Report metrics about negotiated ciphers.

talk/app/webrtc/webrtcsession_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,
  *     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/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"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/fakedtlsidentityservice.h"
 #include "talk/app/webrtc/test/fakemediastreamsignaling.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
@@ skipping 35 matching lines @@
 using cricket::TransportInfo;
 using rtc::SocketAddress;
 using rtc::scoped_ptr;
 using rtc::Thread;
 using webrtc::CreateSessionDescription;
 using webrtc::CreateSessionDescriptionObserver;
 using webrtc::CreateSessionDescriptionRequest;
 using webrtc::DTLSIdentityRequestObserver;
 using webrtc::DTLSIdentityServiceInterface;
 using webrtc::FakeConstraints;
+using webrtc::FakeMetricsObserver;
 using webrtc::IceCandidateCollection;
 using webrtc::JsepIceCandidate;
 using webrtc::JsepSessionDescription;
 using webrtc::PeerConnectionFactoryInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::SessionDescriptionInterface;
 using webrtc::StreamCollection;
 using webrtc::WebRtcSession;
 using webrtc::kBundleWithoutRtcpMux;
 using webrtc::kCreateChannelFailed;
@@ skipping 64 matching lines @@
 
 // Add some extra |newlines| to the |message| after |line|.
 static void InjectAfter(const std::string& line,
                         const std::string& newlines,
                         std::string* message) {
   const std::string tmp = line + newlines;
   rtc::replace_substrs(line.c_str(), line.length(),
                              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()
       : oncandidatesready_(false),
         ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
         ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) {
   }
 
   virtual void OnIceConnectionChange(
       PeerConnectionInterface::IceConnectionState new_state) {
@@ skipping 3669 matching lines @@
   EXPECT_EQ(4000, option_val);
 
   EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.