Rename "PacketTransportInterface" to "PacketTransportInternal".

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"
 // 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/packettransportinterface.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.
 //
 // It's recommended to use the maximum of 65535 in:
 // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
 // However, we use 1024 in order to save memory. usrsctp allocates 104 bytes
 // for each pair of incoming/outgoing streams (on a 64-bit system), so 65535
@@ skipping 13 matching lines @@
 // Abstract SctpTransport interface for use internally (by
 // PeerConnection/WebRtcSession/etc.). Exists to allow mock/fake SctpTransports
 // to be created.
 class SctpTransportInternal {
  public:
   virtual ~SctpTransportInternal() {}
 
   // Changes what underlying DTLS channel is uses. Used when switching which
   // bundled transport the SctpTransport uses.
   // Assumes |channel| is non-null.
-  virtual void SetTransportChannel(rtc::PacketTransportInterface* channel) = 0;
+  virtual void SetTransportChannel(rtc::PacketTransportInternal* channel) = 0;
 
   // When Start is called, connects as soon as possible; this can be called
   // before DTLS completes, in which case the connection will begin when DTLS
   // completes. This method can be called multiple times, though not if either
   // of the ports are changed.
   //
   // |local_sctp_port| and |remote_sctp_port| are passed along the wire and the
   // listener and connector must be using the same port. They are not related
   // to the ports at the IP level. If set to -1, we default to
   // kSctpDefaultPort.
@@ skipping 50 matching lines @@
 
 // Factory class which can be used to allow fake SctpTransports to be injected
 // for testing. Or, theoretically, SctpTransportInternal implementations that
 // use something other than usrsctp.
 class SctpTransportInternalFactory {
  public:
   virtual ~SctpTransportInternalFactory() {}
 
   // Create an SCTP transport using |channel| for the underlying transport.
   virtual std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
-      rtc::PacketTransportInterface* channel) = 0;
+      rtc::PacketTransportInternal* channel) = 0;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_