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

/*
 *  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.
 */

#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_

#include "absl/types/optional.h"
#include "rtc_base/gtest_prod_util.h"
#include "system_wrappers/include/metrics.h"

namespace webrtc {

// Input volume statistics calculator. Computes aggregate stats based on the
// framewise input volume observed by `UpdateStatistics()`. Periodically logs
// the statistics into a histogram.
class InputVolumeStatsReporter { … };

// Updates the histogram that keeps track of recommended input volume changes
// required in order to match the target level in the input volume adaptation
// process.
void UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(int volume);

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_