Files
platform-external-webrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
Alex Loiko 8a3eaddc95 Pre-amplification in the audio processing module.
Added a new sub-module 'GainApplier'. The build target is
'modules/audio_processing/agc2:gain_applier'. A small refactoring
makes the GainApplier used in adaptive-digital AGC2.

The AGC2 now multiplies samples with a gain in 3 places. It's the
GainApplier, the GainCurveApplier, and the FixedGainController. The
GainApplier is used in AdaptiveDigitalGainApplier and will be used as
a pre-amplifier.

Bug: webrtc:9138
Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6
Reviewed-on: https://webrtc-review.googlesource.com/69321
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 10:19:58 +00:00

95 lines
3.6 KiB
C++

/*
* 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 "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);
gain_applier.ApplyGain(fake_audio.float_frame_view());
EXPECT_NEAR(fake_audio.float_frame_view().channel(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.float_frame_view());
EXPECT_NEAR(fake_audio.float_frame_view().channel(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.float_frame_view());
EXPECT_NEAR(fake_audio.float_frame_view().channel(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.float_frame_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.float_frame_view().channel(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.float_frame_view());
// The last sample should have the new gain.
EXPECT_NEAR(next_fake_audio_frame.float_frame_view().channel(0)[0],
initial_signal_level * target_gain_factor, 0.1f);
}
} // namespace webrtc