Removing MessageLoop::QuitTask() from content/

content/test/webrtc_audio_device_test.cc

 // Copyright (c) 2011 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 "content/test/webrtc_audio_device_test.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/file_util.h"
@@ skipping 72 matching lines @@
  public:
   WebRTCMockResourceContext() {}
   virtual ~WebRTCMockResourceContext() {}
   virtual void EnsureInitialized() const OVERRIDE {}
 
  private:
   DISALLOW_COPY_AND_ASSIGN(WebRTCMockResourceContext);
 };
 
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
-  loop_or_proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
+  loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
 }  // end namespace
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
     : render_thread_(NULL), event_(false, false), audio_util_callback_(NULL) {
 }
 
 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
 
@@ skipping 180 matching lines @@
 WebRTCTransportImpl::~WebRTCTransportImpl() {}
 
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }