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diff --git a/dom/media/webrtc/libwebrtcglue/VideoConduit.h b/dom/media/webrtc/libwebrtcglue/VideoConduit.h
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+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.h
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+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef VIDEO_SESSION_H_
+#define VIDEO_SESSION_H_
+
+#include "mozilla/Atomics.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/DataMutex.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/SharedThreadPool.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/UniquePtr.h"
+#include "nsITimer.h"
+
+#include "MediaConduitInterface.h"
+#include "common/MediaEngineWrapper.h"
+#include "RtpRtcpConfig.h"
+#include "RunningStat.h"
+#include "transport/runnable_utils.h"
+
+// conflicts with #include of scoped_ptr.h
+#undef FF
+// Video Engine Includes
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "call/call_basic_stats.h"
+#include "common_video/include/video_frame_buffer_pool.h"
+#include "media/base/video_broadcaster.h"
+#include <functional>
+#include <memory>
+/** This file hosts several structures identifying different aspects
+ * of a RTP Session.
+ */
+
+namespace mozilla {
+
+// Convert (SI) kilobits/sec to (SI) bits/sec
+#define KBPS(kbps) kbps * 1000
+
+const int kViEMinCodecBitrate_bps = KBPS(30);
+const unsigned int kVideoMtu = 1200;
+const int kQpMax = 56;
+
+template <typename T>
+T MinIgnoreZero(const T& a, const T& b) {
+  return std::min(a ? a : b, b ? b : a);
+}
+
+class VideoStreamFactory;
+class WebrtcAudioConduit;
+
+// Interface of external video encoder for WebRTC.
+class WebrtcVideoEncoder : public VideoEncoder, public webrtc::VideoEncoder {};
+
+// Interface of external video decoder for WebRTC.
+class WebrtcVideoDecoder : public VideoDecoder, public webrtc::VideoDecoder {};
+
+/**
+ * Concrete class for Video session. Hooks up
+ *  - media-source and target to external transport
+ */
+class WebrtcVideoConduit
+    : public VideoSessionConduit,
+      public webrtc::RtcpEventObserver,
+      public rtc::VideoSinkInterface<webrtc::VideoFrame>,
+      public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+ public:
+  // Returns true when both encoder and decoder are HW accelerated.
+  static bool HasH264Hardware();
+
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
+  /**
+   * Function to attach Renderer end-point for the Media-Video conduit.
+   * @param aRenderer : Reference to the concrete mozilla Video renderer
+   * implementation Note: Multiple invocations of this API shall remove an
+   * existing renderer and attaches the new to the Conduit.
+   */
+  MediaConduitErrorCode AttachRenderer(
+      RefPtr<mozilla::VideoRenderer> aVideoRenderer) override;
+  void DetachRenderer() override;
+
+  Maybe<uint16_t> RtpSendBaseSeqFor(uint32_t aSsrc) const override;
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const override;
+
+  void StopTransmitting();
+  void StartTransmitting();
+  void StopReceiving();
+  void StartReceiving();
+
+  /**
+   * Function to deliver a capture video frame for encoding and transport.
+   * If the frame's timestamp is 0, it will be automatically generated.
+   *
+   * NOTE: ConfigureSendMediaCodec() must be called before this function can
+   *       be invoked. This ensures the inserted video-frames can be
+   *       transmitted by the conduit.
+   */
+  MediaConduitErrorCode SendVideoFrame(webrtc::VideoFrame aFrame) override;
+
+  bool SendRtp(const uint8_t* aData, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override;
+  bool SendSenderRtcp(const uint8_t* aData, size_t aLength) override;
+  bool SendReceiverRtcp(const uint8_t* aData, size_t aLength) override;
+
+  /*
+   * webrtc:VideoSinkInterface implementation
+   * -------------------------------
+   */
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+  /*
+   * webrtc:VideoSourceInterface implementation
+   * -------------------------------
+   */
+  void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+  bool HasCodecPluginID(uint64_t aPluginID) const override;
+
+  RefPtr<GenericPromise> Shutdown() override;
+
+  bool Denoising() const { return mDenoising; }
+
+  uint8_t SpatialLayers() const { return mSpatialLayers; }
+
+  uint8_t TemporalLayers() const { return mTemporalLayers; }
+
+  webrtc::VideoCodecMode CodecMode() const;
+
+  WebrtcVideoConduit(RefPtr<WebrtcCallWrapper> aCall,
+                     nsCOMPtr<nsISerialEventTarget> aStsThread,
+                     Options aOptions, std::string aPCHandle,
+                     const TrackingId& aRecvTrackingId);
+  virtual ~WebrtcVideoConduit();
+
+  // Call thread.
+  void InitControl(VideoConduitControlInterface* aControl) override;
+
+  // Called when a parameter in mControl has changed. Call thread.
+  void OnControlConfigChange();
+
+  // Necessary Init steps on main thread.
+  MediaConduitErrorCode Init();
+
+  Ssrcs GetLocalSSRCs() const override;
+  Maybe<Ssrc> GetAssociatedLocalRtxSSRC(Ssrc aSsrc) const override;
+  Maybe<Ssrc> GetRemoteSSRC() const override;
+
+  // Call thread.
+  void UnsetRemoteSSRC(uint32_t aSsrc) override;
+
+  static unsigned ToLibwebrtcMaxFramerate(const Maybe<double>& aMaxFramerate);
+
+ private:
+  void NotifyUnsetCurrentRemoteSSRC();
+  void SetRemoteSSRCConfig(uint32_t aSsrc, uint32_t aRtxSsrc);
+  void SetRemoteSSRCAndRestartAsNeeded(uint32_t aSsrc, uint32_t aRtxSsrc);
+  rtc::RefCountedObject<mozilla::VideoStreamFactory>*
+  CreateVideoStreamFactory();
+
+ public:
+  // Creating a recv stream or a send stream requires a local ssrc to be
+  // configured. This method will generate one if needed.
+  void EnsureLocalSSRC();
+  // Creating a recv stream requires a remote ssrc to be configured. This method
+  // will generate one if needed.
+  void EnsureRemoteSSRC();
+
+  Maybe<webrtc::VideoReceiveStreamInterface::Stats> GetReceiverStats()
+      const override;
+  Maybe<webrtc::VideoSendStream::Stats> GetSenderStats() const override;
+  Maybe<webrtc::CallBasicStats> GetCallStats() const override;
+
+  bool AddFrameHistory(dom::Sequence<dom::RTCVideoFrameHistoryInternal>*
+                           outHistories) const override;
+
+  void SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) override;
+
+  uint64_t MozVideoLatencyAvg();
+
+  void DisableSsrcChanges() override {
+    MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+    mAllowSsrcChange = false;
+  }
+
+  void CollectTelemetryData() override;
+
+  void OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                     webrtc::RTPHeader&& aHeader);
+  void OnRtcpReceived(MediaPacket&& aPacket);
+
+  void OnRtcpBye() override;
+  void OnRtcpTimeout() override;
+
+  void SetTransportActive(bool aActive) override;
+
+  MediaEventSourceExc<MediaPacket>& SenderRtpSendEvent() override {
+    return mSenderRtpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& SenderRtcpSendEvent() override {
+    return mSenderRtcpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& ReceiverRtcpSendEvent() override {
+    return mReceiverRtcpSendEvent;
+  }
+  void ConnectReceiverRtpEvent(
+      MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>& aEvent)
+      override {
+    mReceiverRtpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtpReceived);
+  }
+  void ConnectReceiverRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mReceiverRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtcpReceived);
+  }
+  void ConnectSenderRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mSenderRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtcpReceived);
+  }
+
+  std::vector<webrtc::RtpSource> GetUpstreamRtpSources() const override;
+
+ private:
+  // Don't allow copying/assigning.
+  WebrtcVideoConduit(const WebrtcVideoConduit&) = delete;
+  void operator=(const WebrtcVideoConduit&) = delete;
+
+  // Utility function to dump recv codec database
+  void DumpCodecDB() const;
+
+  // Video Latency Test averaging filter
+  void VideoLatencyUpdate(uint64_t aNewSample);
+
+  void CreateSendStream();
+  void DeleteSendStream();
+  void CreateRecvStream();
+  void DeleteRecvStream();
+
+  void DeliverPacket(rtc::CopyOnWriteBuffer packet, PacketType type) override;
+
+  MediaEventSource<void>& RtcpByeEvent() override { return mRtcpByeEvent; }
+  MediaEventSource<void>& RtcpTimeoutEvent() override {
+    return mRtcpTimeoutEvent;
+  }
+  MediaEventSource<void>& RtpPacketEvent() override { return mRtpPacketEvent; }
+
+  bool RequiresNewSendStream(const VideoCodecConfig& newConfig) const;
+
+  mutable mozilla::ReentrantMonitor mRendererMonitor MOZ_UNANNOTATED;
+
+  // Accessed on any thread under mRendererMonitor.
+  RefPtr<mozilla::VideoRenderer> mRenderer;
+
+  // Accessed on any thread under mRendererMonitor.
+  unsigned short mReceivingWidth = 0;
+
+  // Accessed on any thread under mRendererMonitor.
+  unsigned short mReceivingHeight = 0;
+
+  // Call worker thread. All access to mCall->Call() happens here.
+  const nsCOMPtr<nsISerialEventTarget> mCallThread;
+
+  // Socket transport service thread that runs stats queries against us. Any
+  // thread.
+  const nsCOMPtr<nsISerialEventTarget> mStsThread;
+
+  // Thread on which we are fed video frames. Set lazily on first call to
+  // SendVideoFrame().
+  nsCOMPtr<nsISerialEventTarget> mFrameSendingThread;
+
+  struct Control {
+    // Mirrors that map to VideoConduitControlInterface for control. Call thread
+    // only.
+    Mirror<bool> mReceiving;
+    Mirror<bool> mTransmitting;
+    Mirror<Ssrcs> mLocalSsrcs;
+    Mirror<Ssrcs> mLocalRtxSsrcs;
+    Mirror<std::string> mLocalCname;
+    Mirror<std::string> mMid;
+    Mirror<Ssrc> mRemoteSsrc;
+    Mirror<Ssrc> mRemoteRtxSsrc;
+    Mirror<std::string> mSyncGroup;
+    Mirror<RtpExtList> mLocalRecvRtpExtensions;
+    Mirror<RtpExtList> mLocalSendRtpExtensions;
+    Mirror<Maybe<VideoCodecConfig>> mSendCodec;
+    Mirror<Maybe<RtpRtcpConfig>> mSendRtpRtcpConfig;
+    Mirror<std::vector<VideoCodecConfig>> mRecvCodecs;
+    Mirror<Maybe<RtpRtcpConfig>> mRecvRtpRtcpConfig;
+    Mirror<webrtc::VideoCodecMode> mCodecMode;
+
+    // For caching mRemoteSsrc and mRemoteRtxSsrc, since another caller may
+    // change the remote ssrc in the stream config directly.
+    Ssrc mConfiguredRemoteSsrc = 0;
+    Ssrc mConfiguredRemoteRtxSsrc = 0;
+    // For tracking changes to mSendCodec and mSendRtpRtcpConfig.
+    Maybe<VideoCodecConfig> mConfiguredSendCodec;
+    Maybe<RtpRtcpConfig> mConfiguredSendRtpRtcpConfig;
+    // For tracking changes to mRecvCodecs and mRecvRtpRtcpConfig.
+    std::vector<VideoCodecConfig> mConfiguredRecvCodecs;
+    Maybe<RtpRtcpConfig> mConfiguredRecvRtpRtcpConfig;
+
+    Control() = delete;
+    explicit Control(const RefPtr<AbstractThread>& aCallThread);
+  } mControl;
+
+  // WatchManager allowing Mirrors and other watch targets to trigger functions
+  // that will update the webrtc.org configuration.
+  WatchManager<WebrtcVideoConduit> mWatchManager;
+
+  mutable Mutex mMutex MOZ_UNANNOTATED;
+
+  // Decoder factory used by mRecvStream when it needs new decoders. This is
+  // not shared broader like some state in the WebrtcCallWrapper because it
+  // handles CodecPluginID plumbing tied to this VideoConduit.
+  const UniquePtr<WebrtcVideoDecoderFactory> mDecoderFactory;
+
+  // Encoder factory used by mSendStream when it needs new encoders. This is
+  // not shared broader like some state in the WebrtcCallWrapper because it
+  // handles CodecPluginID plumbing tied to this VideoConduit.
+  const UniquePtr<WebrtcVideoEncoderFactory> mEncoderFactory;
+
+  // Our own record of the sinks added to mVideoBroadcaster so we can support
+  // dispatching updates to sinks from off-Call-thread. Call thread only.
+  AutoTArray<rtc::VideoSinkInterface<webrtc::VideoFrame>*, 1> mRegisteredSinks;
+
+  // Broadcaster that distributes our frames to all registered sinks.
+  // Threadsafe.
+  rtc::VideoBroadcaster mVideoBroadcaster;
+
+  // Buffer pool used for scaling frames.
+  // Accessed on the frame-feeding thread only.
+  webrtc::VideoFrameBufferPool mBufferPool;
+
+  // Engine state we are concerned with. Written on the Call thread and read
+  // anywhere.
+  mozilla::Atomic<bool>
+      mEngineTransmitting;  // If true ==> Transmit Subsystem is up and running
+  mozilla::Atomic<bool>
+      mEngineReceiving;  // if true ==> Receive Subsystem up and running
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread.
+  Maybe<VideoCodecConfig> mCurSendCodecConfig;
+
+  // Bookkeeping of stats for telemetry. Call thread only.
+  RunningStat mSendFramerate;
+  RunningStat mSendBitrate;
+  RunningStat mRecvFramerate;
+  RunningStat mRecvBitrate;
+
+  // Must call webrtc::Call::DestroyVideoReceive/SendStream to delete this.
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread.
+  webrtc::VideoReceiveStreamInterface* mRecvStream = nullptr;
+
+  // Must call webrtc::Call::DestroyVideoReceive/SendStream to delete this.
+  webrtc::VideoSendStream* mSendStream = nullptr;
+
+  // Written on the frame feeding thread.
+  // Guarded by mMutex, except for reads on the frame feeding thread.
+  unsigned short mLastWidth = 0;
+
+  // Written on the frame feeding thread.
+  // Guarded by mMutex, except for reads on the frame feeding thread.
+  unsigned short mLastHeight = 0;
+
+  // Written on the frame feeding thread, the timestamp of the last frame on the
+  // send side, in microseconds. This is a local timestamp using the system
+  // clock with a unspecified epoch (Like mozilla::TimeStamp).
+  // Guarded by mMutex.
+  Maybe<uint64_t> mLastTimestampSendUs;
+
+  // Written on the frame receive thread, the rtp timestamp of the last frame
+  // on the receive side, in 90kHz base. This comes from the RTP packet.
+  // Guarded by mMutex.
+  Maybe<uint32_t> mLastRTPTimestampReceive;
+
+  // Accessed from any thread under mRendererMonitor.
+  uint64_t mVideoLatencyAvg = 0;
+
+  const bool mVideoLatencyTestEnable;
+
+  // All in bps.
+  const int mMinBitrate;
+  const int mStartBitrate;
+  const int mPrefMaxBitrate;
+  const int mMinBitrateEstimate;
+
+  // Max bitrate in bps as provided by negotiation. Call thread only.
+  int mNegotiatedMaxBitrate = 0;
+
+  // Set to true to force denoising on.
+  const bool mDenoising;
+
+  // Set to true to ignore sink wants (scaling due to bwe and cpu usage).
+  const bool mLockScaling;
+
+  const uint8_t mSpatialLayers;
+  const uint8_t mTemporalLayers;
+
+  static const unsigned int sAlphaNum = 7;
+  static const unsigned int sAlphaDen = 8;
+  static const unsigned int sRoundingPadding = 1024;
+
+  // Target jitter buffer to be applied to the receive stream in milliseconds.
+  uint16_t mJitterBufferTargetMs = 0;
+
+  // WEBRTC.ORG Call API
+  // Const so can be accessed on any thread. All methods are called on the Call
+  // thread.
+  const RefPtr<WebrtcCallWrapper> mCall;
+
+  // Set up in the ctor and then not touched. Called through by the streams on
+  // any thread. Safe since we own and control the lifetime of the streams.
+  WebrtcSendTransport mSendTransport;
+  WebrtcReceiveTransport mRecvTransport;
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread. Typical non-Call thread access is on the frame delivery
+  // thread.
+  webrtc::VideoSendStream::Config mSendStreamConfig;
+
+  // Call thread only.
+  webrtc::VideoEncoderConfig mEncoderConfig;
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread. Calls can happen under mMutex on any thread.
+  DataMutex<RefPtr<rtc::RefCountedObject<VideoStreamFactory>>>
+      mVideoStreamFactory;
+
+  // Call thread only.
+  webrtc::VideoReceiveStreamInterface::Config mRecvStreamConfig;
+
+  // Are SSRC changes without signaling allowed or not.
+  // Call thread only.
+  bool mAllowSsrcChange = true;
+
+  // Accessed during configuration/signaling (Call thread), and on the frame
+  // delivery thread for frame history tracking. Set only on the Call thread.
+  Atomic<uint32_t> mRecvSSRC =
+      Atomic<uint32_t>(0);  // this can change during a stream!
+
+  // Accessed from both the STS and frame delivery thread for frame history
+  // tracking. Set when receiving packets.
+  Atomic<uint32_t> mRemoteSendSSRC =
+      Atomic<uint32_t>(0);  // this can change during a stream!
+
+  // Main thread only
+  nsTArray<uint64_t> mSendCodecPluginIDs;
+  // Main thread only
+  nsTArray<uint64_t> mRecvCodecPluginIDs;
+
+  // Main thread only
+  MediaEventListener mSendPluginCreated;
+  MediaEventListener mSendPluginReleased;
+  MediaEventListener mRecvPluginCreated;
+  MediaEventListener mRecvPluginReleased;
+
+  // Call thread only. ssrc -> base_seq
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs;
+  // libwebrtc network thread only. ssrc -> base_seq.
+  // To track changes needed to mRtpSendBaseSeqs.
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs_n;
+
+  // Tracking the attributes of received frames over time
+  // Protected by mRendererMonitor
+  dom::RTCVideoFrameHistoryInternal mReceivedFrameHistory;
+
+  // Thread safe
+  Atomic<bool> mTransportActive = Atomic<bool>(false);
+  MediaEventProducer<void> mRtcpByeEvent;
+  MediaEventProducer<void> mRtcpTimeoutEvent;
+  MediaEventProducer<void> mRtpPacketEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtpSendEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpSendEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpSendEvent;
+
+  // Assigned and revoked on mStsThread. Listeners for receiving packets.
+  MediaEventListener mSenderRtcpEventListener;    // Rtp-transmitting pipeline
+  MediaEventListener mReceiverRtcpEventListener;  // Rtp-receiving pipeline
+  MediaEventListener mReceiverRtpEventListener;   // Rtp-receiving pipeline
+};
+}  // namespace mozilla
+
+#endif