source: extensions/gsdl-video/trunk/installed/cmdline/include/ffmpeg/rtp.h@ 18425

Last change on this file since 18425 was 18425, checked in by davidb, 15 years ago

Video extension to Greenstone

File size: 3.6 KB
Line 
1/*
2 * RTP definitions
3 * Copyright (c) 2002 Fabrice Bellard.
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
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
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21#ifndef RTP_H
22#define RTP_H
23
24#include "avcodec.h"
25#include "avformat.h"
26
27#define RTP_MIN_PACKET_LENGTH 12
28#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
29
30int rtp_init(void);
31int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
32
33/** return < 0 if unknown payload type */
34int rtp_get_payload_type(AVCodecContext *codec);
35
36typedef struct RTPDemuxContext RTPDemuxContext;
37typedef struct rtp_payload_data_s rtp_payload_data_s;
38RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
39int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
40 const uint8_t *buf, int len);
41void rtp_parse_close(RTPDemuxContext *s);
42
43extern AVOutputFormat rtp_muxer;
44extern AVInputFormat rtp_demuxer;
45
46int rtp_get_local_port(URLContext *h);
47int rtp_set_remote_url(URLContext *h, const char *uri);
48void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
49
50/**
51 * some rtp servers assume client is dead if they don't hear from them...
52 * so we send a Receiver Report to the provided ByteIO context
53 * (we don't have access to the rtcp handle from here)
54 */
55int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
56
57extern URLProtocol rtp_protocol;
58
59#define RTP_PT_PRIVATE 96
60#define RTP_VERSION 2
61#define RTP_MAX_SDES 256 /**< maximum text length for SDES */
62
63/* RTCP paquets use 0.5 % of the bandwidth */
64#define RTCP_TX_RATIO_NUM 5
65#define RTCP_TX_RATIO_DEN 1000
66
67/** Structure listing useful vars to parse RTP packet payload*/
68typedef struct rtp_payload_data_s
69{
70 int sizelength;
71 int indexlength;
72 int indexdeltalength;
73 int profile_level_id;
74 int streamtype;
75 int objecttype;
76 char *mode;
77
78 /** mpeg 4 AU headers */
79 struct AUHeaders {
80 int size;
81 int index;
82 int cts_flag;
83 int cts;
84 int dts_flag;
85 int dts;
86 int rap_flag;
87 int streamstate;
88 } *au_headers;
89 int nb_au_headers;
90 int au_headers_length_bytes;
91 int cur_au_index;
92} rtp_payload_data_t;
93
94typedef struct AVRtpPayloadType_s
95{
96 int pt;
97 const char enc_name[50]; /* XXX: why 50 ? */
98 enum CodecType codec_type;
99 enum CodecID codec_id;
100 int clock_rate;
101 int audio_channels;
102} AVRtpPayloadType_t;
103
104#if 0
105typedef enum {
106 RTCP_SR = 200,
107 RTCP_RR = 201,
108 RTCP_SDES = 202,
109 RTCP_BYE = 203,
110 RTCP_APP = 204
111} rtcp_type_t;
112
113typedef enum {
114 RTCP_SDES_END = 0,
115 RTCP_SDES_CNAME = 1,
116 RTCP_SDES_NAME = 2,
117 RTCP_SDES_EMAIL = 3,
118 RTCP_SDES_PHONE = 4,
119 RTCP_SDES_LOC = 5,
120 RTCP_SDES_TOOL = 6,
121 RTCP_SDES_NOTE = 7,
122 RTCP_SDES_PRIV = 8,
123 RTCP_SDES_IMG = 9,
124 RTCP_SDES_DOOR = 10,
125 RTCP_SDES_SOURCE = 11
126} rtcp_sdes_type_t;
127#endif
128
129extern AVRtpPayloadType_t AVRtpPayloadTypes[];
130#endif /* RTP_H */
Note: See TracBrowser for help on using the repository browser.