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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.
48 lines
1.4 KiB
C++
48 lines
1.4 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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#ifndef MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#include <memory>
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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namespace webrtc {
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class AudioBuffer;
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class RmsLevel;
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class LevelEstimatorImpl : public LevelEstimator {
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public:
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explicit LevelEstimatorImpl(rtc::CriticalSection* crit);
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~LevelEstimatorImpl() override;
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// TODO(peah): Fold into ctor, once public API is removed.
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void Initialize();
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void ProcessStream(AudioBuffer* audio);
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// LevelEstimator implementation.
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int Enable(bool enable) override;
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bool is_enabled() const override;
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int RMS() override;
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private:
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rtc::CriticalSection* const crit_ = nullptr;
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bool enabled_ RTC_GUARDED_BY(crit_) = false;
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std::unique_ptr<RmsLevel> rms_ RTC_GUARDED_BY(crit_);
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LevelEstimatorImpl);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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