diff --git a/third_party/libwebrtc/README.moz-ff-commit b/third_party/libwebrtc/README.moz-ff-commit index ad20c0e4bef5..bc75b58f06b9 100644 --- a/third_party/libwebrtc/README.moz-ff-commit +++ b/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 diff --git a/third_party/libwebrtc/README.mozilla b/third_party/libwebrtc/README.mozilla index 56c94d1ef566..6ef2d6f2a9d6 100644 --- a/third_party/libwebrtc/README.mozilla +++ b/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. diff --git a/third_party/libwebrtc/pc/BUILD.gn b/third_party/libwebrtc/pc/BUILD.gn index cb9d2978aeb0..fc6950acdbb1 100644 --- a/third_party/libwebrtc/pc/BUILD.gn +++ b/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", diff --git a/third_party/libwebrtc/pc/srtp_session.cc b/third_party/libwebrtc/pc/srtp_session.cc index d01bc3842178..7d1aaf2d6528 100644 --- a/third_party/libwebrtc/pc/srtp_session.cc +++ b/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) { diff --git a/third_party/libwebrtc/pc/srtp_session.h b/third_party/libwebrtc/pc/srtp_session.h index 1c61156fb1eb..048e6656444e 100644 --- a/third_party/libwebrtc/pc/srtp_session.h +++ b/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;