Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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")
 if (is_android) {
@@ skipping 85 matching lines @@
     "jsepsessiondescription.cc",
     "localaudiosource.cc",
     "localaudiosource.h",
     "mediacontroller.cc",
     "mediacontroller.h",
     "mediastream.cc",
     "mediastream.h",
     "mediastreamobserver.cc",
     "mediastreamobserver.h",
     "mediastreamtrack.h",
-    "ortcfactory.cc",
-    "ortcfactory.h",
     "peerconnection.cc",
     "peerconnection.h",
     "peerconnectionfactory.cc",
     "peerconnectionfactory.h",
     "remoteaudiosource.cc",
     "remoteaudiosource.h",
     "rtcstatscollector.cc",
     "rtcstatscollector.h",
     "rtpreceiver.cc",
     "rtpreceiver.h",
@@ skipping 100 matching lines @@
 
     if (rtc_build_libsrtp) {
       deps += [ "//third_party/libsrtp" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
 
+  rtc_source_set("pc_test_utils") {
+    testonly = true
+    sources = [
+      "test/fakeaudiocapturemodule.cc",
+      "test/fakeaudiocapturemodule.h",
+      "test/fakedatachannelprovider.h",
+      "test/fakeperiodicvideocapturer.h",
+      "test/fakertccertificategenerator.h",
+      "test/fakevideotrackrenderer.h",
+      "test/fakevideotracksource.h",
+      "test/mock_datachannel.h",
+      "test/mock_peerconnection.h",
+      "test/mock_webrtcsession.h",
+      "test/mockpeerconnectionobservers.h",
+      "test/peerconnectiontestwrapper.cc",
+      "test/peerconnectiontestwrapper.h",
+      "test/rtcstatsobtainer.h",
+      "test/testsdpstrings.h",
+    ]
+
+    deps = [
+      ":libjingle_peerconnection",
+      "../base:rtc_base_tests_utils",
+      "//testing/gmock",
+    ]
+
+    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" ]
+    }
+  }
+
   config("peerconnection_unittests_config") {
     # The warnings below are enabled by default. Since GN orders compiler flags
     # for a target before flags from configs, the only way to disable such
     # warnings is by having them in a separate config, loaded from the target.
     # TODO(kjellander): Make the code compile without disabling these flags.
     # See https://bugs.webrtc.org/3307.
     if (is_clang && is_win) {
       cflags = [
         # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
         # for -Wno-sign-compare
@@ skipping 11 matching lines @@
     check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
     testonly = true
     sources = [
       "datachannel_unittest.cc",
       "dtmfsender_unittest.cc",
       "fakemediacontroller.h",
       "jsepsessiondescription_unittest.cc",
       "localaudiosource_unittest.cc",
       "mediaconstraintsinterface_unittest.cc",
       "mediastream_unittest.cc",
-      "ortcfactory_unittest.cc",
       "peerconnection_unittest.cc",
       "peerconnectionendtoend_unittest.cc",
       "peerconnectionfactory_unittest.cc",
       "peerconnectioninterface_unittest.cc",
       "proxy_unittest.cc",
       "rtcstats_integrationtest.cc",
       "rtcstatscollector_unittest.cc",
       "rtpsenderreceiver_unittest.cc",
       "sctputils_unittest.cc",
       "statscollector_unittest.cc",
-      "test/fakeaudiocapturemodule.cc",
-      "test/fakeaudiocapturemodule.h",
       "test/fakeaudiocapturemodule_unittest.cc",
-      "test/fakedatachannelprovider.h",
-      "test/fakeperiodicvideocapturer.h",
-      "test/fakertccertificategenerator.h",
-      "test/fakevideotrackrenderer.h",
-      "test/fakevideotracksource.h",
-      "test/mock_datachannel.h",
-      "test/mock_peerconnection.h",
-      "test/mock_webrtcsession.h",
-      "test/mockpeerconnectionobservers.h",
-      "test/peerconnectiontestwrapper.cc",
-      "test/peerconnectiontestwrapper.h",
-      "test/rtcstatsobtainer.h",
       "test/testsdpstrings.h",
       "trackmediainfomap_unittest.cc",
       "videocapturertracksource_unittest.cc",
       "videotrack_unittest.cc",
       "webrtcsdp_unittest.cc",
       "webrtcsession_unittest.cc",
     ]
 
     if (rtc_enable_sctp) {
       defines = [ "HAVE_SCTP" ]
@@ skipping 34 matching lines @@
       ]
       deps += [
         "//testing/android/native_test:native_test_support",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
         "//webrtc/sdk/android:libjingle_peerconnection_jni",
       ]
     }
 
     deps += [
       ":libjingle_peerconnection",
+      ":pc_test_utils",
       "..:webrtc_common",
       "../api:fakemetricsobserver",
       "../base:rtc_base_tests_utils",
       "../media:rtc_unittest_main",
       "../pc:rtc_pc",
       "../system_wrappers:metrics_default",
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }