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 1430 matching lines @@
   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_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(

[tommi, 2016/03/07 23:47:26]

Just curious - Instead of using the "WAIT" macros, is there a callback that
could be used?
The WAIT macros can repeatedly evaluate the expression and expand the expression
into more than one call site, which is not obvious from just reading the code.

[torbjorng (webrtc), 2016/03/08 20:17:05]

Perhaps. The present code is a bit odd, with timeout calls nested in each other.
Since _WAIT also runs an event loop and since I just want to re-enable tests
again, I will for now resist the temptation of improving this code.

+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
+#if 0

[tommi, 2016/03/07 23:47:26]

if this block should be deleted, let's just delete it instead of keeping it
around as dead code.

[torbjorng (webrtc), 2016/03/08 20:17:05]

Done.

   EXPECT_EQ(1, init_observer->GetEnumCounter(
                    webrtc::kEnumCounterAudioSslCipher,
                    rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
                        rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
+#endif
 
   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_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
+#if 0
   EXPECT_EQ(1, init_observer->GetEnumCounter(
                    webrtc::kEnumCounterAudioSslCipher,
                    rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
                        rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));
+#endif
 
   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.
 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_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
+#if 0
   EXPECT_EQ(1, init_observer->GetEnumCounter(
                    webrtc::kEnumCounterAudioSslCipher,
                    rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
                        rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
+#endif
 
   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.
 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_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
+                       initializing_client()->GetDtlsCipherStats(),
+                       rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT),
+                   kMaxWaitForStatsMs);
+#if 0
   EXPECT_EQ(1, init_observer->GetEnumCounter(
                    webrtc::kEnumCounterAudioSslCipher,
                    rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
                        rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
+#endif
 
   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
@@ skipping 438 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)