Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/media/sctp/sctptransportinternal.h

 /*
  *  Copyright (c) 2016 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_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
 #define WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
 
 // TODO(deadbeef): Move SCTP code out of media/, and make it not depend on
 // anything in media/.
 
 #include <memory>  // for unique_ptr
 #include <string>
 #include <vector>
 
-#include "webrtc/base/copyonwritebuffer.h"
-#include "webrtc/base/thread.h"
+#include "webrtc/rtc_base/copyonwritebuffer.h"
+#include "webrtc/rtc_base/thread.h"
 // For SendDataParams/ReceiveDataParams.
 // TODO(deadbeef): Use something else for SCTP. It's confusing that we use an
 // SSRC field for SID.
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/packettransportinternal.h"
 
 namespace cricket {
 
 // The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs)
 // are 0-based, the highest usable SID is 1023.
@@ skipping 95 matching lines @@
   virtual ~SctpTransportInternalFactory() {}
 
   // Create an SCTP transport using |channel| for the underlying transport.
   virtual std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
       rtc::PacketTransportInternal* channel) = 0;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_