Create the SrtpTransportInterface.

webrtc/api/BUILD.gn

 # Copyright (c) 2015 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.
 
 import("../webrtc.gni")
 if (is_android) {
@@ skipping 76 matching lines @@
 
 rtc_source_set("ortc_api") {
   check_includes = false  # TODO(deadbeef): Remove (bugs.webrtc.org/6828)
   sources = [
     "ortc/ortcfactoryinterface.h",
     "ortc/ortcrtpreceiverinterface.h",
     "ortc/ortcrtpsenderinterface.h",
     "ortc/packettransportinterface.h",
     "ortc/rtptransportcontrollerinterface.h",
     "ortc/rtptransportinterface.h",
+    "ortc/srtptransportinterface.h",
     "ortc/udptransportinterface.h",
   ]
 
   # For mediastreaminterface.h, etc.
   # TODO(deadbeef): Create a separate target for the common things ORTC and
   # PeerConnection code shares, so that ortc_api can depend on that instead of
   # libjingle_peerconnection_api.
   public_deps = [
     ":libjingle_peerconnection_api",
   ]
@@ skipping 123 matching lines @@
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
       ":libjingle_peerconnection_api",
       "//webrtc/test:test_support",
     ]
   }
 }