Enabling 'gn check' on webrtc/test.

webrtc/test/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")
 import("//build/config/ui.gni")
@@ skipping 40 matching lines @@
     "video_capturer.cc",
     "video_capturer.h",
   ]
 
   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 = [
+    "..:video_stream_api",
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../base:rtc_task_queue",
     "../common_video",
     "../media:rtc_media_base",
     "../modules/video_capture:video_capture_module",
+    "../system_wrappers",
   ]
 }
 
 rtc_source_set("rtp_test_utils") {
   testonly = true
   sources = [
     "rtcp_packet_parser.cc",
     "rtcp_packet_parser.h",
     "rtp_file_reader.cc",
     "rtp_file_reader.h",
     "rtp_file_writer.cc",
     "rtp_file_writer.h",
   ]
 
   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 = [
     "..:webrtc_common",
+    "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
     "//testing/gtest",
   ]
 }
 
 rtc_source_set("field_trial") {
   testonly = true
   sources = [
     "field_trial.cc",
     "field_trial.h",
@@ skipping 63 matching lines @@
     testonly = true
     sources = [
       "test_main.cc",
     ]
 
     public_deps = [
       ":test_support",
     ]
     deps = [
       ":field_trial",
+      "../base:rtc_base_approved",
       "../system_wrappers:metrics_default",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_source_set("video_test_support") {
     testonly = true
 
     sources = [
       "testsupport/frame_reader.h",
       "testsupport/frame_writer.h",
       "testsupport/metrics/video_metrics.cc",
       "testsupport/metrics/video_metrics.h",
       "testsupport/mock/mock_frame_reader.h",
       "testsupport/mock/mock_frame_writer.h",
       "testsupport/y4m_frame_writer.cc",
       "testsupport/yuv_frame_reader.cc",
       "testsupport/yuv_frame_writer.cc",
     ]
 
     deps = [
+      ":test_support",
+      ":video_test_common",
+      "..:webrtc_common",
       "../base:rtc_base_approved",
       "../common_video",
       "../system_wrappers",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
 
     public_deps = [
       ":fileutils",
@@ skipping 44 matching lines @@
     bundle_data("test_support_unittests_bundle_data") {
       testonly = true
       sources = test_support_unittests_resources
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("test_support_unittests") {
-    deps = []
+    deps = [
+      ":fake_audio_device",
+      ":rtp_test_utils",
+      "../api:video_frame_api",
+      "../base:rtc_base_approved",
+      "../call:call_interfaces",
+      "../common_audio",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
+    ]
     sources = [
       "fake_audio_device_unittest.cc",
       "fake_network_pipe_unittest.cc",
       "frame_generator_unittest.cc",
       "rtp_file_reader_unittest.cc",
       "rtp_file_writer_unittest.cc",
       "testsupport/always_passing_unittest.cc",
       "testsupport/isolated_output_unittest.cc",
       "testsupport/metrics/video_metrics_unittest.cc",
       "testsupport/packet_reader_unittest.cc",
@@ skipping 45 matching lines @@
 rtc_source_set("fileutils") {
   testonly = true
   sources = [
     "testsupport/fileutils.cc",
     "testsupport/fileutils.h",
   ]
   if (is_ios) {
     sources += [ "testsupport/iosfileutils.mm" ]
   }
   visibility = [ ":*" ]
+  deps = [
+    "..:webrtc_common",
+  ]
+  if (is_ios) {
+    deps += [
+      "../base:rtc_base_approved",
+      "../sdk:rtc_sdk_common_objc",
+    ]
+  }
+  if (is_win) {
+    deps += [ "../base:rtc_base" ]
+  }
 }
 
 rtc_source_set("run_test") {
   testonly = true
   sources = [
     "run_test.h",
   ]
   if (is_mac) {
     sources += [ "mac/run_test.mm" ]
   } else {
     sources += [ "run_test.cc" ]
   }
 }
 
 rtc_source_set("fileutils_unittests") {
   testonly = true
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
   sources = [
     "testsupport/fileutils_unittest.cc",
   ]
   deps = [
     ":fileutils",
+    ":test_support",
     "//testing/gmock",
     "//testing/gtest",
   ]
 }
 
 rtc_source_set("direct_transport") {
   testonly = true
   sources = [
     "direct_transport.cc",
     "direct_transport.h",
     "fake_network_pipe.cc",
     "fake_network_pipe.h",
   ]
   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 = [
+    "..:webrtc_common",
     "../api:transport_api",
     "../base:rtc_base_approved",
     "../call",
+    "../modules/rtp_rtcp",
+    "../system_wrappers",
   ]
 }
 
 rtc_source_set("fake_audio_device") {
   testonly = true
   sources = [
     "fake_audio_device.cc",
     "fake_audio_device.h",
   ]
   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 = [
+    "..:webrtc_common",
     "../base:rtc_base_approved",
-    "../modules/audio_device:audio_device",
+    "../common_audio",
+    "../modules/audio_device",
+    "../system_wrappers",
   ]
 }
 
 rtc_source_set("test_common") {
   testonly = true
   sources = [
     "call_test.cc",
     "call_test.h",
     "configurable_frame_size_encoder.cc",
     "configurable_frame_size_encoder.h",
@@ skipping 31 matching lines @@
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     ":direct_transport",
     ":fake_audio_device",
     ":rtp_test_utils",
     ":test_support",
     ":video_test_common",
+    "..:video_stream_api",
     "..:webrtc_common",
+    "../api:transport_api",
+    "../api:video_frame_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_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",
+    "../modules/video_coding:webrtc_h264",
+    "../modules/video_coding:webrtc_vp8",
+    "../modules/video_coding:webrtc_vp9",
+    "../system_wrappers",
     "../video",
+    "../voice_engine",
     "//testing/gmock",
     "//testing/gtest",
   ]
   if (!is_android && !build_with_chromium) {
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
 }
 
 config("test_renderer_exported_config") {
   if (is_win && is_clang) {
@@ skipping 55 matching lines @@
 
   public_configs = [ ":test_renderer_exported_config" ]
 
   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 = [
     ":test_support",
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../common_video",
     "../modules/media_file",
     "//testing/gtest",
   ]
 }
 
 rtc_source_set("audio_codec_mocks") {
   testonly = true
   sources = [
     "mock_audio_decoder.h",
     "mock_audio_decoder_factory.h",
     "mock_audio_encoder.cc",
     "mock_audio_encoder.h",
     "mock_audio_encoder_factory.h",
   ]
 
   deps = [
+    ":test_support",
     "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+    "../base:rtc_base_approved",
     "//testing/gmock",
   ]
 }