Move optional.h to webrtc/api/

webrtc/call/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("../webrtc.gni")
 
 rtc_source_set("call_interfaces") {
   sources = [
     "audio_receive_stream.h",
     "audio_send_stream.cc",
     "audio_send_stream.h",
     "audio_state.h",
     "call.h",
     "callfactoryinterface.h",
     "flexfec_receive_stream.h",
     "syncable.cc",
     "syncable.h",
   ]
   deps = [
     ":rtp_interfaces",
     ":video_stream_api",
     "..: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",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 # TODO(nisse): These RTP targets should be moved elsewhere
 # when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
 rtc_source_set("rtp_interfaces") {
@@ skipping 22 matching lines @@
     "rtp_stream_receiver_controller.cc",
     "rtp_stream_receiver_controller.h",
     "rtx_receive_stream.cc",
     "rtx_receive_stream.h",
     "ssrc_binding_observer.h",
   ]
   deps = [
     ":rtp_interfaces",
     "..:webrtc_common",
     "../api:array_view",
+    "../api:optional",
     "../modules/rtp_rtcp",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_sender") {
   sources = [
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
@@ skipping 26 matching lines @@
     "../api:libjingle_peerconnection_api",
   ]
 
   deps = [
     ":call_interfaces",
     ":rtp_interfaces",
     ":rtp_receiver",
     ":rtp_sender",
     ":video_stream_api",
     "..:webrtc_common",
+    "../api:optional",
     "../api:transport_api",
     "../audio",
     "../logging:rtc_event_log_api",
     "../logging:rtc_event_log_impl",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",
     "../modules/rtp_rtcp",
     "../modules/utility",
     "../rtc_base:rtc_base_approved",
@@ skipping 10 matching lines @@
     "video_config.h",
     "video_receive_stream.cc",
     "video_receive_stream.h",
     "video_send_stream.cc",
     "video_send_stream.h",
   ]
   deps = [
     ":rtp_interfaces",
     "../:webrtc_common",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../common_video:common_video",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
 
@@ skipping 97 matching lines @@
     sources = [
       "test/mock_rtp_packet_sink_interface.h",
     ]
     deps = [
       ":rtp_interfaces",
       "../test:test_support",
       "//testing/gmock",
     ]
   }
 }