Move array_view.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")
 
@@ skipping 25 matching lines @@
 # 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") {
   sources = [
     "rtcp_packet_sink_interface.h",
     "rtp_packet_sink_interface.h",
     "rtp_stream_receiver_controller_interface.h",
     "rtp_transport_controller_send_interface.h",
   ]
   deps = [
+    "../api:array_view",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_receiver") {
   sources = [
     "rtcp_demuxer.cc",
     "rtcp_demuxer.h",
     "rtp_demuxer.cc",
     "rtp_demuxer.h",
     "rtp_rtcp_demuxer_helper.cc",
     "rtp_rtcp_demuxer_helper.h",
     "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",
     "../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 83 matching lines @@
       "rtp_rtcp_demuxer_helper_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
       ":call",
       ":mock_rtp_interfaces",
       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",
       "..:webrtc_common",
+      "../api:array_view",
       "../api:mock_audio_mixer",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../modules/utility:mock_process_thread",
@@ skipping 61 matching lines @@
     sources = [
       "test/mock_rtp_packet_sink_interface.h",
     ]
     deps = [
       ":rtp_interfaces",
       "../test:test_support",
       "//testing/gmock",
     ]
   }
 }