Enable GN check for webrtc/media

webrtc/media/BUILD.gn

 # Copyright (c) 2016 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("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
@@ skipping 15 matching lines @@
 config("rtc_media_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
   # cannot be on the target directly.
   if (!is_win) {
     cflags = [ "-Wno-deprecated-declarations" ]
   }
 }
 
 rtc_static_library("rtc_media_base") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Enabling GN check triggers cyclic dependency error:
+  #   //webrtc/media:rtc_media_base ->
+  #   //webrtc/pc:rtc_pc ->
+  #   //webrtc/media:media ->
+  #   //webrtc/media:rtc_media_base
+  check_includes = false
   defines = []
   libs = []
   deps = []
   sources = [
     "base/adaptedvideotracksource.cc",
     "base/adaptedvideotracksource.h",
     "base/audiosource.h",
     "base/codec.cc",
     "base/codec.h",
     "base/cryptoparams.h",
@@ skipping 38 matching lines @@
     public_deps = [
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
   deps += [
     "..:webrtc_common",
+    "../api:libjingle_peerconnection_api",
+    "../api:video_frame_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../base:rtc_base",
     "../base:rtc_base_approved",
+    "../call:call_interfaces",
+    "../common_video:common_video",
     "../p2p",
   ]
 }
 
 rtc_static_library("rtc_media") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Enabling GN check triggers cyclic dependency error:
+  #   //webrtc/media:media ->
+  #   //webrtc/media:rtc_media ->
+  #   //webrtc/pc:rtc_pc ->
+  #   //webrtc/media:media
+  check_includes = false
   defines = []
   libs = []
   deps = []
   sources = [
     "engine/apm_helpers.cc",
     "engine/apm_helpers.h",
     "engine/internaldecoderfactory.cc",
     "engine/internaldecoderfactory.h",
     "engine/internalencoderfactory.cc",
     "engine/internalencoderfactory.h",
@@ skipping 84 matching lines @@
     deps += [ "../modules/video_capture:video_capture" ]
   } else {
     public_configs += [ ":rtc_media_defines_config" ]
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   deps += [
     ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:transport_api",
+    "../api:video_frame_api",

[magjed_webrtc, 2017/03/03 11:58:16]

How did you generate all these deps? It's difficult to verify.

[kjellander_webrtc, 2017/03/03 19:38:42]

They all come from the complaints of 'gn check' (or 'gn gen out/blah --check').
But then Edward wrote a tool that can be used to avoid most of the manual
labour: tools-webrtc/gn_check_autofix.py.
Example use: python tools-webrtc/gn_check_autofix.py -m "client.webrtc" -b
"Linux64 Debug"

Try patching this CL, remove one of these lines and run 'gn check'. Then you'll
see what source file is having an include that introduces the dependency.

[magjed_webrtc, 2017/03/06 10:11:59]

Cool! Very useful.

+    "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+    "../base:rtc_base",
     "../base:rtc_base_approved",
     "../call",
+    "../common_video:common_video",
+    "../modules/audio_coding:rent_a_codec",
+    "../modules/audio_device:audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
+    "../modules/audio_processing:audio_processing",
+    "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
+    "../modules/video_coding:webrtc_h264",
+    "../modules/video_coding:webrtc_vp8",
+    "../modules/video_coding:webrtc_vp9",
+    "../p2p:rtc_p2p",
     "../system_wrappers",
+    "../video",
     "../voice_engine",
   ]
 }
 
 if (rtc_include_tests) {
   config("rtc_unittest_main_config") {
     # GN orders flags on a target before flags from configs. The default config
     # adds -Wall, and this flag have to be after -Wall -- so they need to
     # come from a config and can"t be on the target directly.
     if (is_clang && is_ios) {
       cflags = [ "-Wno-unused-variable" ]
     }
   }
 
   rtc_source_set("rtc_unittest_main") {
     testonly = true
 
     include_dirs = []
     public_deps = []
-    deps = []
+    deps = [
+      "../modules/audio_coding:rent_a_codec",
+      "../modules/audio_processing:audio_processing",
+      "../p2p:rtc_p2p",
+    ]
     sources = [
       "base/fakemediaengine.h",
       "base/fakenetworkinterface.h",
       "base/fakertp.h",
       "base/fakevideocapturer.h",
       "base/fakevideorenderer.h",
       "base/test/mock_mediachannel.h",
       "base/testutils.cc",
       "base/testutils.h",
       "engine/fakewebrtccall.cc",
@@ skipping 14 matching lines @@
       # Need to add a directory normally exported by libyuv.
       include_dirs += [ "$rtc_libyuv_dir/include" ]
     }
 
     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" ]
     }
 
     deps += [
+      ":rtc_media",
+      ":rtc_media_base",
+      "..:webrtc_common",
+      "../api:call_api",
+      "../api:video_frame_api",
+      "../base:rtc_base",
+      "../base:rtc_base_approved",
       "../base:rtc_base_tests_main",
+      "../base:rtc_base_tests_utils",
+      "../call:call_interfaces",
+      "../test:test_support",
       "//testing/gtest",
     ]
     public_deps += [ "//testing/gmock" ]
   }
 
   config("rtc_media_unittests_config") {
     # GN orders flags on a target before flags from configs. The default config
     # adds -Wall, and this flag have to be after -Wall -- so they need to
     # come from a config and can"t be on the target directly.
     # TODO(kjellander): Make the code compile without disabling these flags.
@@ skipping 28 matching lines @@
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("rtc_media_unittests") {
     testonly = true
 
     defines = []
-    deps = []
+    deps = [
+      "../pc:rtc_pc",
+      "../test:field_trial",
+    ]
     sources = [
       "base/codec_unittest.cc",
       "base/rtpdataengine_unittest.cc",
       "base/rtputils_unittest.cc",
       "base/streamparams_unittest.cc",
       "base/turnutils_unittest.cc",
       "base/videoadapter_unittest.cc",
       "base/videobroadcaster_unittest.cc",
       "base/videocapturer_unittest.cc",
       "base/videocommon_unittest.cc",
@@ skipping 50 matching lines @@
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
       shard_timeout = 900
     }
 
     if (is_ios) {
       deps += [ ":rtc_media_unittests_bundle_data" ]
     }
 
     deps += [
-      # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
       ":rtc_media",
+      ":rtc_media_base",
       ":rtc_unittest_main",
+      "../api:video_frame_api",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
       "../audio",
+      "../base:rtc_base",
+      "../base:rtc_base_approved",
       "../base:rtc_base_tests_utils",
+      "../call:call_interfaces",
+      "../common_video:common_video",
+      "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
+      "../modules/audio_processing:audio_processing",
+      "../modules/video_coding:video_coding_utility",
+      "../modules/video_coding:webrtc_vp8",
+      "../p2p:rtc_p2p_unittests",
       "../system_wrappers:metrics_default",
+      "../test:test_support",
+      "../voice_engine:voice_engine",
     ]
   }
 }