Index: talk/app/webrtc/peerconnection_unittest.cc |
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc |
index 99ee9be867168a99ca9f6fe898ce2517aede4a56..97bb122469fefc3b1bdebba5372437b0ca9ec91c 100644 |
--- a/talk/app/webrtc/peerconnection_unittest.cc |
+++ b/talk/app/webrtc/peerconnection_unittest.cc |
@@ -33,6 +33,7 @@ |
#include <vector> |
#include "talk/app/webrtc/dtmfsender.h" |
+#include "talk/app/webrtc/fakemetricsobserver.h" |
#include "talk/app/webrtc/fakeportallocatorfactory.h" |
#include "talk/app/webrtc/localaudiosource.h" |
#include "talk/app/webrtc/mediastreaminterface.h" |
@@ -1329,17 +1330,25 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) { |
PeerConnectionFactory::Options recv_options; |
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options)); |
+ webrtc::FakeMetricsObserver init_observer; |
+ 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.ssl_cipher_["audio"]); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
+ EXPECT_EQ( |
+ kDefaultSrtpCipher, |
+ init_observer.srtp_cipher_["audio"]); |
} |
// Test that DTLS 1.2 is used if both ends support it. |
@@ -1349,17 +1358,25 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) { |
PeerConnectionFactory::Options recv_options; |
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options)); |
+ webrtc::FakeMetricsObserver init_observer; |
+ 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.ssl_cipher_["audio"]); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
+ EXPECT_EQ( |
+ kDefaultSrtpCipher, |
+ init_observer.srtp_cipher_["audio"]); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the |
@@ -1370,17 +1387,25 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) { |
PeerConnectionFactory::Options recv_options; |
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options)); |
+ webrtc::FakeMetricsObserver init_observer; |
+ 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.ssl_cipher_["audio"]); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
+ EXPECT_EQ( |
+ kDefaultSrtpCipher, |
+ init_observer.srtp_cipher_["audio"]); |
} |
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the |
@@ -1391,17 +1416,25 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) { |
PeerConnectionFactory::Options recv_options; |
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options)); |
+ webrtc::FakeMetricsObserver init_observer; |
+ 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.ssl_cipher_["audio"]); |
EXPECT_EQ_WAIT( |
kDefaultSrtpCipher, |
initializing_client()->GetSrtpCipherStats(), |
kMaxWaitForStatsMs); |
+ EXPECT_EQ( |
+ kDefaultSrtpCipher, |
+ init_observer.srtp_cipher_["audio"]); |
} |
// This test sets up a call between two parties with audio, video and data. |