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

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 35 matching lines @@
 #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;
+  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.
   static const size_t kNumberSamples = 440;
   static const size_t kNumberBytesPerSample = sizeof(Sample);
 
   // Creates a FakeAudioCaptureModule or returns NULL on failure.
   static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
 
   // Returns the number of frames that have been successfully pulled by the
@@ skipping 168 matching lines @@
   // 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();
 
   // 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_;
 
   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_;
+  uint32_t next_frame_time_;
 
   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_