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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc | 94 |
1 files changed, 94 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc new file mode 100644 index 0000000000..8f3e9db4c4 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc @@ -0,0 +1,94 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_processing/agc2/agc2_testing_common.h" + +#include <math.h> + +#include "rtc_base/checks.h" + +namespace webrtc { +namespace test { + +std::vector<double> LinSpace(double l, double r, int num_points) { + RTC_CHECK_GE(num_points, 2); + std::vector<double> points(num_points); + const double step = (r - l) / (num_points - 1.0); + points[0] = l; + for (int i = 1; i < num_points - 1; i++) { + points[i] = static_cast<double>(l) + i * step; + } + points[num_points - 1] = r; + return points; +} + +WhiteNoiseGenerator::WhiteNoiseGenerator(int min_amplitude, int max_amplitude) + : rand_gen_(42), + min_amplitude_(min_amplitude), + max_amplitude_(max_amplitude) { + RTC_DCHECK_LT(min_amplitude_, max_amplitude_); + RTC_DCHECK_LE(kMinS16, min_amplitude_); + RTC_DCHECK_LE(min_amplitude_, kMaxS16); + RTC_DCHECK_LE(kMinS16, max_amplitude_); + RTC_DCHECK_LE(max_amplitude_, kMaxS16); +} + +float WhiteNoiseGenerator::operator()() { + return static_cast<float>(rand_gen_.Rand(min_amplitude_, max_amplitude_)); +} + +SineGenerator::SineGenerator(float amplitude, + float frequency_hz, + int sample_rate_hz) + : amplitude_(amplitude), + frequency_hz_(frequency_hz), + sample_rate_hz_(sample_rate_hz), + x_radians_(0.0f) { + RTC_DCHECK_GT(amplitude_, 0); + RTC_DCHECK_LE(amplitude_, kMaxS16); +} + +float SineGenerator::operator()() { + constexpr float kPi = 3.1415926536f; + x_radians_ += frequency_hz_ / sample_rate_hz_ * 2 * kPi; + if (x_radians_ >= 2 * kPi) { + x_radians_ -= 2 * kPi; + } + // Use sinf instead of std::sinf for libstdc++ compatibility. + return amplitude_ * sinf(x_radians_); +} + +PulseGenerator::PulseGenerator(float pulse_amplitude, + float no_pulse_amplitude, + float frequency_hz, + int sample_rate_hz) + : pulse_amplitude_(pulse_amplitude), + no_pulse_amplitude_(no_pulse_amplitude), + samples_period_( + static_cast<int>(static_cast<float>(sample_rate_hz) / frequency_hz)), + sample_counter_(0) { + RTC_DCHECK_GE(pulse_amplitude_, kMinS16); + RTC_DCHECK_LE(pulse_amplitude_, kMaxS16); + RTC_DCHECK_GT(no_pulse_amplitude_, kMinS16); + RTC_DCHECK_LE(no_pulse_amplitude_, kMaxS16); + RTC_DCHECK_GT(sample_rate_hz, frequency_hz); +} + +float PulseGenerator::operator()() { + sample_counter_++; + if (sample_counter_ >= samples_period_) { + sample_counter_ -= samples_period_; + } + return static_cast<float>(sample_counter_ == 0 ? pulse_amplitude_ + : no_pulse_amplitude_); +} + +} // namespace test +} // namespace webrtc |