Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.

webrtc/modules/audio_processing/agc2/speech_level_estimator.h

@@ -0,0 +1,81 @@
+/*
+ *  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_SPEECH_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <type_traits>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+// Active speech level estimator based on the analysis of the following
+// framewise properties: RMS level (dBFS), peak level (dBFS), speech
+// probability.
+class SpeechLevelEstimator {
+ public:
+  SpeechLevelEstimator(
+      ApmDataDumper* apm_data_dumper,
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int adjacent_speech_frames_threshold);
+  SpeechLevelEstimator(const SpeechLevelEstimator&) = delete;
+  SpeechLevelEstimator& operator=(const SpeechLevelEstimator&) = delete;
+
+  // Updates the level estimation.
+  void Update(float rms_dbfs, float peak_dbfs, float speech_probability);
+  // 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 is_confident() const { return is_confident_; }
+
+  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);
+    }
+    // TODO(bugs.webrtc.org/7494): Remove `time_to_confidence_ms` if redundant.
+    int time_to_confidence_ms;
+    struct Ratio {
+      float numerator;
+      float denominator;
+      float GetRatio() const;
+    } level_dbfs;
+  };
+  static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");
+
+  void UpdateIsConfident();
+
+  void ResetLevelEstimatorState(LevelEstimatorState& state) const;
+
+  void DumpDebugData() const;
+
+  ApmDataDumper* const apm_data_dumper_;
+
+  const float initial_speech_level_dbfs_;
+  const int adjacent_speech_frames_threshold_;
+  LevelEstimatorState preliminary_state_;
+  LevelEstimatorState reliable_state_;
+  float level_dbfs_;
+  bool is_confident_;
+  int num_adjacent_speech_frames_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_