Attachment 342748 Details for Bug 186615 – Patch

[patch] Patch

bug-186615-20180614113805.patch (text/plain), 13.82 KB, created by youenn fablet on 2018-06-14 11:38:06 PDT

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Creator: youenn fablet

Created: 2018-06-14 11:38:06 PDT

Size: 13.82 KB

patch

obsolete

>Subversion Revision: 232840
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index a656b0754484de2f0776e7495abe2f87497d86cc..eca1e50d46b68460891feb98c7822d959044fe47 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,27 @@
>+2018-06-14  Youenn Fablet  <youenn@apple.com>
>+
>+        Activate -Wexit-time-destructors -and Wglobal-constructors in libwebrtc
>+        https://bugs.webkit.org/show_bug.cgi?id=186615
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        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  <youenn@apple.com>
> 
>         Eliminate static initializers in libwebrtc.dylib
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>index 65289559a10eb6da7206fb0402296af767a0b732..c80ec781299181b14e16700951388cbb2571ff57 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>@@ -50,7 +50,7 @@ GCC_WARN_UNUSED_FUNCTION = YES;
> 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);
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>index d8cf487e91246b2e42b0cdadbe5f1bb00370f4c5..c14a46cb8041d403b3a383233bfe2c48bab0dadc 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>@@ -23,7 +23,7 @@ HEADER_SEARCH_PATHS = Source Source/third_party/jsoncpp/source/include Source/th
> 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);
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig
>index b910177b80d036d7090de628308b67e29eb2f0d7..d058c76406ab438765767af1a1883639ba8fb9cf 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/opus.xcconfig
>@@ -12,7 +12,7 @@ HEADER_SEARCH_PATHS = Source/third_party/opus/src/include Source/third_party/opu
> 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);
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig
>index bea021696624249bbd2d862d57d7ebb55b50f1b8..b0919adfee69b5c29af8116ccd13946ccc860f6e 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/usrsctp.xcconfig
>@@ -10,6 +10,6 @@ HEADER_SEARCH_PATHS = Source/ Source/third_party/usrsctp/usrsctplib/usrsctplib/n
> 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__;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h
>index f23492969395a09518622d3194e93bca7e8ab225..a7271fcbacc307156ce64fe0d68955a630e0e6d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h
>@@ -103,7 +103,7 @@ using SphericalPointf = SphericalPoint<float>;
> 
> // 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;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>index c814716b44860994105789a125ad9df20d775413..204b7ed0adb67776da7f9d1d1b78bf6e8fd37abc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>@@ -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 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>index 95e49cc0055cc03db8214310e3821f705f139f11..d5715528169d8302516976caec8b7b1c5b052a48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>@@ -106,7 +106,7 @@ bool LogMessage::log_to_stderr_ = true;
> 
> 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;
>@@ -119,7 +119,17 @@ CriticalSection& logCriticalScope() {
> // 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_;
>@@ -133,7 +143,9 @@ LogMessage::LogMessage(const char* file,
>     : severity_(sev), tag_(kLibjingle) {
> 
>   static std::once_flag callLogCriticalScopeOnce;
>-  std::call_once(callLogCriticalScopeOnce,[] { logCriticalScope(); });
>+  std::call_once(callLogCriticalScopeOnce,[] {
>+    logCriticalScope();
>+  });
> 
>   if (timestamp_) {
>     // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp
>@@ -220,7 +232,7 @@ LogMessage::~LogMessage() {
>   }
> 
>   CritScope cs(&logCriticalScope());
>-  for (auto& kv : streams_) {
>+  for (auto& kv : streamList()) {
>     if (severity_ >= kv.second) {
>       kv.first->OnLogMessage(str);
>     }
>@@ -258,7 +270,7 @@ void LogMessage::SetLogToStderr(bool log_to_stderr) {
> 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 +280,16 @@ int LogMessage::GetLogToStream(LogSink* stream) {
> 
> 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 +361,7 @@ void LogMessage::ConfigureLogging(const char* params) {
> 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;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>index e3879fc05bbf009131219e36b4ef5fb270e608e2..8a012080135d68e356dbd089326865320af6732b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>@@ -207,9 +207,6 @@ class LogMessage {
>   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 @@ class LogMessage {
>   // 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_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>index d8771e728d24d09b3dfdbe2b83d6924ee279c71c..1b12cd49e6e07dc85b4d6ee41506942c308ad120 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>@@ -1043,11 +1043,9 @@ void VirtualSocketServer::UpdateDelayDistribution() {
>   }
> }
> 
>-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
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>index 631974d9bcf36d4871af67d5f06d440d9c63f087..21d963f5d7c0700dd6148e9ce173c2b6c20f8d3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>@@ -221,8 +221,9 @@ Clock* Clock::GetRealTimeClock() {
>   }
>   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)

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Attachments on bug 186615: 342722 | 342724 | 342725 | 342742 | 342748 | 342754 | 342769