commit c1eebfa107d6e7c4f7be6502ec36125553bf578c
author thorcarpenter@google.com <thorcarpenter@google.com> Fri Sep 19 17:54:00 2014 +0000
committer thorcarpenter@google.com <thorcarpenter@google.com> Fri Sep 19 17:54:00 2014 +0000
tree 5cf03d36206b63017488fcdf03365a4ec8ec03af
parent e65812427db6609f4327dabc5336a9d07d7a182d

Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc.

R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/22699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/libjingle_tests.gyp

diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index 45f08a9..44adec3 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -153,6 +153,11 @@
       ],
       'conditions': [
         ['OS=="win"', {
+          'dependencies': [
+            '<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
+            '<(DEPTH)/third_party/nss/nss.gyp:nspr',
+            '<(DEPTH)/third_party/nss/nss.gyp:nss',
+          ],
           'msvs_settings': {
             'VCLinkerTool': {
               'AdditionalDependencies': [

talk/media/sctp/sctpdataengine_unittest.cc

diff --git a/talk/media/sctp/sctpdataengine_unittest.cc b/talk/media/sctp/sctpdataengine_unittest.cc
index eedb3cc..fb00650 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
     rtc::InitializeSSL();
   }