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}
diff --git a/api/ice_transport_interface.h b/api/ice_transport_interface.h index 2ec41aa..6aae38f 100644 --- a/api/ice_transport_interface.h +++ b/api/ice_transport_interface.h
@@ -90,18 +90,11 @@ // 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 - // "WebRTC-UseActiveIceController" as follows: + // Which ICE controller is used is determined as follows: // - // 1. If the field trial is not enabled - // 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 + // 1. 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*
diff --git a/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java index 7bdce00..398b0c3 100644 --- a/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java +++ b/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -102,7 +102,6 @@ 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 @@ 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.");
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc index 70a50d6..60b4db1 100644 --- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -175,8 +175,7 @@ TEST(AudioNetworkAdaptorImplTest, DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) { test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/" - "Enabled/"); + "WebRTC-Audio-FecAdaptation/Enabled/"); rtc::ScopedFakeClock fake_clock; fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs)); auto states = CreateAudioNetworkAdaptor(); @@ -248,8 +247,7 @@ TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) { test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/" - "Enabled/"); + "WebRTC-Audio-FecAdaptation/Enabled/"); auto states = CreateAudioNetworkAdaptor(); AudioEncoderRuntimeConfig config;
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc index 7c12a07..9e50f99 100644 --- a/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -520,9 +520,6 @@ 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 @@ } } -// 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(
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index 0e915c5..feeeea3 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc
@@ -105,12 +105,10 @@ class PeerConnectionIntegrationTest : public PeerConnectionIntegrationBaseTest, - public ::testing::WithParamInterface< - std::tuple<SdpSemantics, std::string>> { + public ::testing::WithParamInterface<SdpSemantics> { protected: PeerConnectionIntegrationTest() - : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()), - std::get<1>(GetParam())) {} + : PeerConnectionIntegrationBaseTest(GetParam()) {} }; // Fake clock must be set before threads are started to prevent race on @@ -3469,21 +3467,15 @@ EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, 12345); } -INSTANTIATE_TEST_SUITE_P( - PeerConnectionIntegrationTest, - PeerConnectionIntegrationTest, - Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), - Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/", - "WebRTC-FrameBuffer3/arm:FrameBuffer3/", - "WebRTC-FrameBuffer3/arm:SyncDecoding/"))); +INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest, + PeerConnectionIntegrationTest, + Values(SdpSemantics::kPlanB_DEPRECATED, + SdpSemantics::kUnifiedPlan)); -INSTANTIATE_TEST_SUITE_P( - PeerConnectionIntegrationTest, - PeerConnectionIntegrationTestWithFakeClock, - Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), - Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/", - "WebRTC-FrameBuffer3/arm:FrameBuffer3/", - "WebRTC-FrameBuffer3/arm:SyncDecoding/"))); +INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest, + PeerConnectionIntegrationTestWithFakeClock, + Values(SdpSemantics::kPlanB_DEPRECATED, + SdpSemantics::kUnifiedPlan)); // Tests that verify interoperability between Plan B and Unified Plan // PeerConnections.
diff --git a/pc/slow_peer_connection_integration_test.cc b/pc/slow_peer_connection_integration_test.cc index b45571e..004b795 100644 --- a/pc/slow_peer_connection_integration_test.cc +++ b/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 @@ class PeerConnectionIntegrationTest : public PeerConnectionIntegrationBaseTest, - public ::testing::WithParamInterface< - std::tuple<SdpSemantics, std::string>> { + public ::testing::WithParamInterface<SdpSemantics> { protected: PeerConnectionIntegrationTest() - : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()), - std::get<1>(GetParam())) {} + : PeerConnectionIntegrationBaseTest(GetParam()) {} }; // Fake clock must be set before threads are started to prevent race on @@ -483,13 +480,10 @@ ASSERT_TRUE(ExpectNewFrames(media_expectations)); } -INSTANTIATE_TEST_SUITE_P( - PeerConnectionIntegrationTest, - PeerConnectionIntegrationTest, - Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), - Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/", - "WebRTC-FrameBuffer3/arm:FrameBuffer3/", - "WebRTC-FrameBuffer3/arm:SyncDecoding/"))); +INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest, + PeerConnectionIntegrationTest, + Values(SdpSemantics::kPlanB_DEPRECATED, + SdpSemantics::kUnifiedPlan)); constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
diff --git a/test/fuzzers/audio_processing_configs_fuzzer.cc b/test/fuzzers/audio_processing_configs_fuzzer.cc index afcb431..331a373 100644 --- a/test/fuzzers/audio_processing_configs_fuzzer.cc +++ b/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -29,8 +29,6 @@ namespace { const std::string kFieldTrialNames[] = { - "WebRTC-Audio-Agc2ForceExtraSaturationMargin", - "WebRTC-Audio-Agc2ForceInitialSaturationMargin", "WebRTC-Aec3MinErleDuringOnsetsKillSwitch", "WebRTC-Aec3ShortHeadroomKillSwitch", };