Convert internal representation of Srtp cryptos from string to int.

talk/app/webrtc/peerconnection_unittest.cc

 /*
  * libjingle
  * Copyright 2012 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 93 matching lines @@
 static const char kVideoTrackLabelBase[] = "video_track";
 static const char kAudioTrackLabelBase[] = "audio_track";
 static const char kDataChannelLabel[] = "data_channel";
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 // This declaration is also #ifdef'd as it causes unused-variable errors.
 #if !defined(THREAD_SANITIZER)
 // SRTP cipher name negotiated by the tests. This must be updated if the
 // default changes.
-static const char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32";
+static const int kDefaultSrtpCipher = rtc::SRTP_AES128_CM_SHA1_32;

[pthatcher1, 2015/11/11 19:59:40]

Should we rename this to kDefaultSrtpCryptoSuite?

[guoweis_webrtc, 2015/11/17 01:21:15]

Done.

 #endif
 
 static void RemoveLinesFromSdp(const std::string& line_start,
                                std::string* sdp) {
   const char kSdpLineEnd[] = "\r\n";
   size_t ssrc_pos = 0;
   while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
       std::string::npos) {
     size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
     sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
@@ skipping 1150 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::GetSslCipherSuiteName(
+  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_EQ_WAIT(kDefaultSrtpCipher,
+  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSrtpCipher,
-                   rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+                   webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
 TEST_F(JsepPeerConnectionP2PTestClient, 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::GetSslCipherSuiteName(
+  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_EQ_WAIT(kDefaultSrtpCipher,
+  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSrtpCipher,
-                   rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+                   webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
 // received supports 1.0.
 TEST_F(JsepPeerConnectionP2PTestClient, 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::GetSslCipherSuiteName(
+  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_EQ_WAIT(kDefaultSrtpCipher,
+  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSrtpCipher,
-                   rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+                   webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
 // received supports 1.2.
 TEST_F(JsepPeerConnectionP2PTestClient, 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::GetSslCipherSuiteName(
+  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_EQ_WAIT(kDefaultSrtpCipher,
+  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),
                  initializing_client()->GetSrtpCipherStats(),
                  kMaxWaitForStatsMs);
   EXPECT_EQ(1, init_observer->GetEnumCounter(
-                   webrtc::kEnumCounterAudioSrtpCipher,
-                   rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+                   webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));
 }
 
 // This test sets up a call between two parties with audio, video and data.
 TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   initializing_client()->CreateDataChannel();
   LocalP2PTest();
   ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
@@ skipping 337 matching lines @@
   server.urls.push_back("stun:hostname");
   server.urls.push_back("turn:hostname");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_configurations_,
                                       &turn_configurations_));
   EXPECT_EQ(1U, stun_configurations_.size());
   EXPECT_EQ(1U, turn_configurations_.size());
 }
 
 #endif // if !defined(THREAD_SANITIZER)