Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )

Reason for revert:
Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6).

Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer.

Original issue's description:
> Enable GN check for webrtc/base
>
> It's not possible to enable it for the rtc_base_approved
> target but since a larger refactoring is ongoing for webrtc/base
> this CL doesn't attempt to fix that.
>
> Changes made:
> * Move webrtc/system_wrappers/include/stringize_macros.h into
>   webrtc/base:rtc_base_approved_unittests (and corresponding
>   unit test to rtc_base_approved_unittests).
> * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
> * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
>   webrtc/base.
> * Remove unused use include of webrtc/base/fileutils.h in
>   webrtc/base/pathutils.cc
>
> BUG=webrtc:6828, webrtc:3806, webrtc:7480
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2717083002
> Cr-Commit-Position: refs/heads/master@{#17766}
> Committed: ed754e71ae

TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6828, webrtc:3806, webrtc:7480
NOTRY=True

Review-Url: https://codereview.webrtc.org/2838683002
Cr-Commit-Position: refs/heads/master@{#17849}

.gn

@@ -24,7 +24,6 @@ secondary_source = "//build/secondary/"
 check_targets = [
   "//webrtc/api/*",
   "//webrtc/audio/*",
-  "//webrtc/base/*",
   "//webrtc/call/*",
   "//webrtc/common_video/*",
   "//webrtc/common_audio/*",

webrtc/base/BUILD.gn

@@ -92,10 +92,6 @@ source_set("protobuf_utils") {
 
 # The subset of rtc_base approved for use outside of libjingle.
 rtc_static_library("rtc_base_approved") {
-  # TODO(kjellander): Remove (bugs.webrtc.org/7480)
-  # Enabling GN check triggers a cyclic dependency caused by rate_limiter.cc:
-  #   :rtc_base_approved -> //webrtc/system_wrappers -> :rtc_base_approved
-  check_includes = false
   defines = []
   libs = []
   deps = []
@@ -140,6 +136,8 @@ rtc_static_library("rtc_base_approved") {
     "location.h",
     "md5.cc",
     "md5.h",
+    "md5digest.cc",
+    "md5digest.h",
     "mod_ops.h",
     "onetimeevent.h",
     "optional.cc",
@@ -173,7 +171,6 @@ rtc_static_library("rtc_base_approved") {
     "string_to_number.h",
     "stringencode.cc",
     "stringencode.h",
-    "stringize_macros.h",
     "stringutils.cc",
     "stringutils.h",
     "swap_queue.h",
@@ -679,11 +676,7 @@ if (rtc_include_tests) {
     ]
     public_configs = [ ":rtc_base_tests_utils_exported_config" ]
     deps = [
-      ":rtc_base",
-      ":rtc_base_approved",
       ":rtc_base_tests_utils",
-      "../test:field_trial",
-      "../test:test_support",
     ]
     public_deps = [
       "//testing/gmock",
@@ -712,8 +705,6 @@ if (rtc_include_tests) {
       "gunit.h",
       "httpserver.cc",
       "httpserver.h",
-      "md5digest.cc",
-      "md5digest.h",
       "memory_usage.cc",
       "memory_usage.h",
       "natserver.cc",
@@ -764,9 +755,6 @@ if (rtc_include_tests) {
     deps = [
       ":rtc_base",
       ":rtc_base_tests_main",
-      ":rtc_base_tests_utils",
-      "../system_wrappers:system_wrappers",
-      "../test:test_support",
       "//testing/gtest",
     ]
     if (is_win) {
@@ -813,7 +801,6 @@ if (rtc_include_tests) {
       "safe_compare_unittest.cc",
       "string_to_number_unittest.cc",
       "stringencode_unittest.cc",
-      "stringize_macros_unittest.cc",
       "stringutils_unittest.cc",
       "swap_queue_unittest.cc",
       "thread_annotations_unittest.cc",
@@ -822,13 +809,8 @@ if (rtc_include_tests) {
       "timeutils_unittest.cc",
     ]
     deps = [
-      ":rtc_base",
       ":rtc_base_approved",
       ":rtc_base_tests_main",
-      ":rtc_base_tests_utils",
-      ":rtc_task_queue",
-      "../system_wrappers:system_wrappers",
-      "../test:test_support",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -845,9 +827,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":rtc_base_tests_main",
-      ":rtc_base_tests_utils",
       ":rtc_task_queue",
-      "../test:test_support",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -862,10 +842,8 @@ if (rtc_include_tests) {
       "numerics/percentile_filter_unittest.cc",
     ]
     deps = [
-      ":rtc_base_approved",
       ":rtc_base_tests_main",
       ":rtc_numerics",
-      "../test:test_support",
     ]
   }
 
@@ -924,8 +902,6 @@ if (rtc_include_tests) {
     }
     deps = [
       ":rtc_base_tests_main",
-      ":rtc_base_tests_utils",
-      "../test:test_support",
     ]
     public_deps = [
       ":rtc_base",

webrtc/base/location.h

@@ -13,7 +13,7 @@
 
 #include <string>
 
-#include "webrtc/base/stringize_macros.h"
+#include "webrtc/system_wrappers/include/stringize_macros.h"
 
 namespace rtc {
 

webrtc/base/pathutils.cc

@@ -16,6 +16,7 @@
 #endif  // WEBRTC_WIN
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/fileutils.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/stringutils.h"

webrtc/common_audio/resampler/sinc_resampler_unittest.cc

@@ -19,11 +19,11 @@
 #include <algorithm>
 #include <memory>
 
-#include "webrtc/base/stringize_macros.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/stringize_macros.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 

webrtc/modules/audio_coding/BUILD.gn

@@ -1577,7 +1577,6 @@ if (rtc_include_tests) {
       "../..:webrtc_common",
       "../../api/audio_codecs:audio_codecs_api",
       "../../base:rtc_base_approved",
-      "../../base:rtc_base_tests_utils",
       "../../common_audio",
       "../../test:rtp_test_utils",
       "../rtp_rtcp",

webrtc/system_wrappers/BUILD.gn

@@ -27,6 +27,7 @@ rtc_static_library("system_wrappers") {
     "include/rw_lock_wrapper.h",
     "include/sleep.h",
     "include/static_instance.h",
+    "include/stringize_macros.h",
     "include/timestamp_extrapolator.h",
     "include/trace.h",
     "source/aligned_malloc.cc",
@@ -210,6 +211,7 @@ if (rtc_include_tests) {
       "source/metrics_unittest.cc",
       "source/ntp_time_unittest.cc",
       "source/rtp_to_ntp_estimator_unittest.cc",
+      "source/stringize_macros_unittest.cc",
     ]
     configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
 

webrtc/system_wrappers/include/stringize_macros.h

@@ -15,8 +15,8 @@
 // symbols (or their output) and manipulating preprocessor symbols
 // that define strings.
 
-#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_
-#define WEBRTC_BASE_STRINGIZE_MACROS_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
 
 // This is not very useful as it does not expand defined symbols if
 // called directly. Use its counterpart without the _NO_EXPANSION
@@ -35,4 +35,4 @@
 //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
 #define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
 
-#endif  // WEBRTC_BASE_STRINGIZE_MACROS_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_

webrtc/system_wrappers/source/stringize_macros_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/stringize_macros.h"
+#include "webrtc/system_wrappers/include/stringize_macros.h"
 
 #include "webrtc/test/gtest.h"
 

webrtc/test/fuzzers/BUILD.gn

@@ -276,7 +276,6 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
   ]
   deps = [
     "../../base:rtc_base_approved",
-    "../../base:rtc_base_tests_utils",
     "../../modules/audio_coding:neteq",
     "../../modules/audio_coding:neteq_test_minimal",
     "../../modules/audio_coding:neteq_unittest_tools",

webrtc/voice_engine/BUILD.gn

@@ -181,7 +181,6 @@ if (rtc_include_tests) {
       ":file_player",
       ":voice_engine",
       "../base:rtc_base_approved",
-      "../base:rtc_base_tests_utils",
       "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",