chromium/third_party/webrtc/modules/audio_processing/agc2/input_volume_controller.h

/*
 *  Copyright (c) 2013 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.
 */

#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_

#include <memory>
#include <vector>

#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc2/clipping_predictor.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/gtest_prod_util.h"

namespace webrtc {

class MonoInputVolumeController;

// The input volume controller recommends what volume to use, handles volume
// changes and clipping detection and prediction. In particular, it handles
// changes triggered by the user (e.g., volume set to zero by a HW mute button).
// This class is not thread-safe.
// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class InputVolumeController final {};

// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class MonoInputVolumeController {};

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_