Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
29
webrtc/modules/audio_processing/level_estimator.cc
Normal file
29
webrtc/modules/audio_processing/level_estimator.cc
Normal file
@ -0,0 +1,29 @@
|
||||
/*
|
||||
* 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
|
Reference in New Issue
Block a user