Ability to specify KeyType (RSA, ECDSA) for SSLIdentity generation in libjingle

talk/app/webrtc/peerconnection_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 22 matching lines @@
 #include <vector>
 
 #include "talk/app/webrtc/dtmfsender.h"
 #include "talk/app/webrtc/fakeportallocatorfactory.h"
 #include "talk/app/webrtc/localaudiosource.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/peerconnectionfactory.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
-#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
+#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
 #include "talk/app/webrtc/videosourceinterface.h"
 #include "talk/media/webrtc/fakewebrtcvideoengine.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
@@ skipping 711 matching lines @@
   }
 
  protected:
   explicit JsepTestClient(const std::string& id)
       : PeerConnectionTestClientBase<JsepMessageReceiver>(id),
         remove_msid_(false),
         remove_bundle_(false),
         remove_sdes_(false) {
   }
 
-  virtual rtc::scoped_refptr<webrtc::PeerConnectionInterface>
-      CreatePeerConnection(webrtc::PortAllocatorFactoryInterface* factory,
-                           const MediaConstraintsInterface* constraints) {
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface>
+      CreatePeerConnection(
+          webrtc::PortAllocatorFactoryInterface* factory,
+          const MediaConstraintsInterface* constraints) override {
     // CreatePeerConnection with IceServers.
     webrtc::PeerConnectionInterface::IceServers ice_servers;
     webrtc::PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:stun.l.google.com:19302";
     ice_servers.push_back(ice_server);
 
-    FakeIdentityService* dtls_service =
-        rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-            new FakeIdentityService() : NULL;
+    dtls_identity_store_.reset(rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
+                               new FakeDtlsIdentityStoreRSA() : nullptr);
     return peer_connection_factory()->CreatePeerConnection(
-        ice_servers, constraints, factory, dtls_service, this);
+        ice_servers, constraints, factory, this, dtls_identity_store_.get());
   }
 
   void HandleIncomingOffer(const std::string& msg) {
     LOG(INFO) << id() << "HandleIncomingOffer ";
     if (NumberOfLocalMediaStreams() == 0) {
       // If we are not sending any streams ourselves it is time to add some.
       AddMediaStream(true, true);
     }
     rtc::scoped_ptr<SessionDescriptionInterface> desc(
          webrtc::CreateSessionDescription("offer", msg, NULL));
@@ skipping 92 matching lines @@
   }
 
  private:
   webrtc::FakeConstraints session_description_constraints_;
   bool remove_msid_;  // True if MSID should be removed in received SDP.
   bool remove_bundle_;  // True if bundle should be removed in received SDP.
   bool remove_sdes_;  // True if a=crypto should be removed in received SDP.
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
   rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
+  rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store_;
 };
 
 template <typename SignalingClass>
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
       : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
         ss_scope_(ss_.get()) {}
 
@@ skipping 665 matching lines @@
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
 TEST_F(JsepPeerConnectionP2PTestClient,
        DISABLED_LocalP2PTestWithVideoDecoderFactory) {
   ASSERT_TRUE(CreateTestClients());
   EnableVideoDecoderFactory();
   LocalP2PTest();
 }
 
 #endif // if !defined(THREAD_SANITIZER)