Create the SrtpTransportInterface.

webrtc/api/ortc/ortcrtpsenderinterface.h

 /*
  *  Copyright 2017 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.
  */
 
 // This file contains interfaces for RtpSenders:
 // http://publications.ortc.org/2016/20161202/#rtcrtpsender*
 //
 // However, underneath the RtpSender is an RtpTransport, rather than a
 // DtlsTransport. This is to allow different types of RTP transports (besides
 // DTLS-SRTP) to be used.
 
 #ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
 #define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
 
+#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediatypes.h"
-#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/ortc/rtptransportinterface.h"
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/api/rtpparameters.h"
 
 namespace webrtc {
 
 // Note: Since sender capabilities may depend on how the OrtcFactory was
 // created, instead of a static "GetCapabilities" method on this interface,
 // there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
 class OrtcRtpSenderInterface {
@@ skipping 35 matching lines @@
 
   // Audio or video sender?
   virtual cricket::MediaType GetKind() const = 0;
 
   // TODO(deadbeef): SSRC conflict signal.
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_