Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
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@ -10,24 +10,85 @@
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#include "modules/audio_processing/agc2/agc2_testing_common.h"
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#include <math.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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std::vector<double> LinSpace(const double l,
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const double r,
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size_t num_points) {
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RTC_CHECK(num_points >= 2);
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std::vector<double> LinSpace(double l, double r, int num_points) {
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RTC_CHECK_GE(num_points, 2);
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std::vector<double> points(num_points);
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const double step = (r - l) / (num_points - 1.0);
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points[0] = l;
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for (size_t i = 1; i < num_points - 1; i++) {
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for (int i = 1; i < num_points - 1; i++) {
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points[i] = static_cast<double>(l) + i * step;
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}
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points[num_points - 1] = r;
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return points;
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}
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WhiteNoiseGenerator::WhiteNoiseGenerator(int min_amplitude, int max_amplitude)
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: rand_gen_(42),
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min_amplitude_(min_amplitude),
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max_amplitude_(max_amplitude) {
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RTC_DCHECK_LT(min_amplitude_, max_amplitude_);
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RTC_DCHECK_LE(kMinS16, min_amplitude_);
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RTC_DCHECK_LE(min_amplitude_, kMaxS16);
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RTC_DCHECK_LE(kMinS16, max_amplitude_);
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RTC_DCHECK_LE(max_amplitude_, kMaxS16);
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}
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float WhiteNoiseGenerator::operator()() {
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return static_cast<float>(rand_gen_.Rand(min_amplitude_, max_amplitude_));
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}
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SineGenerator::SineGenerator(float amplitude,
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float frequency_hz,
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int sample_rate_hz)
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: amplitude_(amplitude),
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frequency_hz_(frequency_hz),
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sample_rate_hz_(sample_rate_hz),
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x_radians_(0.0f) {
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RTC_DCHECK_GT(amplitude_, 0);
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RTC_DCHECK_LE(amplitude_, kMaxS16);
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}
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float SineGenerator::operator()() {
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constexpr float kPi = 3.1415926536f;
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x_radians_ += frequency_hz_ / sample_rate_hz_ * 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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// Use sinf instead of std::sinf for libstdc++ compatibility.
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return amplitude_ * sinf(x_radians_);
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}
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PulseGenerator::PulseGenerator(float pulse_amplitude,
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float no_pulse_amplitude,
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float frequency_hz,
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int sample_rate_hz)
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: pulse_amplitude_(pulse_amplitude),
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no_pulse_amplitude_(no_pulse_amplitude),
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samples_period_(
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static_cast<int>(static_cast<float>(sample_rate_hz) / frequency_hz)),
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sample_counter_(0) {
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RTC_DCHECK_GE(pulse_amplitude_, kMinS16);
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RTC_DCHECK_LE(pulse_amplitude_, kMaxS16);
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RTC_DCHECK_GT(no_pulse_amplitude_, kMinS16);
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RTC_DCHECK_LE(no_pulse_amplitude_, kMaxS16);
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RTC_DCHECK_GT(sample_rate_hz, frequency_hz);
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}
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float PulseGenerator::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>(sample_counter_ == 0 ? pulse_amplitude_
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: no_pulse_amplitude_);
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}
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} // namespace test
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} // namespace webrtc
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