Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule

Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.

BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}

modules/rtp_rtcp/BUILD.gn

@@ -202,6 +202,8 @@ rtc_static_library("rtp_rtcp") {
     "../../api:scoped_refptr",
     "../../api:transport_api",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/transport:field_trial_based_config",
+    "../../api/transport:webrtc_key_value_config",
     "../../api/video:video_bitrate_allocation",
     "../../api/video:video_bitrate_allocator",
     "../../api/video:video_frame",
@@ -222,7 +224,6 @@ rtc_static_library("rtp_rtcp") {
     "../../rtc_base/system:fallthrough",
     "../../rtc_base/time:timestamp_extrapolator",
     "../../system_wrappers",
-    "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
     "../remote_bitrate_estimator",
     "../video_coding:codec_globals_headers",
@@ -432,6 +433,7 @@ if (rtc_include_tests) {
       "../../api:libjingle_peerconnection_api",
       "../../api:scoped_refptr",
       "../../api:transport_api",
+      "../../api/transport:field_trial_based_config",
       "../../api/video:video_bitrate_allocation",
       "../../api/video:video_bitrate_allocator",
       "../../api/video:video_codec_constants",

modules/rtp_rtcp/DEPS

@@ -3,4 +3,6 @@ include_rules = [
   "+common_video",
   "+logging/rtc_event_log",
   "+system_wrappers",
+  # Avoid directly using field_trial. Instead use WebRtcKeyValueConfig.
+  "-system_wrappers/include/field_trial.h",
 ]

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -18,6 +18,7 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/transport/webrtc_key_value_config.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "modules/include/module.h"
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
@@ -107,6 +108,10 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
     // Corresponds to extmap-allow-mixed in SDP negotiation.
     bool extmap_allow_mixed = false;
 
+    // If set, field trials are read from |field_trials|, otherwise
+    // defaults to  webrtc::FieldTrialBasedConfig.
+    WebRtcKeyValueConfig* field_trials = nullptr;
+
    private:
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
   };

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -18,6 +18,7 @@
 #include <utility>
 
 #include "absl/memory/memory.h"
+#include "api/transport/field_trial_based_config.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "rtc_base/checks.h"
@@ -97,6 +98,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       rtt_ms_(0) {
+  FieldTrialBasedConfig default_trials;
   if (!configuration.receiver_only) {
     rtp_sender_.reset(new RTPSender(
         configuration.audio, configuration.clock,
@@ -113,14 +115,17 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
         configuration.overhead_observer,
         configuration.populate_network2_timestamp,
         configuration.frame_encryptor, configuration.require_frame_encryption,
-        configuration.extmap_allow_mixed));
+        configuration.extmap_allow_mixed,
+        configuration.field_trials ? *configuration.field_trials
+                                   : default_trials));
     if (configuration.audio) {
       audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
     } else {
       video_ = absl::make_unique<RTPSenderVideo>(
           clock_, rtp_sender_.get(), configuration.flexfec_sender,
-          configuration.frame_encryptor,
+          configuration.frame_encryptor, configuration.require_frame_encryption,
-          configuration.require_frame_encryption);
+          configuration.field_trials ? *configuration.field_trials
+                                     : default_trials);
     }
     // Make sure rtcp sender use same timestamp offset as rtp sender.
     rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());

modules/rtp_rtcp/source/rtp_sender.cc

@@ -32,7 +32,6 @@
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/time_utils.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ -104,7 +103,8 @@ RTPSender::RTPSender(
     bool populate_network2_timestamp,
     FrameEncryptorInterface* frame_encryptor,
     bool require_frame_encryption,
-    bool extmap_allow_mixed)
+    bool extmap_allow_mixed,
+    const WebRtcKeyValueConfig& field_trials)
     : clock_(clock),
       // TODO(holmer): Remove this conversion?
       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
@@ -148,7 +148,8 @@ RTPSender::RTPSender(
       overhead_observer_(overhead_observer),
       populate_network2_timestamp_(populate_network2_timestamp),
       send_side_bwe_with_overhead_(
-          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
+          field_trials.Lookup("WebRTC-SendSideBwe-WithOverhead")
+              .find("Enabled") == 0) {
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.

modules/rtp_rtcp/source/rtp_sender.h

@@ -21,6 +21,7 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/call/transport.h"
+#include "api/transport/webrtc_key_value_config.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
@@ -60,7 +61,8 @@ class RTPSender {
             bool populate_network2_timestamp,
             FrameEncryptorInterface* frame_encryptor,
             bool require_frame_encryption,
-            bool extmap_allow_mixed);
+            bool extmap_allow_mixed,
+            const WebRtcKeyValueConfig& field_trials);
 
   ~RTPSender();
 

modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc

@@ -10,6 +10,7 @@
 
 #include <vector>
 
+#include "api/transport/field_trial_based_config.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -79,7 +80,8 @@ class RtpSenderAudioTest : public ::testing::Test {
                     false,
                     nullptr,
                     false,
-                    false),
+                    false,
+                    FieldTrialBasedConfig()),
         rtp_sender_audio_(&fake_clock_, &rtp_sender_) {
     rtp_sender_.SetSSRC(kSsrc);
     rtp_sender_.SetSequenceNumber(kSeqNum);

modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -12,6 +12,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "api/transport/field_trial_based_config.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_timing.h"
 #include "logging/rtc_event_log/events/rtc_event.h"
@@ -193,7 +194,7 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
         absl::nullopt, &seq_num_allocator_, nullptr, nullptr, nullptr,
         &mock_rtc_event_log_, &send_packet_observer_,
         &retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
-        false, false));
+        false, false, FieldTrialBasedConfig()));
     rtp_sender_->SetSequenceNumber(kSeqNum);
     rtp_sender_->SetTimestampOffset(0);
     rtp_sender_->SetSSRC(kSsrc);
@@ -333,7 +334,7 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
       kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_,
       absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
-      false));
+      false, FieldTrialBasedConfig()));
   rtp_sender_->SetTimestampOffset(0);
   rtp_sender_->SetSSRC(kSsrc);
 
@@ -376,11 +377,12 @@ TEST_P(RtpSenderTestWithoutPacer,
        TransportFeedbackObserverGetsCorrectByteCount) {
   constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
   testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
                     &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
-      &mock_rtc_event_log_, nullptr, &retransmission_rate_limiter_,
+                    &mock_rtc_event_log_, nullptr,
-      &mock_overhead_observer, false, nullptr, false, false));
+                    &retransmission_rate_limiter_, &mock_overhead_observer,
+                    false, nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
@@ -403,11 +405,12 @@ TEST_P(RtpSenderTestWithoutPacer,
 }
 
 TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
                     &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
                     &mock_rtc_event_log_, &send_packet_observer_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+                    &retransmission_rate_limiter_, nullptr, false, nullptr,
+                    false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
@@ -435,11 +438,12 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
 }
 
 TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
                     &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
                     &mock_rtc_event_log_, &send_packet_observer_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+                    &retransmission_rate_limiter_, nullptr, false, nullptr,
+                    false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
 
   SendGenericPacket();
@@ -490,13 +494,14 @@ TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
 
 TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
   testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
-      nullptr, nullptr, &send_side_delay_observer_, &mock_rtc_event_log_,
+                    nullptr, nullptr, nullptr, &send_side_delay_observer_,
-      nullptr, nullptr, nullptr, false, nullptr, false, false));
+                    &mock_rtc_event_log_, nullptr, nullptr, nullptr, false,
+                    nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
-                                  nullptr, false);
+                                  nullptr, false, FieldTrialBasedConfig());
 
   const uint8_t kPayloadType = 127;
   const char payload_name[] = "GENERIC";
@@ -573,7 +578,8 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
       false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
       &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
       &mock_rtc_event_log_, &send_packet_observer_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+      FieldTrialBasedConfig()));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   rtp_sender_->SetStorePacketsStatus(true, 10);
@@ -958,7 +964,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
       false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
       nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr,
-      false, nullptr, false, false));
+      false, nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ -984,7 +990,8 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt,
       nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+      FieldTrialBasedConfig()));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@@ -1060,7 +1067,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
   const char payload_name[] = "GENERIC";
   const uint8_t payload_type = 127;
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
-                                  nullptr, false);
+                                  nullptr, false, FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
@@ -1109,13 +1116,14 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
       false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc,
       &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
-                                  &flexfec_sender, nullptr, false);
+                                  &flexfec_sender, nullptr, false,
+                                  FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
 
   // Parameters selected to generate a single FEC packet per media packet.
@@ -1180,13 +1188,15 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
       false, &fake_clock_, &transport_, &mock_paced_sender_,
       flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
       &mock_rtc_event_log_, &send_packet_observer_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+      FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
-                                  &flexfec_sender, nullptr, false);
+                                  &flexfec_sender, nullptr, false,
+                                  FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
 
   // Need extension to be registered for timing frames to be sent.
@@ -1277,12 +1287,13 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
       false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(),
       &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
 
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
-                                  &flexfec_sender, nullptr, false);
+                                  &flexfec_sender, nullptr, false,
+                                  FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
 
   // Parameters selected to generate a single FEC packet per media packet.
@@ -1404,12 +1415,14 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
       false, &fake_clock_, &transport_, &mock_paced_sender_,
       flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
       &mock_rtc_event_log_, &send_packet_observer_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+      FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
 
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
-                                  &flexfec_sender, nullptr, false);
+                                  &flexfec_sender, nullptr, false,
+                                  FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
@@ -1469,14 +1482,15 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
     uint32_t total_bitrate_;
     uint32_t retransmit_bitrate_;
   } callback;
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
-      nullptr, &callback, nullptr, nullptr, nullptr,
+                    nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+                    &retransmission_rate_limiter_, nullptr, false, nullptr,
+                    false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
 
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
-                                  nullptr, false);
+                                  nullptr, false, FieldTrialBasedConfig());
   const char payload_name[] = "GENERIC";
   const uint8_t payload_type = 127;
   rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
@@ -1561,7 +1575,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   const char payload_name[] = "GENERIC";
   const uint8_t payload_type = 127;
   RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
-                                  nullptr, false);
+                                  nullptr, false, FieldTrialBasedConfig());
   rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
@@ -1703,7 +1717,7 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
       new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
                     nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
                     &retransmission_rate_limiter_, &mock_overhead_observer,
-                    false, nullptr, false, false));
+                    false, nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
 
   // RTP overhead is 12B.
@@ -1725,7 +1739,7 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
       new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
                     nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
                     &retransmission_rate_limiter_, &mock_overhead_observer,
-                    false, nullptr, false, false));
+                    false, nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSSRC(kSsrc);
 
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
@@ -1735,10 +1749,11 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
 
 TEST_P(RtpSenderTest, SendsKeepAlive) {
   MockTransport transport;
-  rtp_sender_.reset(new RTPSender(
+  rtp_sender_.reset(
-      false, &fake_clock_, &transport, nullptr, absl::nullopt, nullptr, nullptr,
+      new RTPSender(false, &fake_clock_, &transport, nullptr, absl::nullopt,
-      nullptr, nullptr, &mock_rtc_event_log_, nullptr,
+                    nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+                    nullptr, &retransmission_rate_limiter_, nullptr, false,
+                    nullptr, false, false, FieldTrialBasedConfig()));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetTimestampOffset(0);
   rtp_sender_->SetSSRC(kSsrc);

modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -33,7 +33,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/trace_event.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ -188,7 +187,8 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
                                RTPSender* rtp_sender,
                                FlexfecSender* flexfec_sender,
                                FrameEncryptorInterface* frame_encryptor,
-                               bool require_frame_encryption)
+                               bool require_frame_encryption,
+                               const WebRtcKeyValueConfig& field_trials)
     : rtp_sender_(rtp_sender),
       clock_(clock),
       retransmission_settings_(kRetransmitBaseLayer |
@@ -206,7 +206,8 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
       frame_encryptor_(frame_encryptor),
       require_frame_encryption_(require_frame_encryption),
       generic_descriptor_auth_experiment_(
-          field_trial::IsEnabled("WebRTC-GenericDescriptorAuth")) {}
+          field_trials.Lookup("WebRTC-GenericDescriptorAuth").find("Enabled") ==
+          0) {}
 
 RTPSenderVideo::~RTPSenderVideo() {}
 

modules/rtp_rtcp/source/rtp_sender_video.h

@@ -54,7 +54,8 @@ class RTPSenderVideo {
                  RTPSender* rtpSender,
                  FlexfecSender* flexfec_sender,
                  FrameEncryptorInterface* frame_encryptor,
-                 bool require_frame_encryption);
+                 bool require_frame_encryption,
+                 const WebRtcKeyValueConfig& field_trials);
   virtual ~RTPSenderVideo();
 
   bool SendVideo(FrameType frame_type,

modules/rtp_rtcp/source/rtp_sender_video_unittest.cc

@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <string>
 #include <vector>
 
 #include "api/video/video_codec_constants.h"
@@ -24,7 +25,6 @@
 #include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/rate_limiter.h"
-#include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -96,8 +96,14 @@ class TestRtpSenderVideo : public RTPSenderVideo {
  public:
   TestRtpSenderVideo(Clock* clock,
                      RTPSender* rtp_sender,
-                     FlexfecSender* flexfec_sender)
+                     FlexfecSender* flexfec_sender,
-      : RTPSenderVideo(clock, rtp_sender, flexfec_sender, nullptr, false) {}
+                     const WebRtcKeyValueConfig& field_trials)
+      : RTPSenderVideo(clock,
+                       rtp_sender,
+                       flexfec_sender,
+                       nullptr,
+                       false,
+                       field_trials) {}
   ~TestRtpSenderVideo() override {}
 
   StorageType GetStorageType(const RTPVideoHeader& header,
@@ -109,11 +115,26 @@ class TestRtpSenderVideo : public RTPSenderVideo {
   }
 };
 
+class FieldTrials : public WebRtcKeyValueConfig {
+ public:
+  explicit FieldTrials(bool use_send_side_bwe_with_overhead)
+      : use_send_side_bwe_with_overhead_(use_send_side_bwe_with_overhead) {}
+
+  std::string Lookup(absl::string_view key) const override {
+    return key == "WebRTC-SendSideBwe-WithOverhead" &&
+                   use_send_side_bwe_with_overhead_
+               ? "Enabled"
+               : "";
+  }
+
+ private:
+  bool use_send_side_bwe_with_overhead_;
+};
+
 class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
  public:
   RtpSenderVideoTest()
-      : field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
+      : field_trials_(GetParam()),
-                                 : ""),
         fake_clock_(kStartTime),
         retransmission_rate_limiter_(&fake_clock_, 1000),
         // TODO(pbos): Set up to use pacer.
@@ -133,8 +154,9 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
                     false,
                     nullptr,
                     false,
-                    false),
+                    false,
-        rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr) {
+                    field_trials_),
+        rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr, field_trials_) {
     rtp_sender_.SetSequenceNumber(kSeqNum);
     rtp_sender_.SetTimestampOffset(0);
     rtp_sender_.SetSSRC(kSsrc);
@@ -148,7 +170,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
       int version);
 
  protected:
-  test::ScopedFieldTrials field_trials_;
+  FieldTrials field_trials_;
   SimulatedClock fake_clock_;
   LoopbackTransportTest transport_;
   RateLimiter retransmission_rate_limiter_;