Add WebRTC-MixedCodecSimulcast field trial

Disable the checks ensuring we reject mixed-codec simulcast
when the field trial is enabled.
The feature is however not yet implemented.

Bug: webrtc:362277533
Change-Id: Ib1601767c951d61aaa37a3d8767d0a81444d626c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361404
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42942}

experiments/field_trials.py

@@ -101,6 +101,9 @@ ACTIVE_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
     FieldTrial('WebRTC-LibvpxVp8Encoder-AndroidSpecificThreadingSettings',
                42226191,
                date(2024, 9, 1)),
+    FieldTrial('WebRTC-MixedCodecSimulcast',
+               362277533,
+               date(2025, 9, 1)),
     FieldTrial('WebRTC-Pacer-FastRetransmissions',
                40235589,
                date(2024, 4, 1)),

media/base/media_engine.cc

@@ -190,11 +190,13 @@ webrtc::RTCError CheckRtpParametersValues(
                            "requested_resolution simultaniously.");
     }
 
-    if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
+    if (!field_trials.IsEnabled("WebRTC-MixedCodecSimulcast")) {
-                     rtp_parameters.encodings[i].codec) {
+      if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
-      LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
+                       rtp_parameters.encodings[i].codec) {
-                           "Attempted to use different codec values for "
+        LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                           "different encodings.");
+                             "Attempted to use different codec values for "
+                             "different encodings.");
+      }
     }
   }
 

pc/peer_connection_encodings_integrationtest.cc

@@ -23,6 +23,7 @@
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/audio_options.h"
+#include "api/field_trials_view.h"
 #include "api/jsep.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
@@ -51,6 +52,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using ::testing::Eq;
 using ::testing::Optional;
@@ -118,7 +120,8 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
 
   rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() {
     auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>(
-        "pc", &pss_, background_thread_.get(), background_thread_.get());
+        "pc", &pss_, background_thread_.get(), background_thread_.get(),
+        field_trials_);
     pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(),
                          CreateBuiltinAudioDecoderFactory());
     return pc_wrapper;
@@ -445,6 +448,7 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
     return true;
   }
 
+  test::ScopedKeyValueConfig field_trials_;
   rtc::PhysicalSocketServer pss_;
   std::unique_ptr<rtc::Thread> background_thread_;
 };
@@ -2038,6 +2042,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   std::optional<RtpCodecCapability> vp9 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp9");
+  ASSERT_TRUE(vp9);
 
   RtpTransceiverInit init;
   init.direction = RtpTransceiverDirection::kSendOnly;
@@ -2058,6 +2063,41 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
             RTCErrorType::UNSUPPORTED_OPERATION);
 }
 
+TEST_F(PeerConnectionEncodingsIntegrationTest,
+       AddTransceiverAcceptsMixedCodecSimulcast) {
+  // Enable WIP mixed codec simulcast support
+  test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-MixedCodecSimulcast/Enabled/");
+  rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
+  rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
+  ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
+
+  std::optional<RtpCodecCapability> vp8 =
+      local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+                                                   "vp8");
+  ASSERT_TRUE(vp8);
+  std::optional<RtpCodecCapability> vp9 =
+      local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+                                                   "vp9");
+  ASSERT_TRUE(vp9);
+
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
+  encoding_parameters.rid = "h";
+  encoding_parameters.codec = vp8;
+  encoding_parameters.scale_resolution_down_by = 2;
+  init.send_encodings.push_back(encoding_parameters);
+  encoding_parameters.rid = "f";
+  encoding_parameters.codec = vp9;
+  encoding_parameters.scale_resolution_down_by = 1;
+  init.send_encodings.push_back(encoding_parameters);
+
+  auto transceiver_or_error =
+      local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+  ASSERT_TRUE(transceiver_or_error.ok());
+}
+
 // Tests that use the standard path (specifying both `scalability_mode` and
 // `scale_resolution_down_by` or `requested_resolution`) should pass for all
 // codecs.

pc/test/peer_connection_test_wrapper.cc

@@ -24,6 +24,7 @@
 #include "api/audio/audio_processing.h"
 #include "api/create_peerconnection_factory.h"
 #include "api/environment/environment.h"
+#include "api/field_trials_view.h"
 #include "api/media_types.h"
 #include "api/sequence_checker.h"
 #include "api/video_codecs/video_decoder_factory.h"
@@ -50,6 +51,7 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/time_utils.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 namespace {
 
@@ -132,6 +134,21 @@ PeerConnectionTestWrapper::PeerConnectionTestWrapper(
   pc_thread_checker_.Detach();
 }
 
+PeerConnectionTestWrapper::PeerConnectionTestWrapper(
+    const std::string& name,
+    rtc::SocketServer* socket_server,
+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread,
+    webrtc::test::ScopedKeyValueConfig& field_trials)
+    : field_trials_(field_trials, ""),
+      name_(name),
+      socket_server_(socket_server),
+      network_thread_(network_thread),
+      worker_thread_(worker_thread),
+      pending_negotiation_(false) {
+  pc_thread_checker_.Detach();
+}
+
 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {
   RTC_DCHECK_RUN_ON(&pc_thread_checker_);
   // To avoid flaky bot failures, make sure fake sources are stopped prior to
@@ -173,7 +190,8 @@ bool PeerConnectionTestWrapper::CreatePc(
           webrtc::LibvpxVp9DecoderTemplateAdapter,
           webrtc::OpenH264DecoderTemplateAdapter,
           webrtc::Dav1dDecoderTemplateAdapter>>(),
-      nullptr /* audio_mixer */, nullptr /* audio_processing */);
+      nullptr /* audio_mixer */, nullptr /* audio_processing */, nullptr,
+      std::make_unique<webrtc::test::ScopedKeyValueConfig>(field_trials_, ""));
   if (!peer_connection_factory_) {
     return false;
   }

pc/test/peer_connection_test_wrapper.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_options.h"
 #include "api/data_channel_interface.h"
+#include "api/field_trials_view.h"
 #include "api/jsep.h"
 #include "api/media_stream_interface.h"
 #include "api/peer_connection_interface.h"
@@ -48,6 +49,11 @@ class PeerConnectionTestWrapper
                             rtc::SocketServer* socket_server,
                             rtc::Thread* network_thread,
                             rtc::Thread* worker_thread);
+  PeerConnectionTestWrapper(const std::string& name,
+                            rtc::SocketServer* socket_server,
+                            rtc::Thread* network_thread,
+                            rtc::Thread* worker_thread,
+                            webrtc::test::ScopedKeyValueConfig& field_trials);
   virtual ~PeerConnectionTestWrapper();
 
   bool CreatePc(