Clean up PlatformThread.

webrtc/video_engine/vie_channel.cc

 /*
  *  Copyright (c) 2012 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 92 matching lines @@
       vie_sync_(vcm_),
       stats_observer_(new ChannelStatsObserver(this)),
       receive_stats_callback_(nullptr),
       incoming_video_stream_(nullptr),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
       transport_feedback_observer_(transport_feedback_observer),
+      decode_thread_(ChannelDecodeThreadFunction, this, "DecodingThread"),
       nack_history_size_sender_(kMinSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       report_block_stats_sender_(new ReportBlockStats()),
       time_of_first_rtt_ms_(-1),
       rtt_sum_ms_(0),
       last_rtt_ms_(0),
       num_rtts_(0),
       rtp_rtcp_modules_(
           CreateRtpRtcpModules(!sender,
@@ skipping 55 matching lines @@
       vie_receiver_.GetReceiveStatistics());
   module_process_thread_->DeRegisterModule(vcm_);
   module_process_thread_->DeRegisterModule(&vie_sync_);
   send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
   for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
     packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
-  if (decode_thread_) {
-    StopDecodeThread();
-  }
+  StopDecodeThread();
   // Release modules.
   VideoCodingModule::Destroy(vcm_);
 }
 
 void ViEChannel::UpdateHistograms() {
   int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
 
   {
     CriticalSectionScoped cs(crit_.get());
     int64_t elapsed_sec = (now - time_of_first_rtt_ms_) / 1000;
@@ skipping 939 matching lines @@
     // Receive statistics and remote bitrate estimator should only be set for
     // the primary (first) module.
     configuration.receive_statistics = null_receive_statistics;
     configuration.remote_bitrate_estimator = nullptr;
   }
   return modules;
 }
 
 void ViEChannel::StartDecodeThread() {
   RTC_DCHECK(!sender_);
+  if (decode_thread_.IsRunning())
+    return;
   // Start the decode thread
-  if (decode_thread_)
-    return;
-  decode_thread_ = PlatformThread::CreateThread(ChannelDecodeThreadFunction,
-                                                this, "DecodingThread");
-  decode_thread_->Start();
-  decode_thread_->SetPriority(kHighestPriority);
+  decode_thread_.Start();
+  decode_thread_.SetPriority(rtc::kHighestPriority);
 }
 
 void ViEChannel::StopDecodeThread() {
-  if (!decode_thread_)
-    return;
-
   vcm_->TriggerDecoderShutdown();
 
-  decode_thread_->Stop();
-  decode_thread_.reset();
+  decode_thread_.Stop();
 }
 
 int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
                                     VoEVideoSync* ve_sync_interface) {
   return vie_sync_.ConfigureSync(ve_channel_id, ve_sync_interface,
                                  rtp_rtcp_modules_[0],
                                  vie_receiver_.GetRtpReceiver());
 }
 
 int32_t ViEChannel::VoiceChannel() {
@@ skipping 42 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc