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/*
* Copyright (c) 2021 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/capture_levels_adjuster/audio_samples_scaler.h"
#include <tuple>
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
float SampleValueForChannel(int channel) {
constexpr float kSampleBaseValue = 100.f;
constexpr float kSampleChannelOffset = 1.f;
return kSampleBaseValue + channel * kSampleChannelOffset;
}
void PopulateBuffer(AudioBuffer& audio_buffer) {
for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
test::FillBufferChannel(SampleValueForChannel(ch), ch, audio_buffer);
}
}
constexpr int kNumFramesToProcess = 10;
class AudioSamplesScalerTest
: public ::testing::Test,
public ::testing::WithParamInterface<std::tuple<int, int, float>> {
protected:
int sample_rate_hz() const { return std::get<0>(GetParam()); }
int num_channels() const { return std::get<1>(GetParam()); }
float initial_gain() const { return std::get<2>(GetParam()); }
};
INSTANTIATE_TEST_SUITE_P(
AudioSamplesScalerTestSuite,
AudioSamplesScalerTest,
::testing::Combine(::testing::Values(16000, 32000, 48000),
::testing::Values(1, 2, 4),
::testing::Values(0.1f, 1.f, 2.f, 4.f)));
TEST_P(AudioSamplesScalerTest, InitialGainIsRespected) {
AudioSamplesScaler scaler(initial_gain());
AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
num_channels(), sample_rate_hz(), num_channels());
for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
for (int ch = 0; ch < num_channels(); ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
initial_gain() * SampleValueForChannel(ch));
}
}
}
}
TEST_P(AudioSamplesScalerTest, VerifyGainAdjustment) {
const float higher_gain = initial_gain();
const float lower_gain = higher_gain / 2.f;
AudioSamplesScaler scaler(lower_gain);
AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
num_channels(), sample_rate_hz(), num_channels());
// Allow the intial, lower, gain to take effect.
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
// Set the new, higher, gain.
scaler.SetGain(higher_gain);
// Ensure that the new, higher, gain is achieved gradually over one frame.
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
for (int ch = 0; ch < num_channels(); ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
EXPECT_LT(audio_buffer.channels_const()[ch][i],
higher_gain * SampleValueForChannel(ch));
EXPECT_LE(audio_buffer.channels_const()[ch][i],
audio_buffer.channels_const()[ch][i + 1]);
}
EXPECT_LE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
higher_gain * SampleValueForChannel(ch));
}
// Ensure that the new, higher, gain is achieved and stay unchanged.
for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
for (int ch = 0; ch < num_channels(); ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
higher_gain * SampleValueForChannel(ch));
}
}
}
// Set the new, lower, gain.
scaler.SetGain(lower_gain);
// Ensure that the new, lower, gain is achieved gradually over one frame.
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
for (int ch = 0; ch < num_channels(); ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
EXPECT_GT(audio_buffer.channels_const()[ch][i],
lower_gain * SampleValueForChannel(ch));
EXPECT_GE(audio_buffer.channels_const()[ch][i],
audio_buffer.channels_const()[ch][i + 1]);
}
EXPECT_GE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
lower_gain * SampleValueForChannel(ch));
}
// Ensure that the new, lower, gain is achieved and stay unchanged.
for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
PopulateBuffer(audio_buffer);
scaler.Process(audio_buffer);
for (int ch = 0; ch < num_channels(); ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
lower_gain * SampleValueForChannel(ch));
}
}
}
}
TEST(AudioSamplesScaler, UpwardsClamping) {
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
constexpr float kGain = 10.f;
constexpr float kMaxClampedSampleValue = 32767.f;
static_assert(kGain > 1.f, "");
AudioSamplesScaler scaler(kGain);
AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
kNumChannels, kSampleRateHz, kNumChannels);
for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
test::FillBufferChannel(
kMaxClampedSampleValue - audio_buffer.num_channels() + 1.f + ch, ch,
audio_buffer);
}
scaler.Process(audio_buffer);
for (int ch = 0; ch < kNumChannels; ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
kMaxClampedSampleValue);
}
}
}
}
TEST(AudioSamplesScaler, DownwardsClamping) {
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
constexpr float kGain = 10.f;
constexpr float kMinClampedSampleValue = -32768.f;
static_assert(kGain > 1.f, "");
AudioSamplesScaler scaler(kGain);
AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
kNumChannels, kSampleRateHz, kNumChannels);
for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
test::FillBufferChannel(
kMinClampedSampleValue + audio_buffer.num_channels() - 1.f + ch, ch,
audio_buffer);
}
scaler.Process(audio_buffer);
for (int ch = 0; ch < kNumChannels; ++ch) {
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
kMinClampedSampleValue);
}
}
}
}
} // namespace
} // namespace webrtc
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