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/*
* Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file contains tests that verify that field trials do what they're
// supposed to do.
#include <set>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/peer_connection_interface.h"
#include "api/stats/rtcstats_objects.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/frame_generator_capturer_video_track_source.h"
#include "pc/test/peer_connection_test_wrapper.h"
#include "rtc_base/gunit.h"
#include "rtc_base/internal/default_socket_server.h"
#include "rtc_base/physical_socket_server.h"
#include "rtc_base/thread.h"
#include "test/gtest.h"
#include "test/scoped_key_value_config.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
namespace webrtc {
namespace {
static const int kDefaultTimeoutMs = 5000;
bool AddIceCandidates(PeerConnectionWrapper* peer,
std::vector<const IceCandidateInterface*> candidates) {
for (const auto candidate : candidates) {
if (!peer->pc()->AddIceCandidate(candidate)) {
return false;
}
}
return true;
}
} // namespace
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
class PeerConnectionFieldTrialTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapper> WrapperPtr;
PeerConnectionFieldTrialTest()
: clock_(Clock::GetRealTimeClock()),
socket_server_(rtc::CreateDefaultSocketServer()),
main_thread_(socket_server_.get()) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
config_.servers.push_back(ice_server);
config_.sdp_semantics = SdpSemantics::kUnifiedPlan;
}
void TearDown() override { pc_factory_ = nullptr; }
void CreatePCFactory(std::unique_ptr<FieldTrialsView> field_trials) {
PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.signaling_thread = rtc::Thread::Current();
pcf_deps.trials = std::move(field_trials);
pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
pcf_deps.call_factory = webrtc::CreateCallFactory();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
media_deps.adm = FakeAudioCaptureModule::Create();
media_deps.trials = pcf_deps.trials.get();
webrtc::SetMediaEngineDefaults(&media_deps);
pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
// Allow ADAPTER_TYPE_LOOPBACK to create PeerConnections with loopback in
// this test.
RTC_DCHECK(pc_factory_);
PeerConnectionFactoryInterface::Options options;
options.network_ignore_mask = 0;
pc_factory_->SetOptions(options);
}
WrapperPtr CreatePeerConnection() {
auto observer = std::make_unique<MockPeerConnectionObserver>();
auto result = pc_factory_->CreatePeerConnectionOrError(
config_, PeerConnectionDependencies(observer.get()));
RTC_CHECK(result.ok());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
Clock* const clock_;
std::unique_ptr<rtc::SocketServer> socket_server_;
rtc::AutoSocketServerThread main_thread_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
webrtc::PeerConnectionInterface::RTCConfiguration config_;
};
// Tests for the dependency descriptor field trial. The dependency descriptor
// field trial is implemented in media/engine/webrtc_video_engine.cc.
TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) {
std::unique_ptr<test::ScopedKeyValueConfig> field_trials =
std::make_unique<test::ScopedKeyValueConfig>(
"WebRTC-DependencyDescriptorAdvertised/Enabled/");
CreatePCFactory(std::move(field_trials));
WrapperPtr caller = CreatePeerConnection();
caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
auto offer = caller->CreateOffer();
auto contents1 = offer->description()->contents();
ASSERT_EQ(1u, contents1.size());
const cricket::MediaContentDescription* media_description1 =
contents1[0].media_description();
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description1->type());
const cricket::RtpHeaderExtensions& rtp_header_extensions1 =
media_description1->rtp_header_extensions();
bool found = absl::c_find_if(rtp_header_extensions1,
[](const webrtc::RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
EXPECT_TRUE(found);
}
// Tests that dependency descriptor RTP header extensions can be exchanged
// via SDP munging, even if dependency descriptor field trial is disabled.
TEST_F(PeerConnectionFieldTrialTest, InjectDependencyDescriptor) {
std::unique_ptr<test::ScopedKeyValueConfig> field_trials =
std::make_unique<test::ScopedKeyValueConfig>(
"WebRTC-DependencyDescriptorAdvertised/Disabled/");
CreatePCFactory(std::move(field_trials));
WrapperPtr caller = CreatePeerConnection();
WrapperPtr callee = CreatePeerConnection();
caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
auto offer = caller->CreateOffer();
cricket::ContentInfos& contents1 = offer->description()->contents();
ASSERT_EQ(1u, contents1.size());
cricket::MediaContentDescription* media_description1 =
contents1[0].media_description();
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description1->type());
cricket::RtpHeaderExtensions rtp_header_extensions1 =
media_description1->rtp_header_extensions();
bool found1 = absl::c_find_if(rtp_header_extensions1,
[](const webrtc::RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
EXPECT_FALSE(found1);
std::set<int> existing_ids;
for (const webrtc::RtpExtension& rtp_extension : rtp_header_extensions1) {
existing_ids.insert(rtp_extension.id);
}
// Find the currently unused RTP header extension ID.
int insert_id = 1;
std::set<int>::const_iterator iter = existing_ids.begin();
while (true) {
if (iter == existing_ids.end()) {
break;
}
if (*iter != insert_id) {
break;
}
insert_id++;
iter++;
}
rtp_header_extensions1.emplace_back(RtpExtension::kDependencyDescriptorUri,
insert_id);
media_description1->set_rtp_header_extensions(rtp_header_extensions1);
caller->SetLocalDescription(offer->Clone());
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
cricket::ContentInfos& contents2 = answer->description()->contents();
ASSERT_EQ(1u, contents2.size());
cricket::MediaContentDescription* media_description2 =
contents2[0].media_description();
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description2->type());
cricket::RtpHeaderExtensions rtp_header_extensions2 =
media_description2->rtp_header_extensions();
bool found2 = absl::c_find_if(rtp_header_extensions2,
[](const webrtc::RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions2.end();
EXPECT_TRUE(found2);
}
// Test that the ability to emulate degraded networks works without crashing.
TEST_F(PeerConnectionFieldTrialTest, ApplyFakeNetworkConfig) {
std::unique_ptr<test::ScopedKeyValueConfig> field_trials =
std::make_unique<test::ScopedKeyValueConfig>(
"WebRTC-FakeNetworkSendConfig/link_capacity_kbps:500/"
"WebRTC-FakeNetworkReceiveConfig/loss_percent:1/");
CreatePCFactory(std::move(field_trials));
WrapperPtr caller = CreatePeerConnection();
BitrateSettings bitrate_settings;
bitrate_settings.start_bitrate_bps = 1'000'000;
bitrate_settings.max_bitrate_bps = 1'000'000;
caller->pc()->SetBitrate(bitrate_settings);
FrameGeneratorCapturerVideoTrackSource::Config config;
auto video_track_source =
rtc::make_ref_counted<FrameGeneratorCapturerVideoTrackSource>(
config, clock_, /*is_screencast=*/false);
caller->AddTrack(pc_factory_->CreateVideoTrack(video_track_source, "v"));
WrapperPtr callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
// Do the SDP negotiation, and also exchange ice candidates.
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
ASSERT_TRUE_WAIT(
caller->signaling_state() == PeerConnectionInterface::kStable,
kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(callee->IsIceGatheringDone(), kDefaultTimeoutMs);
// Connect an ICE candidate pairs.
ASSERT_TRUE(
AddIceCandidates(callee.get(), caller->observer()->GetAllCandidates()));
ASSERT_TRUE(
AddIceCandidates(caller.get(), callee->observer()->GetAllCandidates()));
// This means that ICE and DTLS are connected.
ASSERT_TRUE_WAIT(callee->IsIceConnected(), kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(caller->IsIceConnected(), kDefaultTimeoutMs);
// Send packets for kDefaultTimeoutMs
WAIT(false, kDefaultTimeoutMs);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtp_stats =
caller->GetStats()->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_GE(outbound_rtp_stats.size(), 1u);
ASSERT_TRUE(outbound_rtp_stats[0]->target_bitrate.is_defined());
// Link capacity is limited to 500k, so BWE is expected to be close to 500k.
ASSERT_LE(*outbound_rtp_stats[0]->target_bitrate, 500'000 * 1.1);
}
} // namespace webrtc