| 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() {
|
|
|