Adding metrics to AEC3

webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc

Index: webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
new file mode 100644
index 0000000000000000000000000000000000000000..417c984cabba1bdc30194d7d437c51c62cd05688
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
@@ -0,0 +1,94 @@
+/*
+ *  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 "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
+
+#include <algorithm>
+
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+RenderDelayControllerMetrics::RenderDelayControllerMetrics() {
+  ResetMetrics();

[hlundin-webrtc, 2017/02/28 10:04:23]

No need.

[peah-webrtc, 2017/02/28 12:57:26]

Done.

+}
+
+void RenderDelayControllerMetrics::ResetMetrics() {

[hlundin-webrtc, 2017/02/28 10:04:23]

Wrong order of methods.

[peah-webrtc, 2017/02/28 12:57:26]

Done.

+  delay_change_counter_ = 0;
+  reliable_delay_estimate_counter_ = 0;
+}
+
+void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
+                                          size_t buffer_delay_blocks) {
+  ++call_counter_;
+
+  if (!initial_update) {
+    if (delay_samples) {
+      ++reliable_delay_estimate_counter_;
+      size_t delay_blocks = (*delay_samples) / kBlockSize;
+
+      if (delay_blocks != delay_blocks_) {
+        ++delay_change_counter_;
+        delay_blocks_ = delay_blocks;
+      }
+    }
+  } else if (++initial_call_counter_ == 5 * 250) {
+    initial_update = false;
+  }
+
+  if (call_counter_ == kMetricsReportingInterval) {
+    int value_to_report = static_cast<int>(delay_blocks_);
+    value_to_report = std::min(124, value_to_report);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
+                                value_to_report, 0, 124, 125);
+
+    value_to_report = static_cast<int>(buffer_delay_blocks);
+    value_to_report = std::min(124, value_to_report);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
+                                value_to_report, 0, 124, 125);
+
+    if (reliable_delay_estimate_counter_ == 0) {
+      value_to_report = 0;

[hlundin-webrtc, 2017/02/28 10:04:23]

Again, some kind of enum?

[peah-webrtc, 2017/02/28 12:57:26]

Done.

+    } else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
+      value_to_report = 4;
+    } else if (reliable_delay_estimate_counter_ > 100) {
+      value_to_report = 3;
+    } else if (reliable_delay_estimate_counter_ > 10) {
+      value_to_report = 2;
+    } else {
+      value_to_report = 1;
+    }
+    RTC_HISTOGRAM_COUNTS_LINEAR(

[hlundin-webrtc, 2017/02/28 10:04:23]

RTC_HISTOGRAM_ENUMERATION

[peah-webrtc, 2017/02/28 12:57:26]

Done.

+        "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates", value_to_report, 0,
+        4, 5);
+
+    if (delay_change_counter_ == 0) {
+      value_to_report = 0;
+    } else if (delay_change_counter_ > 10) {
+      value_to_report = 4;
+    } else if (delay_change_counter_ > 5) {
+      value_to_report = 3;
+    } else if (delay_change_counter_ > 2) {
+      value_to_report = 2;
+    } else {
+      value_to_report = 1;
+    }
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.DelayChanges",

[hlundin-webrtc, 2017/02/28 10:04:23]

RTC_HISTOGRAM_ENUMERATION

[peah-webrtc, 2017/02/28 12:57:26]

Done.

+                                value_to_report, 0, 4, 5);
+
+    metric_reported_ = true;
+    call_counter_ = 0;
+  } else {
+    metric_reported_ = false;
+  }
+}
+
+}  // namespace webrtc