f254e9e9e5
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128. Reason for revert: internal test failures Original change's description: > Simplification and refactoring of the AudioBuffer code > > This CL performs a major refactoring and simplification > of the AudioBuffer code that. > -Removes 7 of the 9 internal buffers of the AudioBuffer. > -Avoids the implicit copying required to keep the > internal buffers in sync. > -Removes all code relating to handling of fixed-point > sample data in the AudioBuffer. > -Changes the naming of the class methods to reflect > that only floating point is handled. > -Corrects some bugs in the code. > -Extends the handling of internal downmixing to be > more generic. > > Bug: webrtc:10882 > Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828 > Commit-Queue: Per Åhgren <peah@webrtc.org> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28928} TBR=gustaf@webrtc.org,peah@webrtc.org Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10882 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28931}
71 lines
1.7 KiB
C++
71 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 float>(audio->channels_const_f()[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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