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
This commit is contained in:
Arun Raghavan
2015-10-13 17:25:22 +05:30
parent 5ae7a5d6cd
commit 753eada3aa
324 changed files with 52533 additions and 16117 deletions

View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -8,56 +8,57 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
class AudioBuffer;
class CriticalSectionWrapper;
class VoiceDetectionImpl : public VoiceDetection,
public ProcessingComponent {
public:
explicit VoiceDetectionImpl(const AudioProcessingImpl* apm);
VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
virtual ~VoiceDetectionImpl();
int ProcessCaptureAudio(AudioBuffer* audio);
// VoiceDetection implementation.
virtual bool is_enabled() const;
bool is_enabled() const override;
// ProcessingComponent implementation.
virtual int Initialize();
virtual int get_version(char* version, int version_len_bytes) const;
int Initialize() override;
private:
// VoiceDetection implementation.
virtual int Enable(bool enable);
virtual int set_stream_has_voice(bool has_voice);
virtual bool stream_has_voice() const;
virtual int set_likelihood(Likelihood likelihood);
virtual Likelihood likelihood() const;
virtual int set_frame_size_ms(int size);
virtual int frame_size_ms() const;
int Enable(bool enable) override;
int set_stream_has_voice(bool has_voice) override;
bool stream_has_voice() const override;
int set_likelihood(Likelihood likelihood) override;
Likelihood likelihood() const override;
int set_frame_size_ms(int size) override;
int frame_size_ms() const override;
// ProcessingComponent implementation.
virtual void* CreateHandle() const;
virtual int InitializeHandle(void* handle) const;
virtual int ConfigureHandle(void* handle) const;
virtual int DestroyHandle(void* handle) const;
virtual int num_handles_required() const;
virtual int GetHandleError(void* handle) const;
void* CreateHandle() const override;
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
int num_handles_required() const override;
int GetHandleError(void* handle) const override;
const AudioProcessingImpl* apm_;
const AudioProcessing* apm_;
CriticalSectionWrapper* crit_;
bool stream_has_voice_;
bool using_external_vad_;
Likelihood likelihood_;
int frame_size_ms_;
int frame_size_samples_;
size_t frame_size_samples_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_