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}
This commit is contained in:
kjellander 2017-04-19 08:37:36 -07:00 committed by Commit bot
parent 800daef50a
commit ed754e71ae
11 changed files with 36 additions and 11 deletions

1
.gn
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@ -24,6 +24,7 @@ secondary_source = "//build/secondary/"
check_targets = [
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/base/*",
"//webrtc/call/*",
"//webrtc/common_video/*",
"//webrtc/common_audio/*",

View File

@ -92,6 +92,10 @@ 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 = []
@ -136,8 +140,6 @@ rtc_static_library("rtc_base_approved") {
"location.h",
"md5.cc",
"md5.h",
"md5digest.cc",
"md5digest.h",
"mod_ops.h",
"onetimeevent.h",
"optional.cc",
@ -171,6 +173,7 @@ rtc_static_library("rtc_base_approved") {
"string_to_number.h",
"stringencode.cc",
"stringencode.h",
"stringize_macros.h",
"stringutils.cc",
"stringutils.h",
"swap_queue.h",
@ -670,7 +673,11 @@ 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",
@ -699,6 +706,8 @@ if (rtc_include_tests) {
"gunit.h",
"httpserver.cc",
"httpserver.h",
"md5digest.cc",
"md5digest.h",
"memory_usage.cc",
"memory_usage.h",
"natserver.cc",
@ -749,6 +758,9 @@ 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) {
@ -795,6 +807,7 @@ 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",
@ -803,8 +816,13 @@ 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).
@ -821,7 +839,9 @@ 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).
@ -836,8 +856,10 @@ if (rtc_include_tests) {
"numerics/percentile_filter_unittest.cc",
]
deps = [
":rtc_base_approved",
":rtc_base_tests_main",
":rtc_numerics",
"../test:test_support",
]
}
@ -897,6 +919,8 @@ if (rtc_include_tests) {
}
deps = [
":rtc_base_tests_main",
":rtc_base_tests_utils",
"../test:test_support",
]
public_deps = [
":rtc_base",

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

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@ -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"

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@ -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_

View File

@ -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"

View File

@ -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"

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@ -1577,6 +1577,7 @@ 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",

View File

@ -30,7 +30,6 @@ 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",
@ -169,7 +168,6 @@ 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" ]

View File

@ -276,6 +276,7 @@ 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",

View File

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