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 22 matching lines @@
 
 // Note P postfix of a function indicates that it should only be called by the
 // processing thread.
 
 #ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
 #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
 
 #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
     : public webrtc::AudioDeviceModule,
       public rtc::MessageHandler {
  public:
   typedef uint16 Sample;
 
   // The value for the following constants have been derived by running VoE
   // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
   static const int kNumberSamples = 440;
   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
   // pulled frame was generated/pushed from a FakeAudioCaptureModule.
   int frames_received() const;
 
   // Following functions are inherited from webrtc::AudioDeviceModule.
   // Only functions called by PeerConnection are implemented, the rest do
   // nothing and return success. If a function is not expected to be called by
   // PeerConnection an assertion is triggered if it is in fact called.
@@ skipping 120 matching lines @@
 
   // The following function is inherited from rtc::MessageHandler.
   void OnMessage(rtc::Message* msg) override;
 
  protected:
   // The constructor is protected because the class needs to be created as a
   // reference counted object (for memory managment reasons). It could be
   // 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();
 
  private:
   // Initializes the state of the FakeAudioCaptureModule. This API is called on
   // creation by the Create() API.
   bool Initialize();
   // SetBuffer() sets all samples in send_buffer_ to |value|.
   void SetSendBuffer(int value);
@@ skipping 12 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.
 
   bool play_is_initialized_; // True when the instance is ready to pull audio.
   bool rec_is_initialized_; // True when the instance is ready to push audio.
 
   // Input to and output from RecordedDataIsAvailable(..) makes it possible to
   // modify the current mic level. The implementation does not care about the
   // mic level so it just feeds back what it receives.
   uint32_t current_mic_level_;
 
   // next_frame_time_ is updated in a non-drifting manner to indicate the next
   // wall clock time the next frame should be generated and received. started_
   // ensures that next_frame_time_ can be initialized properly on first call.
   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];
   // Buffer for samples to send to the webrtc::AudioTransport.
   char send_buffer_[kNumberSamples * kNumberBytesPerSample];
 
   // Counter of frames received that have samples of high enough amplitude to
   // indicate that the frames are not faked somewhere in the audio pipeline
   // (e.g. by a jitter buffer).
   int frames_received_;
 
   // 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_