909118894b
Review URL: http://webrtc-codereview.appspot.com/44002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@111 4adac7df-926f-26a2-2b94-8c16560cd09d
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
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#include "audio_processing.h"
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#include "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 VoiceDetectionImpl : public VoiceDetection,
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public ProcessingComponent {
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public:
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explicit VoiceDetectionImpl(const AudioProcessingImpl* apm);
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virtual ~VoiceDetectionImpl();
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int ProcessCaptureAudio(AudioBuffer* audio);
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// VoiceDetection implementation.
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virtual bool is_enabled() const;
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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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private:
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// VoiceDetection implementation.
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virtual int Enable(bool enable);
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virtual int set_stream_has_voice(bool has_voice);
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virtual bool stream_has_voice() const;
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virtual int set_likelihood(Likelihood likelihood);
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virtual Likelihood likelihood() const;
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virtual int set_frame_size_ms(int size);
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virtual int frame_size_ms() const;
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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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const AudioProcessingImpl* apm_;
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bool stream_has_voice_;
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bool using_external_vad_;
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Likelihood likelihood_;
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int frame_size_ms_;
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int frame_size_samples_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
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