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..d0d14c8af8254a5c6fb32f8cd90d51bca99e3e85 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -1343,20 +1343,21 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
   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));
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10)
+          .rfc_name,
+      initializing_client()->GetDtlsCipherStats(), kMaxWaitForStatsMs);
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSslCipher,
+                   rtc::SSLStreamAdapter::GetDefaultSslCipher(
+                       rtc::SSL_PROTOCOL_DTLS_10).ssl_id));
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
-  EXPECT_EQ(
-      kDefaultSrtpCipher,
-      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSrtpCipher,
+                   webrtc::GetSrtpCipherType(kDefaultSrtpCipher)));
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
@@ -1372,20 +1373,21 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
   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));
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12)
+          .rfc_name,
+      initializing_client()->GetDtlsCipherStats(), kMaxWaitForStatsMs);
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSslCipher,
+                   rtc::SSLStreamAdapter::GetDefaultSslCipher(
+                       rtc::SSL_PROTOCOL_DTLS_12).ssl_id));
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
-  EXPECT_EQ(
-      kDefaultSrtpCipher,
-      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSrtpCipher,
+                   webrtc::GetSrtpCipherType(kDefaultSrtpCipher)));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
@@ -1402,20 +1404,21 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
   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));
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10)
+          .rfc_name,
+      initializing_client()->GetDtlsCipherStats(), kMaxWaitForStatsMs);
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSslCipher,
+                   rtc::SSLStreamAdapter::GetDefaultSslCipher(
+                       rtc::SSL_PROTOCOL_DTLS_10).ssl_id));
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
-  EXPECT_EQ(
-      kDefaultSrtpCipher,
-      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSrtpCipher,
+                   webrtc::GetSrtpCipherType(kDefaultSrtpCipher)));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
@@ -1432,20 +1435,21 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
   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));
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10)
+          .rfc_name,
+      initializing_client()->GetDtlsCipherStats(), kMaxWaitForStatsMs);
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSslCipher,
+                   rtc::SSLStreamAdapter::GetDefaultSslCipher(
+                       rtc::SSL_PROTOCOL_DTLS_10).ssl_id));
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
-  EXPECT_EQ(
-      kDefaultSrtpCipher,
-      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+  EXPECT_EQ(1, init_observer->GetEnumCounter(
+                   webrtc::kEnumCounterAudioSrtpCipher,
+                   webrtc::GetSrtpCipherType(kDefaultSrtpCipher)));
 }
 
 // This test sets up a call between two parties with audio, video and data.