Add IsAcceptableCipher, use instead of GetDefaultCipher.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1418 matching lines @@
       kMaxWaitForStatsMs);
 
   MediaStreamTrackInterface* local_video_track =
       local_streams->at(0)->GetVideoTracks()[0];
   EXPECT_TRUE_WAIT(
       initializing_client()->GetBytesSentStats(local_video_track) > 0,
       kMaxWaitForStatsMs);
 }
 
 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
-// Disabled due to new BoringSSLL version, see  webrtc:5634
-TEST_F(P2PTestConductor, DISABLED_GetDtls12None) {
+TEST_F(P2PTestConductor, GetDtls12None) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   PeerConnectionFactory::Options recv_options;
   recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(
       CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
   rtc::scoped_refptr<webrtc::FakeMetricsObserver>
       init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
   initializing_client()->pc()->RegisterUMAObserver(init_observer);
   LocalP2PTest();
 
-  EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
-                     rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                         rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
-                 initializing_client()->GetDtlsCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSslCipher,
-                   rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
-
+  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1,
             init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
                                           kDefaultSrtpCryptoSuite));
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
 TEST_F(P2PTestConductor, GetDtls12Both) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   PeerConnectionFactory::Options recv_options;
   recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   ASSERT_TRUE(
       CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
   rtc::scoped_refptr<webrtc::FakeMetricsObserver>
       init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
   initializing_client()->pc()->RegisterUMAObserver(init_observer);
   LocalP2PTest();
 
-  EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
-                     rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                         rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)),
-                 initializing_client()->GetDtlsCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSslCipher,
-                   rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                       rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));
-
+  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1,
             init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
                                           kDefaultSrtpCryptoSuite));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
 // received supports 1.0.
-// Disabled due to new BoringSSLL version, see  webrtc:5634
-TEST_F(P2PTestConductor, DISABLED_GetDtls12Init) {
+TEST_F(P2PTestConductor, GetDtls12Init) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   PeerConnectionFactory::Options recv_options;
   recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(
       CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
   rtc::scoped_refptr<webrtc::FakeMetricsObserver>
       init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
   initializing_client()->pc()->RegisterUMAObserver(init_observer);
   LocalP2PTest();
 
-  EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
-                     rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                         rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
-                 initializing_client()->GetDtlsCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSslCipher,
-                   rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
-
+  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1,
             init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
                                           kDefaultSrtpCryptoSuite));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
 // received supports 1.2.
-// Disabled due to new BoringSSLL version, see  webrtc:5634
-TEST_F(P2PTestConductor, DISABLED_GetDtls12Recv) {
+TEST_F(P2PTestConductor, GetDtls12Recv) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   PeerConnectionFactory::Options recv_options;
   recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   ASSERT_TRUE(
       CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
   rtc::scoped_refptr<webrtc::FakeMetricsObserver>
       init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
   initializing_client()->pc()->RegisterUMAObserver(init_observer);
   LocalP2PTest();
 
-  EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
-                     rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                         rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
-                 initializing_client()->GetDtlsCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSslCipher,
-                   rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
-                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
-
+  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1,
             init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
                                           kDefaultSrtpCryptoSuite));
 }
 
 // This test sets up a call between two parties with audio, video and an RTP
 // data channel.
@@ skipping 437 matching lines @@
   PeerConnectionInterface::IceServer server;
   server.urls.push_back("turn:hostname");
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)