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/*
* 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 "webrtc/modules/audio_processing/level_estimator_impl.h"
#include "webrtc/base/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/rms_level.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.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<const int16_t>(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
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