Add new target direct_transport and remove fake_network and direct_transport from test_common.

webrtc/test/BUILD.gn

 # Copyright (c) 2014 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("../webrtc.gni")
 import("//build/config/ui.gni")
@@ skipping 294 matching lines @@
   sources = [
     "testsupport/fileutils_unittest.cc",
   ]
   deps = [
     ":fileutils",
     "//testing/gmock",
     "//testing/gtest",
   ]
 }
 
+rtc_source_set("direct_transport") {
+  testonly = true
+  sources = [
+    "direct_transport.cc",
+    "direct_transport.h",
+    "fake_network_pipe.cc",
+    "fake_network_pipe.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" ]
+  }
+  deps = [
+    "../api:transport_api",
+    "../base:rtc_base_approved",
+    "../call",
+  ]
+}
+
 rtc_source_set("test_common") {
   testonly = true
   sources = [
     "call_test.cc",
     "call_test.h",
     "configurable_frame_size_encoder.cc",
     "configurable_frame_size_encoder.h",
     "constants.cc",
     "constants.h",
-    "direct_transport.cc",
-    "direct_transport.h",
     "drifting_clock.cc",
     "drifting_clock.h",
     "encoder_settings.cc",
     "encoder_settings.h",
     "fake_audio_device.cc",
     "fake_audio_device.h",
     "fake_decoder.cc",
     "fake_decoder.h",
     "fake_encoder.cc",
     "fake_encoder.h",
-    "fake_network_pipe.cc",
-    "fake_network_pipe.h",
     "fake_videorenderer.h",
     "layer_filtering_transport.cc",
     "layer_filtering_transport.h",
     "mock_transport.h",
     "mock_voe_channel_proxy.h",
     "mock_voice_engine.h",
     "null_transport.cc",
     "null_transport.h",
     "rtp_rtcp_observer.h",
     "statistics.cc",
     "statistics.h",
     "win/run_loop_win.cc",
   ]
   if (!is_win) {
     sources += [
       "run_loop.cc",
       "run_loop.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" ]
   }
 
   deps = [
+    ":direct_transport",
     ":rtp_test_utils",
     ":test_support",
     "..:webrtc_common",
-    "../api:transport_api",
     "../audio",
     "../base:rtc_base_approved",
     "../call",
     "../modules/audio_device:mock_audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing",
     "../video",
     "//testing/gmock",
     "//testing/gtest",
   ]
@@ skipping 64 matching lines @@
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     ":test_support",
     ":video_test_common",
     "../modules/media_file",
     "//testing/gtest",
   ]
 }