Get tests working on systems that only support IPv6.

webrtc/pc/peerconnectioninterface_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.
  */
 
 #include <memory>
 #include <sstream>
 #include <string>
 #include <utility>
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/sctp/sctptransportinternal.h"
 #include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/mediastream.h"
 #include "webrtc/pc/peerconnection.h"
 #include "webrtc/pc/streamcollection.h"
 #include "webrtc/pc/test/fakertccertificategenerator.h"
 #include "webrtc/pc/test/fakevideotracksource.h"
@@ skipping 613 matching lines @@
         rtc::Thread::Current(), rtc::Thread::Current(), port_allocator,
         redetermine_role_on_ice_restart, rtc::CryptoOptions());
     return transport_controller;
   }
 
   cricket::TransportController* transport_controller;
 };
 
 class PeerConnectionInterfaceTest : public testing::Test {
  protected:
-  PeerConnectionInterfaceTest() {
+  PeerConnectionInterfaceTest()
+      : pss_(new rtc::PhysicalSocketServer),
+        vss_(new rtc::VirtualSocketServer(pss_.get())),
+        main_(vss_.get()) {
 #ifdef WEBRTC_ANDROID
     webrtc::InitializeAndroidObjects();
 #endif
   }
 
   virtual void SetUp() {
     pc_factory_ = webrtc::CreatePeerConnectionFactory(
         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
         nullptr, nullptr, nullptr);
     ASSERT_TRUE(pc_factory_);
@@ skipping 443 matching lines @@
   const std::string& GetFirstAudioStreamCname(
       const SessionDescriptionInterface* desc) {
     const cricket::ContentInfo* audio_content =
         cricket::GetFirstAudioContent(desc->description());
     const cricket::AudioContentDescription* audio_desc =
         static_cast<const cricket::AudioContentDescription*>(
             audio_content->description);
     return audio_desc->streams()[0].cname;
   }
 
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
+  rtc::AutoSocketServerThread main_;
   cricket::FakePortAllocator* port_allocator_ = nullptr;
   FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
   rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_;
   rtc::scoped_refptr<PeerConnectionInterface> pc_;
   MockPeerConnectionObserver observer_;
   rtc::scoped_refptr<StreamCollection> reference_collection_;
 };
 
 // Test that no callbacks on the PeerConnectionObserver are called after the
@@ skipping 2446 matching lines @@
   EXPECT_NE(a, f);
 
   PeerConnectionInterface::RTCConfiguration g;
   g.disable_ipv6 = true;
   EXPECT_NE(a, g);
 
   PeerConnectionInterface::RTCConfiguration h(
       PeerConnectionInterface::RTCConfigurationType::kAggressive);
   EXPECT_NE(a, h);
 }