diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn index 42be910ca8..6ac07ad622 100644 --- a/test/pc/e2e/BUILD.gn +++ b/test/pc/e2e/BUILD.gn @@ -361,6 +361,7 @@ if (!build_with_chromium) { ] deps = [ ":analyzer_helper", + ":cross_media_metrics_reporter", ":default_audio_quality_analyzer", ":default_video_quality_analyzer", ":media_helper", @@ -659,6 +660,31 @@ if (!build_with_chromium) { 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 0000000000..3bae6c9d0e --- /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 +#include + +#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& report) { + auto inbound_stats = report->GetStatsOfType(); + std::map> + sync_group_stats; + for (const auto& stat : inbound_stats) { + auto media_source_stat = + report->GetAs(*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(*audio_stat->track_id); + auto video_source_stat = + report->GetAs(*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 0000000000..bff5a3167a --- /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 +#include + +#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& 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 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 8d06c2fb16..3b3da674c0 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 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) { RTC_LOG(INFO) << "video_analyzer_threads=" << video_analyzer_threads; quality_metrics_reporters_.push_back( std::make_unique(clock_)); + quality_metrics_reporters_.push_back( + std::make_unique()); video_quality_analyzer_injection_helper_->Start( test_case_name_,