6955870806
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
66 lines
1.6 KiB
C++
66 lines
1.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/level_estimator_impl.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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namespace webrtc {
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LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
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: crit_(crit), rms_(new RMSLevel()) {
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RTC_DCHECK(crit);
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}
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LevelEstimatorImpl::~LevelEstimatorImpl() {}
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void LevelEstimatorImpl::Initialize() {
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rtc::CritScope cs(crit_);
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rms_->Reset();
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}
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void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
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RTC_DCHECK(audio);
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rtc::CritScope cs(crit_);
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if (!enabled_) {
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return;
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}
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for (size_t i = 0; i < audio->num_channels(); i++) {
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rms_->Process(audio->channels_const()[i], audio->num_frames());
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}
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}
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int LevelEstimatorImpl::Enable(bool enable) {
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rtc::CritScope cs(crit_);
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if (enable && !enabled_) {
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rms_->Reset();
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}
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enabled_ = enable;
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return AudioProcessing::kNoError;
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}
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bool LevelEstimatorImpl::is_enabled() const {
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rtc::CritScope cs(crit_);
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return enabled_;
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}
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int LevelEstimatorImpl::RMS() {
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rtc::CritScope cs(crit_);
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if (!enabled_) {
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return AudioProcessing::kNotEnabledError;
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}
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return rms_->RMS();
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}
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} // namespace webrtc
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