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Side by Side Diff: webrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc

Issue 2808513003: Add SafeClamp(), which accepts args of different types (Closed)
Patch Set: rebase Created 3 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2017 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_processing/aec3/adaptive_fir_filter.h" 11 #include "webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h"
12 12
13 #include <math.h> 13 #include <math.h>
14 #include <algorithm> 14 #include <algorithm>
15 #include <numeric> 15 #include <numeric>
16 #include <string> 16 #include <string>
17 #include "webrtc/typedefs.h" 17 #include "webrtc/typedefs.h"
18 #if defined(WEBRTC_ARCH_X86_FAMILY) 18 #if defined(WEBRTC_ARCH_X86_FAMILY)
19 #include <emmintrin.h> 19 #include <emmintrin.h>
20 #endif 20 #endif
21 #include "webrtc/base/arraysize.h" 21 #include "webrtc/base/arraysize.h"
22 #include "webrtc/base/random.h" 22 #include "webrtc/base/random.h"
23 #include "webrtc/base/safe_minmax.h"
23 #include "webrtc/modules/audio_processing/aec3/aec3_fft.h" 24 #include "webrtc/modules/audio_processing/aec3/aec3_fft.h"
24 #include "webrtc/modules/audio_processing/aec3/aec_state.h" 25 #include "webrtc/modules/audio_processing/aec3/aec_state.h"
25 #include "webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h" 26 #include "webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h"
26 #include "webrtc/modules/audio_processing/aec3/render_signal_analyzer.h" 27 #include "webrtc/modules/audio_processing/aec3/render_signal_analyzer.h"
27 #include "webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h" 28 #include "webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h"
28 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" 29 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
29 #include "webrtc/modules/audio_processing/test/echo_canceller_test_tools.h" 30 #include "webrtc/modules/audio_processing/test/echo_canceller_test_tools.h"
30 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h" 31 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
31 #include "webrtc/test/gtest.h" 32 #include "webrtc/test/gtest.h"
32 33
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343 x_hp_filter.Process(x[0]); 344 x_hp_filter.Process(x[0]);
344 y_hp_filter.Process(y); 345 y_hp_filter.Process(y);
345 346
346 render_buffer.Insert(x); 347 render_buffer.Insert(x);
347 render_signal_analyzer.Update(render_buffer, aec_state.FilterDelay()); 348 render_signal_analyzer.Update(render_buffer, aec_state.FilterDelay());
348 349
349 filter.Filter(render_buffer, &S); 350 filter.Filter(render_buffer, &S);
350 fft.Ifft(S, &s); 351 fft.Ifft(S, &s);
351 std::transform(y.begin(), y.end(), s.begin() + kFftLengthBy2, e.begin(), 352 std::transform(y.begin(), y.end(), s.begin() + kFftLengthBy2, e.begin(),
352 [&](float a, float b) { return a - b * kScale; }); 353 [&](float a, float b) { return a - b * kScale; });
353 std::for_each(e.begin(), e.end(), [](float& a) { 354 std::for_each(e.begin(), e.end(),
354 a = std::max(std::min(a, 32767.0f), -32768.0f); 355 [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
355 });
356 fft.ZeroPaddedFft(e, &E); 356 fft.ZeroPaddedFft(e, &E);
357 357
358 gain.Compute(render_buffer, render_signal_analyzer, E, 358 gain.Compute(render_buffer, render_signal_analyzer, E,
359 filter.SizePartitions(), false, &G); 359 filter.SizePartitions(), false, &G);
360 filter.Adapt(render_buffer, G); 360 filter.Adapt(render_buffer, G);
361 aec_state.HandleEchoPathChange(EchoPathVariability(false, false)); 361 aec_state.HandleEchoPathChange(EchoPathVariability(false, false));
362 aec_state.Update(filter.FilterFrequencyResponse(), 362 aec_state.Update(filter.FilterFrequencyResponse(),
363 rtc::Optional<size_t>(), render_buffer, E2_main, Y2, 363 rtc::Optional<size_t>(), render_buffer, E2_main, Y2,
364 x[0], false); 364 x[0], false);
365 } 365 }
366 // Verify that the filter is able to perform well. 366 // Verify that the filter is able to perform well.
367 EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f), 367 EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
368 std::inner_product(y.begin(), y.end(), y.begin(), 0.f)); 368 std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
369 ASSERT_TRUE(aec_state.FilterDelay()); 369 ASSERT_TRUE(aec_state.FilterDelay());
370 EXPECT_EQ(delay_samples / kBlockSize, *aec_state.FilterDelay()); 370 EXPECT_EQ(delay_samples / kBlockSize, *aec_state.FilterDelay());
371 } 371 }
372 } 372 }
373 } // namespace aec3 373 } // namespace aec3
374 } // namespace webrtc 374 } // namespace webrtc
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