| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_ |
| #define MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_ |
| |
| #include <memory> |
| #include <vector> |
| |
| #include "api/array_view.h" |
| #include "modules/audio_processing/agc2/down_sampler.h" |
| #include "modules/audio_processing/agc2/noise_spectrum_estimator.h" |
| #include "modules/audio_processing/utility/ooura_fft.h" |
| #include "rtc_base/constructor_magic.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| class AudioBuffer; |
| |
| class SignalClassifier { |
| public: |
| enum class SignalType { kNonStationary, kStationary }; |
| |
| explicit SignalClassifier(ApmDataDumper* data_dumper); |
| ~SignalClassifier(); |
| |
| void Initialize(int sample_rate_hz); |
| SignalType Analyze(rtc::ArrayView<const float> signal); |
| |
| private: |
| class FrameExtender { |
| public: |
| FrameExtender(size_t frame_size, size_t extended_frame_size); |
| ~FrameExtender(); |
| |
| void ExtendFrame(rtc::ArrayView<const float> x, |
| rtc::ArrayView<float> x_extended); |
| |
| private: |
| std::vector<float> x_old_; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender); |
| }; |
| |
| ApmDataDumper* const data_dumper_; |
| DownSampler down_sampler_; |
| std::unique_ptr<FrameExtender> frame_extender_; |
| NoiseSpectrumEstimator noise_spectrum_estimator_; |
| int sample_rate_hz_; |
| int initialization_frames_left_; |
| int consistent_classification_counter_; |
| SignalType last_signal_type_; |
| const OouraFft ooura_fft_; |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_ |