mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
129f40718c
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 Previous revert: https://webrtc-review.googlesource.com/c/src/+/279040 This CL is identical to the original, except: - the test is updated to spam fewer EXPECT_EQ failures on failure (EXPECT_EQs moved out of inner loop) - the test not resets metrics (metrics::Reset()) at the beginning, like other histogram tests Bug: webrtc:8671, chromium:1349051 Change-Id: Ie802e1f9d03a22ff7018f522a63b19e0b6eec2e8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279046 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38376}
72 lines
2.7 KiB
C++
72 lines
2.7 KiB
C++
/*
|
|
* 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 "absl/types/optional.h"
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
#include "system_wrappers/include/metrics.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Verify the general functionality of RenderDelayControllerMetrics.
|
|
TEST(RenderDelayControllerMetrics, NormalUsage) {
|
|
metrics::Reset();
|
|
|
|
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,
|
|
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);
|
|
|
|
// We expect metric reports every kMetricsReportingIntervalBlocks blocks.
|
|
++expected_num_metric_reports;
|
|
|
|
metrics.Update(absl::nullopt, 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);
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|