@vindral/web-sdk 4.6.0 → 4.7.0

package/README.md

@@ -98,7 +98,7 @@ Easy to customize controls with CSS variables.
 </style>
 ```
 
-[Read the full API reference for all the attributes and CSS variables](https://docs.vindral.com/web-sdk/player)
+[Read the full API reference for all the attributes and CSS variables](https://docs.vindral.com/web-sdk/api/player/interfaces/Player/)
 
 ### Core SDK
 

package/{BddIq5-p.js → Z7yt7Imj.js}

@@ -1,7 +1,7 @@
 import { n as e, t } from "./BsfwXDui.js";
 import { t as n } from "./DrWNJwZx.js";
-import { t as r } from "./BKeRxR4O.js";
-import { i, n as a, t as o } from "./CbyVl6wP.js";
+import { i as r, n as i, t as a } from "./BdMchGeH.js";
+import { t as o } from "./D1kpe8tT.js";
 //#region ../../libs/web-sdk/src/decoders/WebCodecsAudioDecoderContext.ts
 var s, c = globalThis, l = (e) => {
 	var t, n;
@@ -34,26 +34,26 @@ var s, c = globalThis, l = (e) => {
 					return;
 				}
 				this.doneCount++;
-				let { sample: n, decodeStartTime: i, priming: a } = t;
+				let { sample: n, decodeStartTime: r, priming: i } = t;
 				try {
-					let t = e.numberOfFrames, o = e.numberOfChannels, s = e.sampleRate, c = e.duration, l = Math.min(t, Math.round(a.remaining * s / 48e3));
-					if (a.remaining = Math.max(0, a.remaining - Math.round(l * 48e3 / s)), l === t && t > 0) return;
-					let u = new Float32Array((t - l) * o);
+					let t = e.numberOfFrames, a = e.numberOfChannels, s = e.sampleRate, c = e.duration, l = Math.min(t, Math.round(i.remaining * s / 48e3));
+					if (i.remaining = Math.max(0, i.remaining - Math.round(l * 48e3 / s)), l === t && t > 0) return;
+					let u = new Float32Array((t - l) * a);
 					e.copyTo(u, {
 						planeIndex: 0,
 						format: "f32",
 						frameOffset: l
 					});
-					let d = l > 0 ? Math.round((t - l) * n.timescale / s) : typeof c == "number" && Number.isFinite(c) ? Math.max(0, Math.round(c * n.timescale / 1e6)) : n.duration, f = r(r({}, n), {}, {
+					let d = l > 0 ? Math.round((t - l) * n.timescale / s) : typeof c == "number" && Number.isFinite(c) ? Math.max(0, Math.round(c * n.timescale / 1e6)) : n.duration, f = o(o({}, n), {}, {
 						timestamp: n.timestamp + Math.round(l * n.timescale / s),
 						duration: d,
 						type: "audio",
 						format: "f32",
 						data: u,
 						sampleRate: s,
-						channels: o,
+						channels: a,
 						statistics: {
-							decodeTime: Math.max(performance.now() - i, 0),
+							decodeTime: Math.max(performance.now() - r, 0),
 							transportTimeFromWorker: 0,
 							transportTimeToWorker: 0,
 							samplesInBatch: 1
@@ -67,11 +67,11 @@ var s, c = globalThis, l = (e) => {
 			error: (e) => {
 				this.logger.error("Decoder error:", e.name, e.message), this.resetDecoderState(), this.errors.push(e);
 			}
-		})), e(this, "buildCodecString", o), e(this, "getDescription", (e, t) => {
+		})), e(this, "buildCodecString", a), e(this, "getDescription", (e, t) => {
 			let n = this.initSegments.get(t.renditionId);
 			if (n && n.data.byteLength > 0 && n.codec === e) return n.data;
-			if (e === "opus") return a(t.channels, t.sampleRate);
-			if (e === "aac") return i(t.channels, t.sampleRate);
+			if (e === "opus") return i(t.channels, t.sampleRate);
+			if (e === "aac") return r(t.channels, t.sampleRate);
 		}), e(this, "reset", () => {
 			this.decodedSamples = [], this.decoder && this.decoder.state !== "closed" && this.decoder.reset(), this.resetDecoderState();
 		}), e(this, "unload", () => {

package/api-client.js

@@ -1,2 +1,2 @@
-import { t as e } from "./BOdyvABa.js";
+import { t as e } from "./D40eQS8l.js";
 export { e as ApiClient };

package/cast-sender.d.ts

@@ -170,7 +170,12 @@ interface Options {
 	 * Enable bursting for initial connection and channel switches. This makes time to first frame faster at the
 	 * cost of stability (more demanding due to the sudden burst of live content)
 	 *
-	 * Is disabled by default.
+	 * With bursting and adaptive bit rate on, playback starts at a rendition whose bit rate is at most half the
+	 * estimated bandwidth, so the link can catch up with the burst, and adaptive bit rate then steps up. It never
+	 * starts below the lowest rendition. If the rendition it starts at still needs more than half the estimated
+	 * bandwidth (the lowest one on a slow link, or the fixed one with adaptive bit rate off), the burst is skipped.
+	 *
+	 * Is enabled by default.
 	 *
 	 */
 	burstEnabled?: boolean;

package/cast-sender.js

@@ -1,2 +1,2 @@
-import { t as e } from "./xYchr7Lz.js";
+import { t as e } from "./DQ5kuB8C.js";
 export { e as CastSender };

package/core.d.ts

@@ -35,6 +35,15 @@ interface MinMaxAverage {
 	 */
 	min: number;
 }
+type NavigatorEffectiveType = "2g" | "3g" | "4g" | "slow-2g";
+type NavigatorConnectionType = "bluetooth" | "cellular" | "ethernet" | "mixed" | "none" | "other" | "unknown" | "wifi" | "wimax";
+interface NavigatorConnectionStatistics {
+	navigatorRtt?: number;
+	navigatorEffectiveType?: NavigatorEffectiveType;
+	navigatorConnectionType?: NavigatorConnectionType;
+	navigatorSaveData?: boolean;
+	navigatorDownlink?: number;
+}
 type MatchingKeys<TRecord, TMatch, K extends keyof TRecord = keyof TRecord> = K extends (TRecord[K] extends TMatch ? K : never) ? K : never;
 type VoidKeys<Record> = MatchingKeys<Record, void>;
 type EventListenerReturnType = (() => void) | void;
@@ -266,7 +275,12 @@ export interface Options {
 	 * Enable bursting for initial connection and channel switches. This makes time to first frame faster at the
 	 * cost of stability (more demanding due to the sudden burst of live content)
 	 *
-	 * Is disabled by default.
+	 * With bursting and adaptive bit rate on, playback starts at a rendition whose bit rate is at most half the
+	 * estimated bandwidth, so the link can catch up with the burst, and adaptive bit rate then steps up. It never
+	 * starts below the lowest rendition. If the rendition it starts at still needs more than half the estimated
+	 * bandwidth (the lowest one on a slow link, or the fixed one with adaptive bit rate off), the burst is skipped.
+	 *
+	 * Is enabled by default.
 	 *
 	 */
 	burstEnabled?: boolean;
@@ -901,18 +915,13 @@ interface DecoderStatistics {
 	audioRenderer?: "AudioContext" | "MSE";
 	audioDecoder?: "WebCodecs" | "WASM" | "MSE";
 }
-interface DocumentStateModulesStatistics {
+interface DocumentStateModulesStatistics extends NavigatorConnectionStatistics {
 	isVisible: boolean;
 	isOnline: boolean;
 	isVisibleCount: number;
 	isHiddenCount: number;
 	isOnlineCount: number;
 	isOfflineCount: number;
-	navigatorRtt?: number;
-	navigatorEffectiveType?: EffectiveConnectionType;
-	navigatorConnectionType?: ConnectionType;
-	navigatorSaveData?: boolean;
-	navigatorDownlink?: number;
 }
 interface IncomingDataModuleStatistics {
 	/**
@@ -1371,6 +1380,11 @@ export declare class Vindral extends Emitter<PublicVindralEvents> {
 	 * setting targetBufferTime within that range for consistancy if using both min/max.
 	 */
 	set targetBufferTime(bufferTimeMs: number);
+	/**
+	 * Set the burst for joins and channel switches from `burstEnabled` as resolved for the channel, so a channel
+	 * override can turn it off as well as on
+	 */
+	private applyBurstEnabled;
 	/**
 	 * Update the dynamic buffer time configuration at runtime.
 	 * Allows adjusting min and max buffer time without recreating the Vindral instance.
@@ -1544,6 +1558,8 @@ export declare class Vindral extends Emitter<PublicVindralEvents> {
 	private updateTextTracks;
 	private cleanupTextTracks;
 	private filterRenditions;
+	/** Whether the bandwidth leaves room for a join burst of the video and audio the player starts at. */
+	private burstFits;
 	/**
 	 * Patch the subscription with properties from the channel that isn't known until connection
 	 * @param channel Channel with the renditions to patch the subscription based on
@@ -1706,6 +1722,10 @@ interface ConnectionStatistics {
 	 * A very rough initial estimation of minimum available bandwidth.
 	 */
 	estimatedBandwidth: number;
+	/**
+	 * Origin of the edge the client is connected to, or is trying while it reconnects. Once disconnected, the last
+	 * edge it was connected to.
+	 */
 	edgeUrl?: string;
 	/**
 	 * Total number of connections that have been established since instantiation.
@@ -1726,6 +1746,7 @@ interface ConnectionStatistics {
 	 * Does not wait for player bootstrap or media to start.
 	 */
 	timeToConnected?: number;
+	/** Protocol of the current connection. Unset while reconnecting; once disconnected, the last connection's. */
 	connectionProtocol: "vindral_ws" | "moq" | "moq_ws" | undefined;
 	/**
 	 * When connectionProtocol is "moq_ws", indicates why WebSocket was used instead of WebTransport.
@@ -1733,6 +1754,7 @@ interface ConnectionStatistics {
 	 * - "disabled": webtransportEnabled option is false
 	 * - "unavailable": edge did not offer a WebTransport endpoint
 	 * - "fallback": WebTransport was attempted but failed or timed out
+	 * Unset while reconnecting; once disconnected, the last connection's.
 	 */
 	transportSelectionReason: TransportSelectionReason | undefined;
 }

package/core.js

@@ -1,3 +1,2 @@
-import { t as e } from "./CN8w7_en.js";
-import { a as t, c as n, d as r, f as i, i as a, l as o, n as s, o as c, r as l, s as u } from "./BBVSf5AB.js";
-export { s as AUTHENTICATION_EXPIRED_CODE, l as AUTHENTICATION_FAILED_CODE, a as CHANNEL_NOT_FOUND_CODE, t as CHANNEL_NOT_LIVE_CODE, c as CONNECTION_FAILED_AFTER_RETRIES_CODE, u as CONNECTION_FAILED_CODE, n as DISCONNECTED_BY_EDGE, o as INACTIVITY_CODE, r as NO_INCOMING_DATA, e as Vindral, i as VindralError };
+import { a as e, c as t, d as n, f as r, i, l as a, o, r as s, s as c, t as l, u } from "./5u5UZABw.js";
+export { s as AUTHENTICATION_EXPIRED_CODE, i as AUTHENTICATION_FAILED_CODE, e as CHANNEL_NOT_FOUND_CODE, o as CHANNEL_NOT_LIVE_CODE, c as CONNECTION_FAILED_AFTER_RETRIES_CODE, t as CONNECTION_FAILED_CODE, a as DISCONNECTED_BY_EDGE, u as INACTIVITY_CODE, n as NO_INCOMING_DATA, l as Vindral, r as VindralError };

package/legacy.d.ts

@@ -38,6 +38,15 @@ interface MinMaxAverage {
 	 */
 	min: number;
 }
+type NavigatorEffectiveType = "2g" | "3g" | "4g" | "slow-2g";
+type NavigatorConnectionType = "bluetooth" | "cellular" | "ethernet" | "mixed" | "none" | "other" | "unknown" | "wifi" | "wimax";
+interface NavigatorConnectionStatistics {
+	navigatorRtt?: number;
+	navigatorEffectiveType?: NavigatorEffectiveType;
+	navigatorConnectionType?: NavigatorConnectionType;
+	navigatorSaveData?: boolean;
+	navigatorDownlink?: number;
+}
 type MatchingKeys<TRecord, TMatch, K extends keyof TRecord = keyof TRecord> = K extends (TRecord[K] extends TMatch ? K : never) ? K : never;
 type VoidKeys<Record> = MatchingKeys<Record, void>;
 type EventListenerReturnType = (() => void) | void;
@@ -269,7 +278,12 @@ export interface Options {
 	 * Enable bursting for initial connection and channel switches. This makes time to first frame faster at the
 	 * cost of stability (more demanding due to the sudden burst of live content)
 	 *
-	 * Is disabled by default.
+	 * With bursting and adaptive bit rate on, playback starts at a rendition whose bit rate is at most half the
+	 * estimated bandwidth, so the link can catch up with the burst, and adaptive bit rate then steps up. It never
+	 * starts below the lowest rendition. If the rendition it starts at still needs more than half the estimated
+	 * bandwidth (the lowest one on a slow link, or the fixed one with adaptive bit rate off), the burst is skipped.
+	 *
+	 * Is enabled by default.
 	 *
 	 */
 	burstEnabled?: boolean;
@@ -916,18 +930,13 @@ interface DecoderStatistics {
 	audioRenderer?: "AudioContext" | "MSE";
 	audioDecoder?: "WebCodecs" | "WASM" | "MSE";
 }
-interface DocumentStateModulesStatistics {
+interface DocumentStateModulesStatistics extends NavigatorConnectionStatistics {
 	isVisible: boolean;
 	isOnline: boolean;
 	isVisibleCount: number;
 	isHiddenCount: number;
 	isOnlineCount: number;
 	isOfflineCount: number;
-	navigatorRtt?: number;
-	navigatorEffectiveType?: EffectiveConnectionType;
-	navigatorConnectionType?: ConnectionType;
-	navigatorSaveData?: boolean;
-	navigatorDownlink?: number;
 }
 interface IncomingDataModuleStatistics {
 	/**
@@ -1386,6 +1395,11 @@ export declare class Vindral extends Emitter<PublicVindralEvents> {
 	 * setting targetBufferTime within that range for consistancy if using both min/max.
 	 */
 	set targetBufferTime(bufferTimeMs: number);
+	/**
+	 * Set the burst for joins and channel switches from `burstEnabled` as resolved for the channel, so a channel
+	 * override can turn it off as well as on
+	 */
+	private applyBurstEnabled;
 	/**
 	 * Update the dynamic buffer time configuration at runtime.
 	 * Allows adjusting min and max buffer time without recreating the Vindral instance.
@@ -1559,6 +1573,8 @@ export declare class Vindral extends Emitter<PublicVindralEvents> {
 	private updateTextTracks;
 	private cleanupTextTracks;
 	private filterRenditions;
+	/** Whether the bandwidth leaves room for a join burst of the video and audio the player starts at. */
+	private burstFits;
 	/**
 	 * Patch the subscription with properties from the channel that isn't known until connection
 	 * @param channel Channel with the renditions to patch the subscription based on
@@ -1721,6 +1737,10 @@ interface ConnectionStatistics {
 	 * A very rough initial estimation of minimum available bandwidth.
 	 */
 	estimatedBandwidth: number;
+	/**
+	 * Origin of the edge the client is connected to, or is trying while it reconnects. Once disconnected, the last
+	 * edge it was connected to.
+	 */
 	edgeUrl?: string;
 	/**
 	 * Total number of connections that have been established since instantiation.
@@ -1741,6 +1761,7 @@ interface ConnectionStatistics {
 	 * Does not wait for player bootstrap or media to start.
 	 */
 	timeToConnected?: number;
+	/** Protocol of the current connection. Unset while reconnecting; once disconnected, the last connection's. */
 	connectionProtocol: "vindral_ws" | "moq" | "moq_ws" | undefined;
 	/**
 	 * When connectionProtocol is "moq_ws", indicates why WebSocket was used instead of WebTransport.
@@ -1748,6 +1769,7 @@ interface ConnectionStatistics {
 	 * - "disabled": webtransportEnabled option is false
 	 * - "unavailable": edge did not offer a WebTransport endpoint
 	 * - "fallback": WebTransport was attempted but failed or timed out
+	 * Unset while reconnecting; once disconnected, the last connection's.
 	 */
 	transportSelectionReason: TransportSelectionReason | undefined;
 }