Removing MessageLoop::QuitTask() from content/

content/renderer/media/webrtc_audio_device_unittest.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 "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager.h"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using testing::_;
 using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace {
 
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
-  loop_or_proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
+  loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
 class AudioUtil : public AudioUtilInterface {
  public:
   AudioUtil() {}
 
   virtual double GetAudioHardwareSampleRate() OVERRIDE {
     return media::GetAudioHardwareSampleRate();
   }
   virtual double GetAudioInputHardwareSampleRate() OVERRIDE {
@@ skipping 337 matching lines @@
   ScopedWebRTCPtr<webrtc::VoEFile> file(engine.get());
   int duration = 0;
   EXPECT_EQ(0, file->GetFileDuration(file_path.c_str(), duration,
                                      webrtc::kFileFormatPcm16kHzFile));
   EXPECT_NE(0, duration);
 
   EXPECT_EQ(0, file->StartPlayingFileLocally(ch, file_path.c_str(), false,
                                              webrtc::kFileFormatPcm16kHzFile));
 
   message_loop_.PostDelayedTask(FROM_HERE,
-                                new MessageLoop::QuitTask(),
+                                MessageLoop::QuitClosure(),
                                 TestTimeouts::action_timeout_ms());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->Terminate());
 }
 
 // Uses WebRtcAudioDeviceImpl to play out recorded audio in loopback.
 // An external transport implementation is utilized to feed back RTP packets
 // which are recorded, encoded, packetized into RTP packets and finally
 // "transmitted". The RTP packets are then fed back into the VoiceEngine
@@ skipping 38 matching lines @@
 
   ScopedWebRTCPtr<webrtc::VoENetwork> network(engine.get());
   scoped_ptr<WebRTCTransportImpl> transport(
       new WebRTCTransportImpl(network.get()));
   EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
   EXPECT_EQ(0, base->StartPlayout(ch));
   EXPECT_EQ(0, base->StartSend(ch));
 
   LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
-                                new MessageLoop::QuitTask(),
+                                MessageLoop::QuitClosure(),
                                 TestTimeouts::action_timeout_ms());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }