Added DtlsCertificate, a ref counted object owning an SSLIdentity

talk/session/media/mediasession_unittest.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 20 matching lines @@
 #include "talk/media/base/codec.h"
 #include "talk/media/base/testutils.h"
 #include "webrtc/p2p/base/constants.h"
 #include "webrtc/p2p/base/transportdescription.h"
 #include "webrtc/p2p/base/transportinfo.h"
 #include "talk/session/media/mediasession.h"
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/messagedigest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 
 #ifdef HAVE_SRTP
 #define ASSERT_CRYPTO(cd, s, cs) \
     ASSERT_EQ(cricket::CT_NONE, cd->crypto_required()); \
     ASSERT_EQ(s, cd->cryptos().size()); \
     ASSERT_EQ(std::string(cs), cd->cryptos()[0].cipher_suite)
 #else
 #define ASSERT_CRYPTO(cd, s, cs) \
   ASSERT_EQ(cricket::CT_NONE, cd->crypto_required()); \
@@ skipping 158 matching lines @@
   std::vector<std::string> codec_names;
   for (const auto& codec : codecs) {
     codec_names.push_back(codec.name);
   }
   return codec_names;
 }
 
 class MediaSessionDescriptionFactoryTest : public testing::Test {
  public:
   MediaSessionDescriptionFactoryTest()
-      : f1_(&tdf1_), f2_(&tdf2_), id1_("id1"), id2_("id2") {
+      : f1_(&tdf1_),
+        f2_(&tdf2_),
+        cert1_(webrtc::DtlsCertificate::Create(
+            rtc::scoped_ptr<rtc::SSLIdentity>(
+                new rtc::FakeSSLIdentity("id1")).Pass())),
+        cert2_(webrtc::DtlsCertificate::Create(
+            rtc::scoped_ptr<rtc::SSLIdentity>(
+                new rtc::FakeSSLIdentity("id2")).Pass())) {
     f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1));
     f1_.set_data_codecs(MAKE_VECTOR(kDataCodecs1));
     f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2));
     f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2));
     f2_.set_data_codecs(MAKE_VECTOR(kDataCodecs2));
-    tdf1_.set_identity(&id1_);
-    tdf2_.set_identity(&id2_);
+    tdf1_.set_certificate(cert1_);
+    tdf2_.set_certificate(cert2_);
   }
 
   // Create a video StreamParamsVec object with:
   // - one video stream with 3 simulcast streams and FEC,
   StreamParamsVec CreateComplexVideoStreamParamsVec() {
     SsrcGroup sim_group("SIM", MAKE_VECTOR(kSimSsrc));
     SsrcGroup fec_group1("FEC", MAKE_VECTOR(kFec1Ssrc));
     SsrcGroup fec_group2("FEC", MAKE_VECTOR(kFec2Ssrc));
     SsrcGroup fec_group3("FEC", MAKE_VECTOR(kFec3Ssrc));
 
@@ skipping 218 matching lines @@
         return false;
     }
     return true;
   }
 
  protected:
   MediaSessionDescriptionFactory f1_;
   MediaSessionDescriptionFactory f2_;
   TransportDescriptionFactory tdf1_;
   TransportDescriptionFactory tdf2_;
-  rtc::FakeSSLIdentity id1_;
-  rtc::FakeSSLIdentity id2_;
+  rtc::scoped_refptr<webrtc::DtlsCertificate> cert1_;
+  rtc::scoped_refptr<webrtc::DtlsCertificate> cert2_;
 };
 
 // Create a typical audio offer, and ensure it matches what we expect.
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOffer) {

[Henrik Grunell WebRTC, 2015/08/12 14:46:30]

Should we add new tests?

[hbos, 2015/08/14 14:09:39]

Here, the identity -> certificate change should not change what the tests need
to cover.

   f1_.set_secure(SEC_ENABLED);
   rtc::scoped_ptr<SessionDescription> offer(
       f1_.CreateOffer(MediaSessionOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
   const ContentInfo* ac = offer->GetContentByName("audio");
   const ContentInfo* vc = offer->GetContentByName("video");
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc == NULL);
   EXPECT_EQ(std::string(NS_JINGLE_RTP), ac->type);
   const AudioContentDescription* acd =
@@ skipping 1774 matching lines @@
   offer.reset(f1_.CreateOffer(options, NULL));
   ASSERT_TRUE(offer.get() != NULL);
   audio_content = offer->GetContentByName("audio");
   EXPECT_TRUE(VerifyNoCNCodecs(audio_content));
   rtc::scoped_ptr<SessionDescription> answer(
       f1_.CreateAnswer(offer.get(), options, NULL));
   ASSERT_TRUE(answer.get() != NULL);
   audio_content = answer->GetContentByName("audio");
   EXPECT_TRUE(VerifyNoCNCodecs(audio_content));
 }