Mercurial > libavformat.hg
annotate rtp.h @ 1779:de2cf54eb68f libavformat
mxf aes decryption support, patch by Reimar, simplified to only look for first crypto context, will be extended once we get files with multiple cryptocontext, and hope that they won't have broken container ul
author | bcoudurier |
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date | Sun, 11 Feb 2007 12:50:33 +0000 |
parents | 699a422bca90 |
children | 4227bfda0df5 |
rev | line source |
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0 | 1 /* |
2 * RTP definitions | |
3 * Copyright (c) 2002 Fabrice Bellard. | |
4 * | |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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5 * This file is part of FFmpeg. |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
diego
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6 * |
0899bfe4105c
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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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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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18 * License along with FFmpeg; if not, write to the Free Software |
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Update licensing information: The FSF changed postal address.
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
21 #ifndef RTP_H | |
22 #define RTP_H | |
23 | |
24 #define RTP_MIN_PACKET_LENGTH 12 | |
25 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */ | |
26 | |
27 int rtp_init(void); | |
28 int rtp_get_codec_info(AVCodecContext *codec, int payload_type); | |
29 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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30 |
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31 typedef struct RTPDemuxContext RTPDemuxContext; |
774 | 32 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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33 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data); |
885 | 34 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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bellard
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35 const uint8_t *buf, int len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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36 void rtp_parse_close(RTPDemuxContext *s); |
0 | 37 |
1167 | 38 extern AVOutputFormat rtp_muxer; |
39 extern AVInputFormat rtp_demuxer; | |
0 | 40 |
41 int rtp_get_local_port(URLContext *h); | |
42 int rtp_set_remote_url(URLContext *h, const char *uri); | |
43 void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd); | |
1745 | 44 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count); |
0 | 45 |
46 extern URLProtocol rtp_protocol; | |
47 | |
774 | 48 #define RTP_PT_PRIVATE 96 |
49 #define RTP_VERSION 2 | |
50 #define RTP_MAX_SDES 256 /* maximum text length for SDES */ | |
51 | |
52 /* RTCP paquets use 0.5 % of the bandwidth */ | |
53 #define RTCP_TX_RATIO_NUM 5 | |
54 #define RTCP_TX_RATIO_DEN 1000 | |
55 | |
56 /* Structure listing usefull vars to parse RTP packet payload*/ | |
57 typedef struct rtp_payload_data_s | |
58 { | |
59 int sizelength; | |
60 int indexlength; | |
61 int indexdeltalength; | |
62 int profile_level_id; | |
63 int streamtype; | |
64 int objecttype; | |
65 char *mode; | |
66 | |
67 /* mpeg 4 AU headers */ | |
68 struct AUHeaders { | |
69 int size; | |
70 int index; | |
71 int cts_flag; | |
72 int cts; | |
73 int dts_flag; | |
74 int dts; | |
75 int rap_flag; | |
76 int streamstate; | |
77 } *au_headers; | |
78 int nb_au_headers; | |
79 int au_headers_length_bytes; | |
80 int cur_au_index; | |
81 } rtp_payload_data_t; | |
82 | |
83 typedef struct AVRtpPayloadType_s | |
84 { | |
85 int pt; | |
86 const char enc_name[50]; /* XXX: why 50 ? */ | |
87 enum CodecType codec_type; | |
88 enum CodecID codec_id; | |
89 int clock_rate; | |
90 int audio_channels; | |
91 } AVRtpPayloadType_t; | |
92 | |
803
021c8bf015c2
RTCP_* conflict with live.com and they seem not to be used anywhere so commenting them out
michael
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93 #if 0 |
774 | 94 typedef enum { |
95 RTCP_SR = 200, | |
96 RTCP_RR = 201, | |
97 RTCP_SDES = 202, | |
98 RTCP_BYE = 203, | |
99 RTCP_APP = 204 | |
100 } rtcp_type_t; | |
101 | |
102 typedef enum { | |
103 RTCP_SDES_END = 0, | |
104 RTCP_SDES_CNAME = 1, | |
105 RTCP_SDES_NAME = 2, | |
106 RTCP_SDES_EMAIL = 3, | |
107 RTCP_SDES_PHONE = 4, | |
108 RTCP_SDES_LOC = 5, | |
109 RTCP_SDES_TOOL = 6, | |
110 RTCP_SDES_NOTE = 7, | |
111 RTCP_SDES_PRIV = 8, | |
112 RTCP_SDES_IMG = 9, | |
113 RTCP_SDES_DOOR = 10, | |
114 RTCP_SDES_SOURCE = 11 | |
115 } rtcp_sdes_type_t; | |
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021c8bf015c2
RTCP_* conflict with live.com and they seem not to be used anywhere so commenting them out
michael
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116 #endif |
774 | 117 |
118 extern AVRtpPayloadType_t AVRtpPayloadTypes[]; | |
0 | 119 #endif /* RTP_H */ |