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_probability_buffer.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2022 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_PROBABILITY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_
+
+#include <vector>
+
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+
+// This class implements a circular buffer that stores speech probabilities
+// for a speech segment and estimates speech activity for that segment.
+class SpeechProbabilityBuffer {
+ public:
+  // Ctor. The value of `low_probability_threshold` is required to be on the
+  // range [0.0f, 1.0f].
+  explicit SpeechProbabilityBuffer(float low_probability_threshold);
+  ~SpeechProbabilityBuffer() {}
+  SpeechProbabilityBuffer(const SpeechProbabilityBuffer&) = delete;
+  SpeechProbabilityBuffer& operator=(const SpeechProbabilityBuffer&) = delete;
+
+  // Adds `probability` in the buffer and computes an updatds sum of the buffer
+  // probabilities. Value of `probability` is required to be on the range
+  // [0.0f, 1.0f].
+  void Update(float probability);
+
+  // Resets the histogram, forgets the past.
+  void Reset();
+
+  // Returns true if the segment is active (a long enough segment with an
+  // average speech probability above `low_probability_threshold`).
+  bool IsActiveSegment() const;
+
+ private:
+  void RemoveTransient();
+
+  // Use only for testing.
+  float GetSumProbabilities() const { return sum_probabilities_; }
+
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterInitialization);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest, CheckSumAfterUpdate);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest, CheckSumAfterReset);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterTransientNotRemoved);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterTransientRemoved);
+
+  const float low_probability_threshold_;
+
+  // Sum of probabilities stored in `probabilities_`. Must be updated if
+  // `probabilities_` is updated.
+  float sum_probabilities_ = 0.0f;
+
+  // Circular buffer for probabilities.
+  std::vector<float> probabilities_;
+
+  // Current index of the circular buffer, where the newest data will be written
+  // to, therefore, pointing to the oldest data if buffer is full.
+  int buffer_index_ = 0;
+
+  // Indicates if the buffer is full and adding a new value removes the oldest
+  // value.
+  int buffer_is_full_ = false;
+
+  int num_high_probability_observations_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_