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",
};
