mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
62 lines
2.1 KiB
C++
62 lines
2.1 KiB
C++
/*
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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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