Modify PeerConnection for end-to-end QuicDataChannel usage

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 136 matching lines @@
 };
 
 class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
                                  public SignalingMessageReceiver,
                                  public ObserverInterface {
  public:
   static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
       const std::string& id,
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
+      const PeerConnectionInterface::RTCConfiguration& config,
       rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       bool prefer_constraint_apis) {
     PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
-    if (!client->Init(constraints, options, std::move(dtls_identity_store),
-                      prefer_constraint_apis)) {
+    if (!client->Init(constraints, options, config,
+                      std::move(dtls_identity_store), prefer_constraint_apis)) {
       delete client;
       return nullptr;
     }
     return client;
   }
 
   static PeerConnectionTestClient* CreateClient(
       const std::string& id,
       const MediaConstraintsInterface* constraints,
-      const PeerConnectionFactory::Options* options) {
+      const PeerConnectionFactory::Options* options,
+      const PeerConnectionInterface::RTCConfiguration& config) {
     rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
-
     return CreateClientWithDtlsIdentityStore(
-        id, constraints, options, std::move(dtls_identity_store), true);
+        id, constraints, options, config, std::move(dtls_identity_store), true);
   }
 
   static PeerConnectionTestClient* CreateClientPreferNoConstraints(
       const std::string& id,
-      const PeerConnectionFactory::Options* options) {
+      const PeerConnectionFactory::Options* options,
+      const PeerConnectionInterface::RTCConfiguration& config) {
     rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
-
     return CreateClientWithDtlsIdentityStore(
-        id, nullptr, options, std::move(dtls_identity_store), false);
+        id, nullptr, options, config, std::move(dtls_identity_store), false);
   }
 
   ~PeerConnectionTestClient() {
   }
 
   void Negotiate() { Negotiate(true, true); }
 
   void Negotiate(bool audio, bool video) {
     rtc::scoped_ptr<SessionDescriptionInterface> offer;
     ASSERT_TRUE(DoCreateOffer(&offer));
@@ skipping 220 matching lines @@
            offer_answer_options_.offer_to_receive_video ==
                PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
   }
 
   void OnDataChannel(DataChannelInterface* data_channel) override {
     LOG(INFO) << id_ << "OnDataChannel";
     data_channel_ = data_channel;
     data_observer_.reset(new MockDataChannelObserver(data_channel));
   }
 
-  void CreateDataChannel() {
-    data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, nullptr);
+  void CreateDataChannel() { CreateDataChannel(nullptr); }
+
+  void CreateDataChannel(const webrtc::DataChannelInit* init) {
+    data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init);
     ASSERT_TRUE(data_channel_.get() != nullptr);
     data_observer_.reset(new MockDataChannelObserver(data_channel_));
   }
 
   rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
       const std::string& stream_label) {
     FakeConstraints constraints;
     // Disable highpass filter so that we can get all the test audio frames.
     constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
@@ skipping 351 matching lines @@
    private:
     bool completed_;
     std::vector<std::string> tones_;
   };
 
   explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
 
   bool Init(
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
+      const PeerConnectionInterface::RTCConfiguration& config,
       rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       bool prefer_constraint_apis) {
     EXPECT_TRUE(!peer_connection_);
     EXPECT_TRUE(!peer_connection_factory_);
     if (!prefer_constraint_apis) {
       EXPECT_TRUE(!constraints);
     }
     prefer_constraint_apis_ = prefer_constraint_apis;
 
     rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
         new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
 
     if (fake_audio_capture_module_ == nullptr) {
       return false;
     }
     fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
     fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
     peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
         rtc::Thread::Current(), rtc::Thread::Current(),
         fake_audio_capture_module_, fake_video_encoder_factory_,
         fake_video_decoder_factory_);
     if (!peer_connection_factory_) {
       return false;
     }
     if (options) {
       peer_connection_factory_->SetOptions(*options);
     }
-    peer_connection_ = CreatePeerConnection(
-        std::move(port_allocator), constraints, std::move(dtls_identity_store));
+    peer_connection_ =
+        CreatePeerConnection(std::move(port_allocator), constraints, config,
+                             std::move(dtls_identity_store));
     return peer_connection_.get() != nullptr;
   }
 
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
       rtc::scoped_ptr<cricket::PortAllocator> port_allocator,
       const MediaConstraintsInterface* constraints,
+      const PeerConnectionInterface::RTCConfiguration& config,
       rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
-    // CreatePeerConnection with RTCConfiguration.
-    webrtc::PeerConnectionInterface::RTCConfiguration config;
-    webrtc::PeerConnectionInterface::IceServer ice_server;
-    ice_server.uri = "stun:stun.l.google.com:19302";
-    config.servers.push_back(ice_server);
-
     return peer_connection_factory_->CreatePeerConnection(
         config, constraints, std::move(port_allocator),
         std::move(dtls_identity_store), this);
   }
 
   void HandleIncomingOffer(const std::string& msg) {
     LOG(INFO) << id_ << "HandleIncomingOffer ";
     if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
       // If we are not sending any streams ourselves it is time to add some.
       AddMediaStream(true, true);
@@ skipping 157 matching lines @@
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
   rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
 };
 
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
       : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
-        ss_scope_(ss_.get()) {}
+        ss_scope_(ss_.get()) {
+    webrtc::PeerConnectionInterface::IceServer ice_server;
+    ice_server.uri = "stun:stun.l.google.com:19302";
+    config_.servers.push_back(ice_server);
+  }
 
   bool SessionActive() {
     return initiating_client_->SessionActive() &&
            receiving_client_->SessionActive();
   }
 
   // Return true if the number of frames provided have been received
   // on the video and audio tracks provided.
   bool FramesHaveArrived(int audio_frames_to_receive,
                          int video_frames_to_receive) {
@@ skipping 87 matching lines @@
   bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); }
 
   bool CreateTestClients(MediaConstraintsInterface* init_constraints,
                          MediaConstraintsInterface* recv_constraints) {
     return CreateTestClients(init_constraints, nullptr, recv_constraints,
                              nullptr);
   }
 
   bool CreateTestClientsThatPreferNoConstraints() {
     initiating_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints("Caller: ",
-                                                                  nullptr));
+        PeerConnectionTestClient::CreateClientPreferNoConstraints(
+            "Caller: ", nullptr, config_));
     receiving_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints("Callee: ",
-                                                                  nullptr));
+        PeerConnectionTestClient::CreateClientPreferNoConstraints(
+            "Callee: ", nullptr, config_));
     if (!initiating_client_ || !receiving_client_) {
       return false;
     }
     // Remember the choice for possible later resets of the clients.
     prefer_constraint_apis_ = false;
     SetSignalingReceivers();
     return true;
   }
 
   void SetSignalingReceivers() {
     initiating_client_->set_signaling_message_receiver(receiving_client_.get());
     receiving_client_->set_signaling_message_receiver(initiating_client_.get());
   }
 
   bool CreateTestClients(MediaConstraintsInterface* init_constraints,
                          PeerConnectionFactory::Options* init_options,
                          MediaConstraintsInterface* recv_constraints,
                          PeerConnectionFactory::Options* recv_options) {
     initiating_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Caller: ", init_constraints, init_options));
+        "Caller: ", init_constraints, init_options, config_));
     receiving_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Callee: ", recv_constraints, recv_options));
+        "Callee: ", recv_constraints, recv_options, config_));
     if (!initiating_client_ || !receiving_client_) {
       return false;
     }
     SetSignalingReceivers();
     return true;
   }
 
   void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
                            const webrtc::FakeConstraints& recv_constraints) {
     initiating_client_->SetVideoConstraints(init_constraints);
@@ skipping 79 matching lines @@
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
 
     rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
     dtls_identity_store->use_alternate_key();
 
     // Make sure the new client is using a different certificate.
     return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
-        "New Peer: ", &setup_constraints, nullptr,
+        "New Peer: ", &setup_constraints, nullptr, config_,
         std::move(dtls_identity_store), prefer_constraint_apis_);
   }
 
   void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
     // Messages may get lost on the unreliable DataChannel, so we send multiple
     // times to avoid test flakiness.
     static const size_t kSendAttempts = 5;
 
     for (size_t i = 0; i < kSendAttempts; ++i) {
       dc->Send(DataBuffer(data));
@@ skipping 19 matching lines @@
 
   // Set the |receiving_client_| to the |client| passed in and return the
   // original |receiving_client_|.
   PeerConnectionTestClient* set_receiving_client(
       PeerConnectionTestClient* client) {
     PeerConnectionTestClient* old = receiving_client_.release();
     receiving_client_.reset(client);
     return old;
   }
 
+  webrtc::PeerConnectionInterface::RTCConfiguration* config() {
+    return &config_;
+  }
+
  private:
   rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
   rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
   rtc::scoped_ptr<PeerConnectionTestClient> initiating_client_;
   rtc::scoped_ptr<PeerConnectionTestClient> receiving_client_;
   bool prefer_constraint_apis_ = true;
+  webrtc::PeerConnectionInterface::RTCConfiguration config_;
 };
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 #if !defined(THREAD_SANITIZER)
 
 // This test sets up a Jsep call between two parties and test Dtmf.
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
 TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
@@ skipping 451 matching lines @@
   receiving_client()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
                  kMaxWaitMs);
 
   receiving_client()->data_channel()->Close();
   EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
                    kMaxWaitMs);
   EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
 }
 
+#ifdef HAVE_QUIC
+// This test sets up a call between two parties using QUIC instead of DTLS for
+// audio and video, and a QUIC data channel.
+TEST_F(P2PTestConductor, LocalP2PTestQuicDataChannel) {
+  config()->enable_quic = true;
+  ASSERT_TRUE(CreateTestClients());
+  webrtc::DataChannelInit init;
+  init.ordered = false;
+  init.reliable = true;
+  init.id = 1;
+  initializing_client()->CreateDataChannel(&init);
+  receiving_client()->CreateDataChannel(&init);
+  LocalP2PTest();
+  ASSERT_NE(nullptr, initializing_client()->data_channel());
+  ASSERT_NE(nullptr, receiving_client()->data_channel());
+  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
+                   kMaxWaitMs);
+  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
+
+  std::string data = "hello world";
+
+  initializing_client()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
+                 kMaxWaitMs);
+
+  receiving_client()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
+                 kMaxWaitMs);
+}
+#endif  // HAVE_QUIC
+
 // This test sets up a call between two parties and creates a data channel.
 // The test tests that received data is buffered unless an observer has been
 // registered.
 // Rtp data channels can receive data before the underlying
 // transport has detected that a channel is writable and thus data can be
 // received before the data channel state changes to open. That is hard to test
 // but the same buffering is used in that case.
 TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
@@ skipping 66 matching lines @@
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.SetMandatory(
       MediaConstraintsInterface::kEnableDtlsSrtp, true);
   ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
   initializing_client()->CreateDataChannel();
   initializing_client()->Negotiate(false, false);
 }
 #endif
 
+// Tests negotiation of QUIC data channels is completed without error.
+#ifdef HAVE_QUIC
+TEST_F(P2PTestConductor, CreateOfferWithQuicDataChannel) {
+  config()->enable_quic = true;
+  FakeConstraints constraints;
+  constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
+  ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
+  webrtc::DataChannelInit init;
+  init.ordered = false;
+  init.reliable = true;
+  init.id = 1;
+  initializing_client()->CreateDataChannel(&init);
+  initializing_client()->Negotiate(false, false);
+}
+#endif  // HAVE_QUIC
+
 // This test sets up a call between two parties with audio, and video.
 // During the call, the initializing side restart ice and the test verifies that
 // new ice candidates are generated and audio and video still can flow.
 TEST_F(P2PTestConductor, IceRestart) {
   ASSERT_TRUE(CreateTestClients());
 
   // Negotiate and wait for ice completion and make sure audio and video plays.
   LocalP2PTest();
 
   // Create a SDP string of the first audio candidate for both clients.
@@ skipping 285 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace