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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
79 lines
2.0 KiB
C++
79 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/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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