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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 "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) {
const int sample_rate_hz = 48000;
ApmDataDumper apm_data_dumper(0);
Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
kMaxAbsFloatS16Value);
limiter.Process(vectors_with_float_frame.float_frame_view());
}
TEST(Limiter, OutputVolumeAboveThreshold) {
const int sample_rate_hz = 48000;
const float input_level =
(kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
2.f;
ApmDataDumper apm_data_dumper(0);
Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
// Give the level estimator time to adapt.
for (int i = 0; i < 5; ++i) {
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
input_level);
limiter.Process(vectors_with_float_frame.float_frame_view());
}
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
input_level);
limiter.Process(vectors_with_float_frame.float_frame_view());
rtc::ArrayView<const float> channel =
vectors_with_float_frame.float_frame_view().channel(0);
for (const auto& sample : channel) {
EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
}
}
} // namespace webrtc
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