Refactor webrtc/{api,audio} and modules/audio_coding for GN check

webrtc/api/BUILD.gn

 # Copyright (c) 2015 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("../build/webrtc.gni")
 if (is_android) {
@@ skipping 12 matching lines @@
     "call/audio_sink.h",
     "call/flexfec_receive_stream.h",
   ]
 
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
     ":audio_mixer_api",
     ":transport_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
+    "../modules/audio_coding:audio_decoder_factory_interface",
     "../modules/audio_coding:audio_encoder_interface",
   ]
 }
 
 config("libjingle_peerconnection_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
   # cannot be on the target directly.
   if (!is_win && !is_clang) {
     cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
   }
 }
 
 rtc_static_library("libjingle_peerconnection") {
+  check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   cflags = []
   sources = [
     "audiotrack.cc",
     "audiotrack.h",
     "datachannel.cc",
     "datachannel.h",
     "datachannelinterface.h",
     "dtmfsender.cc",
     "dtmfsender.h",
     "dtmfsenderinterface.h",
@@ skipping 154 matching lines @@
         "-Wno-unused-function",
       ]
     }
 
     if (!is_win) {
       cflags = [ "-Wno-sign-compare" ]
     }
   }
 
   rtc_test("peerconnection_unittests") {
+    check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
     testonly = true
     sources = [
       "datachannel_unittest.cc",
       "dtmfsender_unittest.cc",
       "fakemetricsobserver.cc",
       "fakemetricsobserver.h",
       "jsepsessiondescription_unittest.cc",
       "localaudiosource_unittest.cc",
       "mediaconstraintsinterface_unittest.cc",
       "mediastream_unittest.cc",
@@ skipping 93 matching lines @@
     sources = [
       "test/mock_audio_mixer.h",
     ]
 
     public_deps = [
       ":audio_mixer_api",
     ]
 
     deps = [
       "//testing/gmock",
+      "//webrtc/test:test_support",
     ]
   }
 }