annotate rtp.h @ 2357:e9d7db467cf0 libavformat

merge some REGISTER_MUXER/DEMUXER into REGISTER_MUXDEMUX
author aurel
date Fri, 10 Aug 2007 01:44:50 +0000
parents a6d6b2b19341
children 89030c7e2fa7
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1 /*
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2 * RTP definitions
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3 * Copyright (c) 2002 Fabrice Bellard.
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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
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8 * modify it under the terms of the GNU Lesser General Public
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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.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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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
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20 */
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21 #ifndef RTP_H
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22 #define RTP_H
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23
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24 #include "avcodec.h"
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25 #include "avformat.h"
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26
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27 #define RTP_MIN_PACKET_LENGTH 12
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28 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
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29
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30 int rtp_init(void);
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31 int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
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32
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33 /** return < 0 if unknown payload type */
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34 int rtp_get_payload_type(AVCodecContext *codec);
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35
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36 typedef struct RTPDemuxContext RTPDemuxContext;
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37 typedef struct rtp_payload_data_s rtp_payload_data_s;
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38 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
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39 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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40 const uint8_t *buf, int len);
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41 void rtp_parse_close(RTPDemuxContext *s);
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42
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43 extern AVOutputFormat rtp_muxer;
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44 extern AVInputFormat rtp_demuxer;
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45
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46 int rtp_get_local_port(URLContext *h);
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47 int rtp_set_remote_url(URLContext *h, const char *uri);
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48 void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
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49
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50 /**
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51 * some rtp servers assume client is dead if they don't hear from them...
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52 * so we send a Receiver Report to the provided ByteIO context
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53 * (we don't have access to the rtcp handle from here)
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54 */
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55 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
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56
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57 extern URLProtocol rtp_protocol;
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58
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59 #define RTP_PT_PRIVATE 96
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60 #define RTP_VERSION 2
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61 #define RTP_MAX_SDES 256 /**< maximum text length for SDES */
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62
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63 /* RTCP paquets use 0.5 % of the bandwidth */
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64 #define RTCP_TX_RATIO_NUM 5
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65 #define RTCP_TX_RATIO_DEN 1000
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66
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67 /** Structure listing useful vars to parse RTP packet payload*/
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68 typedef struct rtp_payload_data_s
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69 {
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70 int sizelength;
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71 int indexlength;
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72 int indexdeltalength;
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73 int profile_level_id;
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74 int streamtype;
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75 int objecttype;
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76 char *mode;
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77
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78 /** mpeg 4 AU headers */
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79 struct AUHeaders {
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80 int size;
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81 int index;
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82 int cts_flag;
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83 int cts;
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84 int dts_flag;
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85 int dts;
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86 int rap_flag;
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87 int streamstate;
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88 } *au_headers;
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89 int nb_au_headers;
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90 int au_headers_length_bytes;
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91 int cur_au_index;
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92 } rtp_payload_data_t;
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93
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94 typedef struct AVRtpPayloadType_s
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95 {
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96 int pt;
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97 const char enc_name[50]; /* XXX: why 50 ? */
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98 enum CodecType codec_type;
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99 enum CodecID codec_id;
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100 int clock_rate;
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101 int audio_channels;
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102 } AVRtpPayloadType_t;
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103
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104 #if 0
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105 typedef enum {
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106 RTCP_SR = 200,
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107 RTCP_RR = 201,
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108 RTCP_SDES = 202,
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109 RTCP_BYE = 203,
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110 RTCP_APP = 204
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111 } rtcp_type_t;
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112
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113 typedef enum {
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114 RTCP_SDES_END = 0,
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115 RTCP_SDES_CNAME = 1,
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116 RTCP_SDES_NAME = 2,
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117 RTCP_SDES_EMAIL = 3,
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118 RTCP_SDES_PHONE = 4,
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119 RTCP_SDES_LOC = 5,
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120 RTCP_SDES_TOOL = 6,
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121 RTCP_SDES_NOTE = 7,
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122 RTCP_SDES_PRIV = 8,
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123 RTCP_SDES_IMG = 9,
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124 RTCP_SDES_DOOR = 10,
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125 RTCP_SDES_SOURCE = 11
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126 } rtcp_sdes_type_t;
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127 #endif
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128
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129 extern AVRtpPayloadType_t AVRtpPayloadTypes[];
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130 #endif /* RTP_H */