In PeerConnectionTestWrapper, put audio input on a separate thread.

talk/app/webrtc/test/fakeaudiocapturemodule.h

Index: talk/app/webrtc/test/fakeaudiocapturemodule.h
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/talk/app/webrtc/test/fakeaudiocapturemodule.h
index 8ff4aa19e7552eea7a0520bf6f4907ce865a081c..be1df9a7bd7e48c249b8011256994d9757bb7cc7 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h
@@ -40,14 +40,13 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 namespace rtc {
-
 class Thread;
-
 }  // namespace rtc
 
 class FakeAudioCaptureModule
@@ -62,10 +61,7 @@ class FakeAudioCaptureModule
   static const int kNumberBytesPerSample = sizeof(Sample);
 
   // Creates a FakeAudioCaptureModule or returns NULL on failure.
-  // |process_thread| is used to push and pull audio frames to and from the
-  // returned instance. Note: ownership of |process_thread| is not handed over.
-  static rtc::scoped_refptr<FakeAudioCaptureModule> Create(
-      rtc::Thread* process_thread);
+  static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
 
   // Returns the number of frames that have been successfully pulled by the
   // instance. Note that correctly detecting success can only be done if the
@@ -206,7 +202,7 @@ class FakeAudioCaptureModule
   // exposed in which case the burden of proper instantiation would be put on
   // the creator of a FakeAudioCaptureModule instance. To create an instance of
   // this class use the Create(..) API.
-  explicit FakeAudioCaptureModule(rtc::Thread* process_thread);
+  explicit FakeAudioCaptureModule();
   // The destructor is protected because it is reference counted and should not
   // be deleted directly.
   virtual ~FakeAudioCaptureModule();
@@ -239,8 +235,6 @@ class FakeAudioCaptureModule
   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.
@@ -266,8 +260,7 @@ class FakeAudioCaptureModule
   bool started_;
   uint32 next_frame_time_;
 
-  // User provided thread context.
-  rtc::Thread* process_thread_;
+  rtc::scoped_ptr<rtc::Thread> process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
   char rec_buffer_[kNumberSamples * kNumberBytesPerSample];