Fix Gn untracked headers in webrtc/call

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 70 matching lines @@
     "rtp_transport_controller_send.h",
   ]
   deps = [
     ":rtp_interfaces",
     "..:webrtc_common",
     "../modules/congestion_controller",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
+rtc_source_set("bitrate_allocator_header") {

[mbonadei, 2017/09/19 09:00:42]

Is it possible to move this into the "call" target?

If not I think we should remove the _header from the name.

[kjellander_webrtc, 2017/09/19 16:01:43]

Yeah, let's ask owners for advice here.

+  sources = [
+    "bitrate_allocator.h",
+  ]
+  deps = [
+    "../rtc_base:sequenced_task_checker",
+  ]
+}
+
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "callfactory.cc",
     "callfactory.h",
     "flexfec_receive_stream_impl.cc",
     "flexfec_receive_stream_impl.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" ]
   }
 
   public_deps = [
     ":call_interfaces",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
   ]
 
   deps = [
+    ":bitrate_allocator_header",
     ":call_interfaces",
     ":rtp_interfaces",
     ":rtp_receiver",
     ":rtp_sender",
     ":video_stream_api",
     "..:webrtc_common",
     "../api:optional",
     "../api:transport_api",
     "../audio",
     "../logging:rtc_event_log_api",
@@ skipping 38 matching lines @@
     # Skip restricting visibility on mobile platforms since the tests on those
     # gets additional generated targets which would require many lines here to
     # cover (which would be confusing to read and hard to maintain).
     if (!is_android && !is_ios) {
       visibility = [ "..:video_engine_tests" ]
     }
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
+      "fake_rtp_transport_controller_send.h",
       "flexfec_receive_stream_unittest.cc",
       "rtcp_demuxer_unittest.cc",
       "rtp_demuxer_unittest.cc",
       "rtp_rtcp_demuxer_helper_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
+      ":bitrate_allocator_header",
       ":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:congestion_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/pacing:mock_paced_sender",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../modules/utility:mock_process_thread",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers",
       "../test:audio_codec_mocks",
       "../test:direct_transport",
@@ skipping 58 matching lines @@
     sources = [
       "test/mock_rtp_packet_sink_interface.h",
     ]
     deps = [
       ":rtp_interfaces",
       "../test:test_support",
       "//testing/gmock",
     ]
   }
 }