Bug 1766646 - Vendor libwebrtc from 2d6c4d0712

Upstream commit: https://webrtc.googlesource.com/src/+/2d6c4d07129f88d599fc41965c22c7faaf09e793
    Move global libsrtp usage count into a singleton class

    Avoids using webrtc::GlobalMutex. Since state is allocated on first
    use and never destroyed, we avoid an exit-time destructor when
    building with absl::Mutex.

    Bug: webrtc:11567
    Change-Id: Ib9c6480ab0474e37a853460115b35d961b93009c
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258080
    Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
    Commit-Queue: Niels Moller <nisse@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#36455}

third_party/libwebrtc/README.moz-ff-commit

@@ -12903,3 +12903,6 @@ e10a9f609a
 # MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 901bf55ef7
+# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+2d6c4d0712

third_party/libwebrtc/README.mozilla

@@ -8614,3 +8614,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T15:48:02.099292.
 # python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T15:49:07.945576.
+# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
+libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T15:49:49.795451.

third_party/libwebrtc/pc/BUILD.gn

@@ -581,6 +581,7 @@ rtc_source_set("srtp_session") {
     "../rtc_base",
     "../rtc_base:checks",
     "../rtc_base:logging",
+    "../rtc_base:macromagic",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:stringutils",
     "../rtc_base/synchronization:mutex",

third_party/libwebrtc/pc/srtp_session.cc

@@ -27,6 +27,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/string_encode.h"
+#include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/metrics.h"
 #include "third_party/libsrtp/include/srtp.h"
@@ -34,6 +35,81 @@
 
 namespace cricket {
 
+namespace {
+class LibSrtpInitializer {
+ public:
+  // Returns singleton instance of this class. Instance created on first use,
+  // and never destroyed.
+  static LibSrtpInitializer& Get() {
+    static LibSrtpInitializer* const instance = new LibSrtpInitializer();
+    return *instance;
+  }
+  void ProhibitLibsrtpInitialization();
+
+  // These methods are responsible for initializing libsrtp (if the usage count
+  // is incremented from 0 to 1) or deinitializing it (when decremented from 1
+  // to 0).
+  //
+  // Returns true if successful (will always be successful if already inited).
+  bool IncrementLibsrtpUsageCountAndMaybeInit(
+      srtp_event_handler_func_t* handler);
+  void DecrementLibsrtpUsageCountAndMaybeDeinit();
+
+ private:
+  LibSrtpInitializer() = default;
+
+  webrtc::Mutex mutex_;
+  int usage_count_ RTC_GUARDED_BY(mutex_) = 0;
+};
+
+void LibSrtpInitializer::ProhibitLibsrtpInitialization() {
+  webrtc::MutexLock lock(&mutex_);
+  ++usage_count_;
+}
+
+bool LibSrtpInitializer::IncrementLibsrtpUsageCountAndMaybeInit(
+    srtp_event_handler_func_t* handler) {
+  webrtc::MutexLock lock(&mutex_);
+
+  RTC_DCHECK_GE(usage_count_, 0);
+  if (usage_count_ == 0) {
+    int err;
+    err = srtp_init();
+    if (err != srtp_err_status_ok) {
+      RTC_LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;
+      return false;
+    }
+
+    err = srtp_install_event_handler(handler);
+    if (err != srtp_err_status_ok) {
+      RTC_LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
+      return false;
+    }
+
+    err = external_crypto_init();
+    if (err != srtp_err_status_ok) {
+      RTC_LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
+      return false;
+    }
+  }
+  ++usage_count_;
+  return true;
+}
+
+void LibSrtpInitializer::DecrementLibsrtpUsageCountAndMaybeDeinit() {
+  webrtc::MutexLock lock(&mutex_);
+
+  RTC_DCHECK_GE(usage_count_, 1);
+  if (--usage_count_ == 0) {
+    int err = srtp_shutdown();
+    if (err) {
+      RTC_LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
+    }
+  }
+}
+
+}  // namespace
+
 using ::webrtc::ParseRtpSequenceNumber;
 
 // One more than the maximum libsrtp error code. Required by
@@ -53,7 +129,7 @@ SrtpSession::~SrtpSession() {
     srtp_dealloc(session_);
   }
   if (inited_) {
-    DecrementLibsrtpUsageCountAndMaybeDeinit();
+    LibSrtpInitializer::Get().DecrementLibsrtpUsageCountAndMaybeDeinit();
   }
 }
 
@@ -354,7 +430,8 @@ bool SrtpSession::SetKey(int type,
 
   // This is the first time we need to actually interact with libsrtp, so
   // initialize it if needed.
-  if (IncrementLibsrtpUsageCountAndMaybeInit()) {
+  if (LibSrtpInitializer::Get().IncrementLibsrtpUsageCountAndMaybeInit(
+          &SrtpSession::HandleEventThunk)) {
     inited_ = true;
   } else {
     return false;
@@ -377,54 +454,8 @@ bool SrtpSession::UpdateKey(int type,
   return DoSetKey(type, cs, key, len, extension_ids);
 }
 
-ABSL_CONST_INIT int g_libsrtp_usage_count = 0;
-ABSL_CONST_INIT webrtc::GlobalMutex g_libsrtp_lock(absl::kConstInit);
-
 void ProhibitLibsrtpInitialization() {
-  webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
-  ++g_libsrtp_usage_count;
-}
-
-// static
-bool SrtpSession::IncrementLibsrtpUsageCountAndMaybeInit() {
-  webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
-
-  RTC_DCHECK_GE(g_libsrtp_usage_count, 0);
-  if (g_libsrtp_usage_count == 0) {
-    int err;
-    err = srtp_init();
-    if (err != srtp_err_status_ok) {
-      RTC_LOG(LS_ERROR) << "Failed to init SRTP, err=" << err;
-      return false;
-    }
-
-    err = srtp_install_event_handler(&SrtpSession::HandleEventThunk);
-    if (err != srtp_err_status_ok) {
-      RTC_LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
-      return false;
-    }
-
-    err = external_crypto_init();
-    if (err != srtp_err_status_ok) {
-      RTC_LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
-      return false;
-    }
-  }
-  ++g_libsrtp_usage_count;
-  return true;
-}
-
-// static
-void SrtpSession::DecrementLibsrtpUsageCountAndMaybeDeinit() {
-  webrtc::GlobalMutexLock ls(&g_libsrtp_lock);
-
-  RTC_DCHECK_GE(g_libsrtp_usage_count, 1);
-  if (--g_libsrtp_usage_count == 0) {
-    int err = srtp_shutdown();
-    if (err) {
-      RTC_LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
-    }
-  }
+  LibSrtpInitializer::Get().ProhibitLibsrtpInitialization();
 }
 
 void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {

third_party/libwebrtc/pc/srtp_session.h

@@ -120,14 +120,6 @@ class SrtpSession {
   // for debugging.
   void DumpPacket(const void* buf, int len, bool outbound);
 
-  // These methods are responsible for initializing libsrtp (if the usage count
-  // is incremented from 0 to 1) or deinitializing it (when decremented from 1
-  // to 0).
-  //
-  // Returns true if successful (will always be successful if already inited).
-  static bool IncrementLibsrtpUsageCountAndMaybeInit();
-  static void DecrementLibsrtpUsageCountAndMaybeDeinit();
-
   void HandleEvent(const srtp_event_data_t* ev);
   static void HandleEventThunk(srtp_event_data_t* ev);
 
@@ -142,7 +134,6 @@ class SrtpSession {
   int rtcp_auth_tag_len_ = 0;
 
   bool inited_ = false;
-  static webrtc::GlobalMutex lock_;
   int last_send_seq_num_ = -1;
   bool external_auth_active_ = false;
   bool external_auth_enabled_ = false;