| Index: webrtc/pc/peerconnectioninterface_unittest.cc
|
| diff --git a/webrtc/pc/peerconnectioninterface_unittest.cc b/webrtc/pc/peerconnectioninterface_unittest.cc
|
| index de62104acf25bdb16bc9f8947dda4e7c65b61fef..d0e42621cffdeab915e6e4e9fbef0821dcd5b749 100644
|
| --- a/webrtc/pc/peerconnectioninterface_unittest.cc
|
| +++ b/webrtc/pc/peerconnectioninterface_unittest.cc
|
| @@ -22,7 +22,6 @@
|
| #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"
|
| @@ -662,9 +661,7 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
|
| class PeerConnectionInterfaceTest : public testing::Test {
|
| protected:
|
| PeerConnectionInterfaceTest()
|
| - : pss_(new rtc::PhysicalSocketServer),
|
| - vss_(new rtc::VirtualSocketServer(pss_.get())),
|
| - main_(vss_.get()) {
|
| + : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
|
| #ifdef WEBRTC_ANDROID
|
| webrtc::InitializeAndroidObjects();
|
| #endif
|
| @@ -1128,7 +1125,6 @@ class PeerConnectionInterfaceTest : public testing::Test {
|
| 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;
|
|
|