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/transient/voice_probability_delay_unit.h

@@ -0,0 +1,43 @@
+/*
+ *  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_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+
+#include <array>
+
+namespace webrtc {
+
+// Iteratively produces a sequence of delayed voice probability values given a
+// fixed delay between 0 and 20 ms and given a sequence of voice probability
+// values observed every 10 ms. Supports fractional delays, that are delays
+// which are not a multiple integer of 10 ms. Applies interpolation with
+// fractional delays; otherwise, returns a previously observed value according
+// to the given fixed delay.
+class VoiceProbabilityDelayUnit {
+ public:
+  // Ctor. `delay_num_samples` is the delay in number of samples and it must be
+  // non-negative and less than 20 ms.
+  VoiceProbabilityDelayUnit(int delay_num_samples, int sample_rate_hz);
+
+  // Handles delay and sample rate changes and resets the delay unit.
+  void Initialize(int delay_num_samples, int sample_rate_hz);
+
+  // Observes `voice_probability` and returns a delayed voice probability.
+  float Delay(float voice_probability);
+
+ private:
+  std::array<float, 3> weights_;
+  std::array<float, 2> last_probabilities_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_