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/*
* Copyright (c) 2019 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_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/rms_level.h"
namespace webrtc {
// An estimation component used to retrieve level metrics.
class LevelEstimator {
public:
LevelEstimator();
~LevelEstimator();
LevelEstimator(LevelEstimator&) = delete;
LevelEstimator& operator=(LevelEstimator&) = delete;
void ProcessStream(const AudioBuffer& audio);
// Returns the root mean square (RMS) level in dBFs (decibels from digital
// full-scale), or alternately dBov. It is computed over all primary stream
// frames since the last call to RMS(). The returned value is positive but
// should be interpreted as negative. It is constrained to [0, 127].
//
// The computation follows: https://tools.ietf.org/html/rfc6465
// with the intent that it can provide the RTP audio level indication.
//
// Frames passed to ProcessStream() with an |_energy| of zero are considered
// to have been muted. The RMS of the frame will be interpreted as -127.
int RMS() { return rms_.Average(); }
private:
RmsLevel rms_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_
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