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Unified Diff: talk/app/webrtc/peerconnection_unittest.cc

Issue 1337673002: Change WebRTC SslCipher to be exposed as number only. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 3 months ago
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Index: talk/app/webrtc/peerconnection_unittest.cc
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index c077fe003f322b5b1ca846bfe2bb2edcd5dee244..d593f888ff8add8910fa5c6829d720978733da52 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -1342,21 +1342,22 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- initializing_client()->GetDtlsCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
-
- EXPECT_EQ_WAIT(
- kDefaultSrtpCipher,
- initializing_client()->GetSrtpCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- kDefaultSrtpCipher,
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ 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,
+ initializing_client()->GetSrtpCipherStats(),
+ kMaxWaitForStatsMs);
+ EXPECT_EQ(1, init_observer->GetEnumCounter(
+ webrtc::kEnumCounterAudioSrtpCipher,
+ rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
}
// Test that DTLS 1.2 is used if both ends support it.
@@ -1371,21 +1372,22 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12),
- initializing_client()->GetDtlsCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12),
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
-
- EXPECT_EQ_WAIT(
- kDefaultSrtpCipher,
- initializing_client()->GetSrtpCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- kDefaultSrtpCipher,
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ 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,
+ initializing_client()->GetSrtpCipherStats(),
+ kMaxWaitForStatsMs);
+ 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
@@ -1401,21 +1403,22 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- initializing_client()->GetDtlsCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
-
- EXPECT_EQ_WAIT(
- kDefaultSrtpCipher,
- initializing_client()->GetSrtpCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- kDefaultSrtpCipher,
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ 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,
+ initializing_client()->GetSrtpCipherStats(),
+ kMaxWaitForStatsMs);
+ 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
@@ -1431,21 +1434,22 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- initializing_client()->GetDtlsCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
-
- EXPECT_EQ_WAIT(
- kDefaultSrtpCipher,
- initializing_client()->GetSrtpCipherStats(),
- kMaxWaitForStatsMs);
- EXPECT_EQ(
- kDefaultSrtpCipher,
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ 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,
+ initializing_client()->GetSrtpCipherStats(),
+ kMaxWaitForStatsMs);
+ 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.
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