Move optional.h to webrtc/api/

webrtc/video/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")
 
@@ skipping 37 matching lines @@
     "video_stream_encoder.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 = [
     "..:webrtc_common",
+    "../api:optional",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../call:video_stream_api",
     "../common_video",
     "../logging:rtc_event_log_api",
     "../media:rtc_media_base",
     "../modules:module_api",
     "../modules/bitrate_controller",
@@ skipping 18 matching lines @@
 
 if (rtc_include_tests) {
   rtc_source_set("video_quality_test") {
     testonly = true
     visibility = [ ":*" ]  # Only targets in this file can depend on this.
     sources = [
       "video_quality_test.cc",
       "video_quality_test.h",
     ]
     deps = [
+      "../api:optional",
       "../call:call_interfaces",
       "../common_video",
       "../logging:rtc_event_log_api",
       "../media:rtc_media",
       "../media:rtc_media_base",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
       "../modules/video_coding:webrtc_h264",
       "../modules/video_coding:webrtc_vp8",
       "../modules/video_coding:webrtc_vp9",
@@ skipping 149 matching lines @@
       "send_delay_stats_unittest.cc",
       "send_statistics_proxy_unittest.cc",
       "stats_counter_unittest.cc",
       "stream_synchronization_unittest.cc",
       "video_receive_stream_unittest.cc",
       "video_send_stream_tests.cc",
       "video_stream_encoder_unittest.cc",
     ]
     deps = [
       ":video",
+      "../api:optional",
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
       "../call:mock_rtp_interfaces",
       "../call:rtp_receiver",
       "../call:rtp_sender",
       "../call:video_stream_api",
       "../common_video",
       "../logging:rtc_event_log_api",
       "../media:rtc_media",
@@ skipping 27 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" ]
     }
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
   }
 }