Refactoring ./webrtc/modules/BUILD.gn for gn check

webrtc/modules/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/webrtc.gni")
 import("audio_coding/audio_coding.gni")
@@ skipping 47 matching lines @@
 
     videoprocessor_defines = []
     if (rtc_use_h264) {
       videoprocessor_defines += [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
     }
 
     defines = audio_coding_defines + videoprocessor_defines
 
     deps = [
       "..:webrtc_common",
+      "../base:rtc_base",
+      "../base:rtc_base_approved",
       "../common_video",
       "../media:rtc_media_base",
       "../modules/audio_coding",
       "../modules/audio_coding:audio_format_conversion",
+      "../modules/audio_coding:g711",
       "../modules/rtp_rtcp",
       "../modules/utility",
       "../modules/video_coding",
       "../modules/video_coding:video_codecs_test_framework",
+      "../modules/video_coding:webrtc_vp9",
       "../system_wrappers",
       "../test:test_main",
+      "../test:video_test_common",
+      "audio_coding:builtin_audio_decoder_factory",
+      "audio_coding:pcm16b_c",
+      "desktop_capture:primitives",
+      "video_coding:webrtc_h264",
+      "video_coding:webrtc_vp8",
       "//testing/gmock",
       "//testing/gtest",
     ]
 
     sources = [
       "audio_coding/test/APITest.cc",
       "audio_coding/test/Channel.cc",
       "audio_coding/test/EncodeDecodeTest.cc",
       "audio_coding/test/PCMFile.cc",
       "audio_coding/test/PacketLossTest.cc",
@@ skipping 171 matching lines @@
       "audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/channel_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/controller_manager_unittest.cc",
       "audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/fec_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/mock/mock_controller.h",
       "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
     ]
     deps = [
+      "../base:rtc_base_approved",
+      "../system_wrappers:system_wrappers",
+      "../test:field_trial",
+      "../test:test_support",
       "audio_coding:audio_network_adaptor",
       "//testing/gmock",
       "//testing/gtest",
     ]
     if (rtc_enable_protobuf) {
       deps += [ "audio_coding:ana_config_proto" ]
       defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
     }
     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_test("modules_unittests") {
     testonly = true
 
     defines = audio_coding_defines
-    deps = []
+    deps = [
+      "../base:gtest_prod",
+      "../base:rtc_task_queue",
+      "../system_wrappers:metrics_default",
+      "../test:field_trial",
+      "../test:test_support_unittests",
+      "audio_coding:audio_decoder_interface",
+      "audio_coding:audio_encoder_interface",
+      "audio_coding:audio_format_conversion",
+      "audio_coding:g711",
+      "audio_coding:ilbc",
+      "audio_coding:isac_c",
+      "audio_coding:rent_a_codec",
+      "audio_processing:audioproc_debug_proto",
+      "congestion_controller:congestion_controller",
+      "desktop_capture:primitives",
+      "video_coding:video_coding_utility",
+      "video_coding:webrtc_h264",
+    ]
     sources = [
       "audio_coding/acm2/acm_receiver_unittest.cc",
       "audio_coding/acm2/audio_coding_module_unittest.cc",
       "audio_coding/acm2/call_statistics_unittest.cc",
       "audio_coding/acm2/codec_manager_unittest.cc",
       "audio_coding/acm2/rent_a_codec_unittest.cc",
       "audio_coding/codecs/audio_decoder_factory_unittest.cc",
       "audio_coding/codecs/cng/audio_encoder_cng_unittest.cc",
       "audio_coding/codecs/cng/cng_unittest.cc",
       "audio_coding/codecs/ilbc/ilbc_unittest.cc",
@@ skipping 263 matching lines @@
     if (rtc_desktop_capture_supported) {
       deps += [ "desktop_capture:desktop_capture_mock" ]
       sources += [
         "desktop_capture/desktop_and_cursor_composer_unittest.cc",
         "desktop_capture/desktop_capturer_differ_wrapper_unittest.cc",
         "desktop_capture/desktop_frame_rotation_unittest.cc",
         "desktop_capture/mouse_cursor_monitor_unittest.cc",
         "desktop_capture/rgba_color_unittest.cc",
         "desktop_capture/screen_capturer_helper_unittest.cc",
         "desktop_capture/screen_capturer_mac_unittest.cc",
-        "desktop_capture/screen_capturer_mock_objects.h",
         "desktop_capture/screen_capturer_unittest.cc",
         "desktop_capture/test_utils.cc",
         "desktop_capture/test_utils.h",
         "desktop_capture/test_utils_unittest.cc",
         "desktop_capture/win/cursor_unittest.cc",
         "desktop_capture/win/cursor_unittest_resources.h",
         "desktop_capture/win/cursor_unittest_resources.rc",
         "desktop_capture/window_capturer_unittest.cc",
       ]
     }
@@ skipping 103 matching lines @@
       "video_processing",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
 
     data = modules_unittests_resources
 
     if (is_android) {
       deps += [
-        "//testing/android/native_test:native_test_support",
+        "//testing/android/native_test:native_test_native_code",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
 
       # Need to disable error due to the line in
       # base/android/jni_android.h triggering it:
       # const BASE_EXPORT jobject GetApplicationContext()
       # error: type qualifiers ignored on function return type
       cflags = [ "-Wno-ignored-qualifiers" ]
       sources += [
         "audio_device/android/audio_device_unittest.cc",
@@ skipping 17 matching lines @@
         sources += [ "audio_device/ios/audio_device_unittest_ios.cc" ]
       }
 
       ldflags = [ "-ObjC" ]
     }
   }
 
   rtc_test("bwe_simulator") {
     testonly = true
 
-    deps = []
+    deps = [
+      "remote_bitrate_estimator:remote_bitrate_estimator",
+    ]
     sources = [
       "remote_bitrate_estimator/bwe_simulations.cc",
     ]
 
     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 (is_win) {
       cflags = [
         # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
         "/wd4373",  # virtual function override.
       ]
     }
 
     deps += [
       "..:webrtc_common",
       "../base:rtc_base_approved",
       "../test:test_common",
       "../test:test_main",
       "remote_bitrate_estimator:bwe_simulator_lib",
       "video_capture",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 }