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