Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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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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@ -8,46 +8,46 @@
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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 WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#include "audio_processing.h"
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#include "processing_component.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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namespace webrtc {
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class AudioProcessingImpl;
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class AudioBuffer;
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class CriticalSectionWrapper;
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class LevelEstimatorImpl : public LevelEstimator,
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public ProcessingComponent {
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public:
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explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
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LevelEstimatorImpl(const AudioProcessing* apm,
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CriticalSectionWrapper* crit);
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virtual ~LevelEstimatorImpl();
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int AnalyzeReverseStream(AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio);
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int ProcessStream(AudioBuffer* audio);
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// LevelEstimator implementation.
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virtual bool is_enabled() const;
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// ProcessingComponent implementation.
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virtual int get_version(char* version, int version_len_bytes) const;
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bool is_enabled() const override;
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private:
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// LevelEstimator implementation.
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virtual int Enable(bool enable);
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virtual int GetMetrics(Metrics* metrics, Metrics* reverse_metrics);
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int Enable(bool enable) override;
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int RMS() override;
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// ProcessingComponent implementation.
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virtual void* CreateHandle() const;
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virtual int InitializeHandle(void* handle) const;
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virtual int ConfigureHandle(void* handle) const;
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virtual int DestroyHandle(void* handle) const;
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virtual int num_handles_required() const;
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virtual int GetHandleError(void* handle) const;
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void* CreateHandle() const override;
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int InitializeHandle(void* handle) const override;
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int ConfigureHandle(void* handle) const override;
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void DestroyHandle(void* handle) const override;
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int num_handles_required() const override;
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int GetHandleError(void* handle) const override;
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const AudioProcessingImpl* apm_;
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CriticalSectionWrapper* crit_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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