pc/peer_connection_field_trial_tests.cc - src - Git at Google

 /*
  *  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/enable_media_with_defaults.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 "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.adm = FakeAudioCaptureModule::Create();
     EnableMediaWithDefaults(pcf_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);
   video_track_source->Start();
   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
	/*
	* 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/enable_media_with_defaults.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 "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.adm = FakeAudioCaptureModule::Create();
	EnableMediaWithDefaults(pcf_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);
	video_track_source->Start();
	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