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/*
* 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/gain_applier.h"
#include <math.h>
#include <algorithm>
#include <limits>
#include "api/audio/audio_view.h"
#include "modules/audio_processing/agc2/vector_float_frame.h"
#include "rtc_base/gunit.h"
namespace webrtc {
TEST(AutomaticGainController2GainApplier, InitialGainIsRespected) {
constexpr float initial_signal_level = 123.f;
constexpr float gain_factor = 10.f;
VectorFloatFrame fake_audio(1, 1, initial_signal_level);
GainApplier gain_applier(true, gain_factor);
auto fake_view = fake_audio.view();
gain_applier.ApplyGain(fake_audio.view());
EXPECT_NEAR(fake_view[0][0], initial_signal_level * gain_factor, 0.1f);
}
TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
constexpr float initial_signal_level = 30000.f;
constexpr float gain_factor = 10.f;
VectorFloatFrame fake_audio(1, 1, initial_signal_level);
GainApplier gain_applier(true, gain_factor);
gain_applier.ApplyGain(fake_audio.view());
EXPECT_NEAR(fake_audio.view()[0][0], std::numeric_limits<int16_t>::max(),
0.1f);
}
TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
constexpr float initial_signal_level = 30000.f;
constexpr float gain_factor = 10.f;
VectorFloatFrame fake_audio(1, 1, initial_signal_level);
GainApplier gain_applier(false, gain_factor);
gain_applier.ApplyGain(fake_audio.view());
EXPECT_NEAR(fake_audio.view()[0][0], initial_signal_level * gain_factor,
0.1f);
}
TEST(AutomaticGainController2GainApplier, RampingIsDone) {
constexpr float initial_signal_level = 30000.f;
constexpr float initial_gain_factor = 1.f;
constexpr float target_gain_factor = 0.5f;
constexpr int num_channels = 3;
constexpr int samples_per_channel = 4;
VectorFloatFrame fake_audio(num_channels, samples_per_channel,
initial_signal_level);
GainApplier gain_applier(false, initial_gain_factor);
gain_applier.SetGainFactor(target_gain_factor);
gain_applier.ApplyGain(fake_audio.view());
// The maximal gain change should be close to that in linear interpolation.
for (size_t channel = 0; channel < num_channels; ++channel) {
float max_signal_change = 0.f;
float last_signal_level = initial_signal_level;
for (const auto sample : fake_audio.view()[channel]) {
const float current_change = fabs(last_signal_level - sample);
max_signal_change = std::max(max_signal_change, current_change);
last_signal_level = sample;
}
const float total_gain_change =
fabs((initial_gain_factor - target_gain_factor) * initial_signal_level);
EXPECT_NEAR(max_signal_change, total_gain_change / samples_per_channel,
0.1f);
}
// Next frame should have the desired level.
VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel,
initial_signal_level);
gain_applier.ApplyGain(next_fake_audio_frame.view());
// The last sample should have the new gain.
EXPECT_NEAR(next_fake_audio_frame.view()[0][0],
initial_signal_level * target_gain_factor, 0.1f);
}
} // namespace webrtc
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