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Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
284 lines
13 KiB
C++
284 lines
13 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/noise_suppression_impl.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "modules/audio_processing/test/bitexactness_tools.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 1000;
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// Process one frame of data and produce the output.
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void ProcessOneFrame(int sample_rate_hz,
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AudioBuffer* capture_buffer,
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NoiseSuppressionImpl* noise_suppressor) {
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_buffer->SplitIntoFrequencyBands();
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}
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noise_suppressor->AnalyzeCaptureAudio(capture_buffer);
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noise_suppressor->ProcessCaptureAudio(capture_buffer);
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_buffer->MergeFrequencyBands();
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}
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}
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// Processes a specified amount of frames, verifies the results and reports
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// any errors.
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void RunBitexactnessTest(int sample_rate_hz,
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size_t num_channels,
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NoiseSuppressionImpl::Level level,
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float speech_probability_reference,
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rtc::ArrayView<const float> noise_estimate_reference,
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rtc::ArrayView<const float> output_reference) {
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rtc::CriticalSection crit_capture;
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NoiseSuppressionImpl noise_suppressor(&crit_capture);
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noise_suppressor.Initialize(num_channels, sample_rate_hz);
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noise_suppressor.Enable(true);
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noise_suppressor.set_level(level);
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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const StreamConfig capture_config(sample_rate_hz, num_channels, false);
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AudioBuffer capture_buffer(
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames());
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test::InputAudioFile capture_file(
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test::GetApmCaptureTestVectorFileName(sample_rate_hz));
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std::vector<float> capture_input(samples_per_channel * num_channels);
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for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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ProcessOneFrame(sample_rate_hz, &capture_buffer, &noise_suppressor);
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}
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// Extract test results.
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std::vector<float> capture_output;
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test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
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&capture_output);
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float speech_probability = noise_suppressor.speech_probability();
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std::vector<float> noise_estimate = noise_suppressor.NoiseEstimate();
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const float kVectorElementErrorBound = 1.0f / 32768.0f;
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EXPECT_FLOAT_EQ(speech_probability_reference, speech_probability);
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EXPECT_TRUE(test::VerifyArray(noise_estimate_reference, noise_estimate,
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kVectorElementErrorBound));
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// Compare the output with the reference. Only the first values of the output
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// from last frame processed are compared in order not having to specify all
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// preceeding frames as testvectors. As the algorithm being tested has a
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// memory, testing only the last frame implicitly also tests the preceeding
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// frames.
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EXPECT_TRUE(test::VerifyDeinterleavedArray(
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capture_config.num_frames(), capture_config.num_channels(),
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output_reference, capture_output, kVectorElementErrorBound));
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}
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} // namespace
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TEST(NoiseSuppresionBitExactnessTest, Mono8kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1432.341431f, 3321.919922f,
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7677.521973f};
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const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1432.341431f, 3321.919922f,
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7677.521973f};
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const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
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#else
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const float kSpeechProbabilityReference = 0.73421317f;
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const float kNoiseEstimateReference[] = {1175.266113f, 3289.305908f,
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7532.991211f};
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const float kOutputReference[] = {0.003263f, 0.004402f, 0.004537f};
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#endif
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RunBitexactnessTest(8000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2534.461914f, 6277.638672f,
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14367.499023f};
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const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2534.461914f, 6277.638672f,
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14367.499023f};
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const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
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#else
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const float kSpeechProbabilityReference = 0.71672988f;
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const float kNoiseEstimateReference[] = {2151.313965f, 6509.765137f,
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15658.848633f};
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const float kOutputReference[] = {0.003574f, 0.004494f, 0.004499f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono32kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2540.059082f, 6317.822754f,
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14440.845703f};
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const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2540.059082f, 6317.822754f,
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14440.845703f};
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const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
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#else
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const float kSpeechProbabilityReference = 0.67999554f;
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const float kNoiseEstimateReference[] = {2149.780518f, 7076.936035f,
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14939.945312f};
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const float kOutputReference[] = {0.001221f, 0.001984f, 0.002228f};
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#endif
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RunBitexactnessTest(32000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono48kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2564.605713f, 6213.656250f,
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13372.284180f};
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const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2564.605713f, 6213.656250f,
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13372.284180f};
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const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
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#else
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const float kSpeechProbabilityReference = 0.70645678f;
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const float kNoiseEstimateReference[] = {2168.783203f, 6902.895508f,
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13190.677734f};
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const float kOutputReference[] = {-0.013062f, -0.012657f, -0.011934f};
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#endif
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RunBitexactnessTest(48000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Stereo16kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {9992.127930f, 12689.569336f,
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11589.296875f};
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const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
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-0.002441f, 0.000855f, -0.003204f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {10321.353516f, 12133.852539f,
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10923.060547f};
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const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
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-0.002472f, 0.000916f, -0.003235f};
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#else
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const float kSpeechProbabilityReference = 0.67230678f;
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const float kNoiseEstimateReference[] = {9771.250000f, 11329.377930f,
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10503.052734f};
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const float kOutputReference[] = {-0.011459f, -0.008110f, -0.012728f,
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-0.002399f, 0.001018f, -0.003189f};
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#endif
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RunBitexactnessTest(16000, 2, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzModerate) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2057.085938f, 7601.055176f,
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19666.187500f};
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const float kOutputReference[] = {0.004669f, 0.005524f, 0.005432f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2244.497803f, 6864.164062f,
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16726.523438f};
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const float kOutputReference[] = {0.004669f, 0.005615f, 0.005585f};
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#else
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const float kSpeechProbabilityReference = 0.70897013f;
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const float kNoiseEstimateReference[] = {2171.490723f, 6553.567871f,
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15626.562500f};
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const float kOutputReference[] = {0.004513f, 0.005590f, 0.005614f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kModerate,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzHigh) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2095.148193f, 7698.553711f,
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19689.533203f};
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const float kOutputReference[] = {0.004639f, 0.005402f, 0.005310f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2282.515625f, 6984.408203f,
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16920.960938f};
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const float kOutputReference[] = {0.004547f, 0.005432f, 0.005402f};
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#else
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const float kSpeechProbabilityReference = 0.70106733f;
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const float kNoiseEstimateReference[] = {2224.968506f, 6712.025879f,
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15785.087891f};
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const float kOutputReference[] = {0.004394f, 0.005406f, 0.005416f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kHigh,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzVeryHigh) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2677.733398f, 6186.987305f,
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14365.744141f};
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const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2677.733398f, 6186.987305f,
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14365.744141f};
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const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
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#else
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const float kSpeechProbabilityReference = 0.70281971f;
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const float kNoiseEstimateReference[] = {2254.347900f, 6723.699707f,
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15771.625977f};
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const float kOutputReference[] = {0.004321f, 0.005247f, 0.005263f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kVeryHigh,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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} // namespace webrtc
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