Change WebRTC SslCipher to be exposed as number only.

talk/app/webrtc/peerconnection_unittest.cc

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..179d858f6f51ca8c2fce487760ca1fb210d951ce 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::GetSslCipherRfcNameById(

[pthatcher1, 2015/09/30 18:39:35]

I think a good pair of names for these would be:

GetSslCipherName
GetSslCipherFromName

[guoweis_webrtc, 2015/09/30 20:14:21]

There are 2 types of functions for SSL
1. GetSslCipher() ==> get the current cipher on the channel as an id
2. GetSslCipherName(id) ==> translate from id to name

There are 2 types of functions for SRTP
1. GetSrtpCryptoSuite(str*) ==> get the current crypto on the channel as an
string
2. GetSrtpCryptoSuiteFromName(str) ==> translate from name to id

+                     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::GetSrtpCipherByRfcName(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::GetSslCipherRfcNameById(
+                     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::GetSrtpCipherByRfcName(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::GetSslCipherRfcNameById(
+                     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::GetSrtpCipherByRfcName(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::GetSslCipherRfcNameById(
+                     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::GetSrtpCipherByRfcName(kDefaultSrtpCipher)));
 }
 
 // This test sets up a call between two parties with audio, video and data.