зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1812154 - Vendor libwebrtc from 2ae3f7bb60
Upstream commit: https://webrtc.googlesource.com/src/+/2ae3f7bb600a19efcea7e8626d48c7098383da49 Migrate PeerConnectionRampUpTest on new perf metrics export API Bug: b/246095034 Change-Id: I3133f6f7517cc303eeec2e860614e91b2ce4402b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276630 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38207}
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@ -18198,3 +18198,6 @@ f01ceb6f93
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# MOZ_LIBWEBRTC_SRC=/Users/danielbaker/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
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# base of lastest vendoring
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183e9968ce
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# MOZ_LIBWEBRTC_SRC=/Users/danielbaker/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
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# base of lastest vendoring
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2ae3f7bb60
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@ -12154,3 +12154,5 @@ libwebrtc updated from /Users/danielbaker/moz-libwebrtc commit mozpatches on 202
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libwebrtc updated from /Users/danielbaker/moz-libwebrtc commit mozpatches on 2023-02-01T22:35:38.698334.
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# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/moz-libwebrtc --commit mozpatches libwebrtc
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libwebrtc updated from /Users/danielbaker/moz-libwebrtc commit mozpatches on 2023-02-01T22:36:45.384128.
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# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/moz-libwebrtc --commit mozpatches libwebrtc
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libwebrtc updated from /Users/danielbaker/moz-libwebrtc commit mozpatches on 2023-02-01T22:38:06.058882.
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@ -2137,6 +2137,8 @@ if (rtc_include_tests && !build_with_chromium) {
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/test/metrics:global_metrics_logger_and_exporter",
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"../api/test/metrics:metric",
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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_encoder_factory",
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"../api/video_codecs:video_codecs_api",
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@ -2154,7 +2156,6 @@ if (rtc_include_tests && !build_with_chromium) {
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"../rtc_base:task_queue_for_test",
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"../rtc_base:threading",
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"../system_wrappers",
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"../test:perf_test",
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"../test:test_support",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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@ -27,6 +27,8 @@
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#include "api/stats/rtc_stats.h"
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#include "api/stats/rtc_stats_report.h"
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#include "api/stats/rtcstats_objects.h"
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#include "api/test/metrics/global_metrics_logger_and_exporter.h"
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#include "api/test/metrics/metric.h"
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#include "api/video_codecs/builtin_video_decoder_factory.h"
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#include "api/video_codecs/builtin_video_encoder_factory.h"
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#include "modules/audio_device/include/audio_device.h"
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@ -54,11 +56,14 @@
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#include "rtc_base/virtual_socket_server.h"
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#include "system_wrappers/include/clock.h"
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#include "test/gtest.h"
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#include "test/testsupport/perf_test.h"
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namespace webrtc {
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namespace {
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using ::webrtc::test::GetGlobalMetricsLogger;
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using ::webrtc::test::ImprovementDirection;
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using ::webrtc::test::Unit;
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static const int kDefaultTestTimeMs = 15000;
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static const int kRampUpTimeMs = 5000;
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static const int kPollIntervalTimeMs = 50;
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@ -72,6 +77,7 @@ static const int kTurnExternalPort = 0;
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// Setting the network bandwidth to 1 Mbps allows the video's bitrate to push
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// the network's limitations.
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static const int kNetworkBandwidth = 1000000;
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} // namespace
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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@ -265,8 +271,10 @@ class PeerConnectionRampUpTest : public ::testing::Test {
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double average_bandwidth_estimate = total_bwe / number_of_polls;
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std::string value_description =
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"bwe_after_" + std::to_string(kDefaultTestTimeMs / 1000) + "_seconds";
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test::PrintResult("peerconnection_ramp_up_", test_string, value_description,
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average_bandwidth_estimate, "bwe", false);
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GetGlobalMetricsLogger()->LogSingleValueMetric(
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"peerconnection_ramp_up_" + test_string, value_description,
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average_bandwidth_estimate, Unit::kUnitless,
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ImprovementDirection::kNeitherIsBetter);
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}
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rtc::Thread* network_thread() { return network_thread_.get(); }
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