mirror of
https://github.com/danog/libtgvoip.git
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62 lines
2.1 KiB
C
62 lines
2.1 KiB
C
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/*
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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_VOICE_DETECTION_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
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#include <stddef.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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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class AudioBuffer;
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class VoiceDetectionImpl : public VoiceDetection {
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public:
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explicit VoiceDetectionImpl(rtc::CriticalSection* crit);
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~VoiceDetectionImpl() override;
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// TODO(peah): Fold into ctor, once public API is removed.
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void Initialize(int sample_rate_hz);
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void ProcessCaptureAudio(AudioBuffer* audio);
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// VoiceDetection implementation.
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int Enable(bool enable) override;
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bool is_enabled() const override;
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int set_stream_has_voice(bool has_voice) override;
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bool stream_has_voice() const override;
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int set_likelihood(Likelihood likelihood) override;
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Likelihood likelihood() const override;
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int set_frame_size_ms(int size) override;
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int frame_size_ms() const override;
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private:
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class Vad;
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rtc::CriticalSection* const crit_;
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bool enabled_ RTC_GUARDED_BY(crit_) = false;
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bool stream_has_voice_ RTC_GUARDED_BY(crit_) = false;
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bool using_external_vad_ RTC_GUARDED_BY(crit_) = false;
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Likelihood likelihood_ RTC_GUARDED_BY(crit_) = kLowLikelihood;
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int frame_size_ms_ RTC_GUARDED_BY(crit_) = 10;
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size_t frame_size_samples_ RTC_GUARDED_BY(crit_) = 0;
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int sample_rate_hz_ RTC_GUARDED_BY(crit_) = 0;
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std::unique_ptr<Vad> vad_ RTC_GUARDED_BY(crit_);
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VoiceDetectionImpl);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
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