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

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index d86df1e379a9686c2b8243c84dc49f69cb4e6390..0c3c6b693a8743a3d5201cf6fc0f01d09c4a3050 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -16,6 +16,7 @@ rtc_source_set("call_interfaces") {
     "audio_state.h",
     "call.h",
     "callfactoryinterface.h",
+    "config.h",
     "flexfec_receive_stream.h",
     "syncable.cc",
     "syncable.h",
@@ -88,6 +89,7 @@ rtc_static_library("call") {
     "call.cc",
     "callfactory.cc",
     "callfactory.h",
+    "config.cc",
     "flexfec_receive_stream_impl.cc",
     "flexfec_receive_stream_impl.h",
   ]
@@ -100,6 +102,7 @@ rtc_static_library("call") {
   public_deps = [
     ":call_interfaces",
     "../api:call_api",
+    "../api:libjingle_peerconnection_api",
   ]
 
   deps = [
@@ -134,6 +137,7 @@ rtc_source_set("video_stream_api") {
   ]
   deps = [
     "../:webrtc_common",
+    "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../common_video:common_video",
     "../rtc_base:rtc_base_approved",