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Side by Side Diff: webrtc/common_audio/audio_converter_unittest.cc

Issue 2358993004: Enable the -Wundef warning for clang (Closed)
Patch Set: rebase Created 4 years, 2 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
11 #include <cmath> 11 #include <cmath>
12 #include <algorithm> 12 #include <algorithm>
13 #include <memory> 13 #include <memory>
14 #include <vector> 14 #include <vector>
15 15
16 #include "testing/gtest/include/gtest/gtest.h" 16 #include "webrtc/test/gtest.h"
17 #include "webrtc/base/arraysize.h" 17 #include "webrtc/base/arraysize.h"
18 #include "webrtc/base/format_macros.h" 18 #include "webrtc/base/format_macros.h"
19 #include "webrtc/common_audio/audio_converter.h" 19 #include "webrtc/common_audio/audio_converter.h"
20 #include "webrtc/common_audio/channel_buffer.h" 20 #include "webrtc/common_audio/channel_buffer.h"
21 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" 21 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 typedef std::unique_ptr<ChannelBuffer<float>> ScopedBuffer; 25 typedef std::unique_ptr<ChannelBuffer<float>> ScopedBuffer;
26 26
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152 ++dst_channel) { 152 ++dst_channel) {
153 RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate], 153 RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
154 kChannels[dst_channel], kSampleRates[dst_rate]); 154 kChannels[dst_channel], kSampleRates[dst_rate]);
155 } 155 }
156 } 156 }
157 } 157 }
158 } 158 }
159 } 159 }
160 160
161 } // namespace webrtc 161 } // namespace webrtc
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