blob: 27d540a1988e9e8969fcdb08c75ea02bfce4fd72 [file] [log] [blame]
/*
* Copyright (c) 2017 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 <algorithm>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/gain_controller2.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
namespace {
void SetAudioBufferSamples(float value, AudioBuffer* ab) {
// Sets all the samples in |ab| to |value|.
for (size_t k = 0; k < ab->num_channels(); ++k) {
std::fill(ab->channels_f()[k], ab->channels_f()[k] + ab->num_frames(),
value);
}
}
float RunAgc2WithConstantInput(GainController2* agc2,
float input_level,
size_t num_frames,
int sample_rate) {
const int num_samples = rtc::CheckedDivExact(sample_rate, 100);
AudioBuffer ab(num_samples, 1, num_samples, 1, num_samples);
// Give time to the level estimator to converge.
for (size_t i = 0; i < num_frames + 1; ++i) {
SetAudioBufferSamples(input_level, &ab);
agc2->Process(&ab);
}
// Return the last sample from the last processed frame.
return ab.channels_f()[0][num_samples - 1];
}
AudioProcessing::Config::GainController2 CreateAgc2FixedDigitalModeConfig(
float fixed_gain_db) {
AudioProcessing::Config::GainController2 config;
config.adaptive_digital.enabled = false;
config.fixed_digital.gain_db = fixed_gain_db;
// TODO(alessiob): Check why ASSERT_TRUE() below does not compile.
EXPECT_TRUE(GainController2::Validate(config));
return config;
}
std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode(
float fixed_gain_db,
size_t sample_rate_hz) {
auto agc2 = absl::make_unique<GainController2>();
agc2->ApplyConfig(CreateAgc2FixedDigitalModeConfig(fixed_gain_db));
agc2->Initialize(sample_rate_hz);
return agc2;
}
float GainAfterProcessingFile(GainController2* gain_controller) {
// Set up an AudioBuffer to be filled from the speech file.
constexpr size_t kStereo = 2u;
const StreamConfig capture_config(AudioProcessing::kSampleRate48kHz, kStereo,
false);
AudioBuffer ab(capture_config.num_frames(), capture_config.num_channels(),
capture_config.num_frames(), capture_config.num_channels(),
capture_config.num_frames());
test::InputAudioFile capture_file(
test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
std::vector<float> capture_input(capture_config.num_frames() *
capture_config.num_channels());
// The file should contain at least this many frames. Every iteration, we put
// a frame through the gain controller.
const int kNumFramesToProcess = 100;
for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
ReadFloatSamplesFromStereoFile(capture_config.num_frames(),
capture_config.num_channels(), &capture_file,
capture_input);
test::CopyVectorToAudioBuffer(capture_config, capture_input, &ab);
gain_controller->Process(&ab);
}
// Send in a last frame with values constant 1 (It's low enough to detect high
// gain, and for ease of computation). The applied gain is the result.
constexpr float sample_value = 1.f;
SetAudioBufferSamples(sample_value, &ab);
gain_controller->Process(&ab);
return ab.channels_f()[0][0];
}
} // namespace
TEST(GainController2, CreateApplyConfig) {
// Instances GainController2 and applies different configurations.
std::unique_ptr<GainController2> gain_controller2(new GainController2());
// Check that the default config is valid.
AudioProcessing::Config::GainController2 config;
EXPECT_TRUE(GainController2::Validate(config));
gain_controller2->ApplyConfig(config);
// Check that attenuation is not allowed.
config.fixed_digital.gain_db = -5.f;
EXPECT_FALSE(GainController2::Validate(config));
// Check that valid configurations are applied.
for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 40.f}) {
config.fixed_digital.gain_db = fixed_gain_db;
EXPECT_TRUE(GainController2::Validate(config));
gain_controller2->ApplyConfig(config);
}
}
TEST(GainController2, ToString) {
// Tests GainController2::ToString(). Only test the enabled property.
AudioProcessing::Config::GainController2 config;
config.enabled = false;
EXPECT_EQ("{enabled: false", GainController2::ToString(config).substr(0, 15));
config.enabled = true;
EXPECT_EQ("{enabled: true", GainController2::ToString(config).substr(0, 14));
}
TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) {
constexpr float kInputLevel = 1000.f;
constexpr size_t kNumFrames = 5;
constexpr size_t kSampleRateHz = 8000;
constexpr float kGain0Db = 0.f;
constexpr float kGain20Db = 20.f;
auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz);
// Signal level is unchanged with 0 db gain.
EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
kNumFrames, kSampleRateHz),
kInputLevel);
// +20 db should increase signal by a factor of 10.
agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGain20Db));
EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
kNumFrames, kSampleRateHz),
kInputLevel * 10);
}
TEST(GainController2FixedDigital, ChangeFixedGainShouldBeFastAndTimeInvariant) {
// Number of frames required for the fixed gain controller to adapt on the
// input signal when the gain changes.
constexpr size_t kNumFrames = 5;
constexpr float kInputLevel = 1000.f;
constexpr size_t kSampleRateHz = 8000;
constexpr float kGainDbLow = 0.f;
constexpr float kGainDbHigh = 25.f;
static_assert(kGainDbLow < kGainDbHigh, "");
auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz);
// Start with a lower gain.
const float output_level_pre = RunAgc2WithConstantInput(
agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz);
// Increase gain.
agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbHigh));
static_cast<void>(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
kNumFrames, kSampleRateHz));
// Back to the lower gain.
agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbLow));
const float output_level_post = RunAgc2WithConstantInput(
agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz);
EXPECT_EQ(output_level_pre, output_level_post);
}
struct FixedDigitalTestParams {
FixedDigitalTestParams(float gain_db_min,
float gain_db_max,
size_t sample_rate,
bool saturation_expected)
: gain_db_min(gain_db_min),
gain_db_max(gain_db_max),
sample_rate(sample_rate),
saturation_expected(saturation_expected) {}
float gain_db_min;
float gain_db_max;
size_t sample_rate;
bool saturation_expected;
};
class FixedDigitalTest
: public testing::Test,
public testing::WithParamInterface<FixedDigitalTestParams> {};
TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) {
const float kInputLevel = 32767.f;
const size_t kNumFrames = 5;
const auto params = GetParam();
const auto gains_db =
test::LinSpace(params.gain_db_min, params.gain_db_max, 10);
for (const auto gain_db : gains_db) {
SCOPED_TRACE(std::to_string(gain_db));
auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, params.sample_rate);
const float processed_sample = RunAgc2WithConstantInput(
agc2_fixed.get(), kInputLevel, kNumFrames, params.sample_rate);
if (params.saturation_expected) {
EXPECT_FLOAT_EQ(processed_sample, 32767.f);
} else {
EXPECT_LT(processed_sample, 32767.f);
}
}
}
static_assert(test::kLimiterMaxInputLevelDbFs < 10, "");
INSTANTIATE_TEST_CASE_P(
GainController2,
FixedDigitalTest,
::testing::Values(
// When gain < |test::kLimiterMaxInputLevelDbFs|, the limiter will not
// saturate the signal (at any sample rate).
FixedDigitalTestParams(0.1f,
test::kLimiterMaxInputLevelDbFs - 0.01f,
8000,
false),
FixedDigitalTestParams(0.1,
test::kLimiterMaxInputLevelDbFs - 0.01f,
48000,
false),
// When gain > |test::kLimiterMaxInputLevelDbFs|, the limiter will
// saturate the signal (at any sample rate).
FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f,
10.f,
8000,
true),
FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f,
10.f,
48000,
true)));
TEST(GainController2, UsageSaturationMargin) {
GainController2 gain_controller2;
gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);
AudioProcessing::Config::GainController2 config;
// Check that samples are not amplified as much when extra margin is
// high. They should not be amplified at all, but only after convergence. GC2
// starts with a gain, and it takes time until it's down to 0 dB.
config.fixed_digital.gain_db = 0.f;
config.adaptive_digital.enabled = true;
config.adaptive_digital.extra_saturation_margin_db = 50.f;
gain_controller2.ApplyConfig(config);
EXPECT_LT(GainAfterProcessingFile(&gain_controller2), 2.f);
}
TEST(GainController2, UsageNoSaturationMargin) {
GainController2 gain_controller2;
gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);
AudioProcessing::Config::GainController2 config;
// Check that some gain is applied if there is no margin.
config.fixed_digital.gain_db = 0.f;
config.adaptive_digital.enabled = true;
config.adaptive_digital.extra_saturation_margin_db = 0.f;
gain_controller2.ApplyConfig(config);
EXPECT_GT(GainAfterProcessingFile(&gain_controller2), 2.f);
}
} // namespace test
} // namespace webrtc