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..b632e967071cdc2175d5e63a7c4a42dfa4b2a4d1 100644 |
--- a/talk/app/webrtc/peerconnection_unittest.cc |
+++ b/talk/app/webrtc/peerconnection_unittest.cc |
@@ -1347,16 +1347,20 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) { |
initializing_client()->GetDtlsCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10), |
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher)); |
+ init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSslCipher, |
+ webrtc::GetCipherType(rtc::SSLStreamAdapter::GetDefaultSslCipher( |
+ rtc::SSL_PROTOCOL_DTLS_10))), |
+ 1); |
joachim
2015/09/18 22:24:25
Shouldn't this be EXPECT_EQ(1, ...) here and below
guoweis_webrtc
2015/09/22 19:59:39
Done.
|
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- kDefaultSrtpCipher, |
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher)); |
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
+ webrtc::GetCipherType(kDefaultSrtpCipher)), |
+ 1); |
} |
// Test that DTLS 1.2 is used if both ends support it. |
@@ -1376,16 +1380,20 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) { |
initializing_client()->GetDtlsCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12), |
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher)); |
+ init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSslCipher, |
+ webrtc::GetCipherType(rtc::SSLStreamAdapter::GetDefaultSslCipher( |
+ rtc::SSL_PROTOCOL_DTLS_12))), |
+ 1); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- kDefaultSrtpCipher, |
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher)); |
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
+ webrtc::GetCipherType(kDefaultSrtpCipher)), |
+ 1); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the |
@@ -1406,16 +1414,20 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) { |
initializing_client()->GetDtlsCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10), |
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher)); |
+ init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSslCipher, |
+ webrtc::GetCipherType(rtc::SSLStreamAdapter::GetDefaultSslCipher( |
+ rtc::SSL_PROTOCOL_DTLS_10))), |
+ 1); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- kDefaultSrtpCipher, |
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher)); |
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
+ webrtc::GetCipherType(kDefaultSrtpCipher)), |
+ 1); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the |
@@ -1436,16 +1448,20 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) { |
initializing_client()->GetDtlsCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10), |
- init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher)); |
+ init_observer->GetEnumCounter( |
+ webrtc::kEnumCounterAudioSslCipher, |
+ webrtc::GetCipherType(rtc::SSLStreamAdapter::GetDefaultSslCipher( |
+ rtc::SSL_PROTOCOL_DTLS_10))), |
+ 1); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
EXPECT_EQ( |
- kDefaultSrtpCipher, |
- init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher)); |
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
+ webrtc::GetCipherType(kDefaultSrtpCipher)), |
+ 1); |
} |
// This test sets up a call between two parties with audio, video and data. |