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1 /* | 1 /* |
2 * libjingle | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * Copyright 2004 Google Inc. | |
4 * | 3 * |
5 * Redistribution and use in source and binary forms, with or without | 4 * Use of this source code is governed by a BSD-style license |
6 * modification, are permitted provided that the following conditions are met: | 5 * that can be found in the LICENSE file in the root of the source |
7 * | 6 * tree. An additional intellectual property rights grant can be found |
8 * 1. Redistributions of source code must retain the above copyright notice, | 7 * in the file PATENTS. All contributing project authors may |
9 * this list of conditions and the following disclaimer. | 8 * be found in the AUTHORS file in the root of the source tree. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
11 * this list of conditions and the following disclaimer in the documentation | |
12 * and/or other materials provided with the distribution. | |
13 * 3. The name of the author may not be used to endorse or promote products | |
14 * derived from this software without specific prior written permission. | |
15 * | |
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
26 */ | 9 */ |
27 | 10 |
28 // Fake RTP and RTCP packets to use in unit tests. | 11 // Fake RTP and RTCP packets to use in unit tests. |
29 | 12 |
30 #ifndef WEBRTC_MEDIA_BASE_FAKERTP_H_ | 13 #ifndef WEBRTC_MEDIA_BASE_FAKERTP_H_ |
31 #define WEBRTC_MEDIA_BASE_FAKERTP_H_ | 14 #define WEBRTC_MEDIA_BASE_FAKERTP_H_ |
32 | 15 |
33 // A typical PCMU RTP packet. | 16 // A typical PCMU RTP packet. |
34 // PT=0, SN=1, TS=0, SSRC=1 | 17 // PT=0, SN=1, TS=0, SSRC=1 |
35 // all data FF | 18 // all data FF |
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95 }; | 78 }; |
96 | 79 |
97 // PT= 101, SN=2, TS=3, SSRC = 4 | 80 // PT= 101, SN=2, TS=3, SSRC = 4 |
98 static const unsigned char kDataPacket[] = { | 81 static const unsigned char kDataPacket[] = { |
99 0x80, 0x65, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x04, | 82 0x80, 0x65, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x04, |
100 0x00, 0x00, 0x00, 0x00, | 83 0x00, 0x00, 0x00, 0x00, |
101 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, | 84 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, |
102 }; | 85 }; |
103 | 86 |
104 #endif // WEBRTC_MEDIA_BASE_FAKERTP_H_ | 87 #endif // WEBRTC_MEDIA_BASE_FAKERTP_H_ |
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