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Unified Diff: webrtc/modules/audio_processing/level_controller/signal_classifier.h

Issue 2090583002: New module for the adaptive level controlling functionality in the audio processing module (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Temporarily deactivated the level controller until the CL with the proper tuning has been landed Created 4 years, 6 months ago
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Index: webrtc/modules/audio_processing/level_controller/signal_classifier.h
diff --git a/webrtc/modules/audio_processing/level_controller/signal_classifier.h b/webrtc/modules/audio_processing/level_controller/signal_classifier.h
new file mode 100644
index 0000000000000000000000000000000000000000..cfa0fc56b5e9a084da0a17c21ac74b42f79e6b0c
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/signal_classifier.h
@@ -0,0 +1,65 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
+#include "webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class SignalClassifier {
+ public:
+ enum class SignalType { kHighlyNonStationary, kNonStationary, kStationary };
+
+ explicit SignalClassifier(ApmDataDumper* data_dumper);
+ ~SignalClassifier();
+
+ void Initialize(int sample_rate_hz);
+ void Analyze(const AudioBuffer& audio, SignalType* signal_type);
+
+ private:
+ class FrameExtender {
+ public:
+ FrameExtender(size_t frame_size, size_t extended_frame_size)
+ : x_old_(extended_frame_size - frame_size, 0.f) {}
+
+ 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_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_

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