Pass the RtcEventLog instance to ICE via JsepTransportController.
This CL fixes a bug that the RtcEventLog owned by PeerConnection was not
passed to P2PTransportChannel after JsepTransportController was
introduced to deprecate the legacy TransportController.
Bug: webrtc:9337
Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5
Reviewed-on: https://webrtc-review.googlesource.com/79964
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23572}diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index ecd84d1..186da00 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -33,7 +33,12 @@
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/sdp_video_format.h"
+#include "call/call.h"
+#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
+#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/engine/fakewebrtcvideoengine.h"
+#include "media/engine/webrtcmediaengine.h"
+#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/portinterface.h"
#include "p2p/base/teststunserver.h"
@@ -239,7 +244,7 @@
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
if (!client->Init(nullptr, nullptr, nullptr, std::move(dependencies),
- network_thread, worker_thread)) {
+ network_thread, worker_thread, nullptr)) {
delete client;
return nullptr;
}
@@ -555,6 +560,10 @@
}
cricket::PortAllocator* port_allocator() const { return port_allocator_; }
+ webrtc::FakeRtcEventLogFactory* event_log_factory() const {
+ return event_log_factory_;
+ }
+
private:
explicit PeerConnectionWrapper(const std::string& debug_name)
: debug_name_(debug_name) {}
@@ -564,7 +573,8 @@
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
rtc::Thread* network_thread,
- rtc::Thread* worker_thread) {
+ rtc::Thread* worker_thread,
+ std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
// There's an error in this test code if Init ends up being called twice.
RTC_DCHECK(!peer_connection_);
RTC_DCHECK(!peer_connection_factory_);
@@ -580,15 +590,31 @@
return false;
}
rtc::Thread* const signaling_thread = rtc::Thread::Current();
- peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
- network_thread, worker_thread, signaling_thread,
- rtc::scoped_refptr<webrtc::AudioDeviceModule>(
- fake_audio_capture_module_),
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
- webrtc::CreateBuiltinVideoEncoderFactory(),
- webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
- nullptr /* audio_processing */);
+
+ webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
+ pc_factory_dependencies.network_thread = network_thread;
+ pc_factory_dependencies.worker_thread = worker_thread;
+ pc_factory_dependencies.signaling_thread = signaling_thread;
+ pc_factory_dependencies.media_engine =
+ cricket::WebRtcMediaEngineFactory::Create(
+ rtc::scoped_refptr<webrtc::AudioDeviceModule>(
+ fake_audio_capture_module_),
+ webrtc::CreateBuiltinAudioEncoderFactory(),
+ webrtc::CreateBuiltinAudioDecoderFactory(),
+ webrtc::CreateBuiltinVideoEncoderFactory(),
+ webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
+ webrtc::AudioProcessingBuilder().Create());
+ pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
+ if (event_log_factory) {
+ event_log_factory_ = event_log_factory.get();
+ pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
+ } else {
+ pc_factory_dependencies.event_log_factory =
+ webrtc::CreateRtcEventLogFactory();
+ }
+ peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
+ std::move(pc_factory_dependencies));
+
if (!peer_connection_factory_) {
return false;
}
@@ -952,6 +978,8 @@
std::vector<PeerConnectionInterface::IceGatheringState>
ice_gathering_state_history_;
+ webrtc::FakeRtcEventLogFactory* event_log_factory_;
+
rtc::AsyncInvoker invoker_;
friend class PeerConnectionIntegrationBaseTest;
@@ -1114,12 +1142,15 @@
webrtc::PeerConnectionInterface::kIceConnectionCompleted);
}
+ // When |event_log_factory| is null, the default implementation of the event
+ // log factory will be used.
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
const std::string& debug_name,
const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
- webrtc::PeerConnectionDependencies dependencies) {
+ webrtc::PeerConnectionDependencies dependencies,
+ std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
RTCConfiguration modified_config;
if (config) {
modified_config = *config;
@@ -1134,12 +1165,26 @@
if (!client->Init(constraints, options, &modified_config,
std::move(dependencies), network_thread_.get(),
- worker_thread_.get())) {
+ worker_thread_.get(), std::move(event_log_factory))) {
return nullptr;
}
return client;
}
+ std::unique_ptr<PeerConnectionWrapper>
+ CreatePeerConnectionWrapperWithFakeRtcEventLog(
+ const std::string& debug_name,
+ const MediaConstraintsInterface* constraints,
+ const PeerConnectionFactory::Options* options,
+ const RTCConfiguration* config,
+ webrtc::PeerConnectionDependencies dependencies) {
+ std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
+ new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
+ return CreatePeerConnectionWrapper(debug_name, constraints, options, config,
+ std::move(dependencies),
+ std::move(event_log_factory));
+ }
+
bool CreatePeerConnectionWrappers() {
return CreatePeerConnectionWrappersWithConfig(
PeerConnectionInterface::RTCConfiguration(),
@@ -1158,11 +1203,11 @@
sdp_semantics_ = caller_semantics;
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
sdp_semantics_ = callee_semantics;
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
sdp_semantics_ = original_semantics;
return caller_ && callee_;
}
@@ -1172,10 +1217,10 @@
MediaConstraintsInterface* callee_constraints) {
caller_ = CreatePeerConnectionWrapper(
"Caller", caller_constraints, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
callee_ = CreatePeerConnectionWrapper(
"Callee", callee_constraints, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
return caller_ && callee_;
}
@@ -1185,10 +1230,10 @@
const PeerConnectionInterface::RTCConfiguration& callee_config) {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, &caller_config,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, &callee_config,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
return caller_ && callee_;
}
@@ -1199,10 +1244,10 @@
webrtc::PeerConnectionDependencies callee_dependencies) {
caller_ =
CreatePeerConnectionWrapper("Caller", nullptr, nullptr, &caller_config,
- std::move(caller_dependencies));
+ std::move(caller_dependencies), nullptr);
callee_ =
CreatePeerConnectionWrapper("Callee", nullptr, nullptr, &callee_config,
- std::move(callee_dependencies));
+ std::move(callee_dependencies), nullptr);
return caller_ && callee_;
}
@@ -1211,9 +1256,20 @@
const PeerConnectionFactory::Options& callee_options) {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, &caller_options, nullptr,
- webrtc::PeerConnectionDependencies(nullptr));
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, &callee_options, nullptr,
+ webrtc::PeerConnectionDependencies(nullptr), nullptr);
+ return caller_ && callee_;
+ }
+
+ bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
+ PeerConnectionInterface::RTCConfiguration default_config;
+ caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
+ "Caller", nullptr, nullptr, &default_config,
+ webrtc::PeerConnectionDependencies(nullptr));
+ callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
+ "Callee", nullptr, nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
return caller_ && callee_;
}
@@ -1227,7 +1283,7 @@
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr, nullptr,
- std::move(dependencies));
+ std::move(dependencies), nullptr);
}
// Once called, SDP blobs and ICE candidates will be automatically signaled
@@ -4354,6 +4410,39 @@
}
#endif // HAVE_SCTP
+TEST_P(PeerConnectionIntegrationTest,
+ IceEventsGeneratedAndLoggedInRtcEventLog) {
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
+ ConnectFakeSignaling();
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 1;
+ caller()->SetOfferAnswerOptions(options);
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
+ ASSERT_NE(nullptr, caller()->event_log_factory());
+ ASSERT_NE(nullptr, callee()->event_log_factory());
+ webrtc::FakeRtcEventLog* caller_event_log =
+ static_cast<webrtc::FakeRtcEventLog*>(
+ caller()->event_log_factory()->last_log_created());
+ webrtc::FakeRtcEventLog* callee_event_log =
+ static_cast<webrtc::FakeRtcEventLog*>(
+ callee()->event_log_factory()->last_log_created());
+ ASSERT_NE(nullptr, caller_event_log);
+ ASSERT_NE(nullptr, callee_event_log);
+ int caller_ice_config_count = caller_event_log->GetEventCount(
+ webrtc::RtcEvent::Type::IceCandidatePairConfig);
+ int caller_ice_event_count = caller_event_log->GetEventCount(
+ webrtc::RtcEvent::Type::IceCandidatePairEvent);
+ int callee_ice_config_count = callee_event_log->GetEventCount(
+ webrtc::RtcEvent::Type::IceCandidatePairConfig);
+ int callee_ice_event_count = callee_event_log->GetEventCount(
+ webrtc::RtcEvent::Type::IceCandidatePairEvent);
+ EXPECT_LT(0, caller_ice_config_count);
+ EXPECT_LT(0, caller_ice_event_count);
+ EXPECT_LT(0, callee_ice_config_count);
+ EXPECT_LT(0, callee_ice_event_count);
+}
+
INSTANTIATE_TEST_CASE_P(PeerConnectionIntegrationTest,
PeerConnectionIntegrationTest,
Values(SdpSemantics::kPlanB,
