ThreadScope concept to check variable accessed on correct thread

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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.
  */
 
 #ifndef WEBRTC_PC_CHANNEL_H_
 #define WEBRTC_PC_CHANNEL_H_
 
 #include <map>
 #include <memory>
 #include <set>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread_scope.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
 #include "webrtc/pc/bundlefilter.h"
@@ skipping 130 matching lines @@
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
   // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
   // true). Gets the transport channels from |transport_controller_|.
   bool SetTransport_n(const std::string& transport_name);
 
-  void SetTransportChannel_n(TransportChannel* transport);
+  void SetTransportChannel_n(TransportChannel* transport)
+      RTC_MUST_RUN_ON(network_thread_);
   void SetRtcpTransportChannel_n(TransportChannel* transport,
                                  bool update_writablity);
 
   bool was_ever_writable() const { return was_ever_writable_; }
   void set_local_content_direction(MediaContentDirection direction) {
     local_content_direction_ = direction;
   }
   void set_remote_content_direction(MediaContentDirection direction) {
     remote_content_direction_ = direction;
   }
@@ skipping 135 matching lines @@
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
   // Transport related members that should be accessed from network thread.
   TransportController* const transport_controller_;
-  std::string transport_name_;
+  std::string transport_name_ RTC_MUST_BE_ACCESSED_FROM(network_thread_);
   bool rtcp_transport_enabled_;
   TransportChannel* transport_channel_;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   TransportChannel* rtcp_transport_channel_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
   bool rtp_ready_to_send_;
   bool rtcp_ready_to_send_;
@@ skipping 313 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_