Use suffixed {uint,int}{8,16,32,64}_t types.

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 65a03c84045fb2c6a3e90be0104aefe3feb76c9f..4284b9ed51b89778e33aa0f2893be4a98fff9603 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h
@@ -53,7 +53,7 @@ class FakeAudioCaptureModule
     : public webrtc::AudioDeviceModule,
       public rtc::MessageHandler {
  public:
-  typedef uint16 Sample;
+  typedef uint16_t 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.
@@ -242,7 +242,7 @@ class FakeAudioCaptureModule
 
   // The time in milliseconds when Process() was last called or 0 if no call
   // has been made.
-  uint32 last_process_time_ms_;
+  uint32_t last_process_time_ms_;
 
   // Callback for playout and recording.
   webrtc::AudioTransport* audio_callback_;
@@ -262,7 +262,7 @@ class FakeAudioCaptureModule
   // 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_;
+  uint32_t next_frame_time_;
 
   rtc::scoped_ptr<rtc::Thread> process_thread_;