chromium/third_party/webrtc/modules/audio_processing/agc2/saturation_protector_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/saturation_protector_buffer.h"

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

namespace webrtc {
namespace {

Eq;
Optional;

TEST(GainController2SaturationProtectorBuffer, Init) { … }

TEST(GainController2SaturationProtectorBuffer, PushBack) { … }

TEST(GainController2SaturationProtectorBuffer, Reset) { … }

// Checks that the front value does not change until the ring buffer gets full.
TEST(GainController2SaturationProtectorBuffer, FrontUntilBufferIsFull) { … }

// Checks that when the buffer is full it behaves as a shift register.
TEST(GainController2SaturationProtectorBuffer, FrontIsDelayed) { … }

}  // namespace
}  // namespace webrtc