Move ThreadWrapper to ProcessThread in base.

webrtc/test/fake_audio_device.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/test/fake_audio_device.h"
 
 #include <algorithm>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/media_file/media_file_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc {
 namespace test {
 
 FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
     : audio_callback_(NULL),
       capturing_(false),
       captured_audio_(),
       playout_buffer_(),
       last_playout_ms_(-1),
@@ skipping 15 matching lines @@
     thread_->Stop();
 }
 
 int32_t FakeAudioDevice::Init() {
   rtc::CritScope cs(&lock_);
   if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
     return -1;
 
   if (!tick_->StartTimer(true, 10))
     return -1;
-  thread_ = ThreadWrapper::CreateThread(FakeAudioDevice::Run, this,
-                                        "FakeAudioDevice");
+  thread_ = PlatformThread::CreateThread(FakeAudioDevice::Run, this,
+                                         "FakeAudioDevice");
   if (thread_.get() == NULL)
     return -1;
   if (!thread_->Start()) {
     thread_.reset();
     return -1;
   }
   thread_->SetPriority(webrtc::kHighPriority);
   return 0;
 }
 
@@ skipping 74 matching lines @@
   rtc::CritScope cs(&lock_);
   capturing_ = true;
 }
 
 void FakeAudioDevice::Stop() {
   rtc::CritScope cs(&lock_);
   capturing_ = false;
 }
 }  // namespace test
 }  // namespace webrtc