Add Audio support in Chromoting host when using WebRTC.

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_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",
       "webrtc_dummy_video_capturer.cc",
       "webrtc_dummy_video_capturer.h",
       "webrtc_dummy_video_encoder.cc",
       "webrtc_dummy_video_encoder.h",
@@ skipping 21 matching lines @@
         "webrtc_video_stream.cc",
       ]
     }
   }
 }
 
 static_library("test_support") {
   testonly = true
 
   sources = [
+    "fake_audio_source.cc",
+    "fake_audio_source.h",
     "fake_authenticator.cc",
     "fake_authenticator.h",
     "fake_connection_to_client.cc",
     "fake_connection_to_client.h",
     "fake_connection_to_host.cc",
     "fake_connection_to_host.h",
     "fake_datagram_socket.cc",
     "fake_datagram_socket.h",
     "fake_desktop_capturer.cc",
     "fake_desktop_capturer.h",
@@ skipping 47 matching lines @@
     "port_range_unittest.cc",
     "ppapi_module_stub.cc",
     "pseudotcp_adapter_unittest.cc",
     "session_config_unittest.cc",
     "spake2_authenticator_unittest.cc",
     "ssl_hmac_channel_authenticator_unittest.cc",
     "third_party_authenticator_unittest.cc",
     "v2_authenticator_unittest.cc",
     "validating_authenticator_unittest.cc",
     "video_frame_pump_unittest.cc",
+    "webrtc_audio_source_adapter_unittest.cc",
     "webrtc_transport_unittest.cc",
   ]
 
   deps = [
     ":test_support",
     "//net:test_support",
     "//testing/gmock",
     "//testing/gtest",
   ]
 }