Add av sync metrics to pc level tests

Bug: webrtc:11381
Change-Id: I0a44583114401f09425d49dbb36957160b3f149f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178201
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31603}
This commit is contained in:
Andrey Logvin
2020-07-01 11:22:35 +00:00
committed by Commit Bot
parent ba0ba71e93
commit afeb07030e
4 changed files with 228 additions and 0 deletions

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@ -361,6 +361,7 @@ if (!build_with_chromium) {
] ]
deps = [ deps = [
":analyzer_helper", ":analyzer_helper",
":cross_media_metrics_reporter",
":default_audio_quality_analyzer", ":default_audio_quality_analyzer",
":default_video_quality_analyzer", ":default_video_quality_analyzer",
":media_helper", ":media_helper",
@ -659,6 +660,31 @@ if (!build_with_chromium) {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
} }
rtc_library("cross_media_metrics_reporter") {
visibility = [ "*" ]
testonly = true
sources = [
"cross_media_metrics_reporter.cc",
"cross_media_metrics_reporter.h",
]
deps = [
"../..:perf_test",
"../../../api:network_emulation_manager_api",
"../../../api:peer_connection_quality_test_fixture_api",
"../../../api:rtc_stats_api",
"../../../api:track_id_stream_info_map",
"../../../api/units:timestamp",
"../../../rtc_base:criticalsection",
"../../../rtc_base:rtc_event",
"../../../rtc_base:rtc_numerics",
"../../../system_wrappers:field_trial",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("sdp_changer") { rtc_library("sdp_changer") {
visibility = [ "*" ] visibility = [ "*" ]
testonly = true testonly = true

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@ -0,0 +1,129 @@
/*
* Copyright (c) 2020 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 "test/pc/e2e/cross_media_metrics_reporter.h"
#include <utility>
#include <vector>
#include "api/stats/rtc_stats.h"
#include "api/stats/rtcstats_objects.h"
#include "api/units/timestamp.h"
#include "rtc_base/event.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace webrtc_pc_e2e {
void CrossMediaMetricsReporter::Start(
absl::string_view test_case_name,
const TrackIdStreamInfoMap* reporter_helper) {
test_case_name_ = std::string(test_case_name);
reporter_helper_ = reporter_helper;
}
void CrossMediaMetricsReporter::OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) {
auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
std::map<absl::string_view, std::vector<const RTCInboundRTPStreamStats*>>
sync_group_stats;
for (const auto& stat : inbound_stats) {
auto media_source_stat =
report->GetAs<RTCMediaStreamTrackStats>(*stat->track_id);
if (stat->estimated_playout_timestamp.ValueOrDefault(0.) > 0 &&
media_source_stat->track_identifier.is_defined()) {
sync_group_stats[reporter_helper_->GetSyncGroupLabelFromTrackId(
*media_source_stat->track_identifier)]
.push_back(stat);
}
}
rtc::CritScope cs(&lock_);
for (const auto& pair : sync_group_stats) {
// If there is less than two streams, it is not a sync group.
if (pair.second.size() < 2) {
continue;
}
auto sync_group = std::string(pair.first);
const RTCInboundRTPStreamStats* audio_stat = pair.second[0];
const RTCInboundRTPStreamStats* video_stat = pair.second[1];
RTC_CHECK(pair.second.size() == 2 && audio_stat->kind.is_defined() &&
video_stat->kind.is_defined() &&
*audio_stat->kind != *video_stat->kind)
<< "Sync group should consist of one audio and one video stream.";
if (*audio_stat->kind == RTCMediaStreamTrackKind::kVideo) {
std::swap(audio_stat, video_stat);
}
// Stream labels of a sync group are same for all polls, so we need it add
// it only once.
if (stats_info_.find(sync_group) == stats_info_.end()) {
auto audio_source_stat =
report->GetAs<RTCMediaStreamTrackStats>(*audio_stat->track_id);
auto video_source_stat =
report->GetAs<RTCMediaStreamTrackStats>(*video_stat->track_id);
// *_source_stat->track_identifier is always defined here because we
// checked it while grouping stats.
stats_info_[sync_group].audio_stream_label =
std::string(reporter_helper_->GetStreamLabelFromTrackId(
*audio_source_stat->track_identifier));
stats_info_[sync_group].video_stream_label =
std::string(reporter_helper_->GetStreamLabelFromTrackId(
*video_source_stat->track_identifier));
}
double audio_video_playout_diff = *audio_stat->estimated_playout_timestamp -
*video_stat->estimated_playout_timestamp;
if (audio_video_playout_diff > 0) {
stats_info_[sync_group].audio_ahead_ms.AddSample(
audio_video_playout_diff);
stats_info_[sync_group].video_ahead_ms.AddSample(0);
} else {
stats_info_[sync_group].audio_ahead_ms.AddSample(0);
stats_info_[sync_group].video_ahead_ms.AddSample(
std::abs(audio_video_playout_diff));
}
}
}
void CrossMediaMetricsReporter::StopAndReportResults() {
rtc::CritScope cs(&lock_);
for (const auto& pair : stats_info_) {
const std::string& sync_group = pair.first;
ReportResult("audio_ahead_ms",
GetTestCaseName(pair.second.audio_stream_label, sync_group),
pair.second.audio_ahead_ms, "ms",
webrtc::test::ImproveDirection::kSmallerIsBetter);
ReportResult("video_ahead_ms",
GetTestCaseName(pair.second.video_stream_label, sync_group),
pair.second.video_ahead_ms, "ms",
webrtc::test::ImproveDirection::kSmallerIsBetter);
}
}
void CrossMediaMetricsReporter::ReportResult(
const std::string& metric_name,
const std::string& test_case_name,
const SamplesStatsCounter& counter,
const std::string& unit,
webrtc::test::ImproveDirection improve_direction) {
test::PrintResult(metric_name, /*modifier=*/"", test_case_name, counter, unit,
/*important=*/false, improve_direction);
}
std::string CrossMediaMetricsReporter::GetTestCaseName(
const std::string& stream_label,
const std::string& sync_group) const {
return test_case_name_ + "/" + sync_group + "_" + stream_label;
}
} // namespace webrtc_pc_e2e
} // namespace webrtc

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@ -0,0 +1,70 @@
/*
* Copyright (c) 2020 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.
*/
#ifndef TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
#define TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
#include <map>
#include <string>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/test/peerconnection_quality_test_fixture.h"
#include "api/test/track_id_stream_info_map.h"
#include "api/units/timestamp.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/numerics/samples_stats_counter.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace webrtc_pc_e2e {
class CrossMediaMetricsReporter
: public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
public:
CrossMediaMetricsReporter() = default;
~CrossMediaMetricsReporter() override = default;
void Start(absl::string_view test_case_name,
const TrackIdStreamInfoMap* reporter_helper) override;
void OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) override;
void StopAndReportResults() override;
private:
struct StatsInfo {
SamplesStatsCounter audio_ahead_ms;
SamplesStatsCounter video_ahead_ms;
std::string audio_stream_label;
std::string video_stream_label;
};
static void ReportResult(const std::string& metric_name,
const std::string& test_case_name,
const SamplesStatsCounter& counter,
const std::string& unit,
webrtc::test::ImproveDirection improve_direction =
webrtc::test::ImproveDirection::kNone);
std::string GetTestCaseName(const std::string& stream_label,
const std::string& sync_group) const;
std::string test_case_name_;
const TrackIdStreamInfoMap* reporter_helper_;
rtc::CriticalSection lock_;
std::map<std::string, StatsInfo> stats_info_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_

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@ -33,6 +33,7 @@
#include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h" #include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h" #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
#include "test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h" #include "test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h"
#include "test/pc/e2e/cross_media_metrics_reporter.h"
#include "test/pc/e2e/stats_poller.h" #include "test/pc/e2e/stats_poller.h"
#include "test/pc/e2e/test_peer_factory.h" #include "test/pc/e2e/test_peer_factory.h"
#include "test/testsupport/file_utils.h" #include "test/testsupport/file_utils.h"
@ -251,6 +252,8 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
RTC_LOG(INFO) << "video_analyzer_threads=" << video_analyzer_threads; RTC_LOG(INFO) << "video_analyzer_threads=" << video_analyzer_threads;
quality_metrics_reporters_.push_back( quality_metrics_reporters_.push_back(
std::make_unique<VideoQualityMetricsReporter>(clock_)); std::make_unique<VideoQualityMetricsReporter>(clock_));
quality_metrics_reporters_.push_back(
std::make_unique<CrossMediaMetricsReporter>());
video_quality_analyzer_injection_helper_->Start( video_quality_analyzer_injection_helper_->Start(
test_case_name_, test_case_name_,