Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/agc2/gain_applier.h

@@ -0,0 +1,44 @@
+/*
+ *  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_GAIN_APPLIER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+class GainApplier {
+ public:
+  GainApplier(bool hard_clip_samples, float initial_gain_factor);
+
+  void ApplyGain(AudioFrameView<float> signal);
+  void SetGainFactor(float gain_factor);
+  float GetGainFactor() const { return current_gain_factor_; }
+
+ private:
+  void Initialize(size_t samples_per_channel);
+
+  // Whether to clip samples after gain is applied. If 'true', result
+  // will fit in FloatS16 range.
+  const bool hard_clip_samples_;
+  float last_gain_factor_;
+
+  // If this value is not equal to 'last_gain_factor', gain will be
+  // ramped from 'last_gain_factor_' to this value during the next
+  // 'ApplyGain'.
+  float current_gain_factor_;
+  int samples_per_channel_ = -1;
+  float inverse_samples_per_channel_ = -1.f;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_