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Revert "AEC3: clarify render delay controller metrics"

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This reverts commit fd745d3e3c7083cfa52307b9e4fc908930ddf2d2.

Reason for revert: Breaks downstream projects.

Original change's description:
> AEC3: clarify render delay controller metrics
>
> This CL:
> - makes it easier to understand the (nontrivial) metric interpretation
> - corrects the computation of BufferDelay to use 0 for absent delay
> - deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
> - updates the unit test to directly test metric reporting
>
> Corresponding update to histograms.xml:
> https://crrev.com/c/3944909
>
> Bug: webrtc:8671, chromium:1349051
> Change-Id: If73b6fca4de7343bff2c53f72cedda458d36c599
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278782
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38362}

Bug: webrtc:8671, chromium:1349051
Change-Id: I1e2bd0f91acb67532e21f5d2f8526a398711a413
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279040
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38367}
This commit is contained in:
Mirko Bonadei 2022-10-12 13:40:44 +00:00 committed by WebRTC LUCI CQ
parent db955f0f13
commit b2b627701c
4 changed files with 36 additions and 53 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
modules/audio_processing/aec3/render_delay_controller.cc
View File

@ -155,11 +155,9 @@ absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
last_delay_estimate_quality_ = delay_samples_->quality;
}
metrics_.Update(
delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
: absl::nullopt,
delay_ ? absl::optional<size_t>(delay_->delay) : absl::nullopt,
delay_estimator_.Clockdrift());
metrics_.Update(delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
: absl::nullopt,
delay_ ? delay_->delay : 0, 0, delay_estimator_.Clockdrift());
data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
delay_samples ? delay_samples->delay : 0);

29
modules/audio_processing/aec3/render_delay_controller_metrics.cc
View File

@ -37,13 +37,16 @@ enum class DelayChangesCategory {
kNumCategories
};
constexpr int kMaxSkewShiftCount = 20;
} // namespace
RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
void RenderDelayControllerMetrics::Update(
absl::optional<size_t> delay_samples,
absl::optional<size_t> buffer_delay_blocks,
size_t buffer_delay_blocks,
absl::optional<int> skew_shift_blocks,
ClockdriftDetector::Level clockdrift) {
++call_counter_;
@ -51,8 +54,6 @@ void RenderDelayControllerMetrics::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;
@ -63,21 +64,21 @@ void RenderDelayControllerMetrics::Update(
delay_blocks_ = delay_blocks;
}
if (skew_shift_blocks) {
skew_shift_count_ = std::min(kMaxSkewShiftCount, skew_shift_count_);
}
} 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);
value_to_report = static_cast<int>(buffer_delay_blocks + 2);
value_to_report = std::min(124, value_to_report >> 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
value_to_report, 0, 124, 125);
@ -119,8 +120,20 @@ void RenderDelayControllerMetrics::Update(
"WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
static_cast<int>(ClockdriftDetector::Level::kNumCategories));
metrics_reported_ = true;
call_counter_ = 0;
ResetMetrics();
} else {
metrics_reported_ = false;
}
if (!initial_update && ++skew_report_timer_ == 60 * kNumBlocksPerSecond) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxSkewShiftCount",
skew_shift_count_, 0, kMaxSkewShiftCount,
kMaxSkewShiftCount + 1);
skew_shift_count_ = 0;
skew_report_timer_ = 0;
}
}

9
modules/audio_processing/aec3/render_delay_controller_metrics.h
View File

@ -29,9 +29,13 @@ class RenderDelayControllerMetrics {
// Updates the metric with new data.
void Update(absl::optional<size_t> delay_samples,
absl::optional<size_t> buffer_delay_blocks,
size_t buffer_delay_blocks,
absl::optional<int> skew_shift_blocks,
ClockdriftDetector::Level clockdrift);
// Returns true if the metrics have just been reported, otherwise false.
bool MetricsReported() { return metrics_reported_; }
private:
// Resets the metrics.
void ResetMetrics();
@ -40,8 +44,11 @@ class RenderDelayControllerMetrics {
int reliable_delay_estimate_counter_ = 0;
int delay_change_counter_ = 0;
int call_counter_ = 0;
int skew_report_timer_ = 0;
int initial_call_counter_ = 0;
bool metrics_reported_ = false;
bool initial_update = true;
int skew_shift_count_ = 0;
};
} // namespace webrtc

43
modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
View File

@ -12,7 +12,6 @@
#include "absl/types/optional.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "system_wrappers/include/metrics.h"
#include "test/gtest.h"
namespace webrtc {
@ -21,49 +20,15 @@ namespace webrtc {
TEST(RenderDelayControllerMetrics, NormalUsage) {
RenderDelayControllerMetrics metrics;
int expected_num_metric_reports = 0;
for (int j = 0; j < 3; ++j) {
for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
metrics.Update(absl::nullopt, absl::nullopt,
metrics.Update(absl::nullopt, 0, absl::nullopt,
ClockdriftDetector::Level::kNone);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.EchoPathDelay"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.BufferDelay"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(metrics::NumSamples(
"WebRTC.Audio.EchoCanceller.ReliableDelayEstimates"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.DelayChanges"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.Clockdrift"),
expected_num_metric_reports);
EXPECT_FALSE(metrics.MetricsReported());
}
// We expect metric reports every kMetricsReportingIntervalBlocks blocks.
++expected_num_metric_reports;
metrics.Update(absl::nullopt, absl::nullopt,
metrics.Update(absl::nullopt, 0, absl::nullopt,
ClockdriftDetector::Level::kNone);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.EchoPathDelay"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.BufferDelay"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(metrics::NumSamples(
"WebRTC.Audio.EchoCanceller.ReliableDelayEstimates"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.DelayChanges"),
expected_num_metric_reports);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.Clockdrift"),
expected_num_metric_reports);
EXPECT_TRUE(metrics.MetricsReported());
}
}