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/agc2_testing_common.h

@@ -0,0 +1,78 @@
+/*
+ *  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_AGC2_TESTING_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
+
+#include <math.h>
+
+#include <limits>
+#include <vector>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace test {
+
+// Level Estimator test parameters.
+constexpr float kDecayMs = 500.f;
+
+// Limiter parameters.
+constexpr float kLimiterMaxInputLevelDbFs = 1.f;
+constexpr float kLimiterKneeSmoothnessDb = 1.f;
+constexpr float kLimiterCompressionRatio = 5.f;
+constexpr float kPi = 3.1415926536f;
+
+std::vector<double> LinSpace(const double l, const double r, size_t num_points);
+
+class SineGenerator {
+ public:
+  SineGenerator(float frequency, int rate)
+      : frequency_(frequency), rate_(rate) {}
+  float operator()() {
+    x_radians_ += frequency_ / rate_ * 2 * kPi;
+    if (x_radians_ > 2 * kPi) {
+      x_radians_ -= 2 * kPi;
+    }
+    return 1000.f * sinf(x_radians_);
+  }
+
+ private:
+  float frequency_;
+  int rate_;
+  float x_radians_ = 0.f;
+};
+
+class PulseGenerator {
+ public:
+  PulseGenerator(float frequency, int rate)
+      : samples_period_(
+            static_cast<int>(static_cast<float>(rate) / frequency)) {
+    RTC_DCHECK_GT(rate, frequency);
+  }
+  float operator()() {
+    sample_counter_++;
+    if (sample_counter_ >= samples_period_) {
+      sample_counter_ -= samples_period_;
+    }
+    return static_cast<float>(
+        sample_counter_ == 0 ? std::numeric_limits<int16_t>::max() : 10.f);
+  }
+
+ private:
+  int samples_period_;
+  int sample_counter_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_