Index: talk/app/webrtc/peerconnection_unittest.cc |
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc |
index 4faf599907c50fd77bf6bfd45296293d70084646..3193ffd898e71c1cdf23ccb3049c477c6ecd9811 100644 |
--- a/talk/app/webrtc/peerconnection_unittest.cc |
+++ b/talk/app/webrtc/peerconnection_unittest.cc |
@@ -113,7 +113,7 @@ |
#if !defined(THREAD_SANITIZER) |
// SRTP cipher name negotiated by the tests. This must be updated if the |
// default changes. |
-static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32; |
+static const char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32"; |
#endif |
static void RemoveLinesFromSdp(const std::string& line_start, |
@@ -1327,7 +1327,7 @@ |
initializing_client()->pc()->RegisterUMAObserver(init_observer); |
LocalP2PTest(); |
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName( |
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName( |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)), |
initializing_client()->GetDtlsCipherStats(), |
@@ -1337,12 +1337,12 @@ |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT))); |
- EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
+ EXPECT_EQ_WAIT(kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
- EXPECT_EQ(1, |
- init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
- kDefaultSrtpCryptoSuite)); |
+ EXPECT_EQ(1, init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSrtpCipher, |
+ rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher))); |
} |
// Test that DTLS 1.2 is used if both ends support it. |
@@ -1358,7 +1358,7 @@ |
initializing_client()->pc()->RegisterUMAObserver(init_observer); |
LocalP2PTest(); |
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName( |
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName( |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)), |
initializing_client()->GetDtlsCipherStats(), |
@@ -1368,12 +1368,12 @@ |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT))); |
- EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
+ EXPECT_EQ_WAIT(kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
- EXPECT_EQ(1, |
- init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
- kDefaultSrtpCryptoSuite)); |
+ EXPECT_EQ(1, init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSrtpCipher, |
+ rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher))); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the |
@@ -1390,7 +1390,7 @@ |
initializing_client()->pc()->RegisterUMAObserver(init_observer); |
LocalP2PTest(); |
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName( |
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName( |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)), |
initializing_client()->GetDtlsCipherStats(), |
@@ -1400,12 +1400,12 @@ |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT))); |
- EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
+ EXPECT_EQ_WAIT(kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
- EXPECT_EQ(1, |
- init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
- kDefaultSrtpCryptoSuite)); |
+ EXPECT_EQ(1, init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSrtpCipher, |
+ rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher))); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the |
@@ -1422,7 +1422,7 @@ |
initializing_client()->pc()->RegisterUMAObserver(init_observer); |
LocalP2PTest(); |
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName( |
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName( |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)), |
initializing_client()->GetDtlsCipherStats(), |
@@ -1432,12 +1432,12 @@ |
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest( |
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT))); |
- EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
+ EXPECT_EQ_WAIT(kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
- EXPECT_EQ(1, |
- init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
- kDefaultSrtpCryptoSuite)); |
+ EXPECT_EQ(1, init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSrtpCipher, |
+ rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher))); |
} |
// This test sets up a call between two parties with audio, video and data. |