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/adaptive_digital_gain_applier.h

@@ -0,0 +1,69 @@
+/*
+ *  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_DIGITAL_GAIN_APPLIER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
+
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Part of the adaptive digital controller that applies a digital adaptive gain.
+// The gain is updated towards a target. The logic decides when gain updates are
+// allowed, it controls the adaptation speed and caps the target based on the
+// estimated noise level and the speech level estimate confidence.
+class AdaptiveDigitalGainApplier {
+ public:
+  // Information about a frame to process.
+  struct FrameInfo {
+    float input_level_dbfs;        // Estimated speech plus noise level.
+    float input_noise_level_dbfs;  // Estimated noise level.
+    VadLevelAnalyzer::Result vad_result;
+    float limiter_envelope_dbfs;  // Envelope level from the limiter.
+    bool estimate_is_confident;
+  };
+
+  // Ctor.
+  // `adjacent_speech_frames_threshold` indicates how many speech frames are
+  // required before a gain increase is allowed. `max_gain_change_db_per_second`
+  // limits the adaptation speed (uniformly operated across frames).
+  // `max_output_noise_level_dbfs` limits the output noise level.
+  AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper,
+                             int adjacent_speech_frames_threshold,
+                             float max_gain_change_db_per_second,
+                             float max_output_noise_level_dbfs);
+  AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
+  AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
+      delete;
+
+  // Analyzes `info`, updates the digital gain and applies it to a 10 ms
+  // `frame`. Supports any sample rate supported by APM.
+  void Process(const FrameInfo& info, AudioFrameView<float> frame);
+
+ private:
+  ApmDataDumper* const apm_data_dumper_;
+  GainApplier gain_applier_;
+
+  const int adjacent_speech_frames_threshold_;
+  const float max_gain_change_db_per_10ms_;
+  const float max_output_noise_level_dbfs_;
+
+  int calls_since_last_gain_log_;
+  int frames_to_gain_increase_allowed_;
+  float last_gain_db_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_