Move optional.h to webrtc/api/

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 580d606f59eace16e5abfb6851e46619e6f87aac..706d502b213577e0eb16732ad0ef9647c73fbdad 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -26,6 +26,7 @@ rtc_source_set("call_interfaces") {
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../rtc_base:rtc_base",
@@ -68,6 +69,7 @@ rtc_source_set("rtp_receiver") {
     ":rtp_interfaces",
     "..:webrtc_common",
     "../api:array_view",
+    "../api:optional",
     "../modules/rtp_rtcp",
     "../rtc_base:rtc_base_approved",
   ]
@@ -114,6 +116,7 @@ rtc_static_library("call") {
     ":rtp_sender",
     ":video_stream_api",
     "..:webrtc_common",
+    "../api:optional",
     "../api:transport_api",
     "../audio",
     "../logging:rtc_event_log_api",
@@ -144,6 +147,7 @@ rtc_source_set("video_stream_api") {
     ":rtp_interfaces",
     "../:webrtc_common",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../common_video:common_video",
     "../rtc_base:rtc_base_approved",