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diff --git a/chromium/third_party/blink/renderer/platform/peerconnection/rtc_peer_connection_handler_platform.h b/chromium/third_party/blink/renderer/platform/peerconnection/rtc_peer_connection_handler_platform.h
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+/*
+ * Copyright (C) 2012 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *
+ *     * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *     * 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.
+ *     * Neither the name of Google Inc. nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "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 COPYRIGHT
+ * OWNER OR CONTRIBUTORS 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 THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_PEER_CONNECTION_HANDLER_PLATFORM_H_
+#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_PEER_CONNECTION_HANDLER_PLATFORM_H_
+
+#include <memory>
+
+#include "third_party/blink/renderer/platform/peerconnection/rtc_stats.h"
+#include "third_party/blink/renderer/platform/platform_export.h"
+#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h"
+#include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
+#include "third_party/blink/renderer/platform/wtf/vector.h"
+#include "third_party/webrtc/api/peer_connection_interface.h"
+#include "third_party/webrtc/api/rtc_error.h"
+#include "third_party/webrtc/api/rtp_transceiver_interface.h"
+#include "third_party/webrtc/api/stats/rtc_stats.h"
+
+namespace webrtc {
+enum class RTCErrorType;
+struct DataChannelInit;
+}  // namespace webrtc
+
+namespace blink {
+
+class MediaConstraints;
+class RTCAnswerOptionsPlatform;
+class RTCIceCandidatePlatform;
+class RTCOfferOptionsPlatform;
+class RTCRtpSenderPlatform;
+class RTCRtpTransceiverPlatform;
+class RTCSessionDescriptionPlatform;
+class RTCSessionDescriptionRequest;
+class RTCStatsRequest;
+class RTCVoidRequest;
+class WebLocalFrame;
+class WebMediaStream;
+class WebMediaStreamTrack;
+
+class PLATFORM_EXPORT RTCPeerConnectionHandlerPlatform {
+ public:
+  enum class IceConnectionStateVersion {
+    // Only applicable in Unified Plan when the JavaScript-exposed
+    // iceConnectionState is calculated in blink. In this case, kLegacy is used
+    // to report the webrtc::PeerConnectionInterface implementation which is not
+    // visible in JavaScript, but still useful to track for debugging purposes.
+    kLegacy,
+    // The JavaScript-visible iceConnectionState. In Plan B, this is the same as
+    // the webrtc::PeerConnectionInterface implementation.
+    kDefault,
+  };
+
+  virtual ~RTCPeerConnectionHandlerPlatform() = default;
+
+  virtual bool Initialize(
+      const webrtc::PeerConnectionInterface::RTCConfiguration&,
+      const MediaConstraints&,
+      WebLocalFrame*) = 0;
+
+  virtual void Stop() = 0;
+  // This function should be called when the object is taken out of service.
+  // There might be functions that need to return through the object, so it
+  // cannot be deleted yet, but no new operations should be allowed.
+  // All references to the object except the owning reference are deleted
+  // by this function.
+  virtual void StopAndUnregister() = 0;
+
+  // Unified Plan: The list of transceivers after the createOffer() call.
+  // Because of offerToReceive[Audio/Video] it is possible for createOffer() to
+  // create new transceivers or update the direction of existing transceivers.
+  // https://w3c.github.io/webrtc-pc/#legacy-configuration-extensions
+  // Plan B: Returns an empty list.
+  virtual Vector<std::unique_ptr<RTCRtpTransceiverPlatform>> CreateOffer(
+      RTCSessionDescriptionRequest*,
+      const MediaConstraints&) = 0;
+  virtual Vector<std::unique_ptr<RTCRtpTransceiverPlatform>> CreateOffer(
+      RTCSessionDescriptionRequest*,
+      RTCOfferOptionsPlatform*) = 0;
+  virtual void CreateAnswer(RTCSessionDescriptionRequest*,
+                            const MediaConstraints&) = 0;
+  virtual void CreateAnswer(RTCSessionDescriptionRequest*,
+                            RTCAnswerOptionsPlatform*) = 0;
+  virtual void SetLocalDescription(RTCVoidRequest*) = 0;
+  virtual void SetLocalDescription(RTCVoidRequest*,
+                                   RTCSessionDescriptionPlatform*) = 0;
+  virtual void SetRemoteDescription(RTCVoidRequest*,
+                                    RTCSessionDescriptionPlatform*) = 0;
+  virtual RTCSessionDescriptionPlatform* LocalDescription() = 0;
+  virtual RTCSessionDescriptionPlatform* RemoteDescription() = 0;
+  virtual RTCSessionDescriptionPlatform* CurrentLocalDescription() = 0;
+  virtual RTCSessionDescriptionPlatform* CurrentRemoteDescription() = 0;
+  virtual RTCSessionDescriptionPlatform* PendingLocalDescription() = 0;
+  virtual RTCSessionDescriptionPlatform* PendingRemoteDescription() = 0;
+  virtual const webrtc::PeerConnectionInterface::RTCConfiguration&
+  GetConfiguration() const = 0;
+  virtual webrtc::RTCErrorType SetConfiguration(
+      const webrtc::PeerConnectionInterface::RTCConfiguration&) = 0;
+
+  virtual void AddICECandidate(RTCVoidRequest*, RTCIceCandidatePlatform*) = 0;
+  virtual void RestartIce() = 0;
+  virtual void GetStats(RTCStatsRequest*) = 0;
+  // Gets stats using the new stats collection API, see
+  // third_party/webrtc/api/stats/.  These will replace the old stats collection
+  // API when the new API has matured enough.
+  virtual void GetStats(RTCStatsReportCallback,
+                        const Vector<webrtc::NonStandardGroupId>&) = 0;
+  virtual scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
+      const String& label,
+      const webrtc::DataChannelInit&) = 0;
+  virtual webrtc::RTCErrorOr<std::unique_ptr<RTCRtpTransceiverPlatform>>
+  AddTransceiverWithTrack(const WebMediaStreamTrack&,
+                          const webrtc::RtpTransceiverInit&) = 0;
+  virtual webrtc::RTCErrorOr<std::unique_ptr<RTCRtpTransceiverPlatform>>
+  AddTransceiverWithKind(
+      // webrtc::MediaStreamTrackInterface::kAudioKind or kVideoKind
+      const String& kind,
+      const webrtc::RtpTransceiverInit&) = 0;
+  // Adds the track to the peer connection, returning the resulting transceiver
+  // or error.
+  virtual webrtc::RTCErrorOr<std::unique_ptr<RTCRtpTransceiverPlatform>>
+  AddTrack(const WebMediaStreamTrack&, const Vector<WebMediaStream>&) = 0;
+  // Removes the sender.
+  // In Plan B: Returns OK() with value nullptr on success. The sender's track
+  // must be nulled by the caller.
+  // In Unified Plan: Returns OK() with the updated transceiver state.
+  virtual webrtc::RTCErrorOr<std::unique_ptr<RTCRtpTransceiverPlatform>>
+  RemoveTrack(RTCRtpSenderPlatform*) = 0;
+
+  // Returns a pointer to the underlying native PeerConnection object.
+  virtual webrtc::PeerConnectionInterface* NativePeerConnection() = 0;
+
+  virtual void RunSynchronousOnceClosureOnSignalingThread(
+      CrossThreadOnceClosure closure,
+      const char* trace_event_name) = 0;
+  virtual void RunSynchronousRepeatingClosureOnSignalingThread(
+      const base::RepeatingClosure& closure,
+      const char* trace_event_name) = 0;
+
+  // Inform chrome://webrtc-internals/ that the iceConnectionState has changed.
+  virtual void TrackIceConnectionStateChange(
+      IceConnectionStateVersion version,
+      webrtc::PeerConnectionInterface::IceConnectionState state) = 0;
+};
+
+}  // namespace blink
+
+#endif  // THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_PEER_CONNECTION_HANDLER_PLATFORM_H_