Diff
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-06-14 23:21:06 UTC (rev 232858)
@@ -1,5 +1,29 @@
2018-06-14 Youenn Fablet <[email protected]>
+ Activate -Wexit-time-destructors -and Wglobal-constructors in libwebrtc
+ https://bugs.webkit.org/show_bug.cgi?id=186615
+
+ Reviewed by Darin Adler.
+
+ Update xcconfig files to activate these compile flags.
+ Also enable -Wthread-safety since libwebrtc code is using some related attributes.
+ Update libwebrtc code base to accomodate these flags.
+
+ * Configurations/libwebrtc.xcconfig:
+ * Configurations/opus.xcconfig:
+ * Configurations/usrsctp.xcconfig:
+ * Source/webrtc/modules/audio_processing/beamformer/array_util.h:
+ (webrtc::DegreesToRadians): Make function constexpr.
+ * Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc:
+ Make sure the destructor is never called.
+ * Source/webrtc/rtc_base/logging.cc:
+ Update code to move streams_ from a static class member to a regular static function variable.
+ * Source/webrtc/rtc_base/logging.h:
+ * Source/webrtc/system_wrappers/source/clock.cc:
+ Make sure the destructor is never called.
+
+2018-06-14 Youenn Fablet <[email protected]>
+
Eliminate static initializers in libwebrtc.dylib
https://bugs.webkit.org/show_bug.cgi?id=186570
<rdar://problem/41054874>
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig 2018-06-14 23:21:06 UTC (rev 232858)
@@ -50,7 +50,7 @@
GCC_WARN_UNUSED_VARIABLE = YES;
PREBINDING = NO;
STRIP_INSTALLED_PRODUCT = NO;
-WARNING_CFLAGS = -Wexit-time-destructors -Wglobal-constructors;
+WARNING_CFLAGS = -Wexit-time-destructors -Wglobal-constructors -Wthread-safety;
SUPPORTED_PLATFORMS = iphoneos iphonesimulator macosx;
VALID_ARCHS = $(ARCHS_STANDARD_64_BIT);
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig 2018-06-14 23:21:06 UTC (rev 232858)
@@ -23,7 +23,7 @@
PUBLIC_HEADERS_FOLDER_PATH = /usr/local/include/libwebrtc;
USE_HEADERMAP = NO;
INSTALLHDRS_SCRIPT_PHASE = YES;
-WARNING_CFLAGS = -Wno-deprecated-declarations;
+WARNING_CFLAGS = -Wno-deprecated-declarations $(inherited);
// FIXME: Set WEBRTC_USE_BUILTIN_ISAC_FIX and WEBRTC_USE_BUILTIN_ISAC_FLOAT for iOS and Mac
GCC_PREPROCESSOR_DEFINITIONS = GTEST_RELATIVE_PATH WEBRTC_OPUS_SUPPORT_120MS_PTIME=0 WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_USE_BUILTIN_ILBC WEBRTC_CODEC_ILBC WEBRTC_USE_BUILTIN_OPUS WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722 WEBRTC_OPUS_VARIABLE_COMPLEXITY=0 WEBRTC_USE_BUILTIN_ISAC_FIX=1 WEBRTC_USE_BUILTIN_ISAC_FLOAT=0 $(inherited);
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig 2018-06-14 23:21:06 UTC (rev 232858)
@@ -12,7 +12,7 @@
INSTALL_PATH = /usr/local/lib;
PUBLIC_HEADERS_FOLDER_PATH = /usr/local/include/libwebrtc;
USE_HEADERMAP = NO;
-WARNING_CFLAGS = -Wno-incompatible-pointer-types
+WARNING_CFLAGS = -Wno-incompatible-pointer-types $(inherited)
GCC_PREPROCESSOR_DEFINITIONS = VAR_ARRAYS OPUS_BUILD OPUS_EXPORT= HAVE_LRINT HAVE_LRINTF OPUS_X86_MAY_HAVE_SSE2 $(inherited);
GCC_PREPROCESSOR_DEFINITIONS[arch=arm64*] = $(inherited) OPUS_ARM_PRESUME_AARCH64_NEON_INTR;
GCC_PREPROCESSOR_DEFINITIONS[arch=x86_64] = $(inherited) $(SSE4_FLAG);
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig 2018-06-14 23:21:06 UTC (rev 232858)
@@ -10,6 +10,6 @@
INSTALL_PATH = /usr/local/lib;
PUBLIC_HEADERS_FOLDER_PATH = /usr/local/include/libwebrtc;
USE_HEADERMAP = NO;
-WARNING_CFLAGS = -Wno-deprecated-declarations;
+WARNING_CFLAGS = -Wno-deprecated-declarations $(inherited);
GCC_PREPROCESSOR_DEFINITIONS = SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin $(inherited);
OTHER_CFLAGS = -UINET -UINET6 -U__APPLE__;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h 2018-06-14 23:21:06 UTC (rev 232858)
@@ -103,7 +103,7 @@
// Helper functions to transform degrees to radians and the inverse.
template <typename T>
-T DegreesToRadians(T angle_degrees) {
+constexpr T DegreesToRadians(T angle_degrees) {
return M_PI * angle_degrees / 180;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc 2018-06-14 23:21:06 UTC (rev 232858)
@@ -19,8 +19,9 @@
namespace webrtc {
RtpFeedback* NullObjectRtpFeedback() {
- static NullRtpFeedback null_rtp_feedback;
- return &null_rtp_feedback;
+ static std::aligned_storage<sizeof(NullRtpFeedback), std::alignment_of<NullRtpFeedback>::value>::type null_rtp_feedback_storage;
+ static NullRtpFeedback* null_rtp_feedback = new (&null_rtp_feedback_storage) NullRtpFeedback;
+ return null_rtp_feedback;
}
namespace RtpUtility {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc 2018-06-14 23:21:06 UTC (rev 232858)
@@ -106,7 +106,7 @@
namespace {
-// Global lock for log subsystem, only needed to serialize access to streams_.
+// Global lock for log subsystem, only needed to serialize access to streamList().
CriticalSection& logCriticalScope() {
static std::aligned_storage<sizeof(CriticalSection), std::alignment_of<CriticalSection>::value>::type g_log_crit_storage;
static CriticalSection* g_log_crit = new (&g_log_crit_storage) CriticalSection;
@@ -119,8 +119,18 @@
// Note: we explicitly do not clean this up, because of the uncertain ordering
// of destructors at program exit. Let the person who sets the stream trigger
// cleanup by setting to null, or let it leak (safe at program exit).
-LogMessage::StreamList LogMessage::streams_ RTC_GUARDED_BY(logCriticalScope());
+typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
+typedef std::list<StreamAndSeverity> StreamList;
+
+// The output streams and their associated severities
+StreamList& streamList()
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
+ static std::aligned_storage<sizeof(StreamList), std::alignment_of<StreamList>::value>::type stream_list_storage;
+ static StreamList* stream_list = new (&stream_list_storage) StreamList;
+ return *stream_list;
+}
+
// Boolean options default to false (0)
bool LogMessage::thread_, LogMessage::timestamp_;
@@ -220,7 +230,7 @@
}
CritScope cs(&logCriticalScope());
- for (auto& kv : streams_) {
+ for (auto& kv : streamList()) {
if (severity_ >= kv.second) {
kv.first->OnLogMessage(str);
}
@@ -258,7 +268,7 @@
int LogMessage::GetLogToStream(LogSink* stream) {
CritScope cs(&logCriticalScope());
LoggingSeverity sev = LS_NONE;
- for (auto& kv : streams_) {
+ for (auto& kv : streamList()) {
if (!stream || stream == kv.first) {
sev = std::min(sev, kv.second);
}
@@ -268,15 +278,16 @@
void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) {
CritScope cs(&logCriticalScope());
- streams_.push_back(std::make_pair(stream, min_sev));
+ streamList().push_back(std::make_pair(stream, min_sev));
UpdateMinLogSeverity();
}
void LogMessage::RemoveLogToStream(LogSink* stream) {
CritScope cs(&logCriticalScope());
- for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
+ auto& streams = streamList();
+ for (auto it = streams.begin(); it != streams.end(); ++it) {
if (stream == it->first) {
- streams_.erase(it);
+ streams.erase(it);
break;
}
}
@@ -348,7 +359,7 @@
void LogMessage::UpdateMinLogSeverity()
RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
LoggingSeverity min_sev = dbg_sev_;
- for (auto& kv : streams_) {
+ for (auto& kv : streamList()) {
min_sev = std::min(dbg_sev_, kv.second);
}
min_sev_ = min_sev;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h 2018-06-14 23:21:06 UTC (rev 232858)
@@ -207,9 +207,6 @@
static void ConfigureLogging(const char* params);
private:
- typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
- typedef std::list<StreamAndSeverity> StreamList;
-
// Updates min_sev_ appropriately when debug sinks change.
static void UpdateMinLogSeverity();
@@ -238,9 +235,6 @@
// ctx_sev_ is the minimum level at which file context is displayed
static LoggingSeverity min_sev_, dbg_sev_, ctx_sev_;
- // The output streams and their associated severities
- static StreamList streams_;
-
// Flags for formatting options
static bool thread_, timestamp_;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc 2018-06-14 23:21:06 UTC (rev 232858)
@@ -1043,11 +1043,9 @@
}
}
-static double PI = 4 * atan(1.0);
-
static double Normal(double x, double mean, double stddev) {
double a = (x - mean) * (x - mean) / (2 * stddev * stddev);
- return exp(-a) / (stddev * sqrt(2 * PI));
+ return exp(-a) / (stddev * sqrt(2 * M_PI));
}
#if 0 // static unused gives a warning
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc (232857 => 232858)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc 2018-06-14 22:08:22 UTC (rev 232857)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc 2018-06-14 23:21:06 UTC (rev 232858)
@@ -221,8 +221,9 @@
}
return g_shared_clock;
#elif defined(WEBRTC_POSIX)
- static UnixRealTimeClock clock;
- return &clock;
+ static std::aligned_storage<sizeof(UnixRealTimeClock), std::alignment_of<UnixRealTimeClock>::value>::type clock_storage;
+ static UnixRealTimeClock* clock = new (&clock_storage) UnixRealTimeClock;
+ return clock;
#else // defined(WEBRTC_POSIX)
return nullptr;
#endif // !defined(WEBRTC_WIN) || defined(WEBRTC_POSIX)
