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/aec3/api_call_jitter_metrics.cc

@@ -0,0 +1,121 @@
+/*
+ *  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.
+ */
+
+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+bool TimeToReportMetrics(int frames_since_last_report) {
+  constexpr int kNumFramesPerSecond = 100;
+  constexpr int kReportingIntervalFrames = 10 * kNumFramesPerSecond;
+  return frames_since_last_report == kReportingIntervalFrames;
+}
+
+}  // namespace
+
+ApiCallJitterMetrics::Jitter::Jitter()
+    : max_(0), min_(std::numeric_limits<int>::max()) {}
+
+void ApiCallJitterMetrics::Jitter::Update(int num_api_calls_in_a_row) {
+  min_ = std::min(min_, num_api_calls_in_a_row);
+  max_ = std::max(max_, num_api_calls_in_a_row);
+}
+
+void ApiCallJitterMetrics::Jitter::Reset() {
+  min_ = std::numeric_limits<int>::max();
+  max_ = 0;
+}
+
+void ApiCallJitterMetrics::Reset() {
+  render_jitter_.Reset();
+  capture_jitter_.Reset();
+  num_api_calls_in_a_row_ = 0;
+  frames_since_last_report_ = 0;
+  last_call_was_render_ = false;
+  proper_call_observed_ = false;
+}
+
+void ApiCallJitterMetrics::ReportRenderCall() {
+  if (!last_call_was_render_) {
+    // If the previous call was a capture and a proper call has been observed
+    // (containing both render and capture data), storing the last number of
+    // capture calls into the metrics.
+    if (proper_call_observed_) {
+      capture_jitter_.Update(num_api_calls_in_a_row_);
+    }
+
+    // Reset the call counter to start counting render calls.
+    num_api_calls_in_a_row_ = 0;
+  }
+  ++num_api_calls_in_a_row_;
+  last_call_was_render_ = true;
+}
+
+void ApiCallJitterMetrics::ReportCaptureCall() {
+  if (last_call_was_render_) {
+    // If the previous call was a render and a proper call has been observed
+    // (containing both render and capture data), storing the last number of
+    // render calls into the metrics.
+    if (proper_call_observed_) {
+      render_jitter_.Update(num_api_calls_in_a_row_);
+    }
+    // Reset the call counter to start counting capture calls.
+    num_api_calls_in_a_row_ = 0;
+
+    // If this statement is reached, at least one render and one capture call
+    // have been observed.
+    proper_call_observed_ = true;
+  }
+  ++num_api_calls_in_a_row_;
+  last_call_was_render_ = false;
+
+  // Only report and update jitter metrics for when a proper call, containing
+  // both render and capture data, has been observed.
+  if (proper_call_observed_ &&
+      TimeToReportMetrics(++frames_since_last_report_)) {
+    // Report jitter, where the base basic unit is frames.
+    constexpr int kMaxJitterToReport = 50;
+
+    // Report max and min jitter for render and capture, in units of 20 ms.
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MaxRenderJitter",
+        std::min(kMaxJitterToReport, render_jitter().max()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MinRenderJitter",
+        std::min(kMaxJitterToReport, render_jitter().min()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MaxCaptureJitter",
+        std::min(kMaxJitterToReport, capture_jitter().max()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MinCaptureJitter",
+        std::min(kMaxJitterToReport, capture_jitter().min()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+
+    frames_since_last_report_ = 0;
+    Reset();
+  }
+}
+
+bool ApiCallJitterMetrics::WillReportMetricsAtNextCapture() const {
+  return TimeToReportMetrics(frames_since_last_report_ + 1);
+}
+
+}  // namespace webrtc