Linux/ChromeOS Chromium style checker cleanup, content/ edition.

content/renderer/media/rtc_peer_connection_handler_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <string>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/utf_string_conversions.h"
 #include "base/values.h"
 #include "content/renderer/media/media_stream_extra_data.h"
@@ skipping 173 matching lines @@
   MockPeerConnectionImpl* native_peer_connection() {
     return static_cast<MockPeerConnectionImpl*>(native_peer_connection_.get());
   }
 };
 
 class RTCPeerConnectionHandlerTest : public ::testing::Test {
  public:
   RTCPeerConnectionHandlerTest() : mock_peer_connection_(NULL) {
   }
 
-  void SetUp() {
+  virtual void SetUp() {
     mock_client_.reset(new MockWebRTCPeerConnectionHandlerClient());
     mock_dependency_factory_.reset(new MockMediaStreamDependencyFactory());
     mock_dependency_factory_->EnsurePeerConnectionFactory();
     pc_handler_.reset(
         new RTCPeerConnectionHandlerUnderTest(mock_client_.get(),
                                               mock_dependency_factory_.get()));
     mock_tracker_.reset(new NiceMock<MockPeerConnectionTracker>());
     WebKit::WebRTCConfiguration config;
     WebKit::WebMediaConstraints constraints;
     EXPECT_TRUE(pc_handler_->InitializeForTest(config, constraints,
@@ skipping 375 matching lines @@
   EXPECT_FALSE(pc_handler_->createDTMFSender(tracks[0]));
 
   local_stream.audioSources(tracks);
   ASSERT_LE(1ul, tracks.size());
   scoped_ptr<WebKit::WebRTCDTMFSenderHandler> sender(
       pc_handler_->createDTMFSender(tracks[0]));
   EXPECT_TRUE(sender.get());
 }
 
 }  // namespace content