Add WebrtcConnectionToClient.

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")
 import("//remoting/remoting_srcs.gni")
 
 source_set("protocol") {
   sources = rebase_path(remoting_srcs_gypi_values.remoting_protocol_sources,
                         ".",
@@ skipping 28 matching lines @@
     if (enable_webrtc) {
       deps += [
         # TODO(GYP): crbug.com/481633. We should probably not have to depend on
         # libjingle_webrtc; that should be pulled in automatically by
         # libpeerconnection instead.
         "//third_party/libjingle:libjingle_webrtc",
         "//third_party/libjingle:libpeerconnection",
       ]
     } else {
       sources -= [
+        "webrtc_connection_to_client.cc",
         "webrtc_transport.cc",
         "webrtc_video_capturer_adapter.cc",
+        "webrtc_video_stream.cc",
       ]
     }
   }
 }
 
 source_set("test_support") {
   testonly = true
 
   sources = [
     "fake_authenticator.cc",
@@ skipping 48 matching lines @@
     "negotiating_authenticator_unittest.cc",
     "pairing_registry_unittest.cc",
     "port_range_unittest.cc",
     "ppapi_module_stub.cc",
     "pseudotcp_adapter_unittest.cc",
     "session_config_unittest.cc",
     "ssl_hmac_channel_authenticator_unittest.cc",
     "third_party_authenticator_unittest.cc",
     "v2_authenticator_unittest.cc",
     "video_frame_pump_unittest.cc",
+    "webrtc_connection_to_client_unittest.cc",
     "webrtc_transport_unittest.cc",
   ]
 
   deps = [
     ":test_support",
     "//testing/gmock",
     "//testing/gtest",
   ]
 }