saza | 6787f23 | 2019-10-11 19:31:07 +0200 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include "modules/audio_processing/level_estimator.h" |
| 12 | |
| 13 | #include "api/array_view.h" |
| 14 | |
| 15 | namespace webrtc { |
| 16 | |
| 17 | LevelEstimator::LevelEstimator() { |
| 18 | rms_.Reset(); |
| 19 | } |
| 20 | |
| 21 | LevelEstimator::~LevelEstimator() = default; |
| 22 | |
| 23 | void LevelEstimator::ProcessStream(const AudioBuffer& audio) { |
| 24 | for (size_t i = 0; i < audio.num_channels(); i++) { |
| 25 | rms_.Analyze(rtc::ArrayView<const float>(audio.channels_const()[i], |
| 26 | audio.num_frames())); |
| 27 | } |
| 28 | } |
| 29 | } // namespace webrtc |