Remove VirtualSocketServer's dependency on PhysicalSocketServer.

webrtc/pc/peerconnection_integrationtest.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 12 matching lines @@
 #include <vector>
 
 #include "webrtc/api/fakemetricsobserver.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/fakenetwork.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
-#include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/engine/fakewebrtcvideoengine.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/portinterface.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/client/basicportallocator.h"
@@ skipping 886 matching lines @@
   friend class PeerConnectionIntegrationTest;
 };
 
 // Tests two PeerConnections connecting to each other end-to-end, using a
 // virtual network, fake A/V capture and fake encoder/decoders. The
 // PeerConnections share the threads/socket servers, but use separate versions
 // of everything else (including "PeerConnectionFactory"s).
 class PeerConnectionIntegrationTest : public testing::Test {
  public:
   PeerConnectionIntegrationTest()
-      : pss_(new rtc::PhysicalSocketServer),
-        ss_(new rtc::VirtualSocketServer(pss_.get())),
+      : ss_(new rtc::VirtualSocketServer()),
         network_thread_(new rtc::Thread(ss_.get())),
         worker_thread_(rtc::Thread::Create()) {
     RTC_CHECK(network_thread_->Start());
     RTC_CHECK(worker_thread_->Start());
   }
 
   ~PeerConnectionIntegrationTest() {
     if (caller_) {
       caller_->set_signaling_message_receiver(nullptr);
     }
@@ skipping 184 matching lines @@
     EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
                    caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
     EXPECT_EQ(
         1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
                                            expected_cipher_suite));
     caller()->pc()->RegisterUMAObserver(nullptr);
   }
 
  private:
   // |ss_| is used by |network_thread_| so it must be destroyed later.
-  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
   std::unique_ptr<rtc::VirtualSocketServer> ss_;
   // |network_thread_| and |worker_thread_| are used by both
   // |caller_| and |callee_| so they must be destroyed
   // later.
   std::unique_ptr<rtc::Thread> network_thread_;
   std::unique_ptr<rtc::Thread> worker_thread_;
   std::unique_ptr<PeerConnectionWrapper> caller_;
   std::unique_ptr<PeerConnectionWrapper> callee_;
 };
 
@@ skipping 1748 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
                                   kMaxWaitForFramesMs);
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)