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.
68 lines
1.9 KiB
C++
68 lines
1.9 KiB
C++
/*
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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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