blob: 12fa2125f08acdd3820971dddbcb370f3d25cdfa [file] [log] [blame]
* 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.
#include <cstddef>
#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:
// with the intent that it can provide the RTP audio level indication.
// The expected approach is to provide constant-sized chunks of audio to
// Process(). When enough chunks have been accumulated to form a packet, call
// RMS() to get the audio level indicator for the RTP header.
class RMSLevel {
static const int kMinLevel = 127;
// Can be called to reset internal states, but is not required during normal
// operation.
void Reset();
// Pass each chunk of audio to Process() to accumulate the level.
void Process(const int16_t* data, size_t length);
// If all samples with the given |length| have a magnitude of zero, this is
// a shortcut to avoid some computation.
void ProcessMuted(size_t length);
// Computes the RMS level over all data passed to Process() since the last
// call to RMS(). The returned value is positive but should be interpreted as
// negative as per the RFC. It is constrained to [0, 127].
int RMS();
float sum_square_;
size_t sample_count_;
} // namespace webrtc