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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
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 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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