Move optional.h to webrtc/api/

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 44 matching lines @@
     "srtpsession.h",
     "srtptransport.cc",
     "srtptransport.h",
     "voicechannel.h",
   ]
 
   deps = [
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:ortc_api",
     "../media:rtc_data",
     "../media:rtc_h264_profile_id",
     "../media:rtc_media_base",
     "../p2p:rtc_p2p",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_task_queue",
   ]
 
   if (rtc_build_libsrtp) {
@@ skipping 84 matching lines @@
 
   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 = [
     ":rtc_pc_base",
     "..:webrtc_common",
     "../api:call_api",
+    "../api:optional",
     "../api:rtc_stats_api",
     "../api/video_codecs:video_codecs_api",
     "../call:call_interfaces",
     "../logging:rtc_event_log_api",
     "../media:rtc_data",
     "../media:rtc_media_base",
     "../p2p:rtc_p2p",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
     "../stats",
@@ skipping 279 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }