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Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/packet.h

Issue 1345433002: Add RTC_ prefix to contructormagic macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Formatting fix. Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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107 const uint8_t* payload_; // First byte after header. 107 const uint8_t* payload_; // First byte after header.
108 const size_t packet_length_bytes_; // Total length of packet. 108 const size_t packet_length_bytes_; // Total length of packet.
109 size_t payload_length_bytes_; // Length of the payload, after RTP header. 109 size_t payload_length_bytes_; // Length of the payload, after RTP header.
110 // Zero for dummy RTP packets. 110 // Zero for dummy RTP packets.
111 // Virtual lengths are used when parsing RTP header files (dummy RTP files). 111 // Virtual lengths are used when parsing RTP header files (dummy RTP files).
112 const size_t virtual_packet_length_bytes_; 112 const size_t virtual_packet_length_bytes_;
113 size_t virtual_payload_length_bytes_; 113 size_t virtual_payload_length_bytes_;
114 double time_ms_; // Used to denote a packet's arrival time. 114 double time_ms_; // Used to denote a packet's arrival time.
115 bool valid_header_; // Set by the RtpHeaderParser. 115 bool valid_header_; // Set by the RtpHeaderParser.
116 116
117 DISALLOW_COPY_AND_ASSIGN(Packet); 117 RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
118 }; 118 };
119 119
120 } // namespace test 120 } // namespace test
121 } // namespace webrtc 121 } // namespace webrtc
122 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ 122 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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