Add a minimal RtpTransport class for use by BaseChannel.

webrtc/pc/BUILD.gn

 # 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.
 
 import("../webrtc.gni")
 if (is_android) {
@@ skipping 28 matching lines @@
     "currentspeakermonitor.cc",
     "currentspeakermonitor.h",
     "externalhmac.cc",
     "externalhmac.h",
     "mediamonitor.cc",
     "mediamonitor.h",
     "mediasession.cc",
     "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
+    "rtptransport.cc",
+    "rtptransport.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
     "../api:call_api",
     "../base:rtc_base",
     "../media",
   ]
@@ skipping 298 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }