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}
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 42be910..6ac07ad 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -361,6 +361,7 @@
]
deps = [
":analyzer_helper",
+ ":cross_media_metrics_reporter",
":default_audio_quality_analyzer",
":default_video_quality_analyzer",
":media_helper",
@@ -659,6 +660,31 @@
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") {
visibility = [ "*" ]
testonly = true
diff --git a/test/pc/e2e/cross_media_metrics_reporter.cc b/test/pc/e2e/cross_media_metrics_reporter.cc
new file mode 100644
index 0000000..3bae6c9
--- /dev/null
+++ b/test/pc/e2e/cross_media_metrics_reporter.cc
@@ -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
diff --git a/test/pc/e2e/cross_media_metrics_reporter.h b/test/pc/e2e/cross_media_metrics_reporter.h
new file mode 100644
index 0000000..bff5a31
--- /dev/null
+++ b/test/pc/e2e/cross_media_metrics_reporter.h
@@ -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_
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index 8d06c2f..3b3da67 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -33,6 +33,7 @@
#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/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/test_peer_factory.h"
#include "test/testsupport/file_utils.h"
@@ -251,6 +252,8 @@
RTC_LOG(INFO) << "video_analyzer_threads=" << video_analyzer_threads;
quality_metrics_reporters_.push_back(
std::make_unique<VideoQualityMetricsReporter>(clock_));
+ quality_metrics_reporters_.push_back(
+ std::make_unique<CrossMediaMetricsReporter>());
video_quality_analyzer_injection_helper_->Start(
test_case_name_,
