Index: talk/media/sctp/sctpdataengine_unittest.cc |
diff --git a/talk/media/sctp/sctpdataengine_unittest.cc b/talk/media/sctp/sctpdataengine_unittest.cc |
index c540e7c810ca1dfe445c5124a381d60f0a2a1387..d406fa18cdcafe60ffbb31db71572c7235e6d72b 100644 |
--- a/talk/media/sctp/sctpdataengine_unittest.cc |
+++ b/talk/media/sctp/sctpdataengine_unittest.cc |
@@ -45,6 +45,11 @@ |
#include "webrtc/base/ssladapter.h" |
#include "webrtc/base/thread.h" |
+#ifdef HAVE_NSS_SSL_H |
+// TODO(thorcarpenter): Remove after webrtc switches over to BoringSSL. |
+#include "webrtc/base/nssstreamadapter.h" |
+#endif // HAVE_NSS_SSL_H |
+ |
enum { |
MSG_PACKET = 1, |
}; |
@@ -218,6 +223,12 @@ |
// usrsctp uses the NSS random number generator on non-Android platforms, |
// so we need to initialize SSL. |
static void SetUpTestCase() { |
+#ifdef HAVE_NSS_SSL_H |
+ // TODO(thorcarpenter): Remove after webrtc switches over to BoringSSL. |
+ if (!rtc::NSSContext::InitializeSSL(NULL)) { |
+ LOG(LS_WARNING) << "Unabled to initialize NSS."; |
+ } |
+#endif // HAVE_NSS_SSL_H |
} |
virtual void SetUp() { |