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

webrtc/base/sslstreamadapter.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 132 matching lines @@
   //                        part of the RFC defining each exporter
   //                        usage (IN)
   // context/context_len -- a context to bind to for this connection;
   //                        optional, can be NULL, 0 (IN)
   // use_context         -- whether to use the context value
   //                        (needed to distinguish no context from
   //                        zero-length ones).
   // result              -- where to put the computed value
   // result_len          -- the length of the computed value
   virtual bool ExportKeyingMaterial(const std::string& label,
-                                    const uint8* context,
+                                    const uint8_t* context,
                                     size_t context_len,
                                     bool use_context,
-                                    uint8* result,
+                                    uint8_t* result,
                                     size_t result_len);
 
   // DTLS-SRTP interface
   virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers);
   virtual bool GetDtlsSrtpCipher(std::string* cipher);
 
   // Capabilities testing
   static bool HaveDtls();
   static bool HaveDtlsSrtp();
   static bool HaveExporter();
@@ skipping 12 matching lines @@
 
   // If true (default), the client is required to provide a certificate during
   // handshake. If no certificate is given, handshake fails. This applies to
   // server mode only.
   bool client_auth_enabled_;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_SSLSTREAMADAPTER_H_