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+/*
+ * Copyright (c) 2018 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_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
+
+#include <stddef.h>
+#include <type_traits>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/saturation_protector.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+// Level estimator for the digital adaptive gain controller.
+class AdaptiveModeLevelEstimator {
+ public:
+ explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
+ AdaptiveModeLevelEstimator(const AdaptiveModeLevelEstimator&) = delete;
+ AdaptiveModeLevelEstimator& operator=(const AdaptiveModeLevelEstimator&) =
+ delete;
+ AdaptiveModeLevelEstimator(
+ ApmDataDumper* apm_data_dumper,
+ AudioProcessing::Config::GainController2::LevelEstimator level_estimator,
+ int adjacent_speech_frames_threshold,
+ float initial_saturation_margin_db,
+ float extra_saturation_margin_db);
+
+ // Updates the level estimation.
+ void Update(const VadLevelAnalyzer::Result& vad_data);
+ // Returns the estimated speech plus noise level.
+ float level_dbfs() const { return level_dbfs_; }
+ // Returns true if the estimator is confident on its current estimate.
+ bool IsConfident() const;
+
+ void Reset();
+
+ private:
+ // Part of the level estimator state used for check-pointing and restore ops.
+ struct LevelEstimatorState {
+ bool operator==(const LevelEstimatorState& s) const;
+ inline bool operator!=(const LevelEstimatorState& s) const {
+ return !(*this == s);
+ }
+ struct Ratio {
+ float numerator;
+ float denominator;
+ float GetRatio() const;
+ };
+ // TODO(crbug.com/webrtc/7494): Remove time_to_full_buffer_ms if redundant.
+ int time_to_full_buffer_ms;
+ Ratio level_dbfs;
+ SaturationProtectorState saturation_protector;
+ };
+ static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");
+
+ void ResetLevelEstimatorState(LevelEstimatorState& state) const;
+
+ void DumpDebugData() const;
+
+ ApmDataDumper* const apm_data_dumper_;
+
+ const AudioProcessing::Config::GainController2::LevelEstimator
+ level_estimator_type_;
+ const int adjacent_speech_frames_threshold_;
+ const float initial_saturation_margin_db_;
+ const float extra_saturation_margin_db_;
+ LevelEstimatorState preliminary_state_;
+ LevelEstimatorState reliable_state_;
+ float level_dbfs_;
+ int num_adjacent_speech_frames_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_