Clean up PlatformThread.

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.
  */
 
@@ skipping 12 matching lines @@
 namespace test {
 
 FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
     : audio_callback_(NULL),
       capturing_(false),
       captured_audio_(),
       playout_buffer_(),
       last_playout_ms_(-1),
       clock_(clock),
       tick_(EventTimerWrapper::Create()),
+      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
       file_utility_(new ModuleFileUtility(0)),
       input_stream_(FileWrapper::Create()) {
   memset(captured_audio_, 0, sizeof(captured_audio_));
   memset(playout_buffer_, 0, sizeof(playout_buffer_));
   // Open audio input file as read-only and looping.
   EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
       << filename;
 }
 
 FakeAudioDevice::~FakeAudioDevice() {
   Stop();
 
-  if (thread_.get() != NULL)
-    thread_->Stop();
+  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_ = PlatformThread::CreateThread(FakeAudioDevice::Run, this,
-                                         "FakeAudioDevice");
-  if (thread_.get() == NULL)
-    return -1;
-  if (!thread_->Start()) {
-    thread_.reset();
-    return -1;
-  }
-  thread_->SetPriority(webrtc::kHighPriority);
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
   return 0;
 }
 
 int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
   rtc::CritScope cs(&lock_);
   audio_callback_ = callback;
   return 0;
 }
 
 bool FakeAudioDevice::Playing() const {
@@ skipping 67 matching lines @@
   rtc::CritScope cs(&lock_);
   capturing_ = true;
 }
 
 void FakeAudioDevice::Stop() {
   rtc::CritScope cs(&lock_);
   capturing_ = false;
 }
 }  // namespace test
 }  // namespace webrtc