/* * 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 #include #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(AudioBuffer* audio) { RTC_DCHECK(audio); rtc::CritScope cs(crit_); if (!enabled_) { return; } for (size_t i = 0; i < audio->num_channels(); i++) { rms_->Analyze(rtc::ArrayView(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