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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.
*/
#include "modules/audio_processing/level_estimator.h"
#include "api/array_view.h"
namespace webrtc {
LevelEstimator::LevelEstimator() {
rms_.Reset();
}
LevelEstimator::~LevelEstimator() = default;
void LevelEstimator::ProcessStream(const AudioBuffer& audio) {
for (size_t i = 0; i < audio.num_channels(); i++) {
rms_.Analyze(rtc::ArrayView<const float>(audio.channels_const()[i],
audio.num_frames()));
}
}
} // namespace webrtc
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