Moved transport.h from webrtc/ to webrtc/api, created build target and updated dependencies.

webrtc/media/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("//build/config/linux/pkg_config.gni")
 import("../build/webrtc.gni")
@@ skipping 208 matching lines @@
     sources += [
       "devices/gtkvideorenderer.cc",
       "devices/gtkvideorenderer.h",
     ]
     public_configs += [ ":gtk-lib" ]
   }
   deps += [
     ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
+    "../api:transport_api",
     "../base:rtc_base_approved",
     "../call",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/video_coding",
     "../system_wrappers",
     "../voice_engine",
   ]
 }
 
 if (rtc_include_tests) {
@@ skipping 157 matching lines @@
       # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
       ":rtc_media",
       ":rtc_unittest_main",
       "../audio",
       "../base:rtc_base_tests_utils",
       "../modules/audio_device:mock_audio_device",
       "../system_wrappers:metrics_default",
     ]
   }
 }