Enabling 'gn check' on //webrtc/test.

webrtc/test/BUILD.gn

Index: webrtc/test/BUILD.gn
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 4050eaff853e5a935cb9e69c0d8f9fac7013de2b..bc391fb6d4d93b3843fc2cdefb6076146f1dd74d 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -53,9 +53,13 @@ rtc_source_set("video_test_common") {
   }
 
   deps = [
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../base:rtc_task_queue",
     "../common_video",
     "../media:rtc_media_base",
     "../modules/video_capture:video_capture_module",
+    "../system_wrappers:system_wrappers",
   ]
 }
 
@@ -77,6 +81,7 @@ rtc_source_set("rtp_test_utils") {
 
   deps = [
     "..:webrtc_common",
+    "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
     "//testing/gtest",
   ]
@@ -107,6 +112,7 @@ rtc_source_set("test_main") {
   ]
   deps = [
     ":field_trial",
+    "../base:rtc_base_approved",
     "../system_wrappers:metrics_default",
     "//testing/gmock",
     "//testing/gtest",
@@ -125,6 +131,16 @@ rtc_source_set("fileutils") {
     configs += [ "//build/config/compiler:enable_arc" ]
   }
   visibility = [ ":*" ]
+  deps = [
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+  ]
+  if (is_ios) {
+    deps += [ "../sdk:rtc_sdk_common_objc" ]
+  }
+  if (is_win) {
+    deps += [ "../base:rtc_base" ]
+  }
 }
 
 rtc_source_set("video_test_support") {
@@ -143,6 +159,9 @@ rtc_source_set("video_test_support") {
   ]
 
   deps = [
+    ":test_support",
+    ":video_test_common",
+    "..:webrtc_common",
     "../base:rtc_base_approved",
     "../common_video",
     "../system_wrappers",
@@ -191,6 +210,7 @@ rtc_source_set("test_support") {
   ]
 
   deps = [
+    "..:webrtc_common",
     "../base:gtest_prod",
     "../base:rtc_base_approved",
     "../common_video",
@@ -255,7 +275,16 @@ if (is_ios) {
 }
 
 rtc_test("test_support_unittests") {
-  deps = []
+  deps = [
+    ":fake_audio_device",

[kjellander_webrtc, 2017/04/12 13:15:01]

Move these to line 327+.
According to the GN style guide targets should have sources in the beginning and
deps at the end.
They only ended up here because the empty deps list was put here.

[mbonadei, 2017/04/12 13:51:55]

Done, good catch!

+    ":rtp_test_utils",
+    "../api:video_frame_api",
+    "../base:rtc_base_approved",
+    "../call:call_interfaces",
+    "../common_audio:common_audio",
+    "../modules/rtp_rtcp:rtp_rtcp",
+    "../system_wrappers:system_wrappers",
+  ]
   sources = [
     "fake_audio_device_unittest.cc",
     "fake_network_pipe_unittest.cc",
@@ -314,6 +343,7 @@ rtc_source_set("fileutils_unittests") {
   ]
   deps = [
     ":fileutils",
+    ":test_support",
     "//testing/gmock",
     "//testing/gtest",
   ]
@@ -332,9 +362,11 @@ rtc_source_set("direct_transport") {
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
   deps = [
+    "..:webrtc_common",
     "../api:transport_api",
     "../base:rtc_base_approved",
     "../call",
+    "../system_wrappers:system_wrappers",
   ]
 }
 
@@ -349,8 +381,11 @@ rtc_source_set("fake_audio_device") {
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
   deps = [
+    "..:webrtc_common",
     "../base:rtc_base_approved",
+    "../common_audio:common_audio",
     "../modules/audio_device:audio_device",
+    "../system_wrappers:system_wrappers",
   ]
 }
 
@@ -403,13 +438,25 @@ rtc_source_set("test_common") {
     ":test_support",
     ":video_test_common",
     "..:webrtc_common",
+    "../api:transport_api",
+    "../api:video_frame_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/video_codecs:video_codecs_api",
     "../audio",
     "../base:rtc_base_approved",
+    "../base:rtc_task_queue",
     "../call",
+    "../logging:rtc_event_log_api",
+    "../modules/audio_device:mock_audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing",
+    "../modules/rtp_rtcp:rtp_rtcp",

[kjellander_webrtc, 2017/04/12 13:15:01]

whoa, does test_common really pull in this much? I guess it'll improve as more
stuff is moved into api/ in the future.

+    "../modules/video_coding:webrtc_h264",
+    "../modules/video_coding:webrtc_vp8",
+    "../modules/video_coding:webrtc_vp9",
+    "../system_wrappers:system_wrappers",
     "../video",
+    "../voice_engine:voice_engine",
     "//testing/gmock",
     "//testing/gtest",
   ]
@@ -485,6 +532,9 @@ rtc_source_set("test_renderer") {
 
   deps = [
     ":test_support",
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../common_video:common_video",
     "../modules/media_file",
     "//testing/gtest",
   ]