@motionscript/browser 0.1.0-alpha.3 → 0.1.0-alpha.6

package/dist/audio/loop-region.js

@@ -0,0 +1,43 @@
+/**
+ * The region a looping request repeats: from its in-point to its out-point, or to
+ * the end of the buffer when it has none. `null` for a request that does not loop,
+ * and for a degenerate region (an out-point at or before the in-point), where the
+ * device falls back to wrapping the whole buffer.
+ *
+ * Shared by the live device and the offline mixer so a preview and an export loop
+ * the same samples.
+ */
+export function loopRegionOf(request, bufferDuration) {
+    if (!request.loop)
+        return null;
+    const start = Math.min(Math.max(0, request.trimStart), bufferDuration);
+    const end = Math.min(request.trimEnd ?? bufferDuration, bufferDuration);
+    return end > start ? { start, end } : null;
+}
+/**
+ * Configure `source` to play `request`'s loop: on or off, and over the trimmed
+ * region rather than the whole buffer (`loopStart`/`loopEnd` default to the
+ * whole buffer, which is how a trim used to be lost).
+ */
+export function applyLoop(source, request) {
+    source.loop = request.loop;
+    const region = source.buffer ? loopRegionOf(request, source.buffer.duration) : null;
+    if (region) {
+        source.loopStart = region.start;
+        source.loopEnd = region.end;
+    }
+    return region;
+}
+/**
+ * Fold a source offset into a loop region.
+ *
+ * A seek into the middle of a looping clip lands at `trimStart + elapsed`, which
+ * can be many passes past the region's end; the device would not wrap an offset
+ * that starts outside the region, so it is wrapped here.
+ */
+export function wrapIntoRegion(region, offset) {
+    if (!region || offset < region.end)
+        return offset;
+    return region.start + ((offset - region.start) % (region.end - region.start));
+}
+//# sourceMappingURL=loop-region.js.map

package/dist/audio/loop-region.js.map

@@ -0,0 +1 @@
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package/dist/audio/mixer.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"mixer.d.ts","sourceRoot":"","sources":["../../src/audio/mixer.ts"],"names":[],"mappings":"AAAA,OAAO,EAA2B,KAAK,YAAY,EAAE,MAAM,oBAAoB,CAAC;AAChF,OAAO,KAAK,EAAE,UAAU,EAAE,qBAAqB,EAAE,MAAM,kCAAkC,CAAC;AAW1F,yDAAyD;AACzD,eAAO,MAAM,mBAAmB,QAAQ,CAAC;AAEzC,MAAM,WAAW,eAAe;IAC5B,+CAA+C;IAC/C,UAAU,CAAC,EAAE,MAAM,CAAC;IACpB;;;;OAIG;IACH,OAAO,CAAC,EAAE,YAAY,CAAC;CAC1B;AA4BD;;;;;;;;;;;;;;;;;;GAkBG;AACH,wBAAsB,QAAQ,CAC1B,QAAQ,EAAE,SAAS,qBAAqB,EAAE,EAC1C,aAAa,EAAE,MAAM,EACrB,OAAO,GAAE,eAAoB,GAC9B,OAAO,CAAC,WAAW,GAAG,IAAI,CAAC,CA8F7B;AAED;;;;;;;;GAQG;AACH,qBAAa,aAAc,YAAW,UAAU,CAAC,WAAW,CAAC;IAErD,OAAO,CAAC,QAAQ,CAAC,UAAU;IAC3B,OAAO,CAAC,QAAQ,CAAC,OAAO,CAAC;gBADR,UAAU,GAAE,MAA4B,EACxC,OAAO,CAAC,EAAE,YAAY,YAAA;IAG3C,GAAG,CACC,QAAQ,EAAE,SAAS,qBAAqB,EAAE,EAC1C,aAAa,EAAE,MAAM,GACtB,OAAO,CAAC,WAAW,GAAG,IAAI,CAAC;CAMjC;AAED;;;;;;;GAOG;AACH,wBAAgB,SAAS,CAAC,MAAM,EAAE,WAAW,GAAG,UAAU,CAAC,WAAW,CAAC,CA6CtE"}

package/dist/audio/mixer.js

@@ -1,6 +1,7 @@
-import { sourceTimeAtSceneElapsed } from "@motionscript/core";
-import { fetchAssetBytes } from "../asset-bytes";
+import { sourceTimeAtClipElapsed } from "@motionscript/core";
+import { fetchAssetBytes, localAssetBytes } from "../asset-bytes";
 import { AudioBusGraph } from "./bus-graph";
+import { applyLoop } from "./loop-region";
 import { applySpeedToPlaybackRate, buildAudioFilterGraph, effectiveSpeed, speedCurveOf, } from "./filter-graph";
 /** Sample rate used when the caller doesn't pick one. */
 export const DEFAULT_SAMPLE_RATE = 44100;
@@ -20,10 +21,16 @@ function audioUrl(src, catalog) {
         }
     }
 }
-/** Fetches and decodes an audio source against a scratch `OfflineAudioContext` (decoding requires a context but doesn't render through it). */
+/**
+ * Reads and decodes an audio source against a scratch `OfflineAudioContext`
+ * (decoding requires a context but doesn't render through it). The host's byte
+ * source is asked first, as for every other asset read: a mix that went to the
+ * network would fail where the preview played, offline or for bytes not yet
+ * uploaded.
+ */
 async function fetchAudioBuffer(src, ctx) {
-    const arrayBuffer = await (await fetchAssetBytes(src)).arrayBuffer();
-    return ctx.decodeAudioData(arrayBuffer);
+    const blob = (await localAssetBytes(src)) ?? (await fetchAssetBytes(src));
+    return ctx.decodeAudioData(await blob.arrayBuffer());
 }
 /**
  * Render every scheduled request into one timeline-length buffer.
@@ -53,9 +60,18 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         return null;
     const uniqueSrcs = [...new Set(requests.map(r => r.request.src))];
     const scratchCtx = new OfflineAudioContext(2, frames, sampleRate);
+    // A source only a picture's own sound wants may hold no audio at all, which
+    // is silence rather than a failed mix — see `AudioRequest.optional`.
+    const asked = new Set(requests.filter(r => !r.request.optional).map(r => r.request.src));
     const decoded = new Map();
     await Promise.all(uniqueSrcs.map(async (src) => {
-        decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));
+        try {
+            decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));
+        }
+        catch (cause) {
+            if (asked.has(src))
+                throw cause;
+        }
     }));
     const mixCtx = new OfflineAudioContext(2, frames, sampleRate);
     // The same shared-chain rule the live device uses, for the same reason: an
@@ -74,7 +90,7 @@ export async function mixAudio(requests, totalDuration, options = {}) {
             continue;
         const source = mixCtx.createBufferSource();
         source.buffer = srcBuffer;
-        source.loop = req.loop;
+        applyLoop(source, req);
         const scalarSpeed = effectiveSpeed(req.filters);
         const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
         const speedCurve = speedCurveOf(req.filters);
@@ -86,7 +102,7 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         gain.gain.value = Number.isFinite(req.volume) ? req.volume : 1;
         // The trimmed source length the filter and speed curves are authored over.
         const sourceLength = Math.max(0, srcBuffer.duration - req.trimStart);
-        // The clip's own scene-time length — what a curve resolves against.
+        // The clip's own clip-time length — what a curve resolves against.
         // Deliberately *not* `playDuration`, which is that length truncated by the
         // timeline end: resolving against the truncated value would squeeze the
         // whole curve into the surviving part instead of cutting it off.
@@ -110,9 +126,9 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         }
         // start()'s 3rd arg is a source-buffer duration. A constant rate scales it
         // linearly, but under a speed curve the buffer consumed over a span of
-        // scene time is the integral of the rate, not a multiple of it.
+        // clip time is the integral of the rate, not a multiple of it.
         const consumed = speedCurve
-            ? sourceTimeAtSceneElapsed(speedCurve, sourceLength, playDuration)
+            ? sourceTimeAtClipElapsed(speedCurve, sourceLength, playDuration)
             : playDuration * rate;
         source.start(globalStart, req.trimStart, consumed);
         for (const osc of graph.oscillators)

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{ sourceTimeAtSceneElapsed, type AssetCatalog } from \"@motionscript/core\";\nimport type { AudioMixer, ScheduledAudioRequest } from \"@motionscript/skia-render/export\";\nimport { fetchAssetBytes } from \"../asset-bytes\";\nimport { AudioBusGraph } from \"./bus-graph\";\nimport {\n    applySpeedToPlaybackRate,\n    buildAudioFilterGraph,\n    effectiveSpeed,\n    speedCurveOf,\n} from \"./filter-graph\";\n\n/** Sample rate used when the caller doesn't pick one. */\nexport const DEFAULT_SAMPLE_RATE = 44100;\n\nexport interface MixAudioOptions {\n    /** Defaults to {@link DEFAULT_SAMPLE_RATE}. */\n    sampleRate?: number;\n    /**\n     * Resolves a request's `src` to the address its bytes live at, the same job\n     * `WebStorageAdapter.assetUrl` does for the live paths. Omitted, a `src` is\n     * fetched as written — right only while the host keys its manifest by URL.\n     */\n    catalog?: AssetCatalog;\n}\n\n/** Audio first, then video: a clip's own soundtrack is requested under the video's key. */\nfunction audioUrl(src: string, catalog: AssetCatalog | undefined): string {\n    if (!catalog) return src;\n    try {\n        return catalog.getAudioMeta(src).src || src;\n    } catch {\n        try {\n            return catalog.getVideoMeta(src).src || src;\n        } catch {\n            return src;\n        }\n    }\n}\n\n/** Fetches and decodes an audio source against a scratch `OfflineAudioContext` (decoding requires a context but doesn't render through it). */\nasync function fetchAudioBuffer(src: string, ctx: OfflineAudioContext): Promise<AudioBuffer> {\n    const arrayBuffer = await (await fetchAssetBytes(src)).arrayBuffer();\n    return ctx.decodeAudioData(arrayBuffer);\n}\n\n/**\n * Render every scheduled request into one timeline-length buffer.\n *\n * An `OfflineAudioContext` graph — gain per request, looping/trimming, and the\n * same filter chain the live {@link WebAudioSink} builds — rendered in one\n * pass rather than scheduled per frame.\n *\n * Shared by the video exporter and by `mixTimelineAudio`, deliberately: a split\n * export and a whole one have to be the same mix, and two implementations of\n * \"stack these clips\" would eventually disagree.\n *\n * Curve-valued filter params and `speed` curves schedule here exactly as they do\n * in {@link WebAudioSink.playBuffer}, which is the reference implementation —\n * the two must be read side by side when either changes.\n *\n * One deliberate difference: this schedules at absolute context times, so clip\n * starts are sample-accurate here and quantized to `1/fps` in live playback,\n * which starts sources at `currentTime` on a clock tick.\n */\nexport async function mixAudio(\n    requests: readonly ScheduledAudioRequest[],\n    totalDuration: number,\n    options: MixAudioOptions = {},\n): Promise<AudioBuffer | null> {\n    if (requests.length === 0) return null;\n    const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE;\n    const frames = Math.ceil(totalDuration * sampleRate);\n    if (frames <= 0) return null;\n\n    const uniqueSrcs = [...new Set(requests.map(r => r.request.src))];\n    const scratchCtx = new OfflineAudioContext(2, frames, sampleRate);\n\n    const decoded = new Map<string, AudioBuffer>();\n    await Promise.all(uniqueSrcs.map(async (src) => {\n        decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));\n    }));\n\n    const mixCtx = new OfflineAudioContext(2, frames, sampleRate);\n    // The same shared-chain rule the live device uses, for the same reason: an\n    // export that flattened buses into per-clip processing would not match what\n    // the preview sounded like.\n    const buses = new AudioBusGraph(mixCtx, mixCtx.destination);\n\n    for (const { request: req, globalOffset } of requests) {\n        const srcBuffer = decoded.get(req.src);\n        if (!srcBuffer) continue;\n\n        const globalStart = globalOffset + req.startAt;\n        // `endAt` is `Infinity` for an open clip; either way the timeline bounds it.\n        const globalEnd = Math.min(globalOffset + req.endAt, totalDuration);\n        const playDuration = globalEnd - globalStart;\n        if (playDuration <= 0) continue;\n\n        const source = mixCtx.createBufferSource();\n        source.buffer = srcBuffer;\n        source.loop = req.loop;\n\n        const scalarSpeed = effectiveSpeed(req.filters);\n        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;\n        const speedCurve = speedCurveOf(req.filters);\n        // With a curve the scalar is folded into the scheduled ramp below, so\n        // setting it here as well would apply it twice.\n        if (rate !== 1 && !speedCurve) source.playbackRate.value = rate;\n\n        const gain = mixCtx.createGain();\n        gain.gain.value = Number.isFinite(req.volume) ? req.volume : 1;\n\n        // The trimmed source length the filter and speed curves are authored over.\n        const sourceLength = Math.max(0, srcBuffer.duration - req.trimStart);\n        // The clip's own scene-time length — what a curve resolves against.\n        // Deliberately *not* `playDuration`, which is that length truncated by the\n        // timeline end: resolving against the truncated value would squeeze the\n        // whole curve into the surviving part instead of cutting it off.\n        const clipDuration = Number.isFinite(req.endAt)\n            ? Math.max(0, req.endAt - req.startAt)\n            : sourceLength / rate;\n\n        // Filter chain sits between the source and the per-request gain, matching\n        // the live WebAudioSink graph so exports sound identical to preview.\n        // Passing the start time and clip duration is what lets curve-valued params\n        // schedule — with the defaults they collapse to a single static value.\n        // `elapsed` is 0: unlike a live seek, the mix always starts a clip at its\n        // own beginning.\n        const graph = buildAudioFilterGraph(\n            mixCtx, source, req.filters ?? [], globalStart, clipDuration, 0,\n        );\n        graph.output.connect(gain);\n        // From the mix's start, whichever clip asks first: a bus's chain is shared,\n        // so its schedule cannot begin at one clip's start. Context time 0 is\n        // `-globalOffset` in this request's seconds, which are its stops' too.\n        gain.connect(buses.inputFor(req.bus, 0, -globalOffset));\n\n        if (speedCurve) {\n            applySpeedToPlaybackRate(\n                source.playbackRate, speedCurve, sourceLength, clipDuration, rate, globalStart, 0,\n            );\n        }\n\n        // start()'s 3rd arg is a source-buffer duration. A constant rate scales it\n        // linearly, but under a speed curve the buffer consumed over a span of\n        // scene time is the integral of the rate, not a multiple of it.\n        const consumed = speedCurve\n            ? sourceTimeAtSceneElapsed(speedCurve, sourceLength, playDuration)\n            : playDuration * rate;\n\n        source.start(globalStart, req.trimStart, consumed);\n        for (const osc of graph.oscillators) osc.start(globalStart);\n    }\n\n    return mixCtx.startRendering();\n}\n\n/**\n * Adapter presenting {@link mixAudio} as the exporter's injected mixer seam.\n *\n * The sample rate is a constructor argument rather than a third `mix` parameter\n * because {@link AudioMixer} — the seam `renderTimeline` calls through — takes\n * exactly two, so anything passed as a third would be silently dropped. Fixing\n * it here rather than widening the interface keeps the rate a property of *this*\n * mixer, which is whose business it is.\n */\nexport class WebAudioMixer implements AudioMixer<AudioBuffer> {\n    constructor(\n        private readonly sampleRate: number = DEFAULT_SAMPLE_RATE,\n        private readonly catalog?: AssetCatalog,\n    ) { }\n\n    mix(\n        requests: readonly ScheduledAudioRequest[],\n        totalDuration: number,\n    ): Promise<AudioBuffer | null> {\n        return mixAudio(requests, totalDuration, {\n            sampleRate: this.sampleRate,\n            catalog: this.catalog,\n        });\n    }\n}\n\n/**\n * Encode an `AudioBuffer` as a 16-bit PCM WAV.\n *\n * The only other encoder in this package is the AAC track inside the mp4 muxer,\n * which is reachable solely by exporting a video — so without this a mixdown has\n * no file representation at all. WAV because it needs no codec support and is\n * lossless; a host wanting mp3/aac can re-encode from here.\n */\nexport function encodeWav(buffer: AudioBuffer): Uint8Array<ArrayBuffer> {\n    const channelCount = buffer.numberOfChannels;\n    const frameCount = buffer.length;\n    const bytesPerSample = 2;\n    const blockAlign = channelCount * bytesPerSample;\n    const dataBytes = frameCount * blockAlign;\n\n    const out = new ArrayBuffer(44 + dataBytes);\n    const view = new DataView(out);\n\n    const ascii = (offset: number, text: string) => {\n        for (let i = 0; i < text.length; i++) view.setUint8(offset + i, text.charCodeAt(i));\n    };\n\n    ascii(0, \"RIFF\");\n    view.setUint32(4, 36 + dataBytes, true);\n    ascii(8, \"WAVE\");\n    ascii(12, \"fmt \");\n    view.setUint32(16, 16, true);            // PCM header length\n    view.setUint16(20, 1, true);             // format: PCM\n    view.setUint16(22, channelCount, true);\n    view.setUint32(24, buffer.sampleRate, true);\n    view.setUint32(28, buffer.sampleRate * blockAlign, true);\n    view.setUint16(32, blockAlign, true);\n    view.setUint16(34, 8 * bytesPerSample, true);\n    ascii(36, \"data\");\n    view.setUint32(40, dataBytes, true);\n\n    // Pull each channel out once — getChannelData is a live view, but calling it\n    // per sample inside the interleave loop is measurably slower.\n    const channels: Float32Array[] = [];\n    for (let c = 0; c < channelCount; c++) channels.push(buffer.getChannelData(c));\n\n    let offset = 44;\n    for (let frame = 0; frame < frameCount; frame++) {\n        for (let c = 0; c < channelCount; c++) {\n            // Clamp before scaling: the mix bus can exceed [-1,1] where clips\n            // overlap, and letting that wrap turns a loud passage into noise.\n            const sample = Math.max(-1, Math.min(1, channels[c][frame]));\n            view.setInt16(offset, sample < 0 ? sample * 0x8000 : sample * 0x7fff, true);\n            offset += bytesPerSample;\n        }\n    }\n\n    return new Uint8Array(out);\n}\n"]}
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{ sourceTimeAtClipElapsed, type AssetCatalog } from \"@motionscript/core\";\r\nimport type { AudioMixer, ScheduledAudioRequest } from \"@motionscript/skia-render/export\";\r\nimport { fetchAssetBytes, localAssetBytes } from \"../asset-bytes\";\r\nimport { AudioBusGraph } from \"./bus-graph\";\r\nimport { applyLoop } from \"./loop-region\";\r\nimport {\r\n    applySpeedToPlaybackRate,\r\n    buildAudioFilterGraph,\r\n    effectiveSpeed,\r\n    speedCurveOf,\r\n} from \"./filter-graph\";\r\n\r\n/** Sample rate used when the caller doesn't pick one. */\r\nexport const DEFAULT_SAMPLE_RATE = 44100;\r\n\r\nexport interface MixAudioOptions {\r\n    /** Defaults to {@link DEFAULT_SAMPLE_RATE}. */\r\n    sampleRate?: number;\r\n    /**\r\n     * Resolves a request's `src` to the address its bytes live at, the same job\r\n     * `WebStorageAdapter.assetUrl` does for the live paths. Omitted, a `src` is\r\n     * fetched as written — right only while the host keys its manifest by URL.\r\n     */\r\n    catalog?: AssetCatalog;\r\n}\r\n\r\n/** Audio first, then video: a clip's own soundtrack is requested under the video's key. */\r\nfunction audioUrl(src: string, catalog: AssetCatalog | undefined): string {\r\n    if (!catalog) return src;\r\n    try {\r\n        return catalog.getAudioMeta(src).src || src;\r\n    } catch {\r\n        try {\r\n            return catalog.getVideoMeta(src).src || src;\r\n        } catch {\r\n            return src;\r\n        }\r\n    }\r\n}\r\n\r\n/**\r\n * Reads and decodes an audio source against a scratch `OfflineAudioContext`\r\n * (decoding requires a context but doesn't render through it). The host's byte\r\n * source is asked first, as for every other asset read: a mix that went to the\r\n * network would fail where the preview played, offline or for bytes not yet\r\n * uploaded.\r\n */\r\nasync function fetchAudioBuffer(src: string, ctx: OfflineAudioContext): Promise<AudioBuffer> {\r\n    const blob = (await localAssetBytes(src)) ?? (await fetchAssetBytes(src));\r\n    return ctx.decodeAudioData(await blob.arrayBuffer());\r\n}\r\n\r\n/**\r\n * Render every scheduled request into one timeline-length buffer.\r\n *\r\n * An `OfflineAudioContext` graph — gain per request, looping/trimming, and the\r\n * same filter chain the live {@link WebAudioSink} builds — rendered in one\r\n * pass rather than scheduled per frame.\r\n *\r\n * Shared by the video exporter and by `mixTimelineAudio`, deliberately: a split\r\n * export and a whole one have to be the same mix, and two implementations of\r\n * \"stack these clips\" would eventually disagree.\r\n *\r\n * Curve-valued filter params and `speed` curves schedule here exactly as they do\r\n * in {@link WebAudioSink.playBuffer}, which is the reference implementation —\r\n * the two must be read side by side when either changes.\r\n *\r\n * One deliberate difference: this schedules at absolute context times, so clip\r\n * starts are sample-accurate here and quantized to `1/fps` in live playback,\r\n * which starts sources at `currentTime` on a clock tick.\r\n */\r\nexport async function mixAudio(\r\n    requests: readonly ScheduledAudioRequest[],\r\n    totalDuration: number,\r\n    options: MixAudioOptions = {},\r\n): Promise<AudioBuffer | null> {\r\n    if (requests.length === 0) return null;\r\n    const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE;\r\n    const frames = Math.ceil(totalDuration * sampleRate);\r\n    if (frames <= 0) return null;\r\n\r\n    const uniqueSrcs = [...new Set(requests.map(r => r.request.src))];\r\n    const scratchCtx = new OfflineAudioContext(2, frames, sampleRate);\r\n\r\n    // A source only a picture's own sound wants may hold no audio at all, which\r\n    // is silence rather than a failed mix — see `AudioRequest.optional`.\r\n    const asked = new Set(requests.filter(r => !r.request.optional).map(r => r.request.src));\r\n    const decoded = new Map<string, AudioBuffer>();\r\n    await Promise.all(uniqueSrcs.map(async (src) => {\r\n        try {\r\n            decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));\r\n        } catch (cause) {\r\n            if (asked.has(src)) throw cause;\r\n        }\r\n    }));\r\n\r\n    const mixCtx = new OfflineAudioContext(2, frames, sampleRate);\r\n    // The same shared-chain rule the live device uses, for the same reason: an\r\n    // export that flattened buses into per-clip processing would not match what\r\n    // the preview sounded like.\r\n    const buses = new AudioBusGraph(mixCtx, mixCtx.destination);\r\n\r\n    for (const { request: req, globalOffset } of requests) {\r\n        const srcBuffer = decoded.get(req.src);\r\n        if (!srcBuffer) continue;\r\n\r\n        const globalStart = globalOffset + req.startAt;\r\n        // `endAt` is `Infinity` for an open clip; either way the timeline bounds it.\r\n        const globalEnd = Math.min(globalOffset + req.endAt, totalDuration);\r\n        const playDuration = globalEnd - globalStart;\r\n        if (playDuration <= 0) continue;\r\n\r\n        const source = mixCtx.createBufferSource();\r\n        source.buffer = srcBuffer;\r\n        applyLoop(source, req);\r\n\r\n        const scalarSpeed = effectiveSpeed(req.filters);\r\n        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;\r\n        const speedCurve = speedCurveOf(req.filters);\r\n        // With a curve the scalar is folded into the scheduled ramp below, so\r\n        // setting it here as well would apply it twice.\r\n        if (rate !== 1 && !speedCurve) source.playbackRate.value = rate;\r\n\r\n        const gain = mixCtx.createGain();\r\n        gain.gain.value = Number.isFinite(req.volume) ? req.volume : 1;\r\n\r\n        // The trimmed source length the filter and speed curves are authored over.\r\n        const sourceLength = Math.max(0, srcBuffer.duration - req.trimStart);\r\n        // The clip's own clip-time length — what a curve resolves against.\r\n        // Deliberately *not* `playDuration`, which is that length truncated by the\r\n        // timeline end: resolving against the truncated value would squeeze the\r\n        // whole curve into the surviving part instead of cutting it off.\r\n        const clipDuration = Number.isFinite(req.endAt)\r\n            ? Math.max(0, req.endAt - req.startAt)\r\n            : sourceLength / rate;\r\n\r\n        // Filter chain sits between the source and the per-request gain, matching\r\n        // the live WebAudioSink graph so exports sound identical to preview.\r\n        // Passing the start time and clip duration is what lets curve-valued params\r\n        // schedule — with the defaults they collapse to a single static value.\r\n        // `elapsed` is 0: unlike a live seek, the mix always starts a clip at its\r\n        // own beginning.\r\n        const graph = buildAudioFilterGraph(\r\n            mixCtx, source, req.filters ?? [], globalStart, clipDuration, 0,\r\n        );\r\n        graph.output.connect(gain);\r\n        // From the mix's start, whichever clip asks first: a bus's chain is shared,\r\n        // so its schedule cannot begin at one clip's start. Context time 0 is\r\n        // `-globalOffset` in this request's seconds, which are its stops' too.\r\n        gain.connect(buses.inputFor(req.bus, 0, -globalOffset));\r\n\r\n        if (speedCurve) {\r\n            applySpeedToPlaybackRate(\r\n                source.playbackRate, speedCurve, sourceLength, clipDuration, rate, globalStart, 0,\r\n            );\r\n        }\r\n\r\n        // start()'s 3rd arg is a source-buffer duration. A constant rate scales it\r\n        // linearly, but under a speed curve the buffer consumed over a span of\r\n        // clip time is the integral of the rate, not a multiple of it.\r\n        const consumed = speedCurve\r\n            ? sourceTimeAtClipElapsed(speedCurve, sourceLength, playDuration)\r\n            : playDuration * rate;\r\n\r\n        source.start(globalStart, req.trimStart, consumed);\r\n        for (const osc of graph.oscillators) osc.start(globalStart);\r\n    }\r\n\r\n    return mixCtx.startRendering();\r\n}\r\n\r\n/**\r\n * Adapter presenting {@link mixAudio} as the exporter's injected mixer seam.\r\n *\r\n * The sample rate is a constructor argument rather than a third `mix` parameter\r\n * because {@link AudioMixer} — the seam `renderTimeline` calls through — takes\r\n * exactly two, so anything passed as a third would be silently dropped. Fixing\r\n * it here rather than widening the interface keeps the rate a property of *this*\r\n * mixer, which is whose business it is.\r\n */\r\nexport class WebAudioMixer implements AudioMixer<AudioBuffer> {\r\n    constructor(\r\n        private readonly sampleRate: number = DEFAULT_SAMPLE_RATE,\r\n        private readonly catalog?: AssetCatalog,\r\n    ) { }\r\n\r\n    mix(\r\n        requests: readonly ScheduledAudioRequest[],\r\n        totalDuration: number,\r\n    ): Promise<AudioBuffer | null> {\r\n        return mixAudio(requests, totalDuration, {\r\n            sampleRate: this.sampleRate,\r\n            catalog: this.catalog,\r\n        });\r\n    }\r\n}\r\n\r\n/**\r\n * Encode an `AudioBuffer` as a 16-bit PCM WAV.\r\n *\r\n * The only other encoder in this package is the AAC track inside the mp4 muxer,\r\n * which is reachable solely by exporting a video — so without this a mixdown has\r\n * no file representation at all. WAV because it needs no codec support and is\r\n * lossless; a host wanting mp3/aac can re-encode from here.\r\n */\r\nexport function encodeWav(buffer: AudioBuffer): Uint8Array<ArrayBuffer> {\r\n    const channelCount = buffer.numberOfChannels;\r\n    const frameCount = buffer.length;\r\n    const bytesPerSample = 2;\r\n    const blockAlign = channelCount * bytesPerSample;\r\n    const dataBytes = frameCount * blockAlign;\r\n\r\n    const out = new ArrayBuffer(44 + dataBytes);\r\n    const view = new DataView(out);\r\n\r\n    const ascii = (offset: number, text: string) => {\r\n        for (let i = 0; i < text.length; i++) view.setUint8(offset + i, text.charCodeAt(i));\r\n    };\r\n\r\n    ascii(0, \"RIFF\");\r\n    view.setUint32(4, 36 + dataBytes, true);\r\n    ascii(8, \"WAVE\");\r\n    ascii(12, \"fmt \");\r\n    view.setUint32(16, 16, true);            // PCM header length\r\n    view.setUint16(20, 1, true);             // format: PCM\r\n    view.setUint16(22, channelCount, true);\r\n    view.setUint32(24, buffer.sampleRate, true);\r\n    view.setUint32(28, buffer.sampleRate * blockAlign, true);\r\n    view.setUint16(32, blockAlign, true);\r\n    view.setUint16(34, 8 * bytesPerSample, true);\r\n    ascii(36, \"data\");\r\n    view.setUint32(40, dataBytes, true);\r\n\r\n    // Pull each channel out once — getChannelData is a live view, but calling it\r\n    // per sample inside the interleave loop is measurably slower.\r\n    const channels: Float32Array[] = [];\r\n    for (let c = 0; c < channelCount; c++) channels.push(buffer.getChannelData(c));\r\n\r\n    let offset = 44;\r\n    for (let frame = 0; frame < frameCount; frame++) {\r\n        for (let c = 0; c < channelCount; c++) {\r\n            // Clamp before scaling: the mix bus can exceed [-1,1] where clips\r\n            // overlap, and letting that wrap turns a loud passage into noise.\r\n            const sample = Math.max(-1, Math.min(1, channels[c][frame]));\r\n            view.setInt16(offset, sample < 0 ? sample * 0x8000 : sample * 0x7fff, true);\r\n            offset += bytesPerSample;\r\n        }\r\n    }\r\n\r\n    return new Uint8Array(out);\r\n}\r\n"]}

package/dist/audio/player.d.ts

@@ -4,7 +4,7 @@ import { AudioSink, AudioRequest } from "@motionscript/core";
  * buffers, then schedules/plays {@link AudioRequest}s as `AudioBufferSourceNode`s
  * through a shared master gain (global mute and volume). `schedule` registers the
  * active set of requests for the current frame; `syncTo` then starts/stops
- * sources to match `sceneTime`, so playback follows scrubbing and seeks.
+ * sources to match `clipTime`, so playback follows scrubbing and seeks.
  */
 export declare class WebAudioSink extends AudioSink {
     private context;
@@ -33,15 +33,15 @@ export declare class WebAudioSink extends AudioSink {
     retain(keep: ReadonlySet<string>): void;
     /** Replaces the active request set; stops any currently-playing source whose request is no longer scheduled. Actual start/stop-to-match-time happens in `syncTo`. */
     schedule(requests: readonly AudioRequest[]): void;
-    /** Starts sources whose `[startAt, endAt)` window now contains `sceneTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
-    syncTo(sceneTime: number): void;
+    /** Starts sources whose `[startAt, endAt)` window now contains `clipTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
+    syncTo(clipTime: number): void;
     play(time: number, _speed: number, _reverse: boolean): Promise<void>;
     stop(): void;
     setMuted(muted: boolean): void;
     setVolume(volume: number): void;
     /** Mute and volume are held separately so unmuting restores the set level. */
     private applyMasterGain;
-    /** Starts a buffer source mid-clip if `sceneTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
+    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
     private playBuffer;
     private stopSource;
     dispose(): void;

package/dist/audio/player.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"player.d.ts","sourceRoot":"","sources":["../../src/audio/player.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,SAAS,EAAE,YAAY,EAA4B,MAAM,oBAAoB,CAAC;AAcvF;;;;;;GAMG;AACH,qBAAa,YAAa,SAAQ,SAAS;IACvC,OAAO,CAAC,OAAO,CAAe;IAC9B,OAAO,CAAC,WAAW,CAAU;IAC7B,OAAO,CAAC,WAAW,CAAkC;IACrD,OAAO,CAAC,QAAQ,CAAoC;IACpD,OAAO,CAAC,MAAM,CAAmC;IACjD,OAAO,CAAC,SAAS,CAAsB;IACvC,OAAO,CAAC,QAAQ,CAAkB;IAClC,OAAO,CAAC,UAAU,CAAW;IAC7B,oEAAoE;IACpE,OAAO,CAAC,KAAK,CAAgB;IAC7B,OAAO,CAAC,KAAK,CAAS;IACtB,OAAO,CAAC,MAAM,CAAK;gBAEP,OAAO,CAAC,EAAE,YAAY;IASlC,UAAU,IAAI,YAAY;IAI1B;;;;OAIG;IACH,MAAM,IAAI,IAAI;IAMd,GAAG,CAAC,GAAG,EAAE,MAAM,GAAG,OAAO;IAInB,MAAM,CAAC,GAAG,EAAE,MAAM,EAAE,IAAI,EAAE,WAAW,GAAG,OAAO,CAAC,IAAI,CAAC;IAkB3D,uKAAuK;IACvK,MAAM,CAAC,IAAI,EAAE,WAAW,CAAC,MAAM,CAAC,GAAG,IAAI;IASvC,qKAAqK;IACrK,QAAQ,CAAC,QAAQ,EAAE,SAAS,YAAY,EAAE,GAAG,IAAI;IAQjD,kLAAkL;IAClL,MAAM,CAAC,SAAS,EAAE,MAAM,GAAG,IAAI;IAqBzB,IAAI,CAAC,IAAI,EAAE,MAAM,EAAE,MAAM,EAAE,MAAM,EAAE,QAAQ,EAAE,OAAO,GAAG,OAAO,CAAC,IAAI,CAAC;IAO1E,IAAI,IAAI,IAAI;IASZ,QAAQ,CAAC,KAAK,EAAE,OAAO,GAAG,IAAI;IAK9B,SAAS,CAAC,MAAM,EAAE,MAAM,GAAG,IAAI;IAK/B,8EAA8E;IAC9E,OAAO,CAAC,eAAe;IAIvB,6MAA6M;IAC7M,OAAO,CAAC,UAAU;IAqElB,OAAO,CAAC,UAAU;IAclB,OAAO,IAAI,IAAI;CAiBlB"}
+{"version":3,"file":"player.d.ts","sourceRoot":"","sources":["../../src/audio/player.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,SAAS,EAAE,YAAY,EAA2B,MAAM,oBAAoB,CAAC;AAetF;;;;;;GAMG;AACH,qBAAa,YAAa,SAAQ,SAAS;IACvC,OAAO,CAAC,OAAO,CAAe;IAC9B,OAAO,CAAC,WAAW,CAAU;IAC7B,OAAO,CAAC,WAAW,CAAkC;IACrD,OAAO,CAAC,QAAQ,CAAoC;IACpD,OAAO,CAAC,MAAM,CAAmC;IACjD,OAAO,CAAC,SAAS,CAAsB;IACvC,OAAO,CAAC,QAAQ,CAAkB;IAClC,OAAO,CAAC,UAAU,CAAW;IAC7B,oEAAoE;IACpE,OAAO,CAAC,KAAK,CAAgB;IAC7B,OAAO,CAAC,KAAK,CAAS;IACtB,OAAO,CAAC,MAAM,CAAK;gBAEP,OAAO,CAAC,EAAE,YAAY;IASlC,UAAU,IAAI,YAAY;IAI1B;;;;OAIG;IACH,MAAM,IAAI,IAAI;IAMd,GAAG,CAAC,GAAG,EAAE,MAAM,GAAG,OAAO;IAInB,MAAM,CAAC,GAAG,EAAE,MAAM,EAAE,IAAI,EAAE,WAAW,GAAG,OAAO,CAAC,IAAI,CAAC;IAkB3D,uKAAuK;IACvK,MAAM,CAAC,IAAI,EAAE,WAAW,CAAC,MAAM,CAAC,GAAG,IAAI;IASvC,qKAAqK;IACrK,QAAQ,CAAC,QAAQ,EAAE,SAAS,YAAY,EAAE,GAAG,IAAI;IAQjD,iLAAiL;IACjL,MAAM,CAAC,QAAQ,EAAE,MAAM,GAAG,IAAI;IAqBxB,IAAI,CAAC,IAAI,EAAE,MAAM,EAAE,MAAM,EAAE,MAAM,EAAE,QAAQ,EAAE,OAAO,GAAG,OAAO,CAAC,IAAI,CAAC;IAO1E,IAAI,IAAI,IAAI;IASZ,QAAQ,CAAC,KAAK,EAAE,OAAO,GAAG,IAAI;IAK9B,SAAS,CAAC,MAAM,EAAE,MAAM,GAAG,IAAI;IAK/B,8EAA8E;IAC9E,OAAO,CAAC,eAAe;IAIvB,4MAA4M;IAC5M,OAAO,CAAC,UAAU;IAqElB,OAAO,CAAC,UAAU;IAclB,OAAO,IAAI,IAAI;CAiBlB"}

package/dist/audio/player.js

@@ -1,12 +1,13 @@
-import { AudioSink, sourceTimeAtSceneElapsed } from "@motionscript/core";
+import { AudioSink, sourceTimeAtClipElapsed } from "@motionscript/core";
 import { AudioBusGraph } from "./bus-graph";
+import { applyLoop, wrapIntoRegion } from "./loop-region";
 import { buildAudioFilterGraph, effectiveSpeed, speedCurveOf, applySpeedToPlaybackRate } from "./filter-graph";
 /**
  * Web Audio implementation of {@link AudioSink} — decodes and caches source
  * buffers, then schedules/plays {@link AudioRequest}s as `AudioBufferSourceNode`s
  * through a shared master gain (global mute and volume). `schedule` registers the
  * active set of requests for the current frame; `syncTo` then starts/stops
- * sources to match `sceneTime`, so playback follows scrubbing and seeks.
+ * sources to match `clipTime`, so playback follows scrubbing and seeks.
  */
 export class WebAudioSink extends AudioSink {
     context;
@@ -83,20 +84,20 @@ export class WebAudioSink extends AudioSink {
         }
         this.scheduled = requests.slice();
     }
-    /** Starts sources whose `[startAt, endAt)` window now contains `sceneTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
-    syncTo(sceneTime) {
+    /** Starts sources whose `[startAt, endAt)` window now contains `clipTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
+    syncTo(clipTime) {
         const liveIds = new Set();
         for (const req of this.scheduled) {
             // `endAt` is `Infinity` for an open clip, so the comparison needs no
             // nullish fallback — see AudioRequest.endAt.
-            const isInWindow = sceneTime >= req.startAt && sceneTime < req.endAt;
+            const isInWindow = clipTime >= req.startAt && clipTime < req.endAt;
             if (!isInWindow)
                 continue;
             liveIds.add(req.id);
             if (!this.active.has(req.id)) {
                 const buffer = this.bufferCache.get(req.src);
                 if (buffer)
-                    this.playBuffer(buffer, req, sceneTime);
+                    this.playBuffer(buffer, req, clipTime);
             }
         }
         for (const [id, active] of this.active) {
@@ -130,8 +131,8 @@ export class WebAudioSink extends AudioSink {
     applyMasterGain() {
         this.masterGain.gain.value = this.muted ? 0 : this.volume;
     }
-    /** Starts a buffer source mid-clip if `sceneTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
-    playBuffer(buffer, req, sceneTime) {
+    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
+    playBuffer(buffer, req, clipTime) {
         const startTime = this.context.currentTime;
         const gainNode = this.context.createGain();
         // Guard against a missing/NaN volume — a non-finite AudioParam value throws.
@@ -139,19 +140,19 @@ export class WebAudioSink extends AudioSink {
         // Into this clip's bus chain, or straight to the master when it has none.
         // The chain is shared: two clips on one bus reach the same nodes, which is
         // what makes a bus effect one tail rather than one per clip.
-        gainNode.connect(this.buses.inputFor(req.bus, startTime, sceneTime));
+        gainNode.connect(this.buses.inputFor(req.bus, startTime, clipTime));
         const source = this.context.createBufferSource();
         source.buffer = buffer;
-        source.loop = req.loop;
+        const loopRegion = applyLoop(source, req);
         const scalarSpeed = effectiveSpeed(req.filters);
         const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
         const speedCurve = speedCurveOf(req.filters);
         if (rate !== 1 && !speedCurve)
             source.playbackRate.value = rate;
-        const elapsed = Math.max(0, sceneTime - req.startAt);
+        const elapsed = Math.max(0, clipTime - req.startAt);
         // Trimmed source length the filter curves / speed curve are authored over.
         const sourceLength = Math.max(0, buffer.duration - req.trimStart);
-        // Scene-time length of the clip. Prefer the bounded [startAt, endAt) span; if
+        // Clip-time length of the clip. Prefer the bounded [startAt, endAt) span; if
         // endAt is unbounded (an open/looping clip reaching the device unresolved),
         // fall back to the source length scaled by speed so curve params still resolve
         // (a clipDuration of 0 would otherwise silently drop every fade/sweep).
@@ -167,14 +168,14 @@ export class WebAudioSink extends AudioSink {
         // speed curve uses the integral inverse to find the true source position.
         let audioOffset;
         if (speedCurve) {
-            audioOffset = req.trimStart + sourceTimeAtSceneElapsed(speedCurve, sourceLength, elapsed);
+            audioOffset = req.trimStart + sourceTimeAtClipElapsed(speedCurve, sourceLength, elapsed);
             // Schedule the time-varying playbackRate (folds in any scalar speed too).
             applySpeedToPlaybackRate(source.playbackRate, speedCurve, sourceLength, clipDuration, rate, startTime, elapsed);
         }
         else {
             audioOffset = req.trimStart + elapsed * rate;
         }
-        source.start(startTime, audioOffset);
+        source.start(startTime, wrapIntoRegion(loopRegion, audioOffset));
         for (const osc of graph.oscillators)
             osc.start(startTime);
         source.onended = () => {

package/dist/audio/player.js.map

@@ -1 +1 @@
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{ AudioSink, AudioRequest, sourceTimeAtSceneElapsed } from \"@motionscript/core\";\nimport { AudioBusGraph } from \"./bus-graph\";\nimport { buildAudioFilterGraph, effectiveSpeed, speedCurveOf, applySpeedToPlaybackRate } from \"./filter-graph\";\n\ntype ActiveSource = {\n    source: AudioBufferSourceNode;\n    gainNode: GainNode;\n    request: AudioRequest;\n    /** Filter-graph nodes to disconnect on teardown. */\n    filterNodes: AudioNode[];\n    /** Tremolo LFOs to stop on teardown. */\n    oscillators: OscillatorNode[];\n};\n\n/**\n * Web Audio implementation of {@link AudioSink} — decodes and caches source\n * buffers, then schedules/plays {@link AudioRequest}s as `AudioBufferSourceNode`s\n * through a shared master gain (global mute and volume). `schedule` registers the\n * active set of requests for the current frame; `syncTo` then starts/stops\n * sources to match `sceneTime`, so playback follows scrubbing and seeks.\n */\nexport class WebAudioSink extends AudioSink {\n    private context: AudioContext;\n    private ownsContext: boolean;\n    private bufferCache = new Map<string, AudioBuffer>();\n    private decoding = new Map<string, Promise<void>>();\n    private active = new Map<string, ActiveSource>();\n    private scheduled: AudioRequest[] = [];\n    private disposed: boolean = false;\n    private masterGain: GainNode;\n    /** The live bus chains, shared across every clip that names one. */\n    private buses: AudioBusGraph;\n    private muted = false;\n    private volume = 1;\n\n    constructor(context?: AudioContext) {\n        super();\n        this.context = context ?? new AudioContext();\n        this.ownsContext = !context;\n        this.masterGain = this.context.createGain();\n        this.masterGain.connect(this.context.destination);\n        this.buses = new AudioBusGraph(this.context, this.masterGain);\n    }\n\n    getContext(): AudioContext {\n        return this.context;\n    }\n\n    /**\n     * Resume the AudioContext. Must be called from within a user gesture\n     * (e.g. a click handler) the first time, or browsers keep the context\n     * suspended and no audio is heard.\n     */\n    unlock(): void {\n        if (this.context.state === \"suspended\") {\n            this.context.resume();\n        }\n    }\n\n    has(src: string): boolean {\n        return this.bufferCache.has(src);\n    }\n\n    async append(src: string, data: ArrayBuffer): Promise<void> {\n        if (this.bufferCache.has(src)) return;\n        const existing = this.decoding.get(src);\n        if (existing) return existing;\n\n        // decodeAudioData detaches `data` — that's fine, the caller in\n        // AssetManager.fetchAudio discards its reference immediately after\n        // awaiting append. Skipping the defensive .slice(0) saves a multi-MB\n        // allocation per audio source.\n        const job = this.context.decodeAudioData(data).then((buffer) => {\n            this.bufferCache.set(src, buffer);\n        }).finally(() => {\n            this.decoding.delete(src);\n        });\n        this.decoding.set(src, job);\n        await job;\n    }\n\n    /** Drops cached buffers and stops active sources for srcs/requests no longer referenced — called when the asset set changes (e.g. seeking across scene boundaries). */\n    retain(keep: ReadonlySet<string>): void {\n        for (const src of [...this.bufferCache.keys()]) {\n            if (!keep.has(src)) this.bufferCache.delete(src);\n        }\n        for (const [id, active] of this.active) {\n            if (!keep.has(active.request.src)) this.stopSource(id, active);\n        }\n    }\n\n    /** Replaces the active request set; stops any currently-playing source whose request is no longer scheduled. Actual start/stop-to-match-time happens in `syncTo`. */\n    schedule(requests: readonly AudioRequest[]): void {\n        const next = new Set(requests.map(r => r.id));\n        for (const [id, active] of this.active) {\n            if (!next.has(id)) this.stopSource(id, active);\n        }\n        this.scheduled = requests.slice();\n    }\n\n    /** Starts sources whose `[startAt, endAt)` window now contains `sceneTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */\n    syncTo(sceneTime: number): void {\n        const liveIds = new Set<string>();\n\n        for (const req of this.scheduled) {\n            // `endAt` is `Infinity` for an open clip, so the comparison needs no\n            // nullish fallback — see AudioRequest.endAt.\n            const isInWindow = sceneTime >= req.startAt && sceneTime < req.endAt;\n            if (!isInWindow) continue;\n\n            liveIds.add(req.id);\n            if (!this.active.has(req.id)) {\n                const buffer = this.bufferCache.get(req.src);\n                if (buffer) this.playBuffer(buffer, req, sceneTime);\n            }\n        }\n\n        for (const [id, active] of this.active) {\n            if (!liveIds.has(id)) this.stopSource(id, active);\n        }\n    }\n\n    async play(time: number, _speed: number, _reverse: boolean): Promise<void> {\n        if (this.context.state === \"suspended\") {\n            await this.context.resume();\n        }\n        this.syncTo(time);\n    }\n\n    stop(): void {\n        for (const [id, active] of this.active) {\n            this.stopSource(id, active);\n        }\n        // A bus's curves are scheduled against the clock of the run that built\n        // it, so the next play, from wherever it lands, builds them again.\n        this.buses.dispose();\n    }\n\n    setMuted(muted: boolean): void {\n        this.muted = muted;\n        this.applyMasterGain();\n    }\n\n    setVolume(volume: number): void {\n        this.volume = Math.max(0, volume);\n        this.applyMasterGain();\n    }\n\n    /** Mute and volume are held separately so unmuting restores the set level. */\n    private applyMasterGain(): void {\n        this.masterGain.gain.value = this.muted ? 0 : this.volume;\n    }\n\n    /** Starts a buffer source mid-clip if `sceneTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */\n    private playBuffer(buffer: AudioBuffer, req: AudioRequest, sceneTime: number): void {\n        const startTime = this.context.currentTime;\n        const gainNode = this.context.createGain();\n        // Guard against a missing/NaN volume — a non-finite AudioParam value throws.\n        gainNode.gain.value = Number.isFinite(req.volume) ? req.volume : 1;\n        // Into this clip's bus chain, or straight to the master when it has none.\n        // The chain is shared: two clips on one bus reach the same nodes, which is\n        // what makes a bus effect one tail rather than one per clip.\n        gainNode.connect(this.buses.inputFor(req.bus, startTime, sceneTime));\n\n        const source = this.context.createBufferSource();\n        source.buffer = buffer;\n        source.loop = req.loop;\n\n        const scalarSpeed = effectiveSpeed(req.filters);\n        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;\n        const speedCurve = speedCurveOf(req.filters);\n        if (rate !== 1 && !speedCurve) source.playbackRate.value = rate;\n\n        const elapsed = Math.max(0, sceneTime - req.startAt);\n        // Trimmed source length the filter curves / speed curve are authored over.\n        const sourceLength = Math.max(0, buffer.duration - req.trimStart);\n        // Scene-time length of the clip. Prefer the bounded [startAt, endAt) span; if\n        // endAt is unbounded (an open/looping clip reaching the device unresolved),\n        // fall back to the source length scaled by speed so curve params still resolve\n        // (a clipDuration of 0 would otherwise silently drop every fade/sweep).\n        const clipDuration = isFinite(req.endAt)\n            ? Math.max(0, req.endAt - req.startAt)\n            : sourceLength / (rate || 1);\n\n        // Insert the filter chain between the source and the per-request gain. Pass\n        // the start time, clip duration, and elapsed so curve-valued params schedule\n        // and align to a mid-clip seek.\n        const graph = buildAudioFilterGraph(\n            this.context, source, req.filters ?? [], startTime, clipDuration, elapsed,\n        );\n        graph.output.connect(gainNode);\n\n        // Source-time offset for a mid-clip seek. Scalar speed scales linearly; a\n        // speed curve uses the integral inverse to find the true source position.\n        let audioOffset: number;\n        if (speedCurve) {\n            audioOffset = req.trimStart + sourceTimeAtSceneElapsed(speedCurve, sourceLength, elapsed);\n            // Schedule the time-varying playbackRate (folds in any scalar speed too).\n            applySpeedToPlaybackRate(\n                source.playbackRate, speedCurve, sourceLength, clipDuration, rate, startTime, elapsed,\n            );\n        } else {\n            audioOffset = req.trimStart + elapsed * rate;\n        }\n\n        source.start(startTime, audioOffset);\n        for (const osc of graph.oscillators) osc.start(startTime);\n\n        source.onended = () => {\n            if (this.active.get(req.id)?.source === source) {\n                this.active.delete(req.id);\n            }\n        };\n\n        this.active.set(req.id, {\n            source,\n            gainNode,\n            request: req,\n            filterNodes: graph.nodes,\n            oscillators: graph.oscillators,\n        });\n    }\n\n    private stopSource(id: string, active: ActiveSource): void {\n        try {\n            active.source.onended = null;\n            active.source.stop();\n            for (const osc of active.oscillators) osc.stop();\n            active.source.disconnect();\n            for (const node of active.filterNodes) node.disconnect();\n            active.gainNode.disconnect();\n        } catch {\n            // already stopped\n        }\n        this.active.delete(id);\n    }\n\n    dispose(): void {\n        if (this.disposed) return;\n        this.disposed = true;\n        super.dispose();\n        this.bufferCache.clear();\n        this.decoding.clear();\n        this.scheduled.length = 0;\n        this.buses.dispose();\n        try {\n            this.masterGain.disconnect();\n        } catch {\n            // already disconnected\n        }\n        if (this.ownsContext && this.context.state !== \"closed\") {\n            this.context.close();\n        }\n    }\n}\n"]}
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{ AudioSink, AudioRequest, sourceTimeAtClipElapsed } from \"@motionscript/core\";\r\nimport { AudioBusGraph } from \"./bus-graph\";\r\nimport { applyLoop, wrapIntoRegion } from \"./loop-region\";\r\nimport { buildAudioFilterGraph, effectiveSpeed, speedCurveOf, applySpeedToPlaybackRate } from \"./filter-graph\";\r\n\r\ntype ActiveSource = {\r\n    source: AudioBufferSourceNode;\r\n    gainNode: GainNode;\r\n    request: AudioRequest;\r\n    /** Filter-graph nodes to disconnect on teardown. */\r\n    filterNodes: AudioNode[];\r\n    /** Tremolo LFOs to stop on teardown. */\r\n    oscillators: OscillatorNode[];\r\n};\r\n\r\n/**\r\n * Web Audio implementation of {@link AudioSink} — decodes and caches source\r\n * buffers, then schedules/plays {@link AudioRequest}s as `AudioBufferSourceNode`s\r\n * through a shared master gain (global mute and volume). `schedule` registers the\r\n * active set of requests for the current frame; `syncTo` then starts/stops\r\n * sources to match `clipTime`, so playback follows scrubbing and seeks.\r\n */\r\nexport class WebAudioSink extends AudioSink {\r\n    private context: AudioContext;\r\n    private ownsContext: boolean;\r\n    private bufferCache = new Map<string, AudioBuffer>();\r\n    private decoding = new Map<string, Promise<void>>();\r\n    private active = new Map<string, ActiveSource>();\r\n    private scheduled: AudioRequest[] = [];\r\n    private disposed: boolean = false;\r\n    private masterGain: GainNode;\r\n    /** The live bus chains, shared across every clip that names one. */\r\n    private buses: AudioBusGraph;\r\n    private muted = false;\r\n    private volume = 1;\r\n\r\n    constructor(context?: AudioContext) {\r\n        super();\r\n        this.context = context ?? new AudioContext();\r\n        this.ownsContext = !context;\r\n        this.masterGain = this.context.createGain();\r\n        this.masterGain.connect(this.context.destination);\r\n        this.buses = new AudioBusGraph(this.context, this.masterGain);\r\n    }\r\n\r\n    getContext(): AudioContext {\r\n        return this.context;\r\n    }\r\n\r\n    /**\r\n     * Resume the AudioContext. Must be called from within a user gesture\r\n     * (e.g. a click handler) the first time, or browsers keep the context\r\n     * suspended and no audio is heard.\r\n     */\r\n    unlock(): void {\r\n        if (this.context.state === \"suspended\") {\r\n            this.context.resume();\r\n        }\r\n    }\r\n\r\n    has(src: string): boolean {\r\n        return this.bufferCache.has(src);\r\n    }\r\n\r\n    async append(src: string, data: ArrayBuffer): Promise<void> {\r\n        if (this.bufferCache.has(src)) return;\r\n        const existing = this.decoding.get(src);\r\n        if (existing) return existing;\r\n\r\n        // decodeAudioData detaches `data` — that's fine, the caller in\r\n        // AssetManager.fetchAudio discards its reference immediately after\r\n        // awaiting append. Skipping the defensive .slice(0) saves a multi-MB\r\n        // allocation per audio source.\r\n        const job = this.context.decodeAudioData(data).then((buffer) => {\r\n            this.bufferCache.set(src, buffer);\r\n        }).finally(() => {\r\n            this.decoding.delete(src);\r\n        });\r\n        this.decoding.set(src, job);\r\n        await job;\r\n    }\r\n\r\n    /** Drops cached buffers and stops active sources for srcs/requests no longer referenced — called when the asset set changes (e.g. seeking across scene boundaries). */\r\n    retain(keep: ReadonlySet<string>): void {\r\n        for (const src of [...this.bufferCache.keys()]) {\r\n            if (!keep.has(src)) this.bufferCache.delete(src);\r\n        }\r\n        for (const [id, active] of this.active) {\r\n            if (!keep.has(active.request.src)) this.stopSource(id, active);\r\n        }\r\n    }\r\n\r\n    /** Replaces the active request set; stops any currently-playing source whose request is no longer scheduled. Actual start/stop-to-match-time happens in `syncTo`. */\r\n    schedule(requests: readonly AudioRequest[]): void {\r\n        const next = new Set(requests.map(r => r.id));\r\n        for (const [id, active] of this.active) {\r\n            if (!next.has(id)) this.stopSource(id, active);\r\n        }\r\n        this.scheduled = requests.slice();\r\n    }\r\n\r\n    /** Starts sources whose `[startAt, endAt)` window now contains `clipTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */\r\n    syncTo(clipTime: number): void {\r\n        const liveIds = new Set<string>();\r\n\r\n        for (const req of this.scheduled) {\r\n            // `endAt` is `Infinity` for an open clip, so the comparison needs no\r\n            // nullish fallback — see AudioRequest.endAt.\r\n            const isInWindow = clipTime >= req.startAt && clipTime < req.endAt;\r\n            if (!isInWindow) continue;\r\n\r\n            liveIds.add(req.id);\r\n            if (!this.active.has(req.id)) {\r\n                const buffer = this.bufferCache.get(req.src);\r\n                if (buffer) this.playBuffer(buffer, req, clipTime);\r\n            }\r\n        }\r\n\r\n        for (const [id, active] of this.active) {\r\n            if (!liveIds.has(id)) this.stopSource(id, active);\r\n        }\r\n    }\r\n\r\n    async play(time: number, _speed: number, _reverse: boolean): Promise<void> {\r\n        if (this.context.state === \"suspended\") {\r\n            await this.context.resume();\r\n        }\r\n        this.syncTo(time);\r\n    }\r\n\r\n    stop(): void {\r\n        for (const [id, active] of this.active) {\r\n            this.stopSource(id, active);\r\n        }\r\n        // A bus's curves are scheduled against the clock of the run that built\r\n        // it, so the next play, from wherever it lands, builds them again.\r\n        this.buses.dispose();\r\n    }\r\n\r\n    setMuted(muted: boolean): void {\r\n        this.muted = muted;\r\n        this.applyMasterGain();\r\n    }\r\n\r\n    setVolume(volume: number): void {\r\n        this.volume = Math.max(0, volume);\r\n        this.applyMasterGain();\r\n    }\r\n\r\n    /** Mute and volume are held separately so unmuting restores the set level. */\r\n    private applyMasterGain(): void {\r\n        this.masterGain.gain.value = this.muted ? 0 : this.volume;\r\n    }\r\n\r\n    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */\r\n    private playBuffer(buffer: AudioBuffer, req: AudioRequest, clipTime: number): void {\r\n        const startTime = this.context.currentTime;\r\n        const gainNode = this.context.createGain();\r\n        // Guard against a missing/NaN volume — a non-finite AudioParam value throws.\r\n        gainNode.gain.value = Number.isFinite(req.volume) ? req.volume : 1;\r\n        // Into this clip's bus chain, or straight to the master when it has none.\r\n        // The chain is shared: two clips on one bus reach the same nodes, which is\r\n        // what makes a bus effect one tail rather than one per clip.\r\n        gainNode.connect(this.buses.inputFor(req.bus, startTime, clipTime));\r\n\r\n        const source = this.context.createBufferSource();\r\n        source.buffer = buffer;\r\n        const loopRegion = applyLoop(source, req);\r\n\r\n        const scalarSpeed = effectiveSpeed(req.filters);\r\n        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;\r\n        const speedCurve = speedCurveOf(req.filters);\r\n        if (rate !== 1 && !speedCurve) source.playbackRate.value = rate;\r\n\r\n        const elapsed = Math.max(0, clipTime - req.startAt);\r\n        // Trimmed source length the filter curves / speed curve are authored over.\r\n        const sourceLength = Math.max(0, buffer.duration - req.trimStart);\r\n        // Clip-time length of the clip. Prefer the bounded [startAt, endAt) span; if\r\n        // endAt is unbounded (an open/looping clip reaching the device unresolved),\r\n        // fall back to the source length scaled by speed so curve params still resolve\r\n        // (a clipDuration of 0 would otherwise silently drop every fade/sweep).\r\n        const clipDuration = isFinite(req.endAt)\r\n            ? Math.max(0, req.endAt - req.startAt)\r\n            : sourceLength / (rate || 1);\r\n\r\n        // Insert the filter chain between the source and the per-request gain. Pass\r\n        // the start time, clip duration, and elapsed so curve-valued params schedule\r\n        // and align to a mid-clip seek.\r\n        const graph = buildAudioFilterGraph(\r\n            this.context, source, req.filters ?? [], startTime, clipDuration, elapsed,\r\n        );\r\n        graph.output.connect(gainNode);\r\n\r\n        // Source-time offset for a mid-clip seek. Scalar speed scales linearly; a\r\n        // speed curve uses the integral inverse to find the true source position.\r\n        let audioOffset: number;\r\n        if (speedCurve) {\r\n            audioOffset = req.trimStart + sourceTimeAtClipElapsed(speedCurve, sourceLength, elapsed);\r\n            // Schedule the time-varying playbackRate (folds in any scalar speed too).\r\n            applySpeedToPlaybackRate(\r\n                source.playbackRate, speedCurve, sourceLength, clipDuration, rate, startTime, elapsed,\r\n            );\r\n        } else {\r\n            audioOffset = req.trimStart + elapsed * rate;\r\n        }\r\n\r\n        source.start(startTime, wrapIntoRegion(loopRegion, audioOffset));\r\n        for (const osc of graph.oscillators) osc.start(startTime);\r\n\r\n        source.onended = () => {\r\n            if (this.active.get(req.id)?.source === source) {\r\n                this.active.delete(req.id);\r\n            }\r\n        };\r\n\r\n        this.active.set(req.id, {\r\n            source,\r\n            gainNode,\r\n            request: req,\r\n            filterNodes: graph.nodes,\r\n            oscillators: graph.oscillators,\r\n        });\r\n    }\r\n\r\n    private stopSource(id: string, active: ActiveSource): void {\r\n        try {\r\n            active.source.onended = null;\r\n            active.source.stop();\r\n            for (const osc of active.oscillators) osc.stop();\r\n            active.source.disconnect();\r\n            for (const node of active.filterNodes) node.disconnect();\r\n            active.gainNode.disconnect();\r\n        } catch {\r\n            // already stopped\r\n        }\r\n        this.active.delete(id);\r\n    }\r\n\r\n    dispose(): void {\r\n        if (this.disposed) return;\r\n        this.disposed = true;\r\n        super.dispose();\r\n        this.bufferCache.clear();\r\n        this.decoding.clear();\r\n        this.scheduled.length = 0;\r\n        this.buses.dispose();\r\n        try {\r\n            this.masterGain.disconnect();\r\n        } catch {\r\n            // already disconnected\r\n        }\r\n        if (this.ownsContext && this.context.state !== \"closed\") {\r\n            this.context.close();\r\n        }\r\n    }\r\n}\r\n"]}