPDemuxContext *s); |
| 0 | 41 |
| 42 int rtp_get_local_port(URLContext *h); | |
| 43 int rtp_set_remote_url(URLContext *h, const char *uri); | |
| 44 void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd); | |
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45 |
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46 /** |
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47 * some rtp servers assume client is dead if they don't hear from them... |
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48 * so we send a Receiver Report to the provided ByteIO context |
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49 * (we don't have access to the rtcp handle from here) |
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50 */ |
| 1745 | 51 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count); |
| 0 | 52 |
| 774 | 53 #define RTP_PT_PRIVATE 96 |
| 54 #define RTP_VERSION 2 | |
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55 #define RTP_MAX_SDES 256 /**< maximum text length for SDES */ |
| 774 | 56 |
| 57 /* RTCP paquets use 0.5 % of the bandwidth */ | |
| 58 #define RTCP_TX_RATIO_NUM 5 | |
| 59 #define RTCP_TX_RATIO_DEN 1000 | |
| 60 | |
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61 /** Structure listing useful vars to parse RTP packet payload*/ |
| 774 | 62 typedef struct rtp_payload_data_s |
| 63 { | |
| 64 int sizelength; | |
| 65 int indexlength; | |
| 66 int indexdeltalength; | |
| 67 int profile_level_id; | |
| 68 int streamtype; | |
| 69 int objecttype; | |
| 70 char *mode; | |
| 71 | |
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72 /** mpeg 4 AU headers */ |
| 774 | 73 struct AUHeaders { |
| 74 int size; | |
| 75 int index; | |
| 76 int cts_flag; | |
| 77 int cts; | |
| 78 int dts_flag; | |
| 79 int dts; | |
| 80 int rap_flag; | |
| 81 int streamstate; | |
| 82 } *au_headers; | |
| 83 int nb_au_headers; | |
| 84 int au_headers_length_bytes; | |
| 85 int cur_au_index; | |
| 86 } rtp_payload_data_t; | |
| 87 | |
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RTCP_* conflict with live.com and they seem not to be used anywhere so commenting them out
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88 #if 0 |
| 774 | 89 typedef enum { |
| 90 RTCP_SR = 200, | |
| 91 RTCP_RR = 201, | |
| 92 RTCP_SDES = 202, | |
| 93 RTCP_BYE = 203, | |
| 94 RTCP_APP = 204 | |
| 95 } rtcp_type_t; | |
| 96 | |
| 97 typedef enum { | |
| 98 RTCP_SDES_END = 0, | |
| 99 RTCP_SDES_CNAME = 1, | |
| 100 RTCP_SDES_NAME = 2, | |
| 101 RTCP_SDES_EMAIL = 3, | |
| 102 RTCP_SDES_PHONE = 4, | |
| 103 RTCP_SDES_LOC = 5, | |
| 104 RTCP_SDES_TOOL = 6, | |
| 105 RTCP_SDES_NOTE = 7, | |
| 106 RTCP_SDES_PRIV = 8, | |
| 107 RTCP_SDES_IMG = 9, | |
| 108 RTCP_SDES_DOOR = 10, | |
| 109 RTCP_SDES_SOURCE = 11 | |
| 110 } rtcp_sdes_type_t; | |
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RTCP_* conflict with live.com and they seem not to be used anywhere so commenting them out
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111 #endif |
| 774 | 112 |
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113 #endif /* FFMPEG_RTP_H */ |
