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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/limiter.h"
#include <algorithm>
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "modules/audio_processing/agc2/vector_float_frame.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/gunit.h"
namespace webrtc {
TEST(Limiter, LimiterShouldConstructAndRun) {
constexpr size_t kSamplesPerChannel = 480;
ApmDataDumper apm_data_dumper(0);
Limiter limiter(&apm_data_dumper, kSamplesPerChannel, "");
std::array<float, kSamplesPerChannel> buffer;
buffer.fill(kMaxAbsFloatS16Value);
limiter.Process(
DeinterleavedView<float>(buffer.data(), kSamplesPerChannel, 1));
}
TEST(Limiter, OutputVolumeAboveThreshold) {
constexpr size_t kSamplesPerChannel = 480;
const float input_level =
(kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
2.f;
ApmDataDumper apm_data_dumper(0);
Limiter limiter(&apm_data_dumper, kSamplesPerChannel, "");
std::array<float, kSamplesPerChannel> buffer;
// Give the level estimator time to adapt.
for (int i = 0; i < 5; ++i) {
std::fill(buffer.begin(), buffer.end(), input_level);
limiter.Process(
DeinterleavedView<float>(buffer.data(), kSamplesPerChannel, 1));
}
std::fill(buffer.begin(), buffer.end(), input_level);
limiter.Process(
DeinterleavedView<float>(buffer.data(), kSamplesPerChannel, 1));
for (const auto& sample : buffer) {
ASSERT_LT(0.9f * kMaxAbsFloatS16Value, sample);
}
}
} // namespace webrtc
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