Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

9 files changed, 10 insertions, 34 deletions

diff --git a/third_party/libwebrtc/test/pc/e2e/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
index 0eb7aa2c68..22c9ee48d2 100644
--- a/third_party/libwebrtc/test/pc/e2e/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@@ -109,6 +109,7 @@ if (!build_with_chromium) {
         "../../../api/video_codecs:builtin_video_encoder_factory",
         "../../../modules/audio_device:test_audio_device_module",
         "../../../modules/audio_processing/aec_dump",
+        "../../../p2p:basic_port_allocator",
         "../../../p2p:rtc_p2p",
         "../../../rtc_base:threading",
         "analyzer/video:quality_analyzing_video_encoder",
@@ -576,6 +577,7 @@ if (!build_with_chromium) {
       "../../../media:media_constants",
       "../../../media:rid_description",
       "../../../media:rtc_media_base",
+      "../../../p2p:p2p_constants",
       "../../../p2p:rtc_p2p",
       "../../../pc:sdp_utils",
       "../../../pc:session_description",
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
index 17876e54be..6adfc50049 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
@@ -130,6 +130,7 @@ rtc_library("quality_analyzing_video_decoder") {
     ":encoded_image_data_injector_api",
     ":simulcast_dummy_buffer_helper",
     "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/environment",
     "../../../../../api/video:encoded_image",
     "../../../../../api/video:video_frame",
     "../../../../../api/video_codecs:video_codecs_api",
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
index e17b5d5d83..3cd179370f 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
@@ -259,10 +259,10 @@ QualityAnalyzingVideoDecoderFactory::GetSupportedFormats() const {
   return delegate_->GetSupportedFormats();
 }
 
-std::unique_ptr<VideoDecoder>
-QualityAnalyzingVideoDecoderFactory::CreateVideoDecoder(
+std::unique_ptr<VideoDecoder> QualityAnalyzingVideoDecoderFactory::Create(
+    const Environment& env,
     const SdpVideoFormat& format) {
-  std::unique_ptr<VideoDecoder> decoder = delegate_->CreateVideoDecoder(format);
+  std::unique_ptr<VideoDecoder> decoder = delegate_->Create(env, format);
   return std::make_unique<QualityAnalyzingVideoDecoder>(
       peer_name_, std::move(decoder), extractor_, analyzer_);
 }
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
index 2f0c2b9d5d..daa919d7e4 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
@@ -18,6 +18,7 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/environment/environment.h"
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
@@ -136,8 +137,8 @@ class QualityAnalyzingVideoDecoderFactory : public VideoDecoderFactory {
 
   // Methods of VideoDecoderFactory interface.
   std::vector<SdpVideoFormat> GetSupportedFormats() const override;
-  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
-      const SdpVideoFormat& format) override;
+  std::unique_ptr<VideoDecoder> Create(const Environment& env,
+                                       const SdpVideoFormat& format) override;
 
  private:
   const std::string peer_name_;
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
index 257fecf309..3c4f6cabe1 100644
--- a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
@@ -27,11 +27,6 @@ using ::webrtc::test::Unit;
 
 constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
 
-// Field trial which controls whether to report standard-compliant bytes
-// sent/received per stream.  If enabled, padding and headers are not included
-// in bytes sent or received.
-constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
-
 }  // namespace
 
 NetworkQualityMetricsReporter::NetworkQualityMetricsReporter(
@@ -107,11 +102,6 @@ void NetworkQualityMetricsReporter::StopAndReportResults() {
   ReportStats(alice_network_label_, alice_stats, alice_packets_loss);
   ReportStats(bob_network_label_, bob_stats, bob_packets_loss);
 
-  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
-    RTC_LOG(LS_ERROR)
-        << "Non-standard GetStats; \"payload\" counts include RTP headers";
-  }
-
   MutexLock lock(&lock_);
   for (const auto& pair : pc_stats_) {
     ReportPCStats(pair.first, pair.second);
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
index 1348a58943..fd523cc48d 100644
--- a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
@@ -48,8 +48,6 @@ class NetworkQualityMetricsReporter
 
  private:
   struct PCStats {
-    // TODO(nisse): Separate audio and video counters. Depends on standard stat
-    // counters, enabled by field trial "WebRTC-UseStandardBytesStats".
     DataSize payload_received = DataSize::Zero();
     DataSize payload_sent = DataSize::Zero();
   };
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
index 90f201facd..3a6b808167 100644
--- a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
@@ -73,8 +73,6 @@ constexpr TimeDelta kQuickTestModeRunDuration = TimeDelta::Millis(100);
 // Field trials to enable Flex FEC advertising and receiving.
 constexpr char kFlexFecEnabledFieldTrials[] =
     "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
-constexpr char kUseStandardsBytesStats[] =
-    "WebRTC-UseStandardBytesStats/Enabled/";
 
 class FixturePeerConnectionObserver : public MockPeerConnectionObserver {
  public:
@@ -439,8 +437,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
 
 std::string PeerConnectionE2EQualityTest::GetFieldTrials(
     const RunParams& run_params) {
-  std::vector<absl::string_view> default_field_trials = {
-      kUseStandardsBytesStats};
+  std::vector<absl::string_view> default_field_trials = {};
   if (run_params.enable_flex_fec_support) {
     default_field_trials.push_back(kFlexFecEnabledFieldTrials);
   }
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
index b965a7acd8..706224ce08 100644
--- a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -51,11 +51,6 @@ using NetworkLayerStats =
 
 constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
 
-// Field trial which controls whether to report standard-compliant bytes
-// sent/received per stream.  If enabled, padding and headers are not included
-// in bytes sent or received.
-constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
-
 EmulatedNetworkStats PopulateStats(std::vector<EmulatedEndpoint*> endpoints,
                                    NetworkEmulationManager* network_emulation) {
   rtc::Event stats_loaded;
@@ -325,11 +320,6 @@ void StatsBasedNetworkQualityMetricsReporter::OnStatsReports(
 void StatsBasedNetworkQualityMetricsReporter::StopAndReportResults() {
   Timestamp end_time = clock_->CurrentTime();
 
-  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
-    RTC_LOG(LS_ERROR)
-        << "Non-standard GetStats; \"payload\" counts include RTP headers";
-  }
-
   std::map<std::string, NetworkLayerStats> stats = collector_.GetStats();
   for (const auto& entry : stats) {
     LogNetworkLayerStats(entry.first, entry.second);
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
index 60daf40c8c..ba6bf04e18 100644
--- a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
@@ -70,9 +70,6 @@ class StatsBasedNetworkQualityMetricsReporter
 
  private:
   struct PCStats {
-    // TODO(bugs.webrtc.org/10525): Separate audio and video counters. Depends
-    // on standard stat counters, enabled by field trial
-    // "WebRTC-UseStandardBytesStats".
     DataSize payload_received = DataSize::Zero();
     DataSize payload_sent = DataSize::Zero();