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

package/dist/esm/__test__/mockWebSocket.js

@@ -0,0 +1,103 @@
+const spyFunctions = {
+    constructorSpy: null,
+    openSpy: null,
+    errorSpy: null,
+    closeSpy: null,
+    messageSpy: null,
+    messageBinarySpy: null,
+    sendSpy: null
+};
+const triggerFunctions = {
+    triggerOnOpen: null,
+    triggerOnError: null,
+    triggerOnClose: null,
+    triggerOnMessage: null,
+    triggerOnMessageBinary: null,
+    triggerRemoteClose: null
+};
+export class MockWebSocketImpl extends WebSocket {
+    constructor(url, protocols) {
+        var _a;
+        super(url, protocols);
+        this._readyState = this.OPEN;
+        (_a = spyFunctions.constructorSpy) === null || _a === void 0 ? void 0 : _a.call(spyFunctions, this.url);
+        triggerFunctions.triggerOnOpen = this.triggerOnOpen.bind(this);
+        triggerFunctions.triggerOnError = this.triggerOnError.bind(this);
+        triggerFunctions.triggerOnClose = this.triggerOnClose.bind(this);
+        triggerFunctions.triggerOnMessage = this.triggerOnMessage.bind(this);
+        triggerFunctions.triggerOnMessageBinary = this.triggerOnMessageBinary.bind(this);
+        triggerFunctions.triggerRemoteClose = this.triggerRemoteClose.bind(this);
+    }
+    get readyState() {
+        return this._readyState;
+    }
+    close(code, reason) {
+        super.close(code, reason);
+        this._readyState = this.CLOSED;
+        this.triggerOnClose({ code, reason });
+    }
+    send(data) {
+        var _a;
+        (_a = spyFunctions.sendSpy) === null || _a === void 0 ? void 0 : _a.call(spyFunctions, data);
+    }
+    triggerOnOpen() {
+        var _a, _b;
+        const event = new Event('open');
+        (_a = this.onopen) === null || _a === void 0 ? void 0 : _a.call(this, event);
+        (_b = spyFunctions.openSpy) === null || _b === void 0 ? void 0 : _b.call(spyFunctions, event);
+    }
+    triggerOnError() {
+        var _a, _b;
+        const event = new Event('error');
+        (_a = this.onerror) === null || _a === void 0 ? void 0 : _a.call(this, event);
+        (_b = spyFunctions.errorSpy) === null || _b === void 0 ? void 0 : _b.call(spyFunctions, event);
+    }
+    triggerOnClose(closeReason) {
+        var _a, _b;
+        const reason = closeReason !== null && closeReason !== void 0 ? closeReason : { code: 1, reason: 'mock reason' };
+        const event = new CloseEvent('close', reason);
+        (_a = this.onclose) === null || _a === void 0 ? void 0 : _a.call(this, event);
+        (_b = spyFunctions.closeSpy) === null || _b === void 0 ? void 0 : _b.call(spyFunctions, event);
+    }
+    triggerRemoteClose(code, reason) {
+        this.close(code, reason);
+    }
+    triggerOnMessage(message) {
+        var _a, _b;
+        const data = JSON.stringify(message);
+        const event = new MessageEvent('message', { data });
+        (_a = this.onmessage) === null || _a === void 0 ? void 0 : _a.call(this, event);
+        (_b = spyFunctions.messageSpy) === null || _b === void 0 ? void 0 : _b.call(spyFunctions, event);
+    }
+    triggerOnMessageBinary(message) {
+        var _a, _b;
+        const data = message !== null && message !== void 0 ? message : new Blob([JSON.stringify({ type: 'test' })], {
+            type: 'application/json'
+        });
+        const event = new MessageEvent('messagebinary', { data });
+        (_a = this.onmessagebinary) === null || _a === void 0 ? void 0 : _a.call(this, event);
+        (_b = spyFunctions.messageBinarySpy) === null || _b === void 0 ? void 0 : _b.call(spyFunctions, event);
+    }
+}
+const originalWebSocket = WebSocket;
+export const mockWebSocket = () => {
+    spyFunctions.constructorSpy = jest.fn();
+    spyFunctions.openSpy = jest.fn();
+    spyFunctions.errorSpy = jest.fn();
+    spyFunctions.closeSpy = jest.fn();
+    spyFunctions.messageSpy = jest.fn();
+    spyFunctions.messageBinarySpy = jest.fn();
+    spyFunctions.sendSpy = jest.fn();
+    global.WebSocket = MockWebSocketImpl;
+    return [spyFunctions, triggerFunctions];
+};
+export const unmockWebSocket = () => {
+    global.WebSocket = originalWebSocket;
+    spyFunctions.constructorSpy = null;
+    spyFunctions.openSpy = null;
+    spyFunctions.errorSpy = null;
+    spyFunctions.closeSpy = null;
+    spyFunctions.messageSpy = null;
+    spyFunctions.messageBinarySpy = null;
+};
+//# sourceMappingURL=mockWebSocket.js.map

package/dist/esm/__test__/mockWebSocket.js.map

@@ -0,0 +1 @@
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package/dist/esm/pixelstreamingfrontend.js

@@ -18,7 +18,8 @@ export { CandidatePairStats } from './PeerConnectionController/CandidatePairStat
 export { CandidateStat } from './PeerConnectionController/CandidateStat';
 export { DataChannelStats } from './PeerConnectionController/DataChannelStats';
 export { InboundAudioStats, InboundVideoStats } from './PeerConnectionController/InboundRTPStats';
-export { OutBoundVideoStats } from './PeerConnectionController/OutBoundRTPStats';
+export { OutboundRTPStats, RemoteOutboundRTPStats } from './PeerConnectionController/OutBoundRTPStats';
+export * from './PeerConnectionController/LatencyCalculator';
 export * from './DataChannel/DataChannelLatencyTestResults';
 export * from './Util/EventEmitter';
 export * from '@epicgames-ps/lib-pixelstreamingcommon-ue5.5';

package/dist/esm/pixelstreamingfrontend.js.map

@@ -1 +1 @@
-{"version":3,"file":"pixelstreamingfrontend.js","sourceRoot":"","sources":["../../src/pixelstreamingfrontend.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD,OAAO,EAAE,sBAAsB,EAAE,MAAM,uCAAuC,CAAC;AAC/E,OAAO,EAAE,eAAe,EAAE,MAAM,yBAAyB,CAAC;AAC1D,cAAc,iBAAiB,CAAC;AAChC,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,aAAa,EAAE,MAAM,wBAAwB,CAAC;AACvD,OAAO,EAAE,aAAa,EAAE,MAAM,wBAAwB,CAAC;AACvD,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,cAAc,EAA2B,MAAM,iCAAiC,CAAC;AAC1F,OAAO,EAAE,aAAa,IAAI,QAAQ,EAAE,MAAM,qBAAqB,CAAC;AAChE,OAAO,EAAE,kBAAkB,EAAE,MAAM,kCAAkC,CAAC;AACtE,cAAc,+BAA+B,CAAC;AAC9C,OAAO,EAAE,eAAe,EAAE,MAAM,4CAA4C,CAAC;AAC7E,cAAc,6BAA6B,CAAC;AAC5C,OAAO,EAAE,gBAAgB,EAAE,MAAM,6CAA6C,CAAC;AAC/E,OAAO,EAAE,kBAAkB,EAAE,MAAM,+CAA+C,CAAC;AACnF,OAAO,EAAE,aAAa,EAAE,MAAM,0CAA0C,CAAC;AACzE,OAAO,EAAE,gBAAgB,EAAE,MAAM,6CAA6C,CAAC;AAC/E,OAAO,EAAE,iBAAiB,EAAE,iBAAiB,EAAE,MAAM,4CAA4C,CAAC;AAClG,OAAO,EAAE,kBAAkB,EAAE,MAAM,6CAA6C,CAAC;AACjF,cAAc,6CAA6C,CAAC;AAC5D,cAAc,qBAAqB,CAAC;AACpC,cAAc,8CAA8C,CAAC"}
+{"version":3,"file":"pixelstreamingfrontend.js","sourceRoot":"","sources":["../../src/pixelstreamingfrontend.ts"],"names":[],"mappings":"AAAA,kDAAkD;AAElD,OAAO,EAAE,sBAAsB,EAAE,MAAM,uCAAuC,CAAC;AAC/E,OAAO,EAAE,eAAe,EAAE,MAAM,yBAAyB,CAAC;AAC1D,cAAc,iBAAiB,CAAC;AAChC,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,aAAa,EAAE,MAAM,wBAAwB,CAAC;AACvD,OAAO,EAAE,aAAa,EAAE,MAAM,wBAAwB,CAAC;AACvD,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAC;AACnD,OAAO,EAAE,cAAc,EAA2B,MAAM,iCAAiC,CAAC;AAC1F,OAAO,EAAE,aAAa,IAAI,QAAQ,EAAE,MAAM,qBAAqB,CAAC;AAChE,OAAO,EAAE,kBAAkB,EAAE,MAAM,kCAAkC,CAAC;AACtE,cAAc,+BAA+B,CAAC;AAC9C,OAAO,EAAE,eAAe,EAAE,MAAM,4CAA4C,CAAC;AAC7E,cAAc,6BAA6B,CAAC;AAC5C,OAAO,EAAE,gBAAgB,EAAE,MAAM,6CAA6C,CAAC;AAC/E,OAAO,EAAE,kBAAkB,EAAE,MAAM,+CAA+C,CAAC;AACnF,OAAO,EAAE,aAAa,EAAE,MAAM,0CAA0C,CAAC;AACzE,OAAO,EAAE,gBAAgB,EAAE,MAAM,6CAA6C,CAAC;AAC/E,OAAO,EAAE,iBAAiB,EAAE,iBAAiB,EAAE,MAAM,4CAA4C,CAAC;AAClG,OAAO,EAAE,gBAAgB,EAAE,sBAAsB,EAAE,MAAM,6CAA6C,CAAC;AACvG,cAAc,8CAA8C,CAAC;AAC7D,cAAc,6CAA6C,CAAC;AAC5D,cAAc,qBAAqB,CAAC;AACpC,cAAc,8CAA8C,CAAC"}

package/dist/types/Config/Config.d.ts

@@ -28,6 +28,7 @@ export declare class Flags {
     static XRControllerInput: "XRControllerInput";
     static WaitForStreamer: "WaitForStreamer";
     static HideUI: "HideUI";
+    static EnableCaptureTimeExt: "EnableCaptureTimeExt";
 }
 export type FlagsKeys = Exclude<keyof typeof Flags, 'prototype'>;
 export type FlagsIds = (typeof Flags)[FlagsKeys];
@@ -50,6 +51,7 @@ export declare class NumericParameters {
     static WebRTCMaxBitrate: "WebRTCMaxBitrate";
     static MaxReconnectAttempts: "MaxReconnectAttempts";
     static StreamerAutoJoinInterval: "StreamerAutoJoinInterval";
+    static KeepaliveDelay: "KeepaliveDelay";
 }
 export type NumericParametersKeys = Exclude<keyof typeof NumericParameters, 'prototype'>;
 export type NumericParametersIds = (typeof NumericParameters)[NumericParametersKeys];

package/dist/types/PeerConnectionController/AggregatedStats.d.ts

@@ -3,18 +3,24 @@ import { InboundTrackStats } from './InboundTrackStats';
 import { DataChannelStats } from './DataChannelStats';
 import { CandidateStat } from './CandidateStat';
 import { CandidatePairStats } from './CandidatePairStats';
-import { OutBoundRTPStats, OutBoundVideoStats } from './OutBoundRTPStats';
+import { RemoteOutboundRTPStats, OutboundRTPStats } from './OutBoundRTPStats';
 import { SessionStats } from './SessionStats';
 import { StreamStats } from './StreamStats';
 import { CodecStats } from './CodecStats';
+/**
+ * The Aggregated Stats that is generated from the RTC Stats Report
+ */
 export declare class AggregatedStats {
     inboundVideoStats: InboundVideoStats;
     inboundAudioStats: InboundAudioStats;
     candidatePairs: Array<CandidatePairStats>;
-    DataChannelStats: DataChannelStats;
+    datachannelStats: DataChannelStats;
     localCandidates: Array<CandidateStat>;
     remoteCandidates: Array<CandidateStat>;
-    outBoundVideoStats: OutBoundVideoStats;
+    outboundVideoStats: OutboundRTPStats;
+    outboundAudioStats: OutboundRTPStats;
+    remoteOutboundVideoStats: RemoteOutboundRTPStats;
+    remoteOutboundAudioStats: RemoteOutboundRTPStats;
     sessionStats: SessionStats;
     streamStats: StreamStats;
     codecs: Map<string, CodecStats>;
@@ -55,12 +61,17 @@ export declare class AggregatedStats {
      * Process the Inbound RTP Audio and Video Data
      * @param stat - inbound rtp stats
      */
-    handleInBoundRTP(stat: InboundRTPStats): void;
+    handleInboundRTP(stat: InboundRTPStats): void;
     /**
-     * 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: OutBoundRTPStats): void;
+    handleLocalOutbound(stat: OutboundRTPStats): void;
+    /**
+     * Process the "remote" outbound RTP Audio and Video stats.
+     * @param stat - remote outbound rtp stats
+     */
+    handleRemoteOutbound(stat: RemoteOutboundRTPStats): void;
     /**
      * Process the Inbound Video Track Data
      * @param stat - video track stats

package/dist/types/PeerConnectionController/InboundRTPStats.d.ts

@@ -2,140 +2,143 @@
  * Inbound Audio Stats collected from the RTC Stats Report
  */
 export declare class InboundAudioStats {
-    audioLevel: number;
+    audioLevel: number | undefined;
     bytesReceived: number;
     codecId: string;
-    concealedSamples: number;
-    concealmentEvents: number;
-    fecPacketsDiscarded: number;
-    fecPacketsReceived: number;
+    concealedSamples: number | undefined;
+    concealmentEvents: number | undefined;
+    fecPacketsDiscarded: number | undefined;
+    fecPacketsReceived: number | undefined;
     headerBytesReceived: number;
     id: string;
-    insertedSamplesForDeceleration: number;
+    insertedSamplesForDeceleration: number | undefined;
     jitter: number;
     jitterBufferDelay: number;
     jitterBufferEmittedCount: number;
-    jitterBufferMinimumDelay: number;
-    jitterBufferTargetDelay: number;
+    jitterBufferMinimumDelay: number | undefined;
+    jitterBufferTargetDelay: number | undefined;
     kind: string;
     lastPacketReceivedTimestamp: number;
-    mediaType: string;
+    mediaType: string | undefined;
     mid: string;
-    packetsDiscarded: number;
+    packetsDiscarded: number | undefined;
     packetsLost: number;
     packetsReceived: number;
-    removedSamplesForAcceleration: number;
-    silentConcealedSamples: number;
+    removedSamplesForAcceleration: number | undefined;
+    silentConcealedSamples: number | undefined;
     ssrc: number;
     timestamp: number;
-    totalAudioEnergy: number;
-    totalSamplesDuration: number;
-    totalSamplesReceived: number;
-    trackIdentifier: string;
-    transportId: string;
+    totalAudioEnergy: number | undefined;
+    totalSamplesDuration: number | undefined;
+    totalSamplesReceived: number | undefined;
+    trackIdentifier: string | undefined;
+    transportId: string | undefined;
     type: string;
-    bitrate: number;
+    bitrate: number | undefined;
 }
 /**
  * Inbound Video Stats collected from the RTC Stats Report
  */
 export declare class InboundVideoStats {
     bytesReceived: number;
-    codecId: string;
-    firCount: number;
-    frameHeight: number;
-    frameWidth: number;
-    framesAssembledFromMultiplePackets: number;
-    framesDecoded: number;
-    framesDropped: number;
-    framesPerSecond: number;
-    framesReceived: number;
-    freezeCount: number;
-    googTimingFrameInfo: string;
+    codecId: string | undefined;
+    firCount: number | undefined;
+    frameHeight: number | undefined;
+    frameWidth: number | undefined;
+    framesAssembledFromMultiplePackets: number | undefined;
+    framesDecoded: number | undefined;
+    framesDropped: number | undefined;
+    framesPerSecond: number | undefined;
+    framesReceived: number | undefined;
+    freezeCount: number | undefined;
+    googTimingFrameInfo: string | undefined;
     headerBytesReceived: number;
     id: string;
     jitter: number;
     jitterBufferDelay: number;
     jitterBufferEmittedCount: number;
-    keyFramesDecoded: number;
+    keyFramesDecoded: number | undefined;
     kind: string;
-    lastPacketReceivedTimestamp: number;
-    mediaType: string;
+    lastPacketReceivedTimestamp: number | undefined;
+    mediaType: string | undefined;
     mid: string;
-    nackCount: number;
+    nackCount: number | undefined;
     packetsLost: number;
     packetsReceived: number;
-    pauseCount: number;
-    pliCount: number;
+    pauseCount: number | undefined;
+    pliCount: number | undefined;
     ssrc: number;
     timestamp: number;
-    totalAssemblyTime: number;
-    totalDecodeTime: number;
-    totalFreezesDuration: number;
-    totalInterFrameDelay: number;
-    totalPausesDuration: number;
-    totalProcessingDelay: number;
-    totalSquaredInterFrameDelay: number;
-    trackIdentifier: string;
-    transportId: string;
+    totalAssemblyTime: number | undefined;
+    totalDecodeTime: number | undefined;
+    totalFreezesDuration: number | undefined;
+    totalInterFrameDelay: number | undefined;
+    totalPausesDuration: number | undefined;
+    totalProcessingDelay: number | undefined;
+    totalSquaredInterFrameDelay: number | undefined;
+    trackIdentifier: string | undefined;
+    transportId: string | undefined;
     type: string;
-    bitrate: number;
+    bitrate: number | undefined;
 }
 /**
  * Inbound Stats collected from the RTC Stats Report
  */
 export declare class InboundRTPStats {
     bytesReceived: number;
-    codecId: string;
+    codecId: string | undefined;
     headerBytesReceived: number;
     id: string;
     jitter: number;
     jitterBufferDelay: number;
     jitterBufferEmittedCount: number;
     kind: string;
-    lastPacketReceivedTimestamp: number;
-    mediaType: string;
+    lastPacketReceivedTimestamp: number | undefined;
+    mediaType: string | undefined;
     mid: string;
     packetsLost: number;
     packetsReceived: number;
+    playoutId: string | undefined;
+    qpsum: number | undefined;
+    remoteId: string | undefined;
     ssrc: number;
     timestamp: number;
-    trackIdentifier: string;
-    transportId: string;
+    trackIdentifier: string | undefined;
+    transportId: string | undefined;
     type: string;
-    audioLevel: number;
-    concealedSamples: number;
-    concealmentEvents: number;
-    fecPacketsDiscarded: number;
-    fecPacketsReceived: number;
-    insertedSamplesForDeceleration: number;
-    jitterBufferMinimumDelay: number;
-    jitterBufferTargetDelay: number;
-    packetsDiscarded: number;
-    removedSamplesForAcceleration: number;
-    silentConcealedSamples: number;
-    totalAudioEnergy: number;
-    totalSamplesDuration: number;
-    totalSamplesReceived: number;
-    firCount: number;
-    frameHeight: number;
-    frameWidth: number;
-    framesAssembledFromMultiplePackets: number;
-    framesDecoded: number;
-    framesDropped: number;
-    framesPerSecond: number;
-    framesReceived: number;
-    freezeCount: number;
-    googTimingFrameInfo: string;
-    keyFramesDecoded: number;
-    nackCount: number;
-    pauseCount: number;
-    pliCount: number;
-    totalAssemblyTime: number;
-    totalDecodeTime: number;
-    totalFreezesDuration: number;
-    totalInterFrameDelay: number;
-    totalPausesDuration: number;
-    totalProcessingDelay: number;
-    totalSquaredInterFrameDelay: number;
+    audioLevel: number | undefined;
+    concealedSamples: number | undefined;
+    concealmentEvents: number | undefined;
+    fecPacketsDiscarded: number | undefined;
+    fecPacketsReceived: number | undefined;
+    insertedSamplesForDeceleration: number | undefined;
+    jitterBufferMinimumDelay: number | undefined;
+    jitterBufferTargetDelay: number | undefined;
+    packetsDiscarded: number | undefined;
+    removedSamplesForAcceleration: number | undefined;
+    silentConcealedSamples: number | undefined;
+    totalAudioEnergy: number | undefined;
+    totalSamplesDuration: number | undefined;
+    totalSamplesReceived: number | undefined;
+    firCount: number | undefined;
+    frameHeight: number | undefined;
+    frameWidth: number | undefined;
+    framesAssembledFromMultiplePackets: number | undefined;
+    framesDecoded: number | undefined;
+    framesDropped: number | undefined;
+    framesPerSecond: number | undefined;
+    framesReceived: number | undefined;
+    freezeCount: number | undefined;
+    googTimingFrameInfo: string | undefined;
+    keyFramesDecoded: number | undefined;
+    nackCount: number | undefined;
+    pauseCount: number | undefined;
+    pliCount: number | undefined;
+    totalAssemblyTime: number | undefined;
+    totalDecodeTime: number | undefined;
+    totalFreezesDuration: number | undefined;
+    totalInterFrameDelay: number | undefined;
+    totalPausesDuration: number | undefined;
+    totalProcessingDelay: number | undefined;
+    totalSquaredInterFrameDelay: number | undefined;
 }

package/dist/types/PeerConnectionController/LatencyCalculator.d.ts

@@ -0,0 +1,87 @@
+import { AggregatedStats } from './AggregatedStats';
+/**
+ * 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 declare class FrameTimingInfo {
+    rtpTimestamp: number;
+    captureTimestamp: number;
+    encodeStartTimestamp: number;
+    encodeFinishTimestamp: number;
+    packetizerFinishTimestamp: number;
+    pacerExitTimestamp: number;
+    networkTimestamp1: number;
+    networkTimestamp2: number;
+    receiveStart: number;
+    receiveFinish: number;
+    decodeStart: number;
+    decodeFinish: number;
+    renderTime: number;
+    isOutlier: boolean;
+    isTriggeredByTimer: boolean;
+    encoderLatencyMs: number;
+    packetizeLatencyMs: number;
+    pacerLatencyMs: number;
+    captureToSendLatencyMs: number;
+}
+/**
+ * Calculates a combination of latency statistics using purely WebRTC API.
+ */
+export declare class LatencyCalculator {
+    private latestSenderRecvClockOffset;
+    calculate(stats: AggregatedStats, receivers: RTCRtpReceiver[]): LatencyInfo;
+    private extractFrameTimingInfo;
+    private calculateSenderLatency;
+    /**
+     * 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).
+     */
+    private getCaptureSource;
+    private calculateSenderReceiverClockOffset;
+    private getRTTMs;
+}
+/**
+ * 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 declare class LatencyInfo {
+    /**
+     * 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)
+     */
+    senderLatencyMs: number | 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)
+     */
+    senderLatencyAbsCaptureTimeMs: number | undefined;
+    rttMs: number | undefined;
+    averageProcessingDelayMs: number | undefined;
+    averageJitterBufferDelayMs: number | undefined;
+    averageDecodeLatencyMs: number | undefined;
+    averageAssemblyDelayMs: number | undefined;
+    averageE2ELatency: number | undefined;
+    frameTiming: FrameTimingInfo | undefined;
+}

package/dist/types/PeerConnectionController/OutBoundRTPStats.d.ts

@@ -1,23 +1,57 @@
 /**
- * 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 declare class OutBoundVideoStats {
+export declare class OutboundRTPStats {
+    active: boolean | undefined;
+    codecId: string | undefined;
     bytesSent: number;
+    frameHeight: number | undefined;
+    frameWidth: number | undefined;
+    framesEncoded: number | undefined;
+    framesPerSecond: number | undefined;
+    framesSent: number | undefined;
+    headerBytesSent: number;
     id: string;
-    localId: string;
+    keyFramesEncoded: number | undefined;
+    kind: string;
+    mediaSourceId: string | undefined;
+    mid: string | undefined;
+    nackCount: number | undefined;
     packetsSent: number;
-    remoteTimestamp: number;
+    qpSum: number | undefined;
+    qualityLimitationDurations: number | undefined;
+    qualityLimitationReason: string | undefined;
+    remoteId: string | undefined;
+    retransmittedBytesSent: number;
+    rid: string | undefined;
+    scalabilityMode: string | undefined;
+    ssrc: string;
+    targetBitrate: number | undefined;
     timestamp: number;
+    totalEncodeTime: number | undefined;
+    totalEncodeBytesTarget: number | undefined;
+    totalPacketSendDelay: number | undefined;
+    transportId: string | undefined;
 }
 /**
- * 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 declare class OutBoundRTPStats {
+export declare class RemoteOutboundRTPStats {
+    bytesSent: number | undefined;
+    codecId: string;
+    id: string | undefined;
     kind: string;
-    bytesSent: number;
-    id: string;
-    localId: string;
-    packetsSent: number;
-    remoteTimestamp: number;
-    timestamp: number;
+    localId: string | undefined;
+    packetsSent: number | undefined;
+    remoteTimestamp: number | undefined;
+    reportsSent: number | undefined;
+    roundTripTimeMeasurements: number | undefined;
+    ssrc: string;
+    timestamp: number | undefined;
+    totalRoundTripTime: number | undefined;
+    transportId: string | undefined;
 }

package/dist/types/PeerConnectionController/PeerConnectionController.d.ts

@@ -1,5 +1,7 @@
 import { Config } from '../Config/Config';
 import { AggregatedStats } from './AggregatedStats';
+import { LatencyCalculator, LatencyInfo } from './LatencyCalculator';
+export declare const kAbsCaptureTime = "http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time";
 /**
  * Handles the Peer Connection
  */
@@ -11,6 +13,7 @@ export declare class PeerConnectionController {
     updateCodecSelection: boolean;
     videoTrack: MediaStreamTrack;
     audioTrack: MediaStreamTrack;
+    latencyCalculator: LatencyCalculator;
     /**
      * Create a new RTC Peer Connection client
      * @param options - Peer connection Options
@@ -24,7 +27,7 @@ export declare class PeerConnectionController {
      */
     createOffer(offerOptions: RTCOfferOptions, config: Config): Promise<void>;
     /**
-     *
+     * Receive offer from UE side and process it as the remote description of this peer connection
      */
     receiveOffer(offer: RTCSessionDescriptionInit, config: Config): Promise<void>;
     /**
@@ -47,6 +50,7 @@ export declare class PeerConnectionController {
      * @returns A modified Session Descriptor
      */
     mungeSDP(sdp: string, useMic: boolean): string;
+    isFirefox(): boolean;
     /**
      * When a Ice Candidate is received add to the RTC Peer Connection
      * @param iceCandidate - RTC Ice Candidate from the Signaling Server
@@ -121,16 +125,26 @@ export declare class PeerConnectionController {
      * @param event - Aggregated Stats
      */
     onVideoStats(event: AggregatedStats): void;
+    /**
+     * And override event for when latency info is calculated
+     * @param latencyInfo - Calculated latency information.
+     */
+    onLatencyCalculated(latencyInfo: LatencyInfo): void;
     /**
      * Event to send the RTC offer to the Signaling server
      * @param offer - RTC Offer
      */
     onSendWebRTCOffer(offer: RTCSessionDescriptionInit): void;
     /**
-     * Event to send the RTC Answer to the Signaling server
+     * Event fired when remote offer description is set.
+     * @param offer - RTC Offer
+     */
+    onSetRemoteDescription(offer: RTCSessionDescriptionInit): void;
+    /**
+     * Event fire when local description answer is set.
      * @param answer - RTC Answer
      */
-    onSendWebRTCAnswer(answer: RTCSessionDescriptionInit): void;
+    onSetLocalDescription(answer: RTCSessionDescriptionInit): void;
     /**
      * An override for showing the Peer connection connecting Overlay
      */