Enabling 'gn check' on webrtc/video

webrtc/video/BUILD.gn

Index: webrtc/video/BUILD.gn
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 524f84b3688a36e1326839052dc15ae0f7319fdc..8694b2c80c53e9478138c6ca7262ce8ecedb231c 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -60,6 +60,7 @@ rtc_static_library("video") {
     "../base:rtc_base_approved",
     "../base:rtc_numerics",
     "../base:rtc_task_queue",
+    "../call:call_interfaces",
     "../common_video",
     "../logging:rtc_event_log_api",
     "../media:rtc_media_base",
@@ -70,6 +71,8 @@ rtc_static_library("video") {
     "../modules/rtp_rtcp",
     "../modules/utility",
     "../modules/video_coding",
+    "../modules/video_coding:video_coding_utility",
+    "../modules/video_coding:webrtc_vp8",
     "../modules/video_processing",
     "../system_wrappers",
     "../voice_engine",
@@ -86,8 +89,19 @@ if (rtc_include_tests) {
     deps = [
       "../base:rtc_base_tests_utils",
       "../base:rtc_task_queue",
+      "../call:call_interfaces",
+      "../common_video",
+      "../logging:rtc_event_log_api",
       "../media:rtc_media_base",
+      "../modules/audio_mixer:audio_mixer_impl",
+      "../modules/rtp_rtcp",
+      "../modules/video_coding:webrtc_h264",
+      "../modules/video_coding:webrtc_vp8",
+      "../modules/video_coding:webrtc_vp9",
       "../system_wrappers",
+      "../test:test_common",
+      "../test:test_support",
+      "../voice_engine",
       "//testing/gtest",
       "//webrtc/test:test_renderer",
       "//webrtc/test:video_test_common",
@@ -105,6 +119,8 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":video_quality_test",
+      "../test:field_trial",
+      "../test:test_support",
       "//testing/gtest",
       "//webrtc/test:test_common",
     ]
@@ -129,6 +145,7 @@ if (rtc_include_tests) {
       "../test:run_test",
       "../test:test_common",
       "../test:test_renderer",
+      "../test:test_support",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
@@ -147,11 +164,13 @@ if (rtc_include_tests) {
 
     deps = [
       ":video_quality_test",
+      "../base:rtc_base_approved",
       "../system_wrappers:metrics_default",
       "../test:field_trial",
       "../test:run_test",
       "../test:test_common",
       "../test:test_renderer",
+      "../test:test_support",
       "//third_party/gflags",
     ]
     if (!build_with_chromium && is_clang) {
@@ -167,12 +186,22 @@ if (rtc_include_tests) {
       "replay.cc",
     ]
     deps = [
+      "..:webrtc_common",
       "../api/video_codecs:video_codecs_api",
+      "../base:rtc_base_approved",
+      "../call:call_interfaces",
+      "../common_video",
+      "../logging:rtc_event_log_api",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
       "../system_wrappers:metrics_default",
       "../test:field_trial",
+      "../test:rtp_test_utils",
       "../test:run_test",
       "../test:test_common",
       "../test:test_renderer",
+      "../test:test_support",
+      "../test:video_test_common",
       "//third_party/gflags",
     ]
     if (!build_with_chromium && is_clang) {
@@ -183,6 +212,10 @@ if (rtc_include_tests) {
 
   # TODO(pbos): Rename test suite.
   rtc_source_set("video_tests") {
+    # TODO(mbonadei): Fix the double definition of _main. It causes problems
+    # at link time. It is defined in //webrtc/test:test_main and
+    # //webrtc/media/rtc_unittest_main.

[kjellander_webrtc, 2017/04/24 05:14:33]

Can you break rtc_unittest_main into two targets so we can depend on the one
this target needs without pulling in the main part? Or just move that file to
another existing target?

+    check_includes = false
     testonly = true
     defines = []
     sources = [
@@ -205,7 +238,36 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":video",
+      "../api:video_frame_api",
+      "../api/video_codecs:video_codecs_api",
+      "../base:rtc_base_approved",
+      "../base:rtc_base_tests_utils",
+      "../call:call_interfaces",
+      "../common_video",
+      "../logging:rtc_event_log_api",
       "../media:rtc_media_base",
+
+      # Skipped (see comment at the beginning of the target)
+      # "../media:rtc_unittest_main",
+      "../modules/pacing",
+      "../modules/rtp_rtcp",
+      "../modules/rtp_rtcp:rtp_rtcp_unittests",
+      "../modules/utility",
+      "../modules/video_coding",
+      "../modules/video_coding:video_coding_utility",
+      "../modules/video_coding:webrtc_h264",
+      "../modules/video_coding:webrtc_vp8",
+      "../modules/video_coding:webrtc_vp9",
+      "../system_wrappers",
+      "../system_wrappers:field_trial_default",
+      "../system_wrappers:metrics_api",
+      "../system_wrappers:metrics_default",
+      "../test:direct_transport",
+      "../test:field_trial",
+      "../test:rtp_test_utils",
+      "../test:test_common",
+      "../test:test_support",
+      "../test:video_test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]