chromium/third_party/webrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc

/*
 *  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/agc2/clipping_predictor_level_buffer.h"

#include <algorithm>

#include "test/gmock.h"
#include "test/gtest.h"

namespace webrtc {
namespace {

Eq;
Optional;

class ClippingPredictorLevelBufferParametrization
    : public ::testing::TestWithParam<int> { … };

TEST_P(ClippingPredictorLevelBufferParametrization, CheckEmptyBufferSize) { … }

TEST_P(ClippingPredictorLevelBufferParametrization, CheckHalfEmptyBufferSize) { … }

TEST_P(ClippingPredictorLevelBufferParametrization, CheckFullBufferSize) { … }

TEST_P(ClippingPredictorLevelBufferParametrization, CheckLargeBufferSize) { … }

TEST_P(ClippingPredictorLevelBufferParametrization, CheckSizeAfterReset) { … }

INSTANTIATE_TEST_SUITE_P(…);

TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterFullBuffer) { … }

TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterPushBeyondCapacity) { … }

TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterTooFewItems) { … }

TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterReset) { … }

}  // namespace
}  // namespace webrtc