Move RtpExtension to api/ directory and config.h/.cc to call/.

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 20 matching lines @@
     "../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") {
   sources = [
     "rtcp_packet_sink_interface.h",
+    "rtp_config.cc",
+    "rtp_config.h",
     "rtp_packet_sink_interface.h",
     "rtp_stream_receiver_controller_interface.h",
     "rtp_transport_controller_send_interface.h",
   ]
   deps = [
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_receiver") {
@@ skipping 42 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" ]
   }
 
   public_deps = [
     ":call_interfaces",
     "../api:call_api",
+    "../api:libjingle_peerconnection_api",
   ]
 
   deps = [
     ":call_interfaces",
     ":rtp_interfaces",
     ":rtp_receiver",
     ":rtp_sender",
+    ":video_stream_api",
     "..:webrtc_common",
     "../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",
     "../rtc_base:rtc_task_queue",
     "../rtc_base:sequenced_task_checker",
     "../system_wrappers",
     "../video",
   ]
 }
 
 rtc_source_set("video_stream_api") {
   sources = [
+    "video_config.cc",
+    "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:transport_api",
     "../common_video:common_video",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
 
@@ skipping 55 matching lines @@
     if (!is_android && !is_ios) {
       visibility = [ "..:webrtc_perf_tests" ]
     }
     sources = [
       "call_perf_tests.cc",
       "rampup_tests.cc",
       "rampup_tests.h",
     ]
     deps = [
       ":call_interfaces",
+      ":video_stream_api",
       "..:webrtc_common",
       "../api/audio_codecs:builtin_audio_encoder_factory",
       "../logging:rtc_event_log_api",
       "../modules/audio_coding",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers",
       "../system_wrappers:metrics_default",
       "../test:direct_transport",
@@ skipping 19 matching lines @@
     sources = [
       "test/mock_rtp_packet_sink_interface.h",
     ]
     deps = [
       ":rtp_interfaces",
       "../test:test_support",
       "//testing/gmock",
     ]
   }
 }