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Side by Side Diff: webrtc/modules/audio_coding/test/PacketLossTest.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years 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
11 #include "webrtc/modules/audio_coding/main/test/PacketLossTest.h" 11 #include "webrtc/modules/audio_coding/test/PacketLossTest.h"
12 12
13 #include "testing/gtest/include/gtest/gtest.h" 13 #include "testing/gtest/include/gtest/gtest.h"
14 #include "webrtc/common.h" 14 #include "webrtc/common.h"
15 #include "webrtc/test/testsupport/fileutils.h" 15 #include "webrtc/test/testsupport/fileutils.h"
16 16
17 namespace webrtc { 17 namespace webrtc {
18 18
19 ReceiverWithPacketLoss::ReceiverWithPacketLoss() 19 ReceiverWithPacketLoss::ReceiverWithPacketLoss()
20 : loss_rate_(0), 20 : loss_rate_(0),
21 burst_length_(1), 21 burst_length_(1),
(...skipping 136 matching lines...) Expand 10 before | Expand all | Expand 10 after
158 158
159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
160 actual_loss_rate_, burst_length_); 160 actual_loss_rate_, burst_length_);
161 receiver_->Run(); 161 receiver_->Run();
162 receiver_->Teardown(); 162 receiver_->Teardown();
163 rtpFile.Close(); 163 rtpFile.Close();
164 #endif 164 #endif
165 } 165 }
166 166
167 } // namespace webrtc 167 } // namespace webrtc
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