Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(548)

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

Issue 1380603005: Revert of 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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/app/webrtc/fakemetricsobserver.cc ('k') | talk/app/webrtc/peerconnectioninterface.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/app/webrtc/peerconnection_unittest.cc
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index d593f888ff8add8910fa5c6829d720978733da52..c077fe003f322b5b1ca846bfe2bb2edcd5dee244 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -1342,22 +1342,21 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- 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)));
+ 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));
}
// Test that DTLS 1.2 is used if both ends support it.
@@ -1372,22 +1371,21 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- 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)));
+ 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));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
@@ -1403,22 +1401,21 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- 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)));
+ 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));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
@@ -1434,22 +1431,21 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- 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)));
+ 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));
}
// This test sets up a call between two parties with audio, video and data.
« no previous file with comments | « talk/app/webrtc/fakemetricsobserver.cc ('k') | talk/app/webrtc/peerconnectioninterface.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698