In PeerConnectionTestWrapper, put audio input on a separate thread.

talk/app/webrtc/test/fakeaudiocapturemodule.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 221 matching lines @@
 
   // Starts the periodic calling of ProcessFrame() in a thread safe way.
   void StartProcessP();
   // Periodcally called function that ensures that frames are pulled and pushed
   // periodically if enabled/started.
   void ProcessFrameP();
   // Pulls frames from the registered webrtc::AudioTransport.
   void ReceiveFrameP();
   // Pushes frames to the registered webrtc::AudioTransport.
   void SendFrameP();
-  // Stops the periodic calling of ProcessFrame() in a thread safe way.
-  void StopProcessP();
 
   // The time in milliseconds when Process() was last called or 0 if no call
   // has been made.
   uint32 last_process_time_ms_;
 
   // Callback for playout and recording.
   webrtc::AudioTransport* audio_callback_;
 
   bool recording_; // True when audio is being pushed from the instance.
   bool playing_; // True when audio is being pulled by the instance.
@@ skipping 27 matching lines @@
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   mutable rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
 #endif  // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_