Report metrics about negotiated ciphers.

This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).

BUG=428343

Review URL: https://codereview.webrtc.org/1156143005

Cr-Commit-Position: refs/heads/master@{#9537}

talk/app/webrtc/fakemetricsobserver.cc

@@ -0,0 +1,81 @@
+/*
+ * libjingle
+ * Copyright 2015 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,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/fakemetricsobserver.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+FakeMetricsObserver::FakeMetricsObserver() {
+  Reset();
+}
+
+void FakeMetricsObserver::Reset() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  memset(counters_, 0, sizeof(counters_));
+  memset(int_histogram_samples_, 0, sizeof(int_histogram_samples_));
+  for (std::string& type : string_histogram_samples_) {
+    type.clear();
+  }
+}
+
+void FakeMetricsObserver::IncrementCounter(PeerConnectionMetricsCounter type) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  ++counters_[type];
+}
+
+void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
+    int value) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK_EQ(int_histogram_samples_[type], 0);
+  int_histogram_samples_[type] = value;
+}
+
+void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
+    const std::string& value) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  string_histogram_samples_[type].assign(value);
+}
+
+int FakeMetricsObserver::GetCounter(PeerConnectionMetricsCounter type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return counters_[type];
+}
+
+int FakeMetricsObserver::GetIntHistogramSample(
+    PeerConnectionMetricsName type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return int_histogram_samples_[type];
+}
+
+const std::string& FakeMetricsObserver::GetStringHistogramSample(
+    PeerConnectionMetricsName type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return string_histogram_samples_[type];
+}
+
+}  // namespace webrtc

talk/app/webrtc/fakemetricsobserver.h

@@ -0,0 +1,68 @@
+/*
+ * libjingle
+ * Copyright 2015 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,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
+#define TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
+
+#include <map>
+#include <string>
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+#include "webrtc/base/thread_checker.h"
+
+namespace webrtc {
+
+class FakeMetricsObserver : public MetricsObserverInterface {
+ public:
+  FakeMetricsObserver();
+  void Reset();
+
+  void IncrementCounter(PeerConnectionMetricsCounter type) override;
+  void AddHistogramSample(PeerConnectionMetricsName type,
+                          int value) override;
+  void AddHistogramSample(PeerConnectionMetricsName type,
+                          const std::string& value) override;
+
+  // Accessors to be used by the tests.
+  int GetCounter(PeerConnectionMetricsCounter type) const;
+  int GetIntHistogramSample(PeerConnectionMetricsName type) const;
+  const std::string& GetStringHistogramSample(
+      PeerConnectionMetricsName type) const;
+
+ protected:
+  ~FakeMetricsObserver() {}
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  int counters_[kPeerConnectionMetricsCounter_Max];
+  int int_histogram_samples_[kPeerConnectionMetricsCounter_Max];
+  std::string string_histogram_samples_[kPeerConnectionMetricsName_Max];
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_

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"
@@ -1336,17 +1337,26 @@ 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));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  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));
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
@@ -1356,17 +1366,26 @@ 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));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  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));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
@@ -1377,17 +1396,26 @@ 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));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  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));
 }
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
@@ -1398,17 +1426,26 @@ 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));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  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));
 }
 
 // This test sets up a call between two parties with audio, video and data.

talk/app/webrtc/peerconnectioninterface.h

@@ -127,6 +127,9 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
   virtual void IncrementCounter(PeerConnectionMetricsCounter type) = 0;
   virtual void AddHistogramSample(PeerConnectionMetricsName type,
                                   int value) = 0;
+  // TODO(jbauch): Make method abstract when it is implemented by Chromium.
+  virtual void AddHistogramSample(PeerConnectionMetricsName type,
+                                  const std::string& value) {}
 
  protected:
   virtual ~MetricsObserverInterface() {}

talk/app/webrtc/umametrics.h

@@ -65,6 +65,12 @@ enum PeerConnectionMetricsName {
   kTimeToConnect,           // In milliseconds.
   kLocalCandidates_IPv4,    // Number of IPv4 local candidates.
   kLocalCandidates_IPv6,    // Number of IPv6 local candidates.
+  kAudioSrtpCipher,         // Name of SRTP cipher used in audio channel.
+  kAudioSslCipher,          // Name of SSL cipher used in audio channel.
+  kVideoSrtpCipher,         // Name of SRTP cipher used in video channel.
+  kVideoSslCipher,          // Name of SSL cipher used in video channel.
+  kDataSrtpCipher,          // Name of SRTP cipher used in data channel.
+  kDataSslCipher,           // Name of SSL cipher used in data channel.
   kPeerConnectionMetricsName_Max
 };
 

talk/app/webrtc/webrtcsession.cc

@@ -44,6 +44,7 @@
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
@@ -1392,7 +1393,11 @@ void WebRtcSession::OnTransportCompleted(cricket::Transport* transport) {
   SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
   // Only report once when Ice connection is completed.
   if (old_state != PeerConnectionInterface::kIceConnectionCompleted) {
-    ReportBestConnectionState(transport);
+    cricket::TransportStats stats;
+    if (metrics_observer_ && transport->GetStats(&stats)) {
+      ReportBestConnectionState(stats);
+      ReportNegotiatedCiphers(stats);
+    }
   }
 }
 
@@ -1872,16 +1877,9 @@ bool WebRtcSession::ReadyToUseRemoteCandidate(
 
 // Walk through the ConnectionInfos to gather best connection usage
 // for IPv4 and IPv6.
-void WebRtcSession::ReportBestConnectionState(cricket::Transport* transport) {
-  if (!metrics_observer_) {
-    return;
-  }
-
-  cricket::TransportStats stats;
-  if (!transport->GetStats(&stats)) {
-    return;
-  }
-
+void WebRtcSession::ReportBestConnectionState(
+    const cricket::TransportStats& stats) {
+  DCHECK(metrics_observer_ != NULL);
   for (cricket::TransportChannelStatsList::const_iterator it =
          stats.channel_stats.begin();
        it != stats.channel_stats.end(); ++it) {
@@ -1897,11 +1895,48 @@ void WebRtcSession::ReportBestConnectionState(cricket::Transport* transport) {
                  AF_INET6) {
         metrics_observer_->IncrementCounter(kBestConnections_IPv6);
       } else {
-        ASSERT(false);
+        RTC_NOTREACHED();
       }
       return;
     }
   }
 }
 
+void WebRtcSession::ReportNegotiatedCiphers(
+    const cricket::TransportStats& stats) {
+  DCHECK(metrics_observer_ != NULL);
+  if (!dtls_enabled_ || stats.channel_stats.empty()) {
+    return;
+  }
+
+  const std::string& srtp_cipher = stats.channel_stats[0].srtp_cipher;
+  const std::string& ssl_cipher = stats.channel_stats[0].ssl_cipher;
+  if (srtp_cipher.empty() && ssl_cipher.empty()) {
+    return;
+  }
+
+  PeerConnectionMetricsName srtp_name;
+  PeerConnectionMetricsName ssl_name;
+  if (stats.content_name == cricket::CN_AUDIO) {
+    srtp_name = kAudioSrtpCipher;
+    ssl_name = kAudioSslCipher;
+  } else if (stats.content_name == cricket::CN_VIDEO) {
+    srtp_name = kVideoSrtpCipher;
+    ssl_name = kVideoSslCipher;
+  } else if (stats.content_name == cricket::CN_DATA) {
+    srtp_name = kDataSrtpCipher;
+    ssl_name = kDataSslCipher;
+  } else {
+    RTC_NOTREACHED();
+    return;
+  }
+
+  if (!srtp_cipher.empty()) {
+    metrics_observer_->AddHistogramSample(srtp_name, srtp_cipher);
+  }
+  if (!ssl_cipher.empty()) {
+    metrics_observer_->AddHistogramSample(ssl_name, ssl_cipher);
+  }
+}
+
 }  // namespace webrtc

talk/app/webrtc/webrtcsession.h

@@ -370,7 +370,9 @@ class WebRtcSession : public cricket::BaseSession,
 
   // Invoked when OnTransportCompleted is signaled to gather the usage
   // of IPv4/IPv6 as best connection.
-  void ReportBestConnectionState(cricket::Transport* transport);
+  void ReportBestConnectionState(const cricket::TransportStats& stats);
+
+  void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
   rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
   rtc::scoped_ptr<cricket::VideoChannel> video_channel_;

talk/app/webrtc/webrtcsession_unittest.cc

@@ -26,6 +26,7 @@
  */
 
 #include "talk/app/webrtc/audiotrack.h"
+#include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
@@ -81,6 +82,7 @@ using webrtc::CreateSessionDescriptionRequest;
 using webrtc::DTLSIdentityRequestObserver;
 using webrtc::DTLSIdentityServiceInterface;
 using webrtc::FakeConstraints;
+using webrtc::FakeMetricsObserver;
 using webrtc::IceCandidateCollection;
 using webrtc::JsepIceCandidate;
 using webrtc::JsepSessionDescription;
@@ -165,33 +167,6 @@ static void InjectAfter(const std::string& line,
                              tmp.c_str(), tmp.length(), message);
 }
 
-class FakeMetricsObserver : public webrtc::MetricsObserverInterface {
- public:
-  FakeMetricsObserver() { Reset(); }
-  void Reset() {
-    memset(peer_connection_metrics_counters_, 0,
-           sizeof(peer_connection_metrics_counters_));
-    memset(peer_connection_metrics_name_, 0,
-           sizeof(peer_connection_metrics_name_));
-  }
-
-  void IncrementCounter(webrtc::PeerConnectionMetricsCounter type) override {
-    peer_connection_metrics_counters_[type]++;
-  }
-  void AddHistogramSample(webrtc::PeerConnectionMetricsName type,
-                          int value) override {
-    ASSERT(peer_connection_metrics_name_[type] == 0);
-    peer_connection_metrics_name_[type] = value;
-  }
-
-  int peer_connection_metrics_counters_
-      [webrtc::kPeerConnectionMetricsCounter_Max];
-  int peer_connection_metrics_name_[webrtc::kPeerConnectionMetricsCounter_Max];
-
-  int AddRef() override { return 1; }
-  int Release() override { return 1; }
-};
-
 class MockIceObserver : public webrtc::IceObserver {
  public:
   MockIceObserver()
@@ -360,7 +335,8 @@ class WebRtcSessionTest : public testing::Test {
       stun_server_(cricket::TestStunServer::Create(Thread::Current(),
                                                    stun_socket_addr_)),
       turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
-      mediastream_signaling_(channel_manager_.get()) {
+      mediastream_signaling_(channel_manager_.get()),
+      metrics_observer_(new rtc::RefCountedObject<FakeMetricsObserver>()) {
     tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
 
     cricket::ServerAddresses stun_servers;
@@ -398,7 +374,7 @@ class WebRtcSessionTest : public testing::Test {
 
     EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
                                      identity_service, rtc_configuration));
-    session_->set_metrics_observer(&metrics_observer_);
+    session_->set_metrics_observer(metrics_observer_);
   }
 
   void Init() {
@@ -1022,20 +998,18 @@ class WebRtcSessionTest : public testing::Test {
     ExpectedBestConnection best_connection_after_initial_ice_converged_;
 
     void VerifyBestConnectionAfterIceConverge(
-        const FakeMetricsObserver& metrics_observer) const {
+        const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer) const {
       Verify(metrics_observer, best_connection_after_initial_ice_converged_);
     }
 
    private:
-    void Verify(const FakeMetricsObserver& metrics_observer,
+    void Verify(const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer,
                 const ExpectedBestConnection& expected) const {
       EXPECT_EQ(
-          metrics_observer
-              .peer_connection_metrics_counters_[webrtc::kBestConnections_IPv4],
+          metrics_observer->GetCounter(webrtc::kBestConnections_IPv4),
           expected.ipv4_count_);
       EXPECT_EQ(
-          metrics_observer
-              .peer_connection_metrics_counters_[webrtc::kBestConnections_IPv6],
+          metrics_observer->GetCounter(webrtc::kBestConnections_IPv6),
           expected.ipv6_count_);
     }
   };
@@ -1135,7 +1109,7 @@ class WebRtcSessionTest : public testing::Test {
                    observer_.ice_connection_state_,
                    kIceCandidatesTimeout);
 
-    metrics_observer_.Reset();
+    metrics_observer_->Reset();
 
     // Clearing the rules, session should move back to completed state.
     loopback_network_manager.ClearRules(fss_.get());
@@ -1280,7 +1254,7 @@ class WebRtcSessionTest : public testing::Test {
   MockIceObserver observer_;
   cricket::FakeVideoMediaChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_channel_;
-  FakeMetricsObserver metrics_observer_;
+  rtc::scoped_refptr<FakeMetricsObserver> metrics_observer_;
 };
 
 TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {

talk/libjingle_tests.gyp

@@ -199,6 +199,8 @@
         'app/webrtc/datachannel_unittest.cc',
         'app/webrtc/dtlsidentitystore_unittest.cc',
         'app/webrtc/dtmfsender_unittest.cc',
+        'app/webrtc/fakemetricsobserver.cc',
+        'app/webrtc/fakemetricsobserver.h',
         'app/webrtc/jsepsessiondescription_unittest.cc',
         'app/webrtc/localaudiosource_unittest.cc',
         'app/webrtc/mediastream_unittest.cc',