From ebafdc84843256bac7f125ff637d7e006f67d595 Mon Sep 17 00:00:00 2001
From: mbonadei <mbonadei@webrtc.org>
Date: Thu, 22 Dec 2016 07:35:39 -0800
Subject: [PATCH] Refactor webrtc/modules/rtp_rtcp for GN check

This moves some GN check configurations out of .gn to individual
targets.

This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from
the static_library 'rtp_rtcp' because it depends on a 'testonly = true'
target. After a check this seems only included in the unitest code:

$ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/
webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"

This commit also removes the dependency on
'//webrt/modules/video_coding' because it seems that the following
include can be removed:

#include "webrtc/modules/video_coding/include/video_coding_defines.h"

The now checked target is:
"//webrtc/modules/rtp_rtcp/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2598963002
Cr-Commit-Position: refs/heads/master@{#15760}
---
 .gn                                        | 1 +
 webrtc/modules/BUILD.gn                    | 1 +
 webrtc/modules/rtp_rtcp/BUILD.gn           | 6 +++++-
 webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 1 -
 4 files changed, 7 insertions(+), 2 deletions(-)

diff --git a/.gn b/.gn
index 0bd9b08667..c381566dec 100644
--- a/.gn
+++ b/.gn
@@ -31,6 +31,7 @@ check_targets = [
   "//webrtc/modules/audio_processing/*",
   "//webrtc/modules/media_file/*",
   "//webrtc/modules/pacing/*",
+  "//webrtc/modules/rtp_rtcp/*",
   "//webrtc/modules/video_capture/*",
   "//webrtc/modules/video_coding/*",
   "//webrtc/stats:rtc_stats",
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index ca519e7c04..40cbdf62c1 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -421,6 +421,7 @@ if (rtc_include_tests) {
       "remote_bitrate_estimator/test/bwe_unittest.cc",
       "remote_bitrate_estimator/test/estimators/nada_unittest.cc",
       "remote_bitrate_estimator/test/metric_recorder_unittest.cc",
+      "rtp_rtcp/mocks/mock_rtp_rtcp.h",
       "rtp_rtcp/source/byte_io_unittest.cc",
       "rtp_rtcp/source/fec_test_helper.cc",
       "rtp_rtcp/source/fec_test_helper.h",
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 80f79c1764..e806f070a1 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",
     "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",
   ]
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 27e6143ae2..aa5df5f8cf 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -22,7 +22,6 @@
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {