From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../aec3/render_delay_controller_metrics.cc        | 132 +++++++++++++++++++++
 1 file changed, 132 insertions(+)
 create mode 100644 third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc

(limited to 'third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc')

diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
new file mode 100644
index 0000000000..1e0a0f443e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2017 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/render_delay_controller_metrics.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class DelayReliabilityCategory {
+  kNone,
+  kPoor,
+  kMedium,
+  kGood,
+  kExcellent,
+  kNumCategories
+};
+enum class DelayChangesCategory {
+  kNone,
+  kFew,
+  kSeveral,
+  kMany,
+  kConstant,
+  kNumCategories
+};
+
+}  // namespace
+
+RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
+
+void RenderDelayControllerMetrics::Update(
+    absl::optional<size_t> delay_samples,
+    absl::optional<size_t> buffer_delay_blocks,
+    ClockdriftDetector::Level clockdrift) {
+  ++call_counter_;
+
+  if (!initial_update) {
+    size_t delay_blocks;
+    if (delay_samples) {
+      ++reliable_delay_estimate_counter_;
+      // Add an offset by 1 (metric is halved before reporting) to reserve 0 for
+      // absent delay.
+      delay_blocks = (*delay_samples) / kBlockSize + 2;
+    } else {
+      delay_blocks = 0;
+    }
+
+    if (delay_blocks != delay_blocks_) {
+      ++delay_change_counter_;
+      delay_blocks_ = delay_blocks;
+    }
+
+  } else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
+    initial_update = false;
+  }
+
+  if (call_counter_ == kMetricsReportingIntervalBlocks) {
+    int value_to_report = static_cast<int>(delay_blocks_);
+    // Divide by 2 to compress metric range.
+    value_to_report = std::min(124, value_to_report >> 1);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
+                                value_to_report, 0, 124, 125);
+
+    // Divide by 2 to compress metric range.
+    // Offset by 1 to reserve 0 for absent delay.
+    value_to_report = buffer_delay_blocks ? (*buffer_delay_blocks + 2) >> 1 : 0;
+    value_to_report = std::min(124, value_to_report);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
+                                value_to_report, 0, 124, 125);
+
+    DelayReliabilityCategory delay_reliability;
+    if (reliable_delay_estimate_counter_ == 0) {
+      delay_reliability = DelayReliabilityCategory::kNone;
+    } else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
+      delay_reliability = DelayReliabilityCategory::kExcellent;
+    } else if (reliable_delay_estimate_counter_ > 100) {
+      delay_reliability = DelayReliabilityCategory::kGood;
+    } else if (reliable_delay_estimate_counter_ > 10) {
+      delay_reliability = DelayReliabilityCategory::kMedium;
+    } else {
+      delay_reliability = DelayReliabilityCategory::kPoor;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates",
+        static_cast<int>(delay_reliability),
+        static_cast<int>(DelayReliabilityCategory::kNumCategories));
+
+    DelayChangesCategory delay_changes;
+    if (delay_change_counter_ == 0) {
+      delay_changes = DelayChangesCategory::kNone;
+    } else if (delay_change_counter_ > 10) {
+      delay_changes = DelayChangesCategory::kConstant;
+    } else if (delay_change_counter_ > 5) {
+      delay_changes = DelayChangesCategory::kMany;
+    } else if (delay_change_counter_ > 2) {
+      delay_changes = DelayChangesCategory::kSeveral;
+    } else {
+      delay_changes = DelayChangesCategory::kFew;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.DelayChanges",
+        static_cast<int>(delay_changes),
+        static_cast<int>(DelayChangesCategory::kNumCategories));
+
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
+        static_cast<int>(ClockdriftDetector::Level::kNumCategories));
+
+    call_counter_ = 0;
+    ResetMetrics();
+  }
+}
+
+void RenderDelayControllerMetrics::ResetMetrics() {
+  delay_change_counter_ = 0;
+  reliable_delay_estimate_counter_ = 0;
+}
+
+}  // namespace webrtc
-- 
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