@motionscript/browser 0.1.0-alpha.0 → 0.1.0-alpha.10

package/dist/audio/mixer.d.ts

@@ -1,13 +1,13 @@
-import { type AssetCatalog } from "@motionscript/core";
-import type { AudioMixer, ScheduledAudioRequest } from "@motionscript/skia-render/export";
+import type { AssetCatalog } from "@motionscript/core";
+import type { AudioMixer, ScheduledAudioRequest } from "../frame-loop";
 /** Sample rate used when the caller doesn't pick one. */
 export declare const DEFAULT_SAMPLE_RATE = 44100;
 export interface MixAudioOptions {
     /** Defaults to {@link DEFAULT_SAMPLE_RATE}. */
     sampleRate?: number;
     /**
-     * Resolves a request's `src` to the address its bytes live at, the same job
-     * `WebStorageAdapter.assetUrl` does for the live paths. Omitted, a `src` is
+     * Resolves a request's `src` to the address its bytes live at, as the storage
+     * adapter does for the live paths. Omitted, a `src` is
      * fetched as written — right only while the host keys its manifest by URL.
      */
     catalog?: AssetCatalog;

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,KAAK,EAAE,YAAY,EAAE,MAAM,oBAAoB,CAAC;AACvD,OAAO,KAAK,EAAE,UAAU,EAAE,qBAAqB,EAAE,MAAM,eAAe,CAAC;AAMvE,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,CAsE7B;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, localAssetBytes } from "../asset-bytes";
 import { AudioBusGraph } from "./bus-graph";
-import { applySpeedToPlaybackRate, buildAudioFilterGraph, effectiveSpeed, speedCurveOf, } from "./filter-graph";
+import { applyLoop } from "./loop-region";
+import { applySpeedToPlaybackRate, buildAudioEffectGraph, buildLevel, playbackOf, sourceOffsetOf } from "./effect-graph";
 /** Sample rate used when the caller doesn't pick one. */
 export const DEFAULT_SAMPLE_RATE = 44100;
 /** Audio first, then video: a clip's own soundtrack is requested under the video's key. */
@@ -19,11 +20,16 @@ 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 response = await fetch(src);
-    const arrayBuffer = await response.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 +59,18 @@ 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
@@ -74,43 +89,27 @@ export async function mixAudio(requests, totalDuration, options = {}) {
             continue;
         const source = mixCtx.createBufferSource();
         source.buffer = srcBuffer;
-        source.loop = req.loop;
-        const scalarSpeed = effectiveSpeed(req.filters);
-        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
-        const speedCurve = speedCurveOf(req.filters);
-        // With a curve the scalar is folded into the scheduled ramp below, so
-        // setting it here as well would apply it twice.
-        if (rate !== 1 && !speedCurve)
-            source.playbackRate.value = rate;
-        const gain = mixCtx.createGain();
-        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.
-        // 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.
-        const clipDuration = Number.isFinite(req.endAt)
-            ? Math.max(0, req.endAt - req.startAt)
-            : sourceLength / rate;
-        // Filter chain sits between the source and the per-request gain, matching
-        // the live WebAudioSink graph so exports sound identical to preview.
-        // Passing the start time and clip duration is what lets curve-valued params
-        // schedule — with the defaults they collapse to a single static value.
-        // `elapsed` is 0: unlike a live seek, the mix always starts a clip at its
-        // own beginning.
-        const graph = buildAudioFilterGraph(mixCtx, source, req.filters ?? [], globalStart, clipDuration, 0);
-        graph.output.connect(gain);
-        gain.connect(buses.inputFor(req.bus, globalStart));
-        if (speedCurve) {
-            applySpeedToPlaybackRate(source.playbackRate, speedCurve, sourceLength, clipDuration, rate, globalStart, 0);
-        }
-        // 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.
-        const consumed = speedCurve
-            ? sourceTimeAtSceneElapsed(speedCurve, sourceLength, playDuration)
-            : playDuration * rate;
+        applyLoop(source, req);
+        // Ramps resolve against the clip's own length, deliberately *not*
+        // `playDuration`, which is that length truncated by the timeline end:
+        // resolving against the truncated value would squeeze the whole ramp
+        // into the surviving part instead of cutting it off. `elapsed` is 0:
+        // unlike a live seek, the mix always starts a clip at its beginning.
+        const playback = playbackOf(req, srcBuffer.duration);
+        if (playback.rate !== 1)
+            source.playbackRate.value = playback.rate;
+        applySpeedToPlaybackRate(source.playbackRate, playback.speedRamp, playback.sourceLength, playback.clipDuration, globalStart, 0);
+        // The same chain as the live WebAudioSink graph, so exports sound like the preview.
+        const graph = buildAudioEffectGraph(mixCtx, source, req.effects ?? [], globalStart, playback.clipDuration, 0);
+        const level = buildLevel(mixCtx, graph.output, req, playback, globalStart, 0);
+        // From the mix's start, whichever clip asks first: a bus's chain is shared,
+        // so its schedule cannot begin at one clip's start. Context time 0 is
+        // `-globalOffset` in this request's seconds, which are its stops' too.
+        level.output.connect(buses.inputFor(req.bus, 0, -globalOffset));
+        // start()'s 3rd arg is a source-buffer duration: under a speed ramp the
+        // buffer consumed over a span of clip time is the integral of the rate,
+        // not a multiple of it.
+        const consumed = sourceOffsetOf(playback, playDuration);
         source.start(globalStart, req.trimStart, consumed);
         for (const osc of graph.oscillators)
             osc.start(globalStart);

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{ sourceTimeAtSceneElapsed, type AssetCatalog } from \"@motionscript/core\";\nimport type { AudioMixer, ScheduledAudioRequest } from \"@motionscript/skia-render/export\";\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 response = await fetch(src);\n    const arrayBuffer = await response.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        gain.connect(buses.inputFor(req.bus, globalStart));\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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type { AssetCatalog } from \"@motionscript/core\";\nimport type { AudioMixer, ScheduledAudioRequest } from \"../frame-loop\";\nimport { fetchAssetBytes, localAssetBytes } from \"../asset-bytes\";\nimport { AudioBusGraph } from \"./bus-graph\";\nimport { applyLoop } from \"./loop-region\";\nimport { applySpeedToPlaybackRate, buildAudioEffectGraph, buildLevel, playbackOf, sourceOffsetOf } from \"./effect-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, as the storage\n     * adapter 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/**\n * Reads and decodes an audio source against a scratch `OfflineAudioContext`\n * (decoding requires a context but doesn't render through it). The host's byte\n * source is asked first, as for every other asset read: a mix that went to the\n * network would fail where the preview played, offline or for bytes not yet\n * uploaded.\n */\nasync function fetchAudioBuffer(src: string, ctx: OfflineAudioContext): Promise<AudioBuffer> {\n    const blob = (await localAssetBytes(src)) ?? (await fetchAssetBytes(src));\n    return ctx.decodeAudioData(await blob.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    // A source only a picture's own sound wants may hold no audio at all, which\n    // is silence rather than a failed mix — see `AudioRequest.optional`.\n    const asked = new Set(requests.filter(r => !r.request.optional).map(r => r.request.src));\n    const decoded = new Map<string, AudioBuffer>();\n    await Promise.all(uniqueSrcs.map(async (src) => {\n        try {\n            decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));\n        } catch (cause) {\n            if (asked.has(src)) throw cause;\n        }\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        applyLoop(source, req);\n\n        // Ramps resolve against the clip's own length, deliberately *not*\n        // `playDuration`, which is that length truncated by the timeline end:\n        // resolving against the truncated value would squeeze the whole ramp\n        // into the surviving part instead of cutting it off. `elapsed` is 0:\n        // unlike a live seek, the mix always starts a clip at its beginning.\n        const playback = playbackOf(req, srcBuffer.duration);\n        if (playback.rate !== 1) source.playbackRate.value = playback.rate;\n        applySpeedToPlaybackRate(\n            source.playbackRate, playback.speedRamp, playback.sourceLength, playback.clipDuration, globalStart, 0,\n        );\n\n        // The same chain as the live WebAudioSink graph, so exports sound like the preview.\n        const graph = buildAudioEffectGraph(mixCtx, source, req.effects ?? [], globalStart, playback.clipDuration, 0);\n        const level = buildLevel(mixCtx, graph.output, req, playback, globalStart, 0);\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        level.output.connect(buses.inputFor(req.bus, 0, -globalOffset));\n\n        // start()'s 3rd arg is a source-buffer duration: under a speed ramp the\n        // buffer consumed over a span of clip time is the integral of the rate,\n        // not a multiple of it.\n        const consumed = sourceOffsetOf(playback, playDuration);\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"]}

package/dist/audio/player.d.ts

@@ -4,23 +4,29 @@ 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;
-    private ownsContext;
+    private readonly borrowed;
+    private _context;
     private bufferCache;
     private decoding;
     private active;
     private scheduled;
     private disposed;
-    private masterGain;
-    /** The live bus chains, shared across every clip that names one. */
-    private buses;
+    /** The master gain and the live bus chains (shared across every clip that names one), built with the context. */
+    private _graph;
     private muted;
     private volume;
     constructor(context?: AudioContext);
+    /**
+     * The context, created on first use rather than with the sink: a new
+     * AudioContext opens the audio device (50–65 ms on a desktop), and a
+     * session that never plays or decodes sound never needs one.
+     */
     getContext(): AudioContext;
+    private get context();
+    private get graph();
     /**
      * Resume the AudioContext. Must be called from within a user gesture
      * (e.g. a click handler) the first time, or browsers keep the context
@@ -33,15 +39,15 @@ 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 — the offset accounts for the request's trim, the elapsed time since `startAt`, and its speed. */
     private playBuffer;
     private stopSource;
     dispose(): void;

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

@@ -1 +1 @@
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+{"version":3,"file":"player.d.ts","sourceRoot":"","sources":["../../src/audio/player.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,SAAS,EAAE,YAAY,EAAE,MAAM,oBAAoB,CAAC;AAc7D;;;;;;GAMG;AACH,qBAAa,YAAa,SAAQ,SAAS;IACvC,OAAO,CAAC,QAAQ,CAAC,QAAQ,CAA2B;IACpD,OAAO,CAAC,QAAQ,CAA6B;IAC7C,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,iHAAiH;IACjH,OAAO,CAAC,MAAM,CAA2D;IACzE,OAAO,CAAC,KAAK,CAAS;IACtB,OAAO,CAAC,MAAM,CAAK;gBAEP,OAAO,CAAC,EAAE,YAAY;IAKlC;;;;OAIG;IACH,UAAU,IAAI,YAAY;IAK1B,OAAO,KAAK,OAAO,GAElB;IAED,OAAO,KAAK,KAAK,GAQhB;IAED;;;;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,mLAAmL;IACnL,OAAO,CAAC,UAAU;IAwClB,OAAO,CAAC,UAAU;IAalB,OAAO,IAAI,IAAI;CAkBlB"}

package/dist/audio/player.js

@@ -1,36 +1,50 @@
-import { AudioSink, sourceTimeAtSceneElapsed } from "@motionscript/core";
+import { AudioSink } from "@motionscript/core";
 import { AudioBusGraph } from "./bus-graph";
-import { buildAudioFilterGraph, effectiveSpeed, speedCurveOf, applySpeedToPlaybackRate } from "./filter-graph";
+import { applyLoop, wrapIntoRegion } from "./loop-region";
+import { applySpeedToPlaybackRate, buildAudioEffectGraph, buildLevel, playbackOf, sourceOffsetOf } from "./effect-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;
-    ownsContext;
+    borrowed;
+    _context = null;
     bufferCache = new Map();
     decoding = new Map();
     active = new Map();
     scheduled = [];
     disposed = false;
-    masterGain;
-    /** The live bus chains, shared across every clip that names one. */
-    buses;
+    /** The master gain and the live bus chains (shared across every clip that names one), built with the context. */
+    _graph = null;
     muted = false;
     volume = 1;
     constructor(context) {
         super();
-        this.context = context ?? new AudioContext();
-        this.ownsContext = !context;
-        this.masterGain = this.context.createGain();
-        this.masterGain.connect(this.context.destination);
-        this.buses = new AudioBusGraph(this.context, this.masterGain);
+        this.borrowed = context;
     }
+    /**
+     * The context, created on first use rather than with the sink: a new
+     * AudioContext opens the audio device (50–65 ms on a desktop), and a
+     * session that never plays or decodes sound never needs one.
+     */
     getContext() {
-        return this.context;
+        this._context ??= this.borrowed ?? new AudioContext();
+        return this._context;
+    }
+    get context() {
+        return this.getContext();
+    }
+    get graph() {
+        if (!this._graph) {
+            const master = this.context.createGain();
+            master.connect(this.context.destination);
+            this._graph = { master, buses: new AudioBusGraph(this.context, master) };
+            this.applyMasterGain();
+        }
+        return this._graph;
     }
     /**
      * Resume the AudioContext. Must be called from within a user gesture
@@ -83,20 +97,20 @@ 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) {
@@ -114,6 +128,9 @@ export class WebAudioSink extends AudioSink {
         for (const [id, active] of this.active) {
             this.stopSource(id, active);
         }
+        // A bus's curves are scheduled against the clock of the run that built
+        // it, so the next play, from wherever it lands, builds them again.
+        this._graph?.buses.dispose();
     }
     setMuted(muted) {
         this.muted = muted;
@@ -125,53 +142,28 @@ export class WebAudioSink extends AudioSink {
     }
     /** Mute and volume are held separately so unmuting restores the set level. */
     applyMasterGain() {
-        this.masterGain.gain.value = this.muted ? 0 : this.volume;
+        if (this._graph)
+            this._graph.master.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) {
-        const gainNode = this.context.createGain();
-        // Guard against a missing/NaN volume — a non-finite AudioParam value throws.
-        gainNode.gain.value = Number.isFinite(req.volume) ? req.volume : 1;
+    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — the offset accounts for the request's trim, the elapsed time since `startAt`, and its speed. */
+    playBuffer(buffer, req, clipTime) {
+        const startTime = this.context.currentTime;
+        const playback = playbackOf(req, buffer.duration);
+        const elapsed = Math.max(0, clipTime - req.startAt);
+        const source = this.context.createBufferSource();
+        source.buffer = buffer;
+        const loopRegion = applyLoop(source, req);
+        if (playback.rate !== 1)
+            source.playbackRate.value = playback.rate;
+        applySpeedToPlaybackRate(source.playbackRate, playback.speedRamp, playback.sourceLength, playback.clipDuration, startTime, elapsed);
+        const graph = buildAudioEffectGraph(this.context, source, req.effects ?? [], startTime, playback.clipDuration, elapsed);
+        const level = buildLevel(this.context, graph.output, req, playback, startTime, elapsed);
         // 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, this.context.currentTime));
-        const source = this.context.createBufferSource();
-        source.buffer = buffer;
-        source.loop = req.loop;
-        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);
-        // 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
-        // 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).
-        const clipDuration = isFinite(req.endAt)
-            ? Math.max(0, req.endAt - req.startAt)
-            : sourceLength / (rate || 1);
-        // Insert the filter chain between the source and the per-request gain. Pass
-        // the start time, clip duration, and elapsed so curve-valued params schedule
-        // and align to a mid-clip seek.
-        const startTime = this.context.currentTime;
-        const graph = buildAudioFilterGraph(this.context, source, req.filters ?? [], startTime, clipDuration, elapsed);
-        graph.output.connect(gainNode);
-        // Source-time offset for a mid-clip seek. Scalar speed scales linearly; a
-        // speed curve uses the integral inverse to find the true source position.
-        let audioOffset;
-        if (speedCurve) {
-            audioOffset = req.trimStart + sourceTimeAtSceneElapsed(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);
+        level.output.connect(this.graph.buses.inputFor(req.bus, startTime, clipTime));
+        const audioOffset = req.trimStart + sourceOffsetOf(playback, elapsed);
+        source.start(startTime, wrapIntoRegion(loopRegion, audioOffset));
         for (const osc of graph.oscillators)
             osc.start(startTime);
         source.onended = () => {
@@ -181,9 +173,8 @@ export class WebAudioSink extends AudioSink {
         };
         this.active.set(req.id, {
             source,
-            gainNode,
             request: req,
-            filterNodes: graph.nodes,
+            nodes: [...graph.nodes, ...level.nodes],
             oscillators: graph.oscillators,
         });
     }
@@ -194,9 +185,8 @@ export class WebAudioSink extends AudioSink {
             for (const osc of active.oscillators)
                 osc.stop();
             active.source.disconnect();
-            for (const node of active.filterNodes)
+            for (const node of active.nodes)
                 node.disconnect();
-            active.gainNode.disconnect();
         }
         catch {
             // already stopped
@@ -211,15 +201,16 @@ export class WebAudioSink extends AudioSink {
         this.bufferCache.clear();
         this.decoding.clear();
         this.scheduled.length = 0;
-        this.buses.dispose();
+        this._graph?.buses.dispose();
         try {
-            this.masterGain.disconnect();
+            this._graph?.master.disconnect();
         }
         catch {
             // already disconnected
         }
-        if (this.ownsContext && this.context.state !== "closed") {
-            this.context.close();
+        const owned = this.borrowed ? null : this._context;
+        if (owned && owned.state !== "closed") {
+            owned.close();
         }
     }
 }

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    }\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 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, this.context.currentTime));\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 startTime = this.context.currentTime;\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 } from \"@motionscript/core\";\nimport { AudioBusGraph } from \"./bus-graph\";\nimport { applyLoop, wrapIntoRegion } from \"./loop-region\";\nimport { applySpeedToPlaybackRate, buildAudioEffectGraph, buildLevel, playbackOf, sourceOffsetOf } from \"./effect-graph\";\n\ntype ActiveSource = {\n    source: AudioBufferSourceNode;\n    request: AudioRequest;\n    /** Effect and level nodes to disconnect on teardown. */\n    nodes: 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 `clipTime`, so playback follows scrubbing and seeks.\n */\nexport class WebAudioSink extends AudioSink {\n    private readonly borrowed: AudioContext | undefined;\n    private _context: AudioContext | null = null;\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    /** The master gain and the live bus chains (shared across every clip that names one), built with the context. */\n    private _graph: { master: GainNode; buses: AudioBusGraph } | null = null;\n    private muted = false;\n    private volume = 1;\n\n    constructor(context?: AudioContext) {\n        super();\n        this.borrowed = context;\n    }\n\n    /**\n     * The context, created on first use rather than with the sink: a new\n     * AudioContext opens the audio device (50–65 ms on a desktop), and a\n     * session that never plays or decodes sound never needs one.\n     */\n    getContext(): AudioContext {\n        this._context ??= this.borrowed ?? new AudioContext();\n        return this._context;\n    }\n\n    private get context(): AudioContext {\n        return this.getContext();\n    }\n\n    private get graph(): { master: GainNode; buses: AudioBusGraph } {\n        if (!this._graph) {\n            const master = this.context.createGain();\n            master.connect(this.context.destination);\n            this._graph = { master, buses: new AudioBusGraph(this.context, master) };\n            this.applyMasterGain();\n        }\n        return this._graph;\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 `clipTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */\n    syncTo(clipTime: 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 = clipTime >= req.startAt && clipTime < 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, clipTime);\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._graph?.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        if (this._graph) this._graph.master.gain.value = this.muted ? 0 : this.volume;\n    }\n\n    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — the offset accounts for the request's trim, the elapsed time since `startAt`, and its speed. */\n    private playBuffer(buffer: AudioBuffer, req: AudioRequest, clipTime: number): void {\n        const startTime = this.context.currentTime;\n        const playback = playbackOf(req, buffer.duration);\n        const elapsed = Math.max(0, clipTime - req.startAt);\n\n        const source = this.context.createBufferSource();\n        source.buffer = buffer;\n        const loopRegion = applyLoop(source, req);\n        if (playback.rate !== 1) source.playbackRate.value = playback.rate;\n        applySpeedToPlaybackRate(\n            source.playbackRate, playback.speedRamp, playback.sourceLength, playback.clipDuration, startTime, elapsed,\n        );\n\n        const graph = buildAudioEffectGraph(\n            this.context, source, req.effects ?? [], startTime, playback.clipDuration, elapsed,\n        );\n        const level = buildLevel(this.context, graph.output, req, playback, startTime, elapsed);\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        level.output.connect(this.graph.buses.inputFor(req.bus, startTime, clipTime));\n\n        const audioOffset = req.trimStart + sourceOffsetOf(playback, elapsed);\n        source.start(startTime, wrapIntoRegion(loopRegion, 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            request: req,\n            nodes: [...graph.nodes, ...level.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.nodes) node.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._graph?.buses.dispose();\n        try {\n            this._graph?.master.disconnect();\n        } catch {\n            // already disconnected\n        }\n        const owned = this.borrowed ? null : this._context;\n        if (owned && owned.state !== \"closed\") {\n            owned.close();\n        }\n    }\n}\n"]}