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6787f232ae
The new configuration path is via AudioProcessing::ApplyConfig and AudioProcessing::GetStatistics. Bug: webrtc:9878 Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29454}
29 lines
903 B
C++
29 lines
903 B
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/level_estimator.h"
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#include "api/array_view.h"
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namespace webrtc {
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LevelEstimator::LevelEstimator() {
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rms_.Reset();
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}
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LevelEstimator::~LevelEstimator() = default;
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void LevelEstimator::ProcessStream(const AudioBuffer& audio) {
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for (size_t i = 0; i < audio.num_channels(); i++) {
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rms_.Analyze(rtc::ArrayView<const float>(audio.channels_const()[i],
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audio.num_frames()));
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}
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}
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} // namespace webrtc
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