Refactor webrtc/modules/rtp_rtcp for GN check

webrtc/modules/rtp_rtcp/BUILD.gn

Index: webrtc/modules/rtp_rtcp/BUILD.gn
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 80f79c1764b84d3fd0a24fd42cb9ec5818942bac..e806f070a1afc47a4b75b26c85bee833a82e27b9 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -20,7 +20,6 @@ rtc_static_library("rtp_rtcp") {
     "include/rtp_rtcp.h",
     "include/rtp_rtcp_defines.h",
     "include/ulpfec_receiver.h",
-    "mocks/mock_rtp_rtcp.h",

[kjellander_webrtc, 2016/12/22 10:34:54]

Please add this to the target where it's used; in modules_unittests of
https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/BUI...

(long term this should probably move into a target inside webrtc/api instead,
but let's leave that to the owners).

[mbonadei, 2016/12/22 14:55:53]

Done.

     "source/byte_io.h",
     "source/dtmf_queue.cc",
     "source/dtmf_queue.h",
@@ -175,7 +174,12 @@ rtc_static_library("rtp_rtcp") {
   deps = [
     "../..:webrtc_common",
     "../../api:transport_api",
+    "../../base:gtest_prod",
+    "../../base:rtc_base_approved",
+    "../../base:rtc_task_queue",
+    "../../call:call_interfaces",
     "../../common_video",
+    "../../logging:rtc_event_log_api",
     "../../system_wrappers",
     "../remote_bitrate_estimator",
   ]