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https://github.com/danog/libtgvoip.git
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68 lines
1.9 KiB
C
68 lines
1.9 KiB
C
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/*
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* Copyright (c) 2016 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_AGC2_SIGNAL_CLASSIFIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/agc2/down_sampler.h"
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#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
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#include "modules/audio_processing/utility/ooura_fft.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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class AudioBuffer;
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class SignalClassifier {
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public:
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enum class SignalType { kNonStationary, kStationary };
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explicit SignalClassifier(ApmDataDumper* data_dumper);
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~SignalClassifier();
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void Initialize(int sample_rate_hz);
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SignalType Analyze(rtc::ArrayView<const float> signal);
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private:
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class FrameExtender {
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public:
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FrameExtender(size_t frame_size, size_t extended_frame_size);
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~FrameExtender();
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void ExtendFrame(rtc::ArrayView<const float> x,
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rtc::ArrayView<float> x_extended);
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private:
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std::vector<float> x_old_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
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};
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ApmDataDumper* const data_dumper_;
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DownSampler down_sampler_;
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std::unique_ptr<FrameExtender> frame_extender_;
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NoiseSpectrumEstimator noise_spectrum_estimator_;
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int sample_rate_hz_;
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int initialization_frames_left_;
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int consistent_classification_counter_;
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SignalType last_signal_type_;
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const OouraFft ooura_fft_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
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