Mercurial > libavformat.hg
annotate rtp.h @ 1978:9beecb275bae libavformat
cosmetics: Put audio (de)muxer entries in alphabetical order.
author | diego |
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date | Tue, 03 Apr 2007 15:57:38 +0000 |
parents | 48b1d0b70e09 |
children | a6d6b2b19341 |
rev | line source |
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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 */ |
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); | |
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29 |
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30 /** return < 0 if unknown payload type */ |
0 | 31 int rtp_get_payload_type(AVCodecContext *codec); |
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32 |
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33 typedef struct RTPDemuxContext RTPDemuxContext; |
774 | 34 typedef struct rtp_payload_data_s rtp_payload_data_s; |
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35 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data); |
885 | 36 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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37 const uint8_t *buf, int len); |
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38 void rtp_parse_close(RTPDemuxContext *s); |
0 | 39 |
1167 | 40 extern AVOutputFormat rtp_muxer; |
41 extern AVInputFormat rtp_demuxer; | |
0 | 42 |
43 int rtp_get_local_port(URLContext *h); | |
44 int rtp_set_remote_url(URLContext *h, const char *uri); | |
45 void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd); | |
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46 |
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47 /** |
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48 * some rtp servers assume client is dead if they don't hear from them... |
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49 * so we send a Receiver Report to the provided ByteIO context |
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50 * (we don't have access to the rtcp handle from here) |
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51 */ |
1745 | 52 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count); |
0 | 53 |
54 extern URLProtocol rtp_protocol; | |
55 | |
774 | 56 #define RTP_PT_PRIVATE 96 |
57 #define RTP_VERSION 2 | |
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58 #define RTP_MAX_SDES 256 /**< maximum text length for SDES */ |
774 | 59 |
60 /* RTCP paquets use 0.5 % of the bandwidth */ | |
61 #define RTCP_TX_RATIO_NUM 5 | |
62 #define RTCP_TX_RATIO_DEN 1000 | |
63 | |
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64 /** Structure listing useful vars to parse RTP packet payload*/ |
774 | 65 typedef struct rtp_payload_data_s |
66 { | |
67 int sizelength; | |
68 int indexlength; | |
69 int indexdeltalength; | |
70 int profile_level_id; | |
71 int streamtype; | |
72 int objecttype; | |
73 char *mode; | |
74 | |
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75 /** mpeg 4 AU headers */ |
774 | 76 struct AUHeaders { |
77 int size; | |
78 int index; | |
79 int cts_flag; | |
80 int cts; | |
81 int dts_flag; | |
82 int dts; | |
83 int rap_flag; | |
84 int streamstate; | |
85 } *au_headers; | |
86 int nb_au_headers; | |
87 int au_headers_length_bytes; | |
88 int cur_au_index; | |
89 } rtp_payload_data_t; | |
90 | |
91 typedef struct AVRtpPayloadType_s | |
92 { | |
93 int pt; | |
94 const char enc_name[50]; /* XXX: why 50 ? */ | |
95 enum CodecType codec_type; | |
96 enum CodecID codec_id; | |
97 int clock_rate; | |
98 int audio_channels; | |
99 } AVRtpPayloadType_t; | |
100 | |
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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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101 #if 0 |
774 | 102 typedef enum { |
103 RTCP_SR = 200, | |
104 RTCP_RR = 201, | |
105 RTCP_SDES = 202, | |
106 RTCP_BYE = 203, | |
107 RTCP_APP = 204 | |
108 } rtcp_type_t; | |
109 | |
110 typedef enum { | |
111 RTCP_SDES_END = 0, | |
112 RTCP_SDES_CNAME = 1, | |
113 RTCP_SDES_NAME = 2, | |
114 RTCP_SDES_EMAIL = 3, | |
115 RTCP_SDES_PHONE = 4, | |
116 RTCP_SDES_LOC = 5, | |
117 RTCP_SDES_TOOL = 6, | |
118 RTCP_SDES_NOTE = 7, | |
119 RTCP_SDES_PRIV = 8, | |
120 RTCP_SDES_IMG = 9, | |
121 RTCP_SDES_DOOR = 10, | |
122 RTCP_SDES_SOURCE = 11 | |
123 } 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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124 #endif |
774 | 125 |
126 extern AVRtpPayloadType_t AVRtpPayloadTypes[]; | |
0 | 127 #endif /* RTP_H */ |