Test CreatePeerConnectionFactory() with a forwarding mock AudioDecoderFactory

webrtc/pc/rtcstats_integrationtest.cc

 /*
  *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <set>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/datachannelinterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/stats/rtcstats_objects.h"
 #include "webrtc/api/stats/rtcstatsreport.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/refcountedobject.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/virtualsocketserver.h"
@@ skipping 21 matching lines @@
     callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
         "callee", &network_thread_, &worker_thread_);
   }
 
   void StartCall() {
     // Create PeerConnections and "connect" sigslots
     PeerConnectionInterface::RTCConfiguration config;
     PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:1.1.1.1:3478";
     config.servers.push_back(ice_server);
-    EXPECT_TRUE(caller_->CreatePc(nullptr, config));
-    EXPECT_TRUE(callee_->CreatePc(nullptr, config));
+    EXPECT_TRUE(caller_->CreatePc(nullptr, config,
+                                  CreateBuiltinAudioEncoderFactory(),
+                                  CreateBuiltinAudioDecoderFactory()));
+    EXPECT_TRUE(callee_->CreatePc(nullptr, config,
+                                  CreateBuiltinAudioEncoderFactory(),
+                                  CreateBuiltinAudioDecoderFactory()));
     PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
 
     // Get user media for audio and video
     caller_->GetAndAddUserMedia(true, FakeConstraints(),
                                 true, FakeConstraints());
     callee_->GetAndAddUserMedia(true, FakeConstraints(),
                                 true, FakeConstraints());
 
     // Create data channels
     DataChannelInit init;
@@ skipping 574 matching lines @@
   caller_ = nullptr;
   // Any pending stats requests should have completed in the act of destroying
   // the peer connection.
   EXPECT_TRUE(stats_obtainer->report());
 }
 #endif  // HAVE_SCTP
 
 }  // namespace
 
 }  // namespace webrtc