Finalization of the first version of EchoCanceller 3

webrtc/modules/audio_processing/aec3/main_filter_update_gain.h

Index: webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
diff --git a/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h b/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
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+/*
+ *  Copyright (c) 2017 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_AEC3_MAIN_FILTER_UPDATE_GAIN_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MAIN_FILTER_UPDATE_GAIN_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "webrtc/modules/audio_processing/aec3/subtractor_output.h"
+#include "webrtc/modules/audio_processing/aec3/fft_buffer.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Provides functionality for computing the adaptive gain for the main filter.
+class MainFilterUpdateGain {
+ public:
+  MainFilterUpdateGain();
+  ~MainFilterUpdateGain();
+
+  // Takes action in the case of a known echo path change.
+  void HandleEchoPathChange();
+
+  // Computes the gain.
+  void Compute(const FftBuffer& render_buffer,
+               const RenderSignalAnalyzer& render_signal_analyzer,
+               const SubtractorOutput& subtractor_output,
+               const AdaptiveFirFilter& filter,
+               bool saturated_capture_signal,
+               FftData* gain_fft);
+
+ private:
+  static int instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  std::array<float, kFftLengthBy2Plus1> H_error_;
+  size_t poor_excitation_counter_;
+  size_t call_counter_ = 0;
+  RTC_DISALLOW_COPY_AND_ASSIGN(MainFilterUpdateGain);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MAIN_FILTER_UPDATE_GAIN_H_