@decartai/sdk 0.0.66 → 0.0.68

package/dist/realtime/webrtc-manager.js

@@ -21,6 +21,7 @@ var WebRTCManager = class {
 	connection;
 	config;
 	logger;
+	observability;
 	localStream = null;
 	subscribeMode = false;
 	managerState = "disconnected";
@@ -28,6 +29,7 @@ var WebRTCManager = class {
 	isReconnecting = false;
 	intentionalDisconnect = false;
 	reconnectGeneration = 0;
+	statsProviderConnection = null;
 	constructor(config) {
 		this.config = config;
 		this.logger = config.logger ?? {
@@ -36,6 +38,7 @@ var WebRTCManager = class {
 			warn() {},
 			error() {}
 		};
+		this.observability = config.observability;
 		this.connection = new WebRTCConnection({
 			onRemoteStream: config.onRemoteStream,
 			onStateChange: (state) => this.handleConnectionStateChange(state),
@@ -46,7 +49,7 @@ var WebRTCManager = class {
 			initialImage: config.initialImage,
 			initialPrompt: config.initialPrompt,
 			logger: this.logger,
-			onDiagnostic: config.onDiagnostic
+			observability: this.observability
 		});
 	}
 	emitState(state) {
@@ -56,15 +59,28 @@ var WebRTCManager = class {
 			this.config.onConnectionStateChange?.(state);
 		}
 	}
+	syncStatsProvider() {
+		const pc = this.getPeerConnection();
+		const isLive = this.managerState === "connected" || this.managerState === "generating";
+		if (isLive && pc && pc !== this.statsProviderConnection) {
+			this.statsProviderConnection = pc;
+			this.observability.setStatsProvider(pc);
+		} else if (!isLive && this.statsProviderConnection) {
+			this.statsProviderConnection = null;
+			this.observability.setStatsProvider(null);
+		}
+	}
 	handleConnectionStateChange(state) {
 		if (this.intentionalDisconnect) {
 			this.emitState("disconnected");
+			this.syncStatsProvider();
 			return;
 		}
 		if (this.isReconnecting) {
 			if (state === "connected" || state === "generating") {
 				this.isReconnecting = false;
 				this.emitState(state);
+				this.syncStatsProvider();
 			}
 			return;
 		}
@@ -73,6 +89,7 @@ var WebRTCManager = class {
 			return;
 		}
 		this.emitState(state);
+		this.syncStatsProvider();
 	}
 	async reconnect() {
 		if (this.isReconnecting || this.intentionalDisconnect) return;
@@ -80,6 +97,8 @@ var WebRTCManager = class {
 		const reconnectGeneration = ++this.reconnectGeneration;
 		this.isReconnecting = true;
 		this.emitState("reconnecting");
+		this.observability.setStatsProvider(null);
+		this.statsProviderConnection = null;
 		const reconnectStart = performance.now();
 		try {
 			let attemptCount = 0;
@@ -101,7 +120,7 @@ var WebRTCManager = class {
 						error: error.message,
 						attempt: error.attemptNumber
 					});
-					this.config.onDiagnostic?.("reconnect", {
+					this.observability.diagnostic("reconnect", {
 						attempt: error.attemptNumber,
 						maxAttempts: RETRY_OPTIONS.retries + 1,
 						durationMs: performance.now() - reconnectStart,
@@ -116,7 +135,7 @@ var WebRTCManager = class {
 					return !PERMANENT_ERRORS.some((err) => msg.includes(err));
 				}
 			});
-			this.config.onDiagnostic?.("reconnect", {
+			this.observability.diagnostic("reconnect", {
 				attempt: attemptCount,
 				maxAttempts: RETRY_OPTIONS.retries + 1,
 				durationMs: performance.now() - reconnectStart,
@@ -166,6 +185,8 @@ var WebRTCManager = class {
 		this.reconnectGeneration += 1;
 		this.connection.cleanup();
 		this.localStream = null;
+		this.statsProviderConnection = null;
+		this.observability.setStatsProvider(null);
 		this.emitState("disconnected");
 	}
 	isConnected() {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@decartai/sdk",
-  "version": "0.0.66",
+  "version": "0.0.68",
   "description": "Decart's JavaScript SDK",
   "type": "module",
   "license": "MIT",

package/dist/realtime/webrtc-stats.d.ts

@@ -1,59 +0,0 @@
-//#region src/realtime/webrtc-stats.d.ts
-type WebRTCStats = {
-  timestamp: number;
-  video: {
-    framesDecoded: number;
-    framesDropped: number;
-    framesPerSecond: number;
-    frameWidth: number;
-    frameHeight: number;
-    bytesReceived: number;
-    packetsReceived: number;
-    packetsLost: number;
-    jitter: number;
-    /** Estimated inbound bitrate in bits/sec, computed from bytesReceived delta. */
-    bitrate: number;
-    freezeCount: number;
-    totalFreezesDuration: number;
-    /** Delta: packets lost since previous sample. */
-    packetsLostDelta: number;
-    /** Delta: frames dropped since previous sample. */
-    framesDroppedDelta: number;
-    /** Delta: freeze count since previous sample. */
-    freezeCountDelta: number;
-    /** Delta: freeze duration (seconds) since previous sample. */
-    freezeDurationDelta: number;
-  } | null;
-  audio: {
-    bytesReceived: number;
-    packetsReceived: number;
-    packetsLost: number;
-    jitter: number;
-    /** Estimated inbound bitrate in bits/sec, computed from bytesReceived delta. */
-    bitrate: number;
-    /** Delta: packets lost since previous sample. */
-    packetsLostDelta: number;
-  } | null;
-  /** Outbound video track stats (from the local camera/screen share being sent). */
-  outboundVideo: {
-    /** Why the encoder is limiting quality: "none", "bandwidth", "cpu", or "other". */
-    qualityLimitationReason: string;
-    /** Cumulative time (seconds) spent in each quality limitation state. */
-    qualityLimitationDurations: Record<string, number>;
-    bytesSent: number;
-    packetsSent: number;
-    framesPerSecond: number;
-    frameWidth: number;
-    frameHeight: number;
-    /** Estimated outbound bitrate in bits/sec, computed from bytesSent delta. */
-    bitrate: number;
-  } | null;
-  connection: {
-    /** Current round-trip time in seconds, or null if unavailable. */
-    currentRoundTripTime: number | null;
-    /** Available outgoing bitrate estimate in bits/sec, or null if unavailable. */
-    availableOutgoingBitrate: number | null;
-  };
-};
-//#endregion
-export { WebRTCStats };

package/dist/realtime/webrtc-stats.js

@@ -1,154 +0,0 @@
-//#region src/realtime/webrtc-stats.ts
-const DEFAULT_INTERVAL_MS = 1e3;
-const MIN_INTERVAL_MS = 500;
-var WebRTCStatsCollector = class {
-	pc = null;
-	intervalId = null;
-	prevBytesVideo = 0;
-	prevBytesAudio = 0;
-	prevBytesSentVideo = 0;
-	prevTimestamp = 0;
-	prevPacketsLostVideo = 0;
-	prevFramesDropped = 0;
-	prevFreezeCount = 0;
-	prevFreezeDuration = 0;
-	prevPacketsLostAudio = 0;
-	onStats = null;
-	intervalMs;
-	constructor(options = {}) {
-		this.intervalMs = Math.max(options.intervalMs ?? DEFAULT_INTERVAL_MS, MIN_INTERVAL_MS);
-	}
-	/** Attach to a peer connection and start polling. */
-	start(pc, onStats) {
-		this.stop();
-		this.pc = pc;
-		this.onStats = onStats;
-		this.prevBytesVideo = 0;
-		this.prevBytesAudio = 0;
-		this.prevBytesSentVideo = 0;
-		this.prevTimestamp = 0;
-		this.prevPacketsLostVideo = 0;
-		this.prevFramesDropped = 0;
-		this.prevFreezeCount = 0;
-		this.prevFreezeDuration = 0;
-		this.prevPacketsLostAudio = 0;
-		this.intervalId = setInterval(() => this.collect(), this.intervalMs);
-	}
-	/** Stop polling and release resources. */
-	stop() {
-		if (this.intervalId !== null) {
-			clearInterval(this.intervalId);
-			this.intervalId = null;
-		}
-		this.pc = null;
-		this.onStats = null;
-	}
-	isRunning() {
-		return this.intervalId !== null;
-	}
-	async collect() {
-		if (!this.pc || !this.onStats) return;
-		try {
-			const rawStats = await this.pc.getStats();
-			const stats = this.parse(rawStats);
-			this.onStats(stats);
-		} catch {
-			this.stop();
-		}
-	}
-	parse(rawStats) {
-		const now = performance.now();
-		const elapsed = this.prevTimestamp > 0 ? (now - this.prevTimestamp) / 1e3 : 0;
-		let video = null;
-		let audio = null;
-		let outboundVideo = null;
-		const connection = {
-			currentRoundTripTime: null,
-			availableOutgoingBitrate: null
-		};
-		rawStats.forEach((report) => {
-			if (report.type === "inbound-rtp" && report.kind === "video") {
-				const bytesReceived = report.bytesReceived ?? 0;
-				const bitrate = elapsed > 0 ? (bytesReceived - this.prevBytesVideo) * 8 / elapsed : 0;
-				this.prevBytesVideo = bytesReceived;
-				const r = report;
-				const packetsLost = r.packetsLost ?? 0;
-				const framesDropped = r.framesDropped ?? 0;
-				const freezeCount = r.freezeCount ?? 0;
-				const freezeDuration = r.totalFreezesDuration ?? 0;
-				video = {
-					framesDecoded: r.framesDecoded ?? 0,
-					framesDropped,
-					framesPerSecond: r.framesPerSecond ?? 0,
-					frameWidth: r.frameWidth ?? 0,
-					frameHeight: r.frameHeight ?? 0,
-					bytesReceived,
-					packetsReceived: r.packetsReceived ?? 0,
-					packetsLost,
-					jitter: r.jitter ?? 0,
-					bitrate: Math.round(bitrate),
-					freezeCount,
-					totalFreezesDuration: freezeDuration,
-					packetsLostDelta: Math.max(0, packetsLost - this.prevPacketsLostVideo),
-					framesDroppedDelta: Math.max(0, framesDropped - this.prevFramesDropped),
-					freezeCountDelta: Math.max(0, freezeCount - this.prevFreezeCount),
-					freezeDurationDelta: Math.max(0, freezeDuration - this.prevFreezeDuration)
-				};
-				this.prevPacketsLostVideo = packetsLost;
-				this.prevFramesDropped = framesDropped;
-				this.prevFreezeCount = freezeCount;
-				this.prevFreezeDuration = freezeDuration;
-			}
-			if (report.type === "outbound-rtp" && report.kind === "video") {
-				const r = report;
-				const bytesSent = r.bytesSent ?? 0;
-				const outBitrate = elapsed > 0 ? (bytesSent - this.prevBytesSentVideo) * 8 / elapsed : 0;
-				this.prevBytesSentVideo = bytesSent;
-				outboundVideo = {
-					qualityLimitationReason: r.qualityLimitationReason ?? "none",
-					qualityLimitationDurations: r.qualityLimitationDurations ?? {},
-					bytesSent,
-					packetsSent: r.packetsSent ?? 0,
-					framesPerSecond: r.framesPerSecond ?? 0,
-					frameWidth: r.frameWidth ?? 0,
-					frameHeight: r.frameHeight ?? 0,
-					bitrate: Math.round(outBitrate)
-				};
-			}
-			if (report.type === "inbound-rtp" && report.kind === "audio") {
-				const bytesReceived = report.bytesReceived ?? 0;
-				const bitrate = elapsed > 0 ? (bytesReceived - this.prevBytesAudio) * 8 / elapsed : 0;
-				this.prevBytesAudio = bytesReceived;
-				const r = report;
-				const audioPacketsLost = r.packetsLost ?? 0;
-				audio = {
-					bytesReceived,
-					packetsReceived: r.packetsReceived ?? 0,
-					packetsLost: audioPacketsLost,
-					jitter: r.jitter ?? 0,
-					bitrate: Math.round(bitrate),
-					packetsLostDelta: Math.max(0, audioPacketsLost - this.prevPacketsLostAudio)
-				};
-				this.prevPacketsLostAudio = audioPacketsLost;
-			}
-			if (report.type === "candidate-pair") {
-				const r = report;
-				if (r.state === "succeeded") {
-					connection.currentRoundTripTime = r.currentRoundTripTime ?? null;
-					connection.availableOutgoingBitrate = r.availableOutgoingBitrate ?? null;
-				}
-			}
-		});
-		this.prevTimestamp = now;
-		return {
-			timestamp: Date.now(),
-			video,
-			audio,
-			outboundVideo,
-			connection
-		};
-	}
-};
-
-//#endregion
-export { WebRTCStatsCollector };