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/noise_level_estimator.cc

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_processing/agc2/noise_level_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "api/array_view.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kFramesPerSecond = 100;
+
+float FrameEnergy(const AudioFrameView<const float>& audio) {
+  float energy = 0.f;
+  for (size_t k = 0; k < audio.num_channels(); ++k) {
+    float channel_energy =
+        std::accumulate(audio.channel(k).begin(), audio.channel(k).end(), 0.f,
+                        [](float a, float b) -> float { return a + b * b; });
+    energy = std::max(channel_energy, energy);
+  }
+  return energy;
+}
+
+float EnergyToDbfs(float signal_energy, size_t num_samples) {
+  const float rms = std::sqrt(signal_energy / num_samples);
+  return FloatS16ToDbfs(rms);
+}
+}  // namespace
+
+NoiseLevelEstimator::NoiseLevelEstimator(ApmDataDumper* data_dumper)
+    : signal_classifier_(data_dumper) {
+  Initialize(48000);
+}
+
+NoiseLevelEstimator::~NoiseLevelEstimator() {}
+
+void NoiseLevelEstimator::Initialize(int sample_rate_hz) {
+  sample_rate_hz_ = sample_rate_hz;
+  noise_energy_ = 1.f;
+  first_update_ = true;
+  min_noise_energy_ = sample_rate_hz * 2.f * 2.f / kFramesPerSecond;
+  noise_energy_hold_counter_ = 0;
+  signal_classifier_.Initialize(sample_rate_hz);
+}
+
+float NoiseLevelEstimator::Analyze(const AudioFrameView<const float>& frame) {
+  const int rate =
+      static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
+  if (rate != sample_rate_hz_) {
+    Initialize(rate);
+  }
+  const float frame_energy = FrameEnergy(frame);
+  if (frame_energy <= 0.f) {
+    RTC_DCHECK_GE(frame_energy, 0.f);
+    return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
+  }
+
+  if (first_update_) {
+    // Initialize the noise energy to the frame energy.
+    first_update_ = false;
+    return EnergyToDbfs(
+        noise_energy_ = std::max(frame_energy, min_noise_energy_),
+        frame.samples_per_channel());
+  }
+
+  const SignalClassifier::SignalType signal_type =
+      signal_classifier_.Analyze(frame.channel(0));
+
+  // Update the noise estimate in a minimum statistics-type manner.
+  if (signal_type == SignalClassifier::SignalType::kStationary) {
+    if (frame_energy > noise_energy_) {
+      // Leak the estimate upwards towards the frame energy if no recent
+      // downward update.
+      noise_energy_hold_counter_ = std::max(noise_energy_hold_counter_ - 1, 0);
+
+      if (noise_energy_hold_counter_ == 0) {
+        noise_energy_ = std::min(noise_energy_ * 1.01f, frame_energy);
+      }
+    } else {
+      // Update smoothly downwards with a limited maximum update magnitude.
+      noise_energy_ =
+          std::max(noise_energy_ * 0.9f,
+                   noise_energy_ + 0.05f * (frame_energy - noise_energy_));
+      noise_energy_hold_counter_ = 1000;
+    }
+  } else {
+    // For a non-stationary signal, leak the estimate downwards in order to
+    // avoid estimate locking due to incorrect signal classification.
+    noise_energy_ = noise_energy_ * 0.99f;
+  }
+
+  // Ensure a minimum of the estimate.
+  return EnergyToDbfs(
+      noise_energy_ = std::max(noise_energy_, min_noise_energy_),
+      frame.samples_per_channel());
+}
+
+}  // namespace webrtc