| /* |
| * Copyright (c) 2012 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_impl.h" |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include "api/array_view.h" |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/rms_level.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit) |
| : crit_(crit), rms_(new RmsLevel()) { |
| RTC_DCHECK(crit); |
| } |
| |
| LevelEstimatorImpl::~LevelEstimatorImpl() {} |
| |
| void LevelEstimatorImpl::Initialize() { |
| rtc::CritScope cs(crit_); |
| rms_->Reset(); |
| } |
| |
| void LevelEstimatorImpl::ProcessStream(const AudioBuffer& audio) { |
| rtc::CritScope cs(crit_); |
| if (!enabled_) { |
| return; |
| } |
| |
| for (size_t i = 0; i < audio.num_channels(); i++) { |
| rms_->Analyze(rtc::ArrayView<const float>(audio.channels_const()[i], |
| audio.num_frames())); |
| } |
| } |
| |
| int LevelEstimatorImpl::Enable(bool enable) { |
| rtc::CritScope cs(crit_); |
| if (enable && !enabled_) { |
| rms_->Reset(); |
| } |
| enabled_ = enable; |
| return AudioProcessing::kNoError; |
| } |
| |
| bool LevelEstimatorImpl::is_enabled() const { |
| rtc::CritScope cs(crit_); |
| return enabled_; |
| } |
| |
| int LevelEstimatorImpl::RMS() { |
| rtc::CritScope cs(crit_); |
| if (!enabled_) { |
| return AudioProcessing::kNotEnabledError; |
| } |
| |
| return rms_->Average(); |
| } |
| } // namespace webrtc |