@epicgames-ps/lib-pixelstreamingfrontend-ue5.5 0.4.7 → 1.0.1

package/dist/esm/PeerConnectionController/AggregatedStats.js

@@ -2,16 +2,22 @@
 import { InboundVideoStats, InboundAudioStats } from './InboundRTPStats';
 import { DataChannelStats } from './DataChannelStats';
 import { CandidateStat } from './CandidateStat';
-import { OutBoundVideoStats } from './OutBoundRTPStats';
+import { RemoteOutboundRTPStats, OutboundRTPStats } from './OutBoundRTPStats';
 import { SessionStats } from './SessionStats';
 import { StreamStats } from './StreamStats';
 import { Logger } from '@epicgames-ps/lib-pixelstreamingcommon-ue5.5';
+/**
+ * The Aggregated Stats that is generated from the RTC Stats Report
+ */
 export class AggregatedStats {
     constructor() {
         this.inboundVideoStats = new InboundVideoStats();
         this.inboundAudioStats = new InboundAudioStats();
-        this.DataChannelStats = new DataChannelStats();
-        this.outBoundVideoStats = new OutBoundVideoStats();
+        this.datachannelStats = new DataChannelStats();
+        this.outboundVideoStats = new OutboundRTPStats();
+        this.outboundAudioStats = new OutboundRTPStats();
+        this.remoteOutboundAudioStats = new RemoteOutboundRTPStats();
+        this.remoteOutboundVideoStats = new RemoteOutboundRTPStats();
         this.sessionStats = new SessionStats();
         this.streamStats = new StreamStats();
         this.codecs = new Map();
@@ -39,7 +45,7 @@ export class AggregatedStats {
                     this.handleDataChannel(stat);
                     break;
                 case 'inbound-rtp':
-                    this.handleInBoundRTP(stat);
+                    this.handleInboundRTP(stat);
                     break;
                 case 'local-candidate':
                     this.handleLocalCandidate(stat);
@@ -49,6 +55,7 @@ export class AggregatedStats {
                 case 'media-playout':
                     break;
                 case 'outbound-rtp':
+                    this.handleLocalOutbound(stat);
                     break;
                 case 'peer-connection':
                     break;
@@ -58,7 +65,7 @@ export class AggregatedStats {
                 case 'remote-inbound-rtp':
                     break;
                 case 'remote-outbound-rtp':
-                    this.handleRemoteOutBound(stat);
+                    this.handleRemoteOutbound(stat);
                     break;
                 case 'track':
                     this.handleTrack(stat);
@@ -97,16 +104,16 @@ export class AggregatedStats {
      * @param stat - the stats coming in from the data channel
      */
     handleDataChannel(stat) {
-        this.DataChannelStats.bytesReceived = stat.bytesReceived;
-        this.DataChannelStats.bytesSent = stat.bytesSent;
-        this.DataChannelStats.dataChannelIdentifier = stat.dataChannelIdentifier;
-        this.DataChannelStats.id = stat.id;
-        this.DataChannelStats.label = stat.label;
-        this.DataChannelStats.messagesReceived = stat.messagesReceived;
-        this.DataChannelStats.messagesSent = stat.messagesSent;
-        this.DataChannelStats.protocol = stat.protocol;
-        this.DataChannelStats.state = stat.state;
-        this.DataChannelStats.timestamp = stat.timestamp;
+        this.datachannelStats.bytesReceived = stat.bytesReceived;
+        this.datachannelStats.bytesSent = stat.bytesSent;
+        this.datachannelStats.dataChannelIdentifier = stat.dataChannelIdentifier;
+        this.datachannelStats.id = stat.id;
+        this.datachannelStats.label = stat.label;
+        this.datachannelStats.messagesReceived = stat.messagesReceived;
+        this.datachannelStats.messagesSent = stat.messagesSent;
+        this.datachannelStats.protocol = stat.protocol;
+        this.datachannelStats.state = stat.state;
+        this.datachannelStats.timestamp = stat.timestamp;
     }
     /**
      * Process the Local Ice Candidate Data
@@ -129,22 +136,22 @@ export class AggregatedStats {
      * @param stat - ice candidate stats
      */
     handleRemoteCandidate(stat) {
-        const RemoteCandidate = new CandidateStat();
-        RemoteCandidate.label = 'remote-candidate';
-        RemoteCandidate.address = stat.address;
-        RemoteCandidate.port = stat.port;
-        RemoteCandidate.protocol = stat.protocol;
-        RemoteCandidate.id = stat.id;
-        RemoteCandidate.candidateType = stat.candidateType;
-        RemoteCandidate.relayProtocol = stat.relayProtocol;
-        RemoteCandidate.transportId = stat.transportId;
-        this.remoteCandidates.push(RemoteCandidate);
+        const remoteCandidate = new CandidateStat();
+        remoteCandidate.label = 'remote-candidate';
+        remoteCandidate.address = stat.address;
+        remoteCandidate.port = stat.port;
+        remoteCandidate.protocol = stat.protocol;
+        remoteCandidate.id = stat.id;
+        remoteCandidate.candidateType = stat.candidateType;
+        remoteCandidate.relayProtocol = stat.relayProtocol;
+        remoteCandidate.transportId = stat.transportId;
+        this.remoteCandidates.push(remoteCandidate);
     }
     /**
      * Process the Inbound RTP Audio and Video Data
      * @param stat - inbound rtp stats
      */
-    handleInBoundRTP(stat) {
+    handleInboundRTP(stat) {
         switch (stat.kind) {
             case 'video':
                 // Calculate bitrate between stat updates
@@ -179,24 +186,60 @@ export class AggregatedStats {
         }
     }
     /**
-     * Process the outbound RTP Audio and Video Data
-     * @param stat - remote outbound stats
+     * Process the "local" outbound RTP Audio and Video stats.
+     * @param stat - local outbound rtp stats
      */
-    handleRemoteOutBound(stat) {
-        switch (stat.kind) {
-            case 'video':
-                this.outBoundVideoStats.bytesSent = stat.bytesSent;
-                this.outBoundVideoStats.id = stat.id;
-                this.outBoundVideoStats.localId = stat.localId;
-                this.outBoundVideoStats.packetsSent = stat.packetsSent;
-                this.outBoundVideoStats.remoteTimestamp = stat.remoteTimestamp;
-                this.outBoundVideoStats.timestamp = stat.timestamp;
-                break;
-            case 'audio':
-                break;
-            default:
-                break;
-        }
+    handleLocalOutbound(stat) {
+        const localOutboundStats = stat.kind === 'audio' ? this.outboundAudioStats : this.outboundVideoStats;
+        localOutboundStats.active = stat.active;
+        localOutboundStats.codecId = stat.codecId;
+        localOutboundStats.bytesSent = stat.bytesSent;
+        localOutboundStats.frameHeight = stat.frameHeight;
+        localOutboundStats.frameWidth = stat.frameWidth;
+        localOutboundStats.framesEncoded = stat.framesEncoded;
+        localOutboundStats.framesPerSecond = stat.framesPerSecond;
+        localOutboundStats.headerBytesSent = stat.headerBytesSent;
+        localOutboundStats.id = stat.id;
+        localOutboundStats.keyFramesEncoded = stat.keyFramesEncoded;
+        localOutboundStats.kind = stat.kind;
+        localOutboundStats.mediaSourceId = stat.mediaSourceId;
+        localOutboundStats.mid = stat.mid;
+        localOutboundStats.nackCount = stat.nackCount;
+        localOutboundStats.packetsSent = stat.packetsSent;
+        localOutboundStats.qpSum = stat.qpSum;
+        localOutboundStats.qualityLimitationDurations = stat.qualityLimitationDurations;
+        localOutboundStats.qualityLimitationReason = stat.qualityLimitationReason;
+        localOutboundStats.remoteId = stat.remoteId;
+        localOutboundStats.retransmittedBytesSent = stat.retransmittedBytesSent;
+        localOutboundStats.rid = stat.rid;
+        localOutboundStats.scalabilityMode = stat.scalabilityMode;
+        localOutboundStats.ssrc = stat.ssrc;
+        localOutboundStats.targetBitrate = stat.targetBitrate;
+        localOutboundStats.timestamp = stat.timestamp;
+        localOutboundStats.totalEncodeTime = stat.totalEncodeTime;
+        localOutboundStats.totalEncodeBytesTarget = stat.totalEncodeBytesTarget;
+        localOutboundStats.totalPacketSendDelay = stat.totalPacketSendDelay;
+        localOutboundStats.transportId = stat.transportId;
+    }
+    /**
+     * Process the "remote" outbound RTP Audio and Video stats.
+     * @param stat - remote outbound rtp stats
+     */
+    handleRemoteOutbound(stat) {
+        const remoteOutboundStats = stat.kind === 'audio' ? this.remoteOutboundAudioStats : this.remoteOutboundVideoStats;
+        remoteOutboundStats.bytesSent = stat.bytesSent;
+        remoteOutboundStats.codecId = stat.codecId;
+        remoteOutboundStats.id = stat.id;
+        remoteOutboundStats.kind = stat.kind;
+        remoteOutboundStats.localId = stat.localId;
+        remoteOutboundStats.packetsSent = stat.packetsSent;
+        remoteOutboundStats.remoteTimestamp = stat.remoteTimestamp;
+        remoteOutboundStats.reportsSent = stat.reportsSent;
+        remoteOutboundStats.roundTripTimeMeasurements = stat.roundTripTimeMeasurements;
+        remoteOutboundStats.ssrc = stat.ssrc;
+        remoteOutboundStats.timestamp = stat.timestamp;
+        remoteOutboundStats.totalRoundTripTime = stat.totalRoundTripTime;
+        remoteOutboundStats.transportId = stat.transportId;
     }
     /**
      * Process the Inbound Video Track Data
@@ -237,13 +280,28 @@ export class AggregatedStats {
      * @returns The candidate pair that is currently receiving data
      */
     getActiveCandidatePair() {
+        if (this.candidatePairs === undefined) {
+            return null;
+        }
         // Check if the RTCTransport stat is not undefined
         if (this.transportStats) {
             // Return the candidate pair that matches the transport candidate pair id
-            return this.candidatePairs.find((candidatePair) => candidatePair.id === this.transportStats.selectedCandidatePairId, null);
+            const selectedPair = this.candidatePairs.find((candidatePair) => candidatePair.id === this.transportStats.selectedCandidatePairId);
+            if (selectedPair === undefined) {
+                return null;
+            }
+            else {
+                return selectedPair;
+            }
+        }
+        // Fall back to the `.selected` member of the candidate pair
+        const selectedPair = this.candidatePairs.find((candidatePair) => candidatePair.selected);
+        if (selectedPair === undefined) {
+            return null;
+        }
+        else {
+            return selectedPair;
         }
-        // Fall back to the selected candidate pair
-        return this.candidatePairs.find((candidatePair) => candidatePair.selected, null);
     }
 }
 //# sourceMappingURL=AggregatedStats.js.map

package/dist/esm/PeerConnectionController/AggregatedStats.js.map

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package/dist/esm/PeerConnectionController/InboundRTPStats.js.map

@@ -1 +1 @@
-{"version":3,"file":"InboundRTPStats.js","sourceRoot":"","sources":["../../../src/PeerConnectionController/InboundRTPStats.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD;;GAEG;AACH,MAAM,OAAO,iBAAiB;CAoC7B;AAED;;GAEG;AACH,MAAM,OAAO,iBAAiB;CA2C7B;AAED;;GAEG;AACH,MAAM,OAAO,eAAe;CA2D3B"}
+{"version":3,"file":"InboundRTPStats.js","sourceRoot":"","sources":["../../../src/PeerConnectionController/InboundRTPStats.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD;;GAEG;AACH,MAAM,OAAO,iBAAiB;CAoC7B;AAED;;GAEG;AACH,MAAM,OAAO,iBAAiB;CA2C7B;AAED;;GAEG;AACH,MAAM,OAAO,eAAe;CA8D3B"}

package/dist/esm/PeerConnectionController/LatencyCalculator.js

@@ -0,0 +1,284 @@
+// Copyright Epic Games, Inc. All Rights Reserved.
+/**
+ * Represents either a:
+ * - synchronization source: https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/getSynchronizationSources
+ * - contributing source: https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
+ * Which also (if browser supports it) may optionall contain fields for captureTimestamp + senderCaptureTimeOffset
+ * if the abs-capture-time RTP header extension is enabled (currently this only works in Chromium based browsers).
+ */
+class RTCRtpCaptureSource {
+}
+/**
+ * FrameTimingInfo is a Chromium-specific set of WebRTC stats useful for latency calculation. It is stored in WebRTC stats as `googTimingFrameInfo`.
+ * It is defined as an RTP header extension here: https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/video-timing/README.md
+ * It is defined in source code here: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video/video_timing.cc;l=82;drc=8d399817282e3c12ed54eb23ec42a5e418298ec6
+ * It is discussed by its author here: https://github.com/w3c/webrtc-provisional-stats/issues/40#issuecomment-1272916692
+ * In summary it a comma-delimited string that contains the following (in this order):
+ * 1)  RTP timestamp: the RTP timestamp of the frame
+ * 2)  Capture time: timestamp when this frame was captured
+ * 3)  Encode start: timestamp when this frame started to be encoded
+ * 4)  Encode finish: timestamp when this frame finished encoding
+ * 5)  Packetization finish: timestamp when this frame was split into packets and was ready to be sent over the network
+ * 6)  Pacer exit: timestamp when last packet of this frame was sent over the network by the sender at this timestamp
+ * 7)  Network timestamp1: place for the SFU to mark when the frame started being forwarded. Application specific.
+ * 8)  Network timestamp2: place for the SFU to mark when the frame finished being forwarded. Application specific.
+ * 9)  Receive start: timestamp when the first packet of this frame was received
+ * 10) Receive finish: timestamp when the last packet of this frame was received
+ * 11) Decode start:  timestamp when the frame was passed to decoder
+ * 12) Decode finish:  timestamp when the frame was decoded
+ * 13) Render time:  timestamp of the projected render time for this frame
+ * 14) "is outlier": a flag for if this frame is bigger in encoded size than the average frame by at least 5x.
+ * 15) "triggered by timer": a flag for if this report was triggered by the timer (The report is sent every 200ms)
+ */
+export class FrameTimingInfo {
+}
+/**
+ * Calculates a combination of latency statistics using purely WebRTC API.
+ */
+export class LatencyCalculator {
+    constructor() {
+        /* Clock offset between peer clocks cannot always be calculated as it relies of latest sender reports.
+         * so we store the last time we had a valid clock offset in the assumption that clocks haven't drifted too much since then.
+         */
+        this.latestSenderRecvClockOffset = null;
+    }
+    calculate(stats, receivers) {
+        const latencyInfo = new LatencyInfo();
+        const rttMS = this.getRTTMs(stats);
+        if (rttMS != null) {
+            latencyInfo.rttMs = rttMS;
+            // Calculate sender latency using the first valid video ssrc/csrc
+            const captureSource = this.getCaptureSource(receivers);
+            if (captureSource != null) {
+                const senderLatencyMs = this.calculateSenderLatency(stats, captureSource);
+                if (senderLatencyMs !== null) {
+                    latencyInfo.senderLatencyMs = senderLatencyMs;
+                }
+            }
+        }
+        // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
+        if (stats.inboundVideoStats.totalProcessingDelay !== undefined &&
+            stats.inboundVideoStats.framesDecoded !== undefined) {
+            latencyInfo.averageProcessingDelayMs =
+                (stats.inboundVideoStats.totalProcessingDelay / stats.inboundVideoStats.framesDecoded) * 1000;
+        }
+        // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay
+        if (stats.inboundVideoStats.jitterBufferDelay !== undefined &&
+            stats.inboundVideoStats.jitterBufferEmittedCount !== undefined) {
+            latencyInfo.averageJitterBufferDelayMs =
+                (stats.inboundVideoStats.jitterBufferDelay /
+                    stats.inboundVideoStats.jitterBufferEmittedCount) *
+                    1000;
+        }
+        // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
+        if (stats.inboundVideoStats.framesDecoded !== undefined &&
+            stats.inboundVideoStats.totalDecodeTime !== undefined) {
+            latencyInfo.averageDecodeLatencyMs =
+                (stats.inboundVideoStats.totalDecodeTime / stats.inboundVideoStats.framesDecoded) * 1000;
+        }
+        // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesassembledfrommultiplepackets
+        if (stats.inboundVideoStats.totalAssemblyTime !== undefined &&
+            stats.inboundVideoStats.framesAssembledFromMultiplePackets !== undefined) {
+            latencyInfo.averageAssemblyDelayMs =
+                (stats.inboundVideoStats.totalAssemblyTime /
+                    stats.inboundVideoStats.framesAssembledFromMultiplePackets) *
+                    1000;
+        }
+        // Extract extra Chrome-specific stats like encoding latency
+        if (stats.inboundVideoStats.googTimingFrameInfo !== undefined &&
+            stats.inboundVideoStats.googTimingFrameInfo.length > 0) {
+            latencyInfo.frameTiming = this.extractFrameTimingInfo(stats.inboundVideoStats.googTimingFrameInfo);
+        }
+        // Calculate E2E latency using video-timing capture to send time + one way network latency + receiver-side latency
+        if (latencyInfo.frameTiming !== undefined &&
+            latencyInfo.frameTiming.captureToSendLatencyMs !== undefined &&
+            latencyInfo.averageProcessingDelayMs !== undefined &&
+            latencyInfo.rttMs !== undefined) {
+            latencyInfo.averageE2ELatency =
+                latencyInfo.frameTiming.captureToSendLatencyMs +
+                    latencyInfo.rttMs * 0.5 +
+                    latencyInfo.averageProcessingDelayMs;
+        }
+        // Calculate E2E latency as abs-capture-time capture to send latency + one way network latency + receiver-side latency
+        if (latencyInfo.senderLatencyMs != undefined &&
+            latencyInfo.averageProcessingDelayMs !== undefined &&
+            latencyInfo.rttMs !== undefined) {
+            latencyInfo.averageE2ELatency =
+                latencyInfo.senderLatencyMs + latencyInfo.rttMs * 0.5 + latencyInfo.averageProcessingDelayMs;
+        }
+        return latencyInfo;
+    }
+    extractFrameTimingInfo(googTimingFrameInfo) {
+        const timingInfo = new FrameTimingInfo();
+        const timingInfoArr = googTimingFrameInfo.split(',');
+        // Should have exactly 15 elements according to:
+        // https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video/video_timing.cc;l=82;drc=8d399817282e3c12ed54eb23ec42a5e418298ec6
+        if (timingInfoArr.length === 15) {
+            timingInfo.rtpTimestamp = Number.parseInt(timingInfoArr[0]);
+            timingInfo.captureTimestamp = Number.parseInt(timingInfoArr[1]);
+            timingInfo.encodeStartTimestamp = Number.parseInt(timingInfoArr[2]);
+            timingInfo.encodeFinishTimestamp = Number.parseInt(timingInfoArr[3]);
+            timingInfo.packetizerFinishTimestamp = Number.parseInt(timingInfoArr[4]);
+            timingInfo.pacerExitTimestamp = Number.parseInt(timingInfoArr[5]);
+            timingInfo.networkTimestamp1 = Number.parseInt(timingInfoArr[6]);
+            timingInfo.networkTimestamp2 = Number.parseInt(timingInfoArr[7]);
+            timingInfo.receiveStart = Number.parseInt(timingInfoArr[8]);
+            timingInfo.receiveFinish = Number.parseInt(timingInfoArr[9]);
+            timingInfo.decodeStart = Number.parseInt(timingInfoArr[10]);
+            timingInfo.decodeFinish = Number.parseInt(timingInfoArr[11]);
+            timingInfo.renderTime = Number.parseInt(timingInfoArr[12]);
+            timingInfo.isOutlier = Number.parseInt(timingInfoArr[13]) > 0;
+            timingInfo.isTriggeredByTimer = Number.parseInt(timingInfoArr[14]) > 0;
+            // Calculate some latency stats
+            timingInfo.encoderLatencyMs = timingInfo.encodeFinishTimestamp - timingInfo.encodeStartTimestamp;
+            timingInfo.packetizeLatencyMs =
+                timingInfo.packetizerFinishTimestamp - timingInfo.encodeFinishTimestamp;
+            timingInfo.pacerLatencyMs = timingInfo.pacerExitTimestamp - timingInfo.packetizerFinishTimestamp;
+            timingInfo.captureToSendLatencyMs = timingInfo.pacerExitTimestamp - timingInfo.captureTimestamp;
+        }
+        return timingInfo;
+    }
+    calculateSenderLatency(stats, captureSource) {
+        // The calculation performed in this function is as per the procedure defined here:
+        // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpcontributingsource-sendercapturetimeoffset
+        // Get the sender capture in the sender's clock
+        const senderCaptureTimestamp = captureSource.captureTimestamp + captureSource.senderCaptureTimeOffset;
+        let sendRecvClockOffset = this.calculateSenderReceiverClockOffset(stats);
+        // Use latest clock offset if we couldn't calculate one now
+        if (sendRecvClockOffset == null) {
+            if (this.latestSenderRecvClockOffset != null) {
+                sendRecvClockOffset = this.latestSenderRecvClockOffset;
+            }
+            else {
+                return null;
+            }
+        }
+        else {
+            this.latestSenderRecvClockOffset = sendRecvClockOffset;
+        }
+        // This brings sender clock roughly inline with recv clock
+        const recvCaptureTimestampNTP = senderCaptureTimestamp + sendRecvClockOffset;
+        // As defined in Chrome source: https://chromium.googlesource.com/external/webrtc/+/master/system_wrappers/include/clock.h#26
+        const ntp1970 = 2208988800000;
+        const recvCaptureTimestamp = recvCaptureTimestampNTP - ntp1970;
+        const senderLatency = captureSource.timestamp - recvCaptureTimestamp;
+        return senderLatency;
+    }
+    /**
+     * Find the first valid ssrc or csrc that has capture time fields present from abs-capture-time header extension.
+     * @param receivers The RTP receviers this peer connection has.
+     * @returns A single valid ssrc or csrc that has capture time fields or null if there is none (e.g. in non-chromium browsers it will be null).
+     */
+    getCaptureSource(receivers) {
+        // We only want video receivers
+        receivers = receivers.filter((receiver) => receiver.track.kind === 'video');
+        for (const receiver of receivers) {
+            // Go through all ssrc and csrc to check for capture timestamp
+            // Note: Conversion to `any` here is because TS does not have captureTimestamp etc defined in the types
+            // these fields only exist in Chromium currently.
+            const sources = receiver
+                .getSynchronizationSources()
+                .concat(receiver.getContributingSources());
+            for (const src of sources) {
+                if (src.captureTimestamp !== undefined &&
+                    src.senderCaptureTimeOffset !== undefined &&
+                    src.timestamp !== undefined) {
+                    const captureSrc = new RTCRtpCaptureSource();
+                    captureSrc.timestamp = src.timestamp;
+                    captureSrc.captureTimestamp = src.captureTimestamp;
+                    captureSrc.senderCaptureTimeOffset = src.senderCaptureTimeOffset;
+                    return captureSrc;
+                }
+            }
+        }
+        return null;
+    }
+    calculateSenderReceiverClockOffset(stats) {
+        // The calculation performed in this function is as per the procedure defined here:
+        // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpcontributingsource-sendercapturetimeoffset
+        const hasRemoteOutboundVideoStats = stats.remoteOutboundVideoStats !== undefined &&
+            stats.remoteOutboundVideoStats.timestamp !== undefined &&
+            stats.remoteOutboundVideoStats.remoteTimestamp !== undefined;
+        // Note: As of Chrome 132, remote-outbound-rtp stats for video are not yet implemented (audio works).
+        // This codepath should activate once they do begin to work.
+        if (!hasRemoteOutboundVideoStats) {
+            return null;
+        }
+        const remoteStatsArrivedTimestamp = stats.remoteOutboundVideoStats.timestamp;
+        const remoteStatsSentTimestamp = stats.remoteOutboundVideoStats.remoteTimestamp;
+        const rttMs = this.getRTTMs(stats);
+        if (remoteStatsArrivedTimestamp !== undefined &&
+            remoteStatsSentTimestamp !== undefined &&
+            rttMs !== null) {
+            const onewayDelay = rttMs * 0.5;
+            return remoteStatsArrivedTimestamp - (remoteStatsSentTimestamp + onewayDelay);
+        }
+        // Could not get stats to calculate sender/receiver clock offset
+        else {
+            return null;
+        }
+    }
+    getRTTMs(stats) {
+        // Try to get it from the active candidate pair
+        const activeCandidatePair = stats.getActiveCandidatePair();
+        if (!!activeCandidatePair && activeCandidatePair.currentRoundTripTime !== undefined) {
+            const curRTTSeconds = activeCandidatePair.currentRoundTripTime;
+            return curRTTSeconds * 1000;
+        }
+        // Next try to get it from remote-outbound-rtp video stats
+        if (!!stats.remoteOutboundVideoStats &&
+            stats.remoteOutboundVideoStats.totalRoundTripTime !== undefined &&
+            stats.remoteOutboundVideoStats.roundTripTimeMeasurements !== undefined &&
+            stats.remoteOutboundVideoStats.roundTripTimeMeasurements > 0) {
+            const avgRttSeconds = stats.remoteOutboundVideoStats.totalRoundTripTime /
+                stats.remoteOutboundVideoStats.roundTripTimeMeasurements;
+            return avgRttSeconds * 1000;
+        }
+        // Next try to get it from remote-outbound-rtp audio stats
+        if (!!stats.remoteOutboundAudioStats &&
+            stats.remoteOutboundAudioStats.totalRoundTripTime !== undefined &&
+            stats.remoteOutboundAudioStats.roundTripTimeMeasurements !== undefined &&
+            stats.remoteOutboundAudioStats.roundTripTimeMeasurements > 0) {
+            const avgRttSeconds = stats.remoteOutboundAudioStats.totalRoundTripTime /
+                stats.remoteOutboundAudioStats.roundTripTimeMeasurements;
+            return avgRttSeconds * 1000;
+        }
+        return null;
+    }
+}
+/**
+ * A collection of latency information calculated using the WebRTC API.
+ * Most stats are calculated following the spec:
+ * https://w3c.github.io/webrtc-stats/#dictionary-rtcinboundrtpstreamstats-members
+ */
+export class LatencyInfo {
+    constructor() {
+        /**
+         * The time taken from the moment a frame is done capturing to the moment it is sent over the network.
+         * Note: This can only be calculated if both offer and answer contain the
+         * the RTP header extension for `video-timing` (Chrome only for now)
+         */
+        this.senderLatencyMs = undefined;
+        /**
+         * The time taken from the moment a frame is done capturing to the moment it is sent over the network.
+         * Note: This can only be calculated if both offer and answer contain the
+         * the RTP header extension for `abs-capture-time` (Chrome only for now)
+         */
+        this.senderLatencyAbsCaptureTimeMs = undefined;
+        /* The round trip time (milliseconds) between each sender->receiver->sender */
+        this.rttMs = undefined;
+        /* Average time taken (milliseconds) from video packet receipt to post-decode. */
+        this.averageProcessingDelayMs = undefined;
+        /* Average time taken (milliseconds) inside the jitter buffer (which is post-receipt but pre-decode). */
+        this.averageJitterBufferDelayMs = undefined;
+        /* Average time taken (milliseconds) to decode a video frame. */
+        this.averageDecodeLatencyMs = undefined;
+        /* Average time taken (milliseconds) to between receipt of the first and last video packet of a. */
+        this.averageAssemblyDelayMs = undefined;
+        /* The sender latency + RTT/2 + processing delay */
+        this.averageE2ELatency = undefined;
+        /* Timing information about the worst performing frame since the last getStats call (only works on Chrome) */
+        this.frameTiming = undefined;
+    }
+}
+//# sourceMappingURL=LatencyCalculator.js.map

package/dist/esm/PeerConnectionController/LatencyCalculator.js.map

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package/dist/esm/PeerConnectionController/OutBoundRTPStats.js

@@ -1,12 +1,16 @@
 // Copyright Epic Games, Inc. All Rights Reserved.
 /**
- * Outbound Video Stats collected from the RTC Stats Report
+ * Outbound RTP stats collected from the RTC Stats Report under `outbound-rtp`.
+ * Wrapper around: https://developer.mozilla.org/en-US/docs/Web/API/RTCOutboundRtpStreamStats
+ * These are stats for video we are sending to a remote peer.
  */
-export class OutBoundVideoStats {
+export class OutboundRTPStats {
 }
 /**
- * Outbound Stats collected from the RTC Stats Report
+ * Remote outbound stats collected from the RTC Stats Report under `remote-outbound-rtp`.
+ * Wrapper around: https://developer.mozilla.org/en-US/docs/Web/API/RTCRemoteOutboundRtpStreamStats
+ * These are stats for media we are receiving from a remote peer.
  */
-export class OutBoundRTPStats {
+export class RemoteOutboundRTPStats {
 }
 //# sourceMappingURL=OutBoundRTPStats.js.map

package/dist/esm/PeerConnectionController/OutBoundRTPStats.js.map

@@ -1 +1 @@
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+{"version":3,"file":"OutBoundRTPStats.js","sourceRoot":"","sources":["../../../src/PeerConnectionController/OutBoundRTPStats.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD;;;;GAIG;AACH,MAAM,OAAO,gBAAgB;CA+B5B;AAED;;;;GAIG;AACH,MAAM,OAAO,sBAAsB;CAclC"}