Add audio support in WebrtcConnectionToHost, audio unittest

remoting/protocol/BUILD.gn

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 import("//build/config/features.gni")
 
 static_library("protocol") {
   sources = [
     "audio_decode_scheduler.cc",
     "audio_decode_scheduler.h",
@@ skipping 188 matching lines @@
       "chromium_port_allocator_factory.cc",
       "chromium_port_allocator_factory.h",
       "chromium_socket_factory.cc",
       "chromium_socket_factory.h",
       "ice_connection_to_client.cc",
       "ice_connection_to_client.h",
       "video_frame_pump.cc",
       "video_frame_pump.h",
       "webrtc_audio_module.cc",
       "webrtc_audio_module.h",
+      "webrtc_audio_sink_adapter.cc",
+      "webrtc_audio_sink_adapter.h",
       "webrtc_audio_source_adapter.cc",
       "webrtc_audio_source_adapter.h",
       "webrtc_audio_stream.cc",
       "webrtc_audio_stream.h",
       "webrtc_connection_to_client.cc",
       "webrtc_connection_to_client.h",
       "webrtc_connection_to_host.cc",
       "webrtc_connection_to_host.h",
       "webrtc_data_stream_adapter.cc",
       "webrtc_data_stream_adapter.h",
@@ skipping 105 matching lines @@
     "webrtc_transport_unittest.cc",
   ]
 
   deps = [
     ":test_support",
     "//net:test_support",
     "//testing/gmock",
     "//testing/gtest",
   ]
 }