Index: talk/media/sctp/sctpdataengine_unittest.cc |
diff --git a/talk/media/sctp/sctpdataengine_unittest.cc b/talk/media/sctp/sctpdataengine_unittest.cc |
index d406fa18cdcafe60ffbb31db71572c7235e6d72b..c540e7c810ca1dfe445c5124a381d60f0a2a1387 100644 |
--- a/talk/media/sctp/sctpdataengine_unittest.cc |
+++ b/talk/media/sctp/sctpdataengine_unittest.cc |
@@ -45,11 +45,6 @@ |
#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, |
}; |
@@ -223,12 +218,6 @@ class SctpDataMediaChannelTest : public testing::Test, |
// 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() { |