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/level_estimator.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2019 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_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/rms_level.h"
+
+namespace webrtc {
+
+// An estimation component used to retrieve level metrics.
+class LevelEstimator {
+ public:
+  LevelEstimator();
+  ~LevelEstimator();
+
+  LevelEstimator(LevelEstimator&) = delete;
+  LevelEstimator& operator=(LevelEstimator&) = delete;
+
+  void ProcessStream(const AudioBuffer& audio);
+
+  // Returns the root mean square (RMS) level in dBFs (decibels from digital
+  // full-scale), or alternately dBov. It is computed over all primary stream
+  // frames since the last call to RMS(). The returned value is positive but
+  // should be interpreted as negative. It is constrained to [0, 127].
+  //
+  // The computation follows: https://tools.ietf.org/html/rfc6465
+  // with the intent that it can provide the RTP audio level indication.
+  //
+  // Frames passed to ProcessStream() with an |_energy| of zero are considered
+  // to have been muted. The RMS of the frame will be interpreted as -127.
+  int RMS() { return rms_.Average(); }
+
+ private:
+  RmsLevel rms_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_