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}
diff --git a/.gn b/.gn
index a1fca6d..edfad1a 100644
--- a/.gn
+++ b/.gn
@@ -24,6 +24,7 @@
check_targets = [
"//webrtc/api/*",
"//webrtc/audio/*",
+ "//webrtc/base/*",
"//webrtc/call/*",
"//webrtc/common_video/*",
"//webrtc/common_audio/*",
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index 6280903..61d9061 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -92,6 +92,10 @@
# 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 = []
@@ -136,8 +140,6 @@
"location.h",
"md5.cc",
"md5.h",
- "md5digest.cc",
- "md5digest.h",
"mod_ops.h",
"onetimeevent.h",
"optional.cc",
@@ -171,6 +173,7 @@
"string_to_number.h",
"stringencode.cc",
"stringencode.h",
+ "stringize_macros.h",
"stringutils.cc",
"stringutils.h",
"swap_queue.h",
@@ -670,7 +673,11 @@
]
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",
@@ -699,6 +706,8 @@
"gunit.h",
"httpserver.cc",
"httpserver.h",
+ "md5digest.cc",
+ "md5digest.h",
"memory_usage.cc",
"memory_usage.h",
"natserver.cc",
@@ -749,6 +758,9 @@
deps = [
":rtc_base",
":rtc_base_tests_main",
+ ":rtc_base_tests_utils",
+ "../system_wrappers:system_wrappers",
+ "../test:test_support",
"//testing/gtest",
]
if (is_win) {
@@ -795,6 +807,7 @@
"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",
@@ -803,8 +816,13 @@
"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).
@@ -821,7 +839,9 @@
]
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).
@@ -836,8 +856,10 @@
"numerics/percentile_filter_unittest.cc",
]
deps = [
+ ":rtc_base_approved",
":rtc_base_tests_main",
":rtc_numerics",
+ "../test:test_support",
]
}
@@ -897,6 +919,8 @@
}
deps = [
":rtc_base_tests_main",
+ ":rtc_base_tests_utils",
+ "../test:test_support",
]
public_deps = [
":rtc_base",
diff --git a/webrtc/base/location.h b/webrtc/base/location.h
index a541bbe..541be9a 100644
--- a/webrtc/base/location.h
+++ b/webrtc/base/location.h
@@ -13,7 +13,7 @@
#include <string>
-#include "webrtc/system_wrappers/include/stringize_macros.h"
+#include "webrtc/base/stringize_macros.h"
namespace rtc {
diff --git a/webrtc/base/pathutils.cc b/webrtc/base/pathutils.cc
index 75dabb5..3036774 100644
--- a/webrtc/base/pathutils.cc
+++ b/webrtc/base/pathutils.cc
@@ -16,7 +16,6 @@
#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"
diff --git a/webrtc/system_wrappers/include/stringize_macros.h b/webrtc/base/stringize_macros.h
similarity index 86%
rename from webrtc/system_wrappers/include/stringize_macros.h
rename to webrtc/base/stringize_macros.h
index 9c8e7e9..7e2f44d 100644
--- a/webrtc/system_wrappers/include/stringize_macros.h
+++ b/webrtc/base/stringize_macros.h
@@ -15,8 +15,8 @@
// symbols (or their output) and manipulating preprocessor symbols
// that define strings.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
+#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_
+#define WEBRTC_BASE_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_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
+#endif // WEBRTC_BASE_STRINGIZE_MACROS_H_
diff --git a/webrtc/system_wrappers/source/stringize_macros_unittest.cc b/webrtc/base/stringize_macros_unittest.cc
similarity index 94%
rename from webrtc/system_wrappers/source/stringize_macros_unittest.cc
rename to webrtc/base/stringize_macros_unittest.cc
index 8b103c5..d0ba113 100644
--- a/webrtc/system_wrappers/source/stringize_macros_unittest.cc
+++ b/webrtc/base/stringize_macros_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/system_wrappers/include/stringize_macros.h"
+#include "webrtc/base/stringize_macros.h"
#include "webrtc/test/gtest.h"
diff --git a/webrtc/common_audio/resampler/sinc_resampler_unittest.cc b/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
index 82546d0..990580d 100644
--- a/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
+++ b/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"
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 9c7e5ee..9bdd1fc 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1577,6 +1577,7 @@
"../..: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",
diff --git a/webrtc/system_wrappers/BUILD.gn b/webrtc/system_wrappers/BUILD.gn
index bbb1b12..1ecb9e8 100644
--- a/webrtc/system_wrappers/BUILD.gn
+++ b/webrtc/system_wrappers/BUILD.gn
@@ -30,7 +30,6 @@
"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",
@@ -169,7 +168,6 @@
"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" ]
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index ed84e48..c184452 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -276,6 +276,7 @@
]
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",
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index ca774f2..40d9ed3 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -181,6 +181,7 @@
":file_player",
":voice_engine",
"../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../test:test_common",
"//testing/gmock",
"//testing/gtest",