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diff --git a/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h
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+++ b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h
@@ -0,0 +1,135 @@
+/*
+ *  Copyright 2013 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.
+ */
+
+#ifndef PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
+#define PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_options.h"
+#include "api/data_channel_interface.h"
+#include "api/jsep.h"
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "pc/test/fake_audio_capture_module.h"
+#include "pc/test/fake_video_track_renderer.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread.h"
+#include "test/scoped_key_value_config.h"
+
+class PeerConnectionTestWrapper
+    : public webrtc::PeerConnectionObserver,
+      public webrtc::CreateSessionDescriptionObserver,
+      public sigslot::has_slots<> {
+ public:
+  static void Connect(PeerConnectionTestWrapper* caller,
+                      PeerConnectionTestWrapper* callee);
+
+  PeerConnectionTestWrapper(const std::string& name,
+                            rtc::SocketServer* socket_server,
+                            rtc::Thread* network_thread,
+                            rtc::Thread* worker_thread);
+  virtual ~PeerConnectionTestWrapper();
+
+  bool CreatePc(
+      const webrtc::PeerConnectionInterface::RTCConfiguration& config,
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory);
+
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory()
+      const {
+    return peer_connection_factory_;
+  }
+  webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
+
+  rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
+      const std::string& label,
+      const webrtc::DataChannelInit& init);
+
+  void WaitForNegotiation();
+
+  // Implements PeerConnectionObserver.
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override;
+  void OnAddTrack(
+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+      const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
+          streams) override;
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
+  void OnRenegotiationNeeded() override {}
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+
+  // Implements CreateSessionDescriptionObserver.
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError) override {}
+
+  void CreateOffer(
+      const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
+  void CreateAnswer(
+      const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
+  void ReceiveOfferSdp(const std::string& sdp);
+  void ReceiveAnswerSdp(const std::string& sdp);
+  void AddIceCandidate(const std::string& sdp_mid,
+                       int sdp_mline_index,
+                       const std::string& candidate);
+  void WaitForCallEstablished();
+  void WaitForConnection();
+  void WaitForAudio();
+  void WaitForVideo();
+  void GetAndAddUserMedia(bool audio,
+                          const cricket::AudioOptions& audio_options,
+                          bool video);
+
+  // sigslots
+  sigslot::signal3<const std::string&, int, const std::string&>
+      SignalOnIceCandidateReady;
+  sigslot::signal1<const std::string&> SignalOnSdpReady;
+  sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
+
+ private:
+  void SetLocalDescription(webrtc::SdpType type, const std::string& sdp);
+  void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp);
+  bool CheckForConnection();
+  bool CheckForAudio();
+  bool CheckForVideo();
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
+      bool audio,
+      const cricket::AudioOptions& audio_options,
+      bool video);
+
+  webrtc::test::ScopedKeyValueConfig field_trials_;
+  std::string name_;
+  rtc::SocketServer* const socket_server_;
+  rtc::Thread* const network_thread_;
+  rtc::Thread* const worker_thread_;
+  webrtc::SequenceChecker pc_thread_checker_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
+      peer_connection_factory_;
+  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
+  std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
+  int num_get_user_media_calls_ = 0;
+  bool pending_negotiation_;
+};
+
+#endif  // PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_