Split ByteBuffer into writer/reader objects.

webrtc/libjingle/xmpp/xmppsocket.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.
  */
 
 #ifndef WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
 #define WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
 
 #include "webrtc/libjingle/xmpp/asyncsocket.h"
 #include "webrtc/libjingle/xmpp/xmppengine.h"
 #include "webrtc/base/asyncsocket.h"
-#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/sigslot.h"
 
 // The below define selects the SSLStreamAdapter implementation for
 // SSL, as opposed to the SSLAdapter socket adapter.
 // #define USE_SSLSTREAM 
 
 namespace rtc {
   class StreamInterface;
   class SocketAddress;
 };
@@ skipping 27 matching lines @@
   void OnCloseEvent(rtc::AsyncSocket * socket, int error);
 #else  // USE_SSLSTREAM
   void OnEvent(rtc::StreamInterface* stream, int events, int err);
 #endif  // USE_SSLSTREAM
 
   rtc::AsyncSocket * cricket_socket_;
 #ifdef USE_SSLSTREAM
   rtc::StreamInterface *stream_;
 #endif  // USE_SSLSTREAM
   buzz::AsyncSocket::State state_;
-  rtc::ByteBuffer buffer_;
+  rtc::Buffer buffer_;
   buzz::TlsOptions tls_;
 };
 
 }  // namespace buzz
 
 #endif // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_