Report metrics about negotiated ciphers.

talk/app/webrtc/peerconnection_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 15 matching lines @@
  */
 
 #include <stdio.h>
 
 #include <algorithm>
 #include <list>
 #include <map>
 #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"
 #include "talk/app/webrtc/peerconnectionfactory.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/fakedtlsidentityservice.h"
 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
@@ skipping 1276 matching lines @@
       kMaxWaitForStatsMs);
 }
 
 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
 TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   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.peer_connection_metrics_name_string_
+          [webrtc::kAudioSslCipher]);
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer.peer_connection_metrics_name_string_
+          [webrtc::kAudioSrtpCipher]);
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
 TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   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.peer_connection_metrics_name_string_
+          [webrtc::kAudioSslCipher]);
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer.peer_connection_metrics_name_string_
+          [webrtc::kAudioSrtpCipher]);
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
 // received supports 1.0.
 TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   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.peer_connection_metrics_name_string_
+          [webrtc::kAudioSslCipher]);
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer.peer_connection_metrics_name_string_
+          [webrtc::kAudioSrtpCipher]);
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
 // received supports 1.2.
 TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   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.peer_connection_metrics_name_string_
+          [webrtc::kAudioSslCipher]);
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer.peer_connection_metrics_name_string_
+          [webrtc::kAudioSrtpCipher]);
 }
 
 // This test sets up a call between two parties with audio, video and data.
 TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   initializing_client()->CreateDataChannel();
   LocalP2PTest();
   ASSERT_TRUE(initializing_client()->data_channel() != NULL);
@@ skipping 161 matching lines @@
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
 TEST_F(JsepPeerConnectionP2PTestClient,
        DISABLED_LocalP2PTestWithVideoDecoderFactory) {
   ASSERT_TRUE(CreateTestClients());
   EnableVideoDecoderFactory();
   LocalP2PTest();
 }
 
 #endif // if !defined(THREAD_SANITIZER)