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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
70 lines
1.7 KiB
C++
70 lines
1.7 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 "modules/audio_processing/level_estimator_impl.h"
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#include <stddef.h>
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#include <stdint.h>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/rms_level.h"
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#include "rtc_base/checks.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_->Analyze(rtc::ArrayView<const int16_t>(audio->channels_const()[i],
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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_->Average();
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}
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} // namespace webrtc
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