We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
80 lines
2.0 KiB
C++
80 lines
2.0 KiB
C++
/*
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* Copyright (c) 2018 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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#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
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#include <math.h>
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#include <limits>
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#include <vector>
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#include "rtc_base/basictypes.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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// Level Estimator test parameters.
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constexpr float kDecayMs = 500.f;
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// Limiter parameters.
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constexpr float kLimiterMaxInputLevelDbFs = 1.f;
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constexpr float kLimiterKneeSmoothnessDb = 1.f;
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constexpr float kLimiterCompressionRatio = 5.f;
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constexpr float kPi = 3.1415926536f;
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std::vector<double> LinSpace(const double l, const double r, size_t num_points);
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class SineGenerator {
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public:
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SineGenerator(float frequency, int rate)
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: frequency_(frequency), rate_(rate) {}
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float operator()() {
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x_radians_ += frequency_ / rate_ * 2 * kPi;
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if (x_radians_ > 2 * kPi) {
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x_radians_ -= 2 * kPi;
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}
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return 1000.f * sinf(x_radians_);
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}
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private:
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float frequency_;
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int rate_;
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float x_radians_ = 0.f;
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};
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class PulseGenerator {
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public:
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PulseGenerator(float frequency, int rate)
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: samples_period_(
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static_cast<int>(static_cast<float>(rate) / frequency)) {
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RTC_DCHECK_GT(rate, frequency);
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}
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float operator()() {
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sample_counter_++;
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if (sample_counter_ >= samples_period_) {
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sample_counter_ -= samples_period_;
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}
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return static_cast<float>(
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sample_counter_ == 0 ? std::numeric_limits<int16_t>::max() : 10.f);
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}
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private:
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int samples_period_;
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int sample_counter_ = 0;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
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