Enabling 'gn check' on webrtc/video

webrtc/video/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
 
@@ skipping 42 matching lines @@
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "..:webrtc_common",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
     "../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",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",
     "../modules/remote_bitrate_estimator",
     "../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",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_source_set("video_quality_test") {
     testonly = true
     sources = [
       "video_quality_test.cc",
       "video_quality_test.h",
     ]
     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",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   rtc_source_set("video_full_stack_tests") {
     testonly = true
     sources = [
       "full_stack_tests.cc",
     ]
     deps = [
       ":video_quality_test",
+      "../test:field_trial",
+      "../test:test_support",
       "//testing/gtest",
       "//webrtc/test:test_common",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
     if (rtc_use_h264) {
       defines = [ "WEBRTC_USE_H264" ]
     }
   }
 
   rtc_executable("video_loopback") {
     testonly = true
     sources = [
       "video_loopback.cc",
     ]
     deps = [
       ":video_quality_test",
       "../system_wrappers:metrics_default",
       "../test:field_trial",
       "../test:run_test",
       "../test:test_common",
       "../test:test_renderer",
+      "../test:test_support",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   rtc_executable("screenshare_loopback") {
     testonly = true
     sources = [
       "screenshare_loopback.cc",
     ]
 
     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) {
       # Suppress warnings from Chrome's Clang plugins.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   rtc_executable("video_replay") {
     testonly = true
     sources = [
       "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) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   # 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 = [
       "call_stats_unittest.cc",
       "encoder_rtcp_feedback_unittest.cc",
       "end_to_end_tests.cc",
       "overuse_frame_detector_unittest.cc",
       "payload_router_unittest.cc",
       "quality_threshold_unittest.cc",
       "receive_statistics_proxy_unittest.cc",
       "report_block_stats_unittest.cc",
       "rtp_stream_receiver_unittest.cc",
       "send_delay_stats_unittest.cc",
       "send_statistics_proxy_unittest.cc",
       "stats_counter_unittest.cc",
       "stream_synchronization_unittest.cc",
       "video_receive_stream_unittest.cc",
       "video_send_stream_tests.cc",
       "vie_encoder_unittest.cc",
     ]
     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",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
   }
 }