Clean up use of WebRTC-UseStandardBytesStats trial in tests

BUG=webrtc:10525

Change-Id: Ia0ec88d5b561ec98af540f849182805d49a327e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337520
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41663}
diff --git a/test/pc/e2e/network_quality_metrics_reporter.cc b/test/pc/e2e/network_quality_metrics_reporter.cc
index 257fecf..3c4f6ca 100644
--- a/test/pc/e2e/network_quality_metrics_reporter.cc
+++ b/test/pc/e2e/network_quality_metrics_reporter.cc
@@ -27,11 +27,6 @@
 
 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 @@
   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/test/pc/e2e/network_quality_metrics_reporter.h b/test/pc/e2e/network_quality_metrics_reporter.h
index 1348a58..fd523cc 100644
--- a/test/pc/e2e/network_quality_metrics_reporter.h
+++ b/test/pc/e2e/network_quality_metrics_reporter.h
@@ -48,8 +48,6 @@
 
  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/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index 90f201f..3a6b808 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -73,8 +73,6 @@
 // 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 @@
 
 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/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
index b965a7a..706224c 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -51,11 +51,6 @@
 
 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::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/test/pc/e2e/stats_based_network_quality_metrics_reporter.h b/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
index 60daf40..ba6bf04 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
@@ -70,9 +70,6 @@
 
  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();