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

webrtc/modules/BUILD.gn

Index: webrtc/modules/BUILD.gn
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 40cbdf62c19b35eee985fc093c13df42ed36c7e1..412ddde682e1b5a8af31de51e8bffd33c2024723 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -65,16 +65,26 @@ if (rtc_include_tests) {
 
     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",
     ]
@@ -266,6 +276,10 @@ if (rtc_include_tests) {
       "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",
@@ -284,7 +298,25 @@ if (rtc_include_tests) {
     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",
@@ -568,7 +600,6 @@ if (rtc_include_tests) {
         "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",
@@ -692,7 +723,7 @@ if (rtc_include_tests) {
 
     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",
       ]
 
@@ -730,7 +761,9 @@ if (rtc_include_tests) {
   rtc_test("bwe_simulator") {
     testonly = true
 
-    deps = []
+    deps = [
+      "remote_bitrate_estimator:remote_bitrate_estimator",
+    ]
     sources = [
       "remote_bitrate_estimator/bwe_simulations.cc",
     ]