chromium/third_party/webrtc/modules/audio_processing/agc2/input_volume_stats_reporter_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/input_volume_stats_reporter.h"

#include "absl/strings/string_view.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"

namespace webrtc {
namespace {

InputVolumeType;

constexpr int kFramesIn60Seconds =;

constexpr absl::string_view kLabelPrefix =;

class InputVolumeStatsReporterTest
    : public ::testing::TestWithParam<InputVolumeType> {};

TEST_P(InputVolumeStatsReporterTest, CheckVolumeOnChangeIsEmpty) {}

TEST_P(InputVolumeStatsReporterTest, CheckRateAverageStatsEmpty) {}

TEST_P(InputVolumeStatsReporterTest, CheckSamples) {}
}  // namespace

TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {}

TEST_P(InputVolumeStatsReporterTest,
       CheckVolumeUpdateStatsAfterNoVolumeChange) {}

TEST_P(InputVolumeStatsReporterTest,
       CheckVolumeUpdateStatsAfterVolumeIncrease) {}

TEST_P(InputVolumeStatsReporterTest,
       CheckVolumeUpdateStatsAfterVolumeDecrease) {}

TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {}

INSTANTIATE_TEST_SUITE_P();

}  // namespace webrtc