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

webrtc/base/socket.h

 /*
  *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 140 matching lines @@
   virtual void SetError(int error) = 0;
   inline bool IsBlocking() const { return IsBlockingError(GetError()); }
 
   enum ConnState {
     CS_CLOSED,
     CS_CONNECTING,
     CS_CONNECTED
   };
   virtual ConnState GetState() const = 0;
 
-  // Fills in the given uint16 with the current estimate of the MTU along the
+  // Fills in the given uint16_t with the current estimate of the MTU along the
   // path to the address to which this socket is connected. NOTE: This method
   // can block for up to 10 seconds on Windows.
-  virtual int EstimateMTU(uint16* mtu) = 0;
+  virtual int EstimateMTU(uint16_t* mtu) = 0;
 
   enum Option {
     OPT_DONTFRAGMENT,
     OPT_RCVBUF,      // receive buffer size
     OPT_SNDBUF,      // send buffer size
     OPT_NODELAY,     // whether Nagle algorithm is enabled
     OPT_IPV6_V6ONLY, // Whether the socket is IPv6 only.
     OPT_DSCP,        // DSCP code
     OPT_RTP_SENDTIME_EXTN_ID,  // This is a non-traditional socket option param.
                                // This is specific to libjingle and will be used
                                // if SendTime option is needed at socket level.
   };
   virtual int GetOption(Option opt, int* value) = 0;
   virtual int SetOption(Option opt, int value) = 0;
 
  protected:
   Socket() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(Socket);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_SOCKET_H__