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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@ -1,5 +1,5 @@
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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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@ -8,22 +8,24 @@
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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_GAIN_CONTROL_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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#include <vector>
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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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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 GainControlImpl : public GainControl,
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public ProcessingComponent {
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public:
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explicit GainControlImpl(const AudioProcessingImpl* apm);
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GainControlImpl(const AudioProcessing* apm,
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CriticalSectionWrapper* crit);
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virtual ~GainControlImpl();
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int ProcessRenderAudio(AudioBuffer* audio);
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@ -31,39 +33,39 @@ class GainControlImpl : public GainControl,
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int ProcessCaptureAudio(AudioBuffer* audio);
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// ProcessingComponent implementation.
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virtual int Initialize();
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virtual int get_version(char* version, int version_len_bytes) const;
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int Initialize() override;
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// GainControl implementation.
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virtual bool is_enabled() const;
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virtual int stream_analog_level();
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bool is_enabled() const override;
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int stream_analog_level() override;
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bool is_limiter_enabled() const override;
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Mode mode() const override;
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private:
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// GainControl implementation.
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virtual int Enable(bool enable);
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virtual int set_stream_analog_level(int level);
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virtual int set_mode(Mode mode);
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virtual Mode mode() const;
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virtual int set_target_level_dbfs(int level);
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virtual int target_level_dbfs() const;
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virtual int set_compression_gain_db(int gain);
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virtual int compression_gain_db() const;
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virtual int enable_limiter(bool enable);
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virtual bool is_limiter_enabled() const;
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virtual int set_analog_level_limits(int minimum, int maximum);
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virtual int analog_level_minimum() const;
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virtual int analog_level_maximum() const;
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virtual bool stream_is_saturated() const;
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int Enable(bool enable) override;
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int set_stream_analog_level(int level) override;
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int set_mode(Mode mode) override;
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int set_target_level_dbfs(int level) override;
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int target_level_dbfs() const override;
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int set_compression_gain_db(int gain) override;
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int compression_gain_db() const override;
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int enable_limiter(bool enable) override;
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int set_analog_level_limits(int minimum, int maximum) override;
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int analog_level_minimum() const override;
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int analog_level_maximum() const override;
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bool stream_is_saturated() const 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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const AudioProcessing* apm_;
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CriticalSectionWrapper* crit_;
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Mode mode_;
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int minimum_capture_level_;
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int maximum_capture_level_;
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@ -77,4 +79,4 @@ class GainControlImpl : public GainControl,
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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