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/*
* Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#include "webrtc/base/array_view.h"
#include "webrtc/base/optional.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Computes the root mean square (RMS) level in dBFs (decibels from digital
// full-scale) of audio data. The computation follows RFC 6465:
// https://tools.ietf.org/html/rfc6465
// with the intent that it can provide the RTP audio level indication.
//
// The expected approach is to provide constant-sized chunks of audio to
// Analyze(). When enough chunks have been accumulated to form a packet, call
// Average() to get the audio level indicator for the RTP header.
class RmsLevel {
public:
struct Levels {
int average;
int peak;
};
static constexpr int kMinLevelDb = 127;
RmsLevel();
~RmsLevel();
// Can be called to reset internal states, but is not required during normal
// operation.
void Reset();
// Pass each chunk of audio to Analyze() to accumulate the level.
void Analyze(rtc::ArrayView<const int16_t> data);
// If all samples with the given |length| have a magnitude of zero, this is
// a shortcut to avoid some computation.
void AnalyzeMuted(size_t length);
// Computes the RMS level over all data passed to Analyze() since the last
// call to Average(). The returned value is positive but should be interpreted
// as negative as per the RFC. It is constrained to [0, 127]. Resets the
// internal state to start a new measurement period.
int Average();
// Like Average() above, but also returns the RMS peak value. Resets the
// internal state to start a new measurement period.
Levels AverageAndPeak();
private:
// Compares |block_size| with |block_size_|. If they are different, calls
// Reset() and stores the new size.
void CheckBlockSize(size_t block_size);
float sum_square_;
size_t sample_count_;
float max_sum_square_;
rtc::Optional<size_t> block_size_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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