Added an empty AudioTransportProxy to AudioState.

webrtc/call/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("../build/webrtc.gni")
 
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "flexfec_receive_stream.cc",
     "flexfec_receive_stream.h",
   ]
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
+  public_deps = [
+    "../api:call_api",
+  ]
+
   deps = [
     "..:webrtc_common",
-    "../api:call_api",
     "../audio",
     "../base:rtc_task_queue",
     "../logging:rtc_event_log_impl",
     "../modules/congestion_controller",
     "../modules/rtp_rtcp",
     "../system_wrappers",
     "../video",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
       "packet_injection_tests.cc",
     ]
     deps = [
       ":call",
+      "../modules/audio_device:mock_audio_device",
       "//testing/gmock",
       "//testing/gtest",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }