Enable -Wexit-time-destructors and -Wglobal-constructors.

This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.

It also adds the possibility to suppress these warnings because
they trigger in a few places.

The long term goal is to avoid regressions on this and remove all the
suppressions.

Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}

BUILD.gn

@@ -166,6 +166,32 @@ config("common_inherited_config") {
   }
 }
 
+# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
+# as soon as WebRTC compiles without it.
+config("no_exit_time_destructors") {
+  if (is_clang) {
+    cflags = [ "-Wno-exit-time-destructors" ]
+  }
+}
+
+# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
+# as soon as WebRTC compiles without it.
+config("no_global_constructors") {
+  if (is_clang) {
+    cflags = [ "-Wno-global-constructors" ]
+  }
+}
+
+config("rtc_prod_config") {
+  # Ideally, WebRTC production code (but not test code) should have these flags.
+  if (is_clang) {
+    cflags = [
+      "-Wexit-time-destructors",
+      "-Wglobal-constructors",
+    ]
+  }
+}
+
 config("common_config") {
   cflags = []
   cflags_c = []

modules/audio_coding/BUILD.gn

@@ -1156,6 +1156,10 @@ rtc_source_set("neteq_test_tools") {
 
 rtc_source_set("neteq_tools") {
   visibility += webrtc_default_visibility
+  configs += [
+    "../..:no_exit_time_destructors",
+    "../..:no_global_constructors",
+  ]
   sources = [
     "neteq/tools/fake_decode_from_file.cc",
     "neteq/tools/fake_decode_from_file.h",

modules/audio_device/BUILD.gn

@@ -147,6 +147,7 @@ rtc_source_set("audio_device_name") {
 }
 
 rtc_source_set("windows_core_audio_utility") {
+  public_configs = [ "../..:no_global_constructors" ]
   if (is_win && !build_with_chromium) {
     sources = [
       "win/core_audio_utility_win.cc",
@@ -303,6 +304,7 @@ rtc_source_set("audio_device_impl") {
       defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
     } else {
       if (is_linux) {
+        configs += [ "../../..:no_global_constructors" ]
         sources += [
           "linux/alsasymboltable_linux.cc",
           "linux/alsasymboltable_linux.h",
@@ -320,6 +322,7 @@ rtc_source_set("audio_device_impl") {
           defines += [ "WEBRTC_USE_X11" ]
         }
         if (rtc_include_pulse_audio) {
+          configs += [ "../..:no_exit_time_destructors" ]
           sources += [
             "linux/audio_device_pulse_linux.cc",
             "linux/audio_device_pulse_linux.h",

modules/audio_processing/agc2/rnn_vad/BUILD.gn

@@ -9,6 +9,7 @@
 import("../../../../webrtc.gni")
 
 rtc_source_set("rnn_vad") {
+  configs += [ "../../../..:no_global_constructors" ]
   sources = [
     "common.h",
     "features_extraction.cc",

modules/congestion_controller/bbr/BUILD.gn

@@ -23,6 +23,7 @@ rtc_static_library("bbr") {
 
 rtc_source_set("bbr_controller") {
   visibility = [ ":*" ]
+  configs += [ "../../..:no_global_constructors" ]
   sources = [
     "bbr_network_controller.cc",
     "bbr_network_controller.h",

modules/congestion_controller/goog_cc/BUILD.gn

@@ -17,7 +17,10 @@ config("bwe_test_logging") {
 }
 
 rtc_static_library("goog_cc") {
-  configs += [ ":bwe_test_logging" ]
+  configs += [
+    ":bwe_test_logging",
+    "../../..:no_global_constructors",
+  ]
   sources = [
     "goog_cc_factory.cc",
     "goog_cc_network_control.cc",

modules/congestion_controller/pcc/BUILD.gn

@@ -26,6 +26,7 @@ rtc_static_library("pcc_controller") {
     "pcc_network_controller.cc",
     "pcc_network_controller.h",
   ]
+  configs += [ "../../..:no_global_constructors" ]
   deps = [
     ":bitrate_controller",
     ":monitor_interval",

modules/remote_bitrate_estimator/BUILD.gn

@@ -33,6 +33,10 @@ rtc_static_library("remote_bitrate_estimator") {
 
   if (rtc_enable_bwe_test_logging) {
     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+    configs += [
+      "../..:no_exit_time_destructors",
+      "../..:no_global_constructors",
+    ]
     sources += [ "test/bwe_test_logging.cc" ]
   } else {
     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]

modules/video_capture/BUILD.gn

@@ -71,6 +71,8 @@ if (!build_with_chromium) {
     rtc_source_set("video_capture_internal_impl_objc") {
       visibility = [ ":video_capture_internal_impl" ]
 
+      configs += [ "../..:no_global_constructors" ]
+
       deps = [
         ":video_capture_module",
         "../../rtc_base:rtc_base_approved",

ortc/BUILD.gn

@@ -14,6 +14,7 @@ if (is_android) {
 
 rtc_static_library("ortc") {
   defines = []
+  configs += [ "..:no_exit_time_destructors" ]
   sources = [
     "ortcfactory.cc",
     "ortcfactory.h",

pc/BUILD.gn

@@ -117,6 +117,7 @@ rtc_source_set("rtc_pc") {
 rtc_static_library("peerconnection") {
   visibility = [ "*" ]
   cflags = []
+  configs += [ "..:no_exit_time_destructors" ]
   sources = [
     "audiotrack.cc",
     "audiotrack.h",

rtc_base/BUILD.gn

@@ -208,6 +208,10 @@ rtc_source_set("logging") {
       "../../webrtc_overrides/rtc_base/logging.h",
     ]
   } else {
+    configs += [
+      "..:no_exit_time_destructors",
+      "..:no_global_constructors",
+    ]
     sources = [
       "logging.cc",
       "logging.h",
@@ -881,6 +885,10 @@ rtc_static_library("rtc_base_generic") {
     ]
 
     if (is_win) {
+      configs += [
+        "..:no_exit_time_destructors",
+        "..:no_global_constructors",
+      ]
       sources += [
         "win32socketinit.cc",
         "win32socketinit.h",

sdk/BUILD.gn

@@ -531,6 +531,7 @@ if (is_ios || is_mac) {
 
     rtc_static_library("videocodec_objc") {
       visibility = [ "*" ]
+      configs += [ "..:no_global_constructors" ]
       sources = [
         "objc/components/video_codec/RTCCodecSpecificInfoH264+Private.h",
         "objc/components/video_codec/RTCCodecSpecificInfoH264.h",
@@ -585,6 +586,7 @@ if (is_ios || is_mac) {
     }
 
     rtc_static_library("vpx_codec_constants") {
+      configs += [ "..:no_global_constructors" ]
       sources = [
         "objc/api/video_codec/RTCVideoCodecConstants.h",
         "objc/api/video_codec/RTCVideoCodecConstants.mm",
@@ -664,6 +666,7 @@ if (is_ios || is_mac) {
     }
 
     rtc_static_library("mediaconstraints_objc") {
+      configs += [ "..:no_global_constructors" ]
       sources = [
         "objc/api/peerconnection/RTCMediaConstraints+Private.h",
         "objc/api/peerconnection/RTCMediaConstraints.h",
@@ -775,6 +778,10 @@ if (is_ios || is_mac) {
         "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
         "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
       ]
+      configs += [
+        "..:no_exit_time_destructors",
+        "..:no_global_constructors",
+      ]
       sources = [
         "objc/api/peerconnection/RTCAudioSource+Private.h",
         "objc/api/peerconnection/RTCAudioSource.h",

sdk/android/BUILD.gn

@@ -602,6 +602,11 @@ if (is_android) {
     # considered public and are subject to change.
     visibility = [ "*" ]
 
+    configs += [
+      "../..:no_exit_time_destructors",
+      "../..:no_global_constructors",
+    ]
+
     sources = [
       "src/jni/androidnetworkmonitor_jni.h",
       "src/jni/pc/androidnetworkmonitor.h",

system_wrappers/BUILD.gn

@@ -14,6 +14,7 @@ import("../webrtc.gni")
 
 rtc_static_library("system_wrappers") {
   visibility = [ "*" ]
+  configs += [ "..:no_exit_time_destructors" ]
   sources = [
     "include/clock.h",
     "include/cpu_info.h",
@@ -144,6 +145,7 @@ rtc_source_set("field_trial_default") {
 
 rtc_source_set("runtime_enabled_features_default") {
   visibility = [ "*" ]
+  configs += [ "..:no_global_constructors" ]
   sources = [
     "source/runtime_enabled_features_default.cc",
   ]

webrtc.gni

@@ -266,6 +266,7 @@ rtc_common_inherited_config = webrtc_root + ":common_inherited_config"
 # Common configs to remove or add in all rtc targets.
 rtc_remove_configs = []
 rtc_add_configs = rtc_common_configs
+rtc_prod_configs = [ webrtc_root + ":rtc_prod_config" ]
 
 set_defaults("rtc_test") {
   configs = rtc_add_configs
@@ -426,6 +427,10 @@ template("rtc_source_set") {
       }
     }
 
+    if (!defined(testonly) || !testonly) {
+      configs += rtc_prod_configs
+    }
+
     configs += invoker.configs
     configs -= rtc_remove_configs
     configs -= invoker.suppressed_configs
@@ -530,6 +535,10 @@ template("rtc_static_library") {
       }
     }
 
+    if (!defined(testonly) || !testonly) {
+      configs += rtc_prod_configs
+    }
+
     configs += invoker.configs
     configs -= rtc_remove_configs
     configs -= invoker.suppressed_configs