Remove mentions of already deleted field trials

- WebRTC-Audio-Agc2ForceExtraSaturationMargin
- WebRTC-Audio-Agc2ForceInitialSaturationMargin
- WebRTC-Audio-BitrateAdaptation
- WebRTC-Audio-TransientSuppressorVadMode
- WebRTC-FrameBuffer3
- WebRTC-IntelVP8
- WebRTC-UseActiveIceController

Bug: None
Change-Id: I3545727c09f761867f2f4c2bb5c400012ce146d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39444}

api/ice_transport_interface.h

@@ -90,18 +90,11 @@ struct IceTransportInit final {
   // best connection to use or ping, and lets the transport decide when and
   // whether to switch.
   //
-  // Which ICE controller is used is determined based on the field trial
+  // Which ICE controller is used is determined as follows:
-  // "WebRTC-UseActiveIceController" as follows:
   //
-  //   1. If the field trial is not enabled
+  //   1. If an active ICE controller factory is supplied, it is used and
-  //      a. The legacy ICE controller factory is used if one is supplied.
-  //      b. If not, a default ICE controller (BasicIceController) is
-  //      constructed and used.
-  //
-  //   2. If the field trial is enabled
-  //      a. If an active ICE controller factory is supplied, it is used and
   //      the legacy ICE controller factory is not used.
-  //      b. If not, a default active ICE controller is used, wrapping over the
+  //   2. If not, a default active ICE controller is used, wrapping over the
   //      supplied or the default legacy ICE controller.
   void set_active_ice_controller_factory(
       cricket::ActiveIceControllerFactoryInterface*

examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java

@@ -102,7 +102,6 @@ public class PeerConnectionClient {
   private static final String VIDEO_CODEC_PARAM_START_BITRATE = "x-google-start-bitrate";
   private static final String VIDEO_FLEXFEC_FIELDTRIAL =
       "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
-  private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/";
   private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL =
       "WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/";
   private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate";
@@ -1002,7 +1001,6 @@ public class PeerConnectionClient {
       fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL;
       Log.d(TAG, "Enable FlexFEC field trial.");
     }
-    fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL;
     if (peerConnectionParameters.disableWebRtcAGCAndHPF) {
       fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL;
       Log.d(TAG, "Disable WebRTC AGC field trial.");

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

@@ -175,8 +175,7 @@ TEST(AudioNetworkAdaptorImplTest,
 TEST(AudioNetworkAdaptorImplTest,
      DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
   test::ScopedFieldTrials override_field_trials(
-      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "WebRTC-Audio-FecAdaptation/Enabled/");
-      "Enabled/");
   rtc::ScopedFakeClock fake_clock;
   fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
   auto states = CreateAudioNetworkAdaptor();
@@ -248,8 +247,7 @@ TEST(AudioNetworkAdaptorImplTest,
 
 TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
   test::ScopedFieldTrials override_field_trials(
-      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "WebRTC-Audio-FecAdaptation/Enabled/");
-      "Enabled/");
   auto states = CreateAudioNetworkAdaptor();
 
   AudioEncoderRuntimeConfig config;

modules/audio_processing/audio_processing_impl_unittest.cc

@@ -520,9 +520,6 @@ TEST(AudioProcessingImplTest,
   apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
 }
 
-// Tests that a stream is successfully processed when AGC2 adaptive digital is
-// used and when the field trial
-// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/` is set.
 TEST(AudioProcessingImplTest,
      ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
   webrtc::test::ScopedFieldTrials field_trials(
@@ -553,9 +550,6 @@ TEST(AudioProcessingImplTest,
   }
 }
 
-// Tests that a stream is successfully processed when AGC2 adaptive digital is
-// used and when the field trial
-// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` is set.
 TEST(AudioProcessingImplTest,
      ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
   webrtc::test::ScopedFieldTrials field_trials(

pc/peer_connection_integrationtest.cc

@@ -105,12 +105,10 @@ namespace {
 
 class PeerConnectionIntegrationTest
     : public PeerConnectionIntegrationBaseTest,
-      public ::testing::WithParamInterface<
+      public ::testing::WithParamInterface<SdpSemantics> {
-          std::tuple<SdpSemantics, std::string>> {
  protected:
   PeerConnectionIntegrationTest()
-      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()),
+      : PeerConnectionIntegrationBaseTest(GetParam()) {}
-                                          std::get<1>(GetParam())) {}
 };
 
 // Fake clock must be set before threads are started to prevent race on
@@ -3469,21 +3467,15 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
   EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, 12345);
 }
 
-INSTANTIATE_TEST_SUITE_P(
+INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
-    PeerConnectionIntegrationTest,
+                         PeerConnectionIntegrationTest,
-    PeerConnectionIntegrationTest,
+                         Values(SdpSemantics::kPlanB_DEPRECATED,
-    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
+                                SdpSemantics::kUnifiedPlan));
-            Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/",
-                   "WebRTC-FrameBuffer3/arm:FrameBuffer3/",
-                   "WebRTC-FrameBuffer3/arm:SyncDecoding/")));
 
-INSTANTIATE_TEST_SUITE_P(
+INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
-    PeerConnectionIntegrationTest,
+                         PeerConnectionIntegrationTestWithFakeClock,
-    PeerConnectionIntegrationTestWithFakeClock,
+                         Values(SdpSemantics::kPlanB_DEPRECATED,
-    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
+                                SdpSemantics::kUnifiedPlan));
-            Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/",
-                   "WebRTC-FrameBuffer3/arm:FrameBuffer3/",
-                   "WebRTC-FrameBuffer3/arm:SyncDecoding/")));
 
 // Tests that verify interoperability between Plan B and Unified Plan
 // PeerConnections.

pc/slow_peer_connection_integration_test.cc

@@ -15,7 +15,6 @@
 
 #include <memory>
 #include <string>
-#include <tuple>
 #include <utility>
 #include <vector>
 
@@ -50,12 +49,10 @@ namespace {
 
 class PeerConnectionIntegrationTest
     : public PeerConnectionIntegrationBaseTest,
-      public ::testing::WithParamInterface<
+      public ::testing::WithParamInterface<SdpSemantics> {
-          std::tuple<SdpSemantics, std::string>> {
  protected:
   PeerConnectionIntegrationTest()
-      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()),
+      : PeerConnectionIntegrationBaseTest(GetParam()) {}
-                                          std::get<1>(GetParam())) {}
 };
 
 // Fake clock must be set before threads are started to prevent race on
@@ -483,13 +480,10 @@ TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }
 
-INSTANTIATE_TEST_SUITE_P(
+INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
-    PeerConnectionIntegrationTest,
+                         PeerConnectionIntegrationTest,
-    PeerConnectionIntegrationTest,
+                         Values(SdpSemantics::kPlanB_DEPRECATED,
-    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
+                                SdpSemantics::kUnifiedPlan));
-            Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/",
-                   "WebRTC-FrameBuffer3/arm:FrameBuffer3/",
-                   "WebRTC-FrameBuffer3/arm:SyncDecoding/")));
 
 constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
                                       cricket::PORTALLOCATOR_DISABLE_STUN |

test/fuzzers/audio_processing_configs_fuzzer.cc

@@ -29,8 +29,6 @@ namespace webrtc {
 namespace {
 
 const std::string kFieldTrialNames[] = {
-    "WebRTC-Audio-Agc2ForceExtraSaturationMargin",
-    "WebRTC-Audio-Agc2ForceInitialSaturationMargin",
     "WebRTC-Aec3MinErleDuringOnsetsKillSwitch",
     "WebRTC-Aec3ShortHeadroomKillSwitch",
 };