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}
diff --git a/experiments/field_trials.py b/experiments/field_trials.py
index fe3ca7d..ebcace0 100755
--- a/experiments/field_trials.py
+++ b/experiments/field_trials.py
@@ -101,6 +101,9 @@
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)),
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index 21ca8a2..acb95d1 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -190,11 +190,13 @@
"requested_resolution simultaniously.");
}
- if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
- rtp_parameters.encodings[i].codec) {
- LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
- "Attempted to use different codec values for "
- "different encodings.");
+ if (!field_trials.IsEnabled("WebRTC-MixedCodecSimulcast")) {
+ if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
+ rtp_parameters.encodings[i].codec) {
+ LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
+ "Attempted to use different codec values for "
+ "different encodings.");
+ }
}
}
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index 9b4bfba..bce27fc 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/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 @@
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 @@
return true;
}
+ test::ScopedKeyValueConfig field_trials_;
rtc::PhysicalSocketServer pss_;
std::unique_ptr<rtc::Thread> background_thread_;
};
@@ -2038,6 +2042,7 @@
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 @@
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.
diff --git a/pc/test/peer_connection_test_wrapper.cc b/pc/test/peer_connection_test_wrapper.cc
index dc926c7..d77924f 100644
--- a/pc/test/peer_connection_test_wrapper.cc
+++ b/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 @@
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 @@
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;
}
diff --git a/pc/test/peer_connection_test_wrapper.h b/pc/test/peer_connection_test_wrapper.h
index 966bee0..70337b8 100644
--- a/pc/test/peer_connection_test_wrapper.h
+++ b/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 @@
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(
