Adding metrics to AEC3

webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc

+/*
+ *  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/echo_remover_metrics.h"
+
+#include <math.h>
+#include <algorithm>
+#include <numeric>
+
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kOneByMetricsCollectionBlocks = 1.f / kMetricsCollectionBlocks;
+
+}  // namespace
+
+EchoRemoverMetrics::DbMetric::DbMetric() : DbMetric(0.f, 0.f, 0.f) {}
+EchoRemoverMetrics::DbMetric::DbMetric(float sum_value,
+                                       float floor_value,
+                                       float ceil_value)
+    : sum_value(sum_value), floor_value(floor_value), ceil_value(ceil_value) {}
+
+void EchoRemoverMetrics::DbMetric::Update(float value) {
+  sum_value += value;
+  floor_value = std::min(floor_value, value);
+  ceil_value = std::max(ceil_value, value);
+}
+
+void EchoRemoverMetrics::DbMetric::FormatForReporting(bool negate,
+                                                      float min_value,
+                                                      float max_value,
+                                                      float offset,
+                                                      float scaling) {
+  aec3::TransformDbMetricForReporting(negate, min_value, max_value, offset,
+                                      scaling * kOneByMetricsCollectionBlocks,
+                                      &sum_value);
+  aec3::TransformDbMetricForReporting(negate, min_value, max_value, offset,
+                                      scaling, &ceil_value);
+  aec3::TransformDbMetricForReporting(negate, min_value, max_value, offset,
+                                      scaling, &floor_value);
+}
+
+EchoRemoverMetrics::EchoRemoverMetrics() {
+  ResetMetrics();
+}
+
+void EchoRemoverMetrics::ResetMetrics() {
+  erl_.fill(DbMetric(0.f, 10000.f, 0.000f));
+  erle_.fill(DbMetric(0.f, 0.f, 1000.f));
+  comfort_noise_.fill(DbMetric(0.f, 100000000.f, 0.f));
+  suppressor_gain_.fill(DbMetric(0.f, 1.f, 0.f));
+  active_render_ = 0;
+  saturated_capture_ = false;
+}
+
+void EchoRemoverMetrics::Update(
+    const AecState& aec_state,
+    const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
+    const std::array<float, kFftLengthBy2Plus1>& suppressor_gain) {
+  if (++block_counter_ <= kMetricsCollectionBlocks) {
+    aec3::UpdateDbMetric(aec_state.Erl(), &erl_);
+    aec3::UpdateDbMetric(aec_state.Erle(), &erle_);
+    aec3::UpdateDbMetric(comfort_noise_spectrum, &comfort_noise_);
+    aec3::UpdateDbMetric(suppressor_gain, &suppressor_gain_);
+    active_render_ += (aec_state.ActiveRender() ? 1 : 0);

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

active_render_count_

[peah-webrtc, 2017/02/28 12:57:25]

Done.

+    saturated_capture_ = saturated_capture_ || aec_state.SaturatedCapture();
+  } else {
+    // Report the metrics over several frames in order to lower the impact of
+    // the logarithms involved on the computational complexity.
+    constexpr int kMetricsCollectionBlocksBy2 = kMetricsCollectionBlocks / 2;
+    constexpr float kComfortNoiseScaling = 1.f / (kBlockSize * kBlockSize);
+    switch (block_counter_) {
+      case kMetricsCollectionBlocks + 1:
+        erle_[0].FormatForReporting(true, 0.f, 19.f, 0.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageErleBand0",
+            static_cast<int>(erle_[0].sum_value), 0, 19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxErleBand0",
+                                    static_cast<int>(erle_[0].ceil_value), 0,
+                                    19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MinErleBand0",
+                                    static_cast<int>(erle_[0].floor_value), 0,
+                                    19, 20);
+        break;
+      case kMetricsCollectionBlocks + 2:
+        erle_[1].FormatForReporting(true, 0.f, 19.f, 0.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageErleBand1",
+            static_cast<int>(erle_[1].sum_value), 0, 19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxErleBand1",
+                                    static_cast<int>(erle_[1].ceil_value), 0,
+                                    19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MinErleBand1",
+                                    static_cast<int>(erle_[1].floor_value), 0,
+                                    19, 20);
+        break;
+      case kMetricsCollectionBlocks + 3:
+        erl_[0].FormatForReporting(true, 0.f, 59.f, 30.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageErlBand0",
+            static_cast<int>(erl_[0].sum_value), 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxErlBand0",
+                                    static_cast<int>(erl_[0].ceil_value), 0, 59,
+                                    30);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MinErlBand0",
+                                    static_cast<int>(erl_[0].floor_value), 0,
+                                    59, 30);
+        break;
+      case kMetricsCollectionBlocks + 4:
+        erl_[1].FormatForReporting(true, 0.f, 59.f, 30.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageErlBand1",
+            static_cast<int>(erl_[1].sum_value), 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxErlBand1",
+                                    static_cast<int>(erl_[1].ceil_value), 0, 59,
+                                    30);
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MinErlBand1",
+                                    static_cast<int>(erl_[1].floor_value), 0,
+                                    59, 30);
+        break;
+      case kMetricsCollectionBlocks + 5:
+        comfort_noise_[0].FormatForReporting(true, 0.f, 89.f, -90.3f,
+                                             kComfortNoiseScaling);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageComfortNoiseBand0",
+            comfort_noise_[0].sum_value, 0, 89, 45);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MaxComfortNoiseBand0",
+            comfort_noise_[0].ceil_value, 0, 89, 45);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MinComfortNoiseBand0",
+            comfort_noise_[0].floor_value, 0, 89, 45);
+        break;
+      case kMetricsCollectionBlocks + 6:
+        comfort_noise_[1].FormatForReporting(true, 0.f, 89.f, -90.3f,
+                                             kComfortNoiseScaling);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageComfortNoiseBand1",
+            comfort_noise_[1].sum_value, 0, 89, 45);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MaxComfortNoiseBand1",
+            comfort_noise_[1].ceil_value, 0, 89, 45);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MinComfortNoiseBand1",
+            comfort_noise_[1].floor_value, 0, 89, 45);
+        break;
+      case kMetricsCollectionBlocks + 7:
+        suppressor_gain_[0].FormatForReporting(true, 0.f, 59.f, 0.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageSuppressorGainBand0",
+            suppressor_gain_[0].sum_value, 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MaxSuppressorGainBand0",
+            suppressor_gain_[0].ceil_value, 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MinSuppressorGainBand0",
+            suppressor_gain_[0].floor_value, 0, 59, 30);
+        break;
+      case kMetricsCollectionBlocks + 8:
+        suppressor_gain_[1].FormatForReporting(true, 0.f, 59.f, 0.f, 1.f);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.AverageSuppressorGainBand1",
+            suppressor_gain_[1].sum_value, 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MaxSuppressorGainBand1",
+            suppressor_gain_[1].ceil_value, 0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.MinSuppressorGainBand1",
+            suppressor_gain_[1].floor_value, 0, 59, 30);
+        break;
+      case kMetricsCollectionBlocks + 9:
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.UsableLinearEstimate",
+            aec_state.UsableLinearEstimate() ? 1 : 0, 0, 1, 2);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.ModelBasedAecFeasible",
+            aec_state.ModelBasedAecFeasible() ? 1 : 0, 0, 1, 2);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.ActiveRender",
+            active_render_ > kMetricsCollectionBlocksBy2 ? 1 : 0, 0, 1, 2);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.FilterDelay",
+            aec_state.FilterDelay() ? *aec_state.FilterDelay() + 1 : 0, 0, 30,
+            31);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.CaptureSaturation",
+            saturated_capture_ ? 1 : 0, 0, 1, 2);
+        metric_reported_ = true;
+        break;
+      case kMetricsCollectionBlocks + 10:
+        RTC_DCHECK_EQ(kMetricsReportingInterval, block_counter_);

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

So every 11 times you are not reporting anything. Why don't you incorporate this
into the kMetricsCollectionBlocks + 9 case?

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

Done.

+        block_counter_ = 0;
+        ResetMetrics();
+        metric_reported_ = false;
+        break;
+      default:
+        RTC_NOTREACHED();
+        break;
+    }
+  }
+}
+
+namespace aec3 {
+
+void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
+                    std::array<EchoRemoverMetrics::DbMetric, 2>* statistic) {
+  RTC_DCHECK(statistic);
+  // Truncation is intended in the band width computation.
+  constexpr int kNumBands = 2;
+  constexpr int kBandWidth = 65 / kNumBands;
+  constexpr float kOneByBandWidth = 1.f / kBandWidth;
+  RTC_DCHECK_EQ(kNumBands, statistic->size());
+  RTC_DCHECK_EQ(65, value.size());
+  for (size_t k = 0; k < statistic->size(); ++k) {
+    float average_band =
+        std::accumulate(value.begin() + kBandWidth * k,
+                        value.begin() + kBandWidth * (k + 1), 0.f) *
+        kOneByBandWidth;
+    (*statistic)[k].Update(average_band);
+  }
+}
+
+void TransformDbMetricForReporting(bool negate,

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

I'm not fond of this method altering the value. Calling it twice will be really
strange (not intended to be used like that, I know).

I suggest you make this method return the modified value, and take the input
value "by value" instead of as a pointer. Also, you can skip the
FormatForReporting method, and simply call TransformDbMetricForReporting where
the metrics are actually reported to UMA.

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

Done.

+                                   float min_value,
+                                   float max_value,
+                                   float offset,
+                                   float scaleing,

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

scaleing -> scaling

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

Done.

+                                   float* value) {
+  RTC_DCHECK(value);
+  *value = 10.f * log10((*value) * scaleing + 1e-10f) + offset;
+  if (negate) {
+    *value = -*value;
+  }
+  *value = std::max(min_value, std::min(max_value, *value));
+}
+
+}  // namespace aec3
+
+}  // namespace webrtc