modules/javafx.web/src/main/native/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp

*** 33,42 ****
--- 33,43 ----
  #include "LibWebRTCPeerConnectionBackend.h"
  #include "LibWebRTCProvider.h"
  #include "Logging.h"
  #include "MediaStreamEvent.h"
  #include "NotImplemented.h"
+ #include "Performance.h"
  #include "PlatformStrategies.h"
  #include "RTCDataChannel.h"
  #include "RTCDataChannelEvent.h"
  #include "RTCOfferOptions.h"
  #include "RTCPeerConnection.h"
*** 48,57 ****
--- 49,59 ----
  #include "RuntimeEnabledFeatures.h"
  #include <webrtc/base/physicalsocketserver.h>
  #include <webrtc/p2p/base/basicpacketsocketfactory.h>
  #include <webrtc/p2p/client/basicportallocator.h>
  #include <webrtc/pc/peerconnectionfactory.h>
+ #include <wtf/CurrentTime.h>
  #include <wtf/MainThread.h>
  
  #include "CoreMediaSoftLink.h"
  
  namespace WebCore {
*** 301,311 ****
      return String(value.data(), value.length());
  }
  
  static inline void fillRTCStats(RTCStatsReport::Stats& stats, const webrtc::RTCStats& rtcStats)
  {
!     stats.timestamp = rtcStats.timestamp_us() / 1000.0;
      stats.id = fromStdString(rtcStats.id());
  }
  
  static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
  {
--- 303,313 ----
      return String(value.data(), value.length());
  }
  
  static inline void fillRTCStats(RTCStatsReport::Stats& stats, const webrtc::RTCStats& rtcStats)
  {
!     stats.timestamp = Performance::reduceTimeResolution(Seconds::fromMicroseconds(rtcStats.timestamp_us())).milliseconds();
      stats.id = fromStdString(rtcStats.id());
  }
  
  static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
  {