@motionscript/browser 0.1.0-alpha.7 → 0.1.0-alpha.9

package/dist/browser/chunks/chunk-2MAFJWBZ.js.map

@@ -1,7 +0,0 @@
-{
-  "version": 3,
-  "sources": ["../../../src/asset-bytes.ts", "../../../src/audio/bus-graph.ts", "../../../src/audio/filter-graph.ts", "../../../src/audio/mixer.ts", "../../../src/audio/loop-region.ts", "../../../src/audio/player.ts"],
-  "sourcesContent": ["/**\r\n * Where a source URL's bytes come from: a host's local store, then the network.\r\n *\r\n * ## Why the web adapter needs this at all\r\n *\r\n * `WebStorageAdapter` addresses every asset by URL and fetches it. That is\r\n * right for an asset that lives somewhere, and wrong for one whose bytes the\r\n * host already has in hand — a file the user has just chosen, a render a local\r\n * model has just produced, a picture whose upload has not finished or cannot\r\n * start because there is no network. In all three the adapter downloads bytes\r\n * that never needed to travel.\r\n *\r\n * `NodeStorageAdapter` has had the answer since it was written: an injected\r\n * `AssetLoader`, with an `inline` map consulted ahead of the disk. This is the\r\n * same seam for the browser.\r\n *\r\n * ## Why it is module-level\r\n *\r\n * `WebStorageAdapter` is constructed at five sites *inside* these packages —\r\n * the React player backend, the exporter, the audio mixer, and both still\r\n * renderers — none of which a host can reach. A constructor argument would\r\n * therefore cover the preview and leave export and stills on the network,\r\n * which is the worst of both: the picture a user sees would come from local\r\n * bytes and the one they render would not.\r\n *\r\n * So this is an *injection point*, not a cache. The adapter still owns every\r\n * decode, every cache, every size ceiling and every eviction rule; all this\r\n * changes is where the compressed bytes came from. A source that answers `null`\r\n * — which is what an unregistered URL, an evicted entry or a host with no local\r\n * store does — leaves the adapter exactly as it was.\r\n */\r\n\r\n/** Local bytes for a source URL, or `null` to go to the network. */\r\nexport type AssetByteSource = (src: string) => Promise<Blob | null>;\r\n\r\nlet source: AssetByteSource | null = null;\r\n\r\n/**\r\n * Installs the byte source for this browsing context, or clears it with `null`.\r\n *\r\n * Called once per window that renders. Later calls replace the previous source\r\n * rather than stacking, because two sources disagreeing about one URL has no\r\n * sensible answer.\r\n */\r\nexport function setAssetByteSource(next: AssetByteSource | null): void {\r\n    source = next;\r\n}\r\n\r\n/**\r\n * The bytes for `src`, or `null`.\r\n *\r\n * Swallows a throwing source on purpose. A host's local store failing is a\r\n * cache miss, not a render failure — the caller falls through to `fetch`, which\r\n * is what it would have done had the source never been installed.\r\n */\r\nexport async function localAssetBytes(src: string): Promise<Blob | null> {\r\n    if (!source) return null;\r\n    try {\r\n        return await source(src);\r\n    } catch {\r\n        return null;\r\n    }\r\n}\r\n\r\n/**\r\n * The network's bytes for `url`. A response that is not OK is a failed load,\r\n * never bytes: a 404 page handed to the font manager marked a family loaded\r\n * with no face in it.\r\n */\r\nexport async function fetchAssetBytes(url: string): Promise<Blob> {\r\n    const response = await fetch(url);\r\n    if (!response.ok) {\r\n        const status = response.statusText ? `${response.status} ${response.statusText}` : `${response.status}`;\r\n        throw new Error(`Fetching \"${url}\" failed: HTTP ${status}`);\r\n    }\r\n    return response.blob();\r\n}\r\n", "import { busStopKey, type AudioBusStop, type AudioFilterItem } from \"@motionscript/core\";\r\nimport { buildAudioFilterGraph } from \"./filter-graph\";\r\n\r\n/**\r\n * The live chains for one audio context's buses.\r\n *\r\n * Held per context rather than per clip, because that is the whole difference\r\n * between a bus and per-clip processing: every clip naming a bus connects into\r\n * the *same* nodes, so a reverb on a bus is one tail the group shares rather\r\n * than one tail each. Flattening the levels into per-clip gain would sound\r\n * right; flattening the effects would not.\r\n */\r\nexport class AudioBusGraph {\r\n    constructor(\r\n        private readonly ctx: BaseAudioContext,\r\n        /** Where the last bus in a chain connects — the master, or the destination. */\r\n        private readonly destination: AudioNode,\r\n    ) { }\r\n\r\n    /**\r\n     * The input node a clip on `bus` should connect to, building the chain the\r\n     * first time it is asked for.\r\n     *\r\n     * `bus` is source-adjacent first, so `bus[0]` is the one the clip feeds and\r\n     * each stop after it is that bus's own parent. An empty or absent chain\r\n     * returns the destination, which is what every clip did before buses\r\n     * existed.\r\n     *\r\n     * `startTime` is the context time a new chain's schedule starts from, and\r\n     * `clipTime` the same instant in the stops' own seconds. A curve on a stop\r\n     * runs over its `startAt`–`endAt`, so the two are what place it.\r\n     */\r\n    inputFor(bus: readonly AudioBusStop[] | undefined, startTime: number, clipTime: number): AudioNode {\r\n        if (!bus || bus.length === 0) return this.destination;\r\n        return this.chainFrom(bus, 0, startTime, clipTime);\r\n    }\r\n\r\n    /** Disconnect every live bus node, so the next {@link inputFor} builds afresh. Safe to call more than once. */\r\n    dispose(): void {\r\n        for (const node of this.nodes) {\r\n            try { node.disconnect(); } catch { /* already torn down */ }\r\n        }\r\n        for (const osc of this.oscillators) {\r\n            try { osc.stop(); } catch { /* never started, or already stopped */ }\r\n        }\r\n        this.nodes.length = 0;\r\n        this.oscillators.length = 0;\r\n        this.inputs.clear();\r\n    }\r\n\r\n    private readonly inputs = new Map<string, AudioNode>();\r\n    private readonly nodes: AudioNode[] = [];\r\n    private readonly oscillators: OscillatorNode[] = [];\r\n\r\n    /**\r\n     * Build (or reuse) the chain for `bus[index]`, connected to whatever\r\n     * `bus[index + 1]` resolves to.\r\n     *\r\n     * Recursive rather than iterative because a bus's *output* has to exist\r\n     * before its own nodes can connect to it, and the outermost bus is the last\r\n     * entry.\r\n     *\r\n     * Reused by everything that changes what it sounds like, down to the master,\r\n     * not by id alone: a bus edited under a playing device, or one id on two\r\n     * tracks, is a different chain, and one built for the first would play the\r\n     * second through stale levels and a curve anchored to the wrong span.\r\n     */\r\n    private chainFrom(bus: readonly AudioBusStop[], index: number, startTime: number, clipTime: number): AudioNode {\r\n        if (index >= bus.length) return this.destination;\r\n\r\n        const stop = bus[index];\r\n        const key = bus.slice(index).map(busStopKey).join(\">\");\r\n        const existing = this.inputs.get(key);\r\n        if (existing) return existing;\r\n\r\n        const output = this.chainFrom(bus, index + 1, startTime, clipTime);\r\n\r\n        // A pass-through head every clip connects to, so the bus has one input\r\n        // whatever its filters turn out to be.\r\n        const input = this.ctx.createGain();\r\n        input.gain.value = 1;\r\n\r\n        // The level is a gain after the filters, as a sound's own level is, so a\r\n        // curve on it is scheduled exactly as one in `filters` is.\r\n        const level: AudioFilterItem = { type: \"gain\", value: stop.mute ? 0 : stop.volume };\r\n        const graph = buildAudioFilterGraph(\r\n            this.ctx, input, [...(stop.filters ?? []), level],\r\n            startTime, stop.endAt - stop.startAt, clipTime - stop.startAt,\r\n        );\r\n        graph.output.connect(output);\r\n\r\n        this.nodes.push(input, ...graph.nodes);\r\n        this.oscillators.push(...graph.oscillators);\r\n        for (const osc of graph.oscillators) osc.start(startTime);\r\n\r\n        this.inputs.set(key, input);\r\n        return input;\r\n    }\r\n}\r\n", "import { AudioFilterItem, Param, isCurve, sourceTimeAtClipElapsed } from \"@motionscript/core\";\r\n\r\n/**\r\n * Result of building an audio-filter graph for one source.\r\n *\r\n * - `output` is the last node in the chain; connect it downstream (e.g. to the\r\n *   per-request gain node).\r\n * - `nodes` are every intermediate node, so callers can `disconnect()` them on\r\n *   teardown.\r\n * - `oscillators` are LFOs (from `tremolo`) the caller must `start()` — and may\r\n *   `stop()` — alongside the source, since oscillators need an explicit start.\r\n */\r\nexport interface AudioFilterGraph {\r\n    output: AudioNode;\r\n    nodes: AudioNode[];\r\n    oscillators: OscillatorNode[];\r\n}\r\n\r\n/**\r\n * How many sample points per second to bake an eased curve segment into for\r\n * `setValueCurveAtTime`. 200/s (≈ every 5ms) is smooth to the ear and cheap.\r\n */\r\nconst CURVE_SAMPLES_PER_SECOND = 200;\r\n\r\n/**\r\n * Apply a (possibly time-varying) {@link Param} to a Web Audio {@link AudioParam}.\r\n *\r\n * - **number** → set once (today's behavior; zero scheduling overhead).\r\n * - **Curve** → resolve against `clipDuration` and schedule each segment:\r\n *   - linear segments via `setValueAtTime` + `linearRampToValueAtTime`,\r\n *   - eased segments baked into a `Float32Array` via `setValueCurveAtTime`.\r\n *\r\n * `elapsed` (clip-seconds already played when a mid-clip seek starts the source)\r\n * shifts the schedule back by `elapsed` so \"now\" reads the curve's value at the\r\n * clip's *current* position, not its start. A `clamp(v)` hook lets a param keep a\r\n * valid range (e.g. echo mix 0–1) at every sampled point.\r\n */\r\nfunction applyParam(\r\n    ctx: BaseAudioContext,\r\n    audioParam: AudioParam,\r\n    param: Param,\r\n    startTime: number,\r\n    clipDuration: number,\r\n    elapsed: number,\r\n    clamp: (v: number) => number = (v) => v,\r\n): void {\r\n    // Coerce to a finite value before touching an AudioParam — the platform throws\r\n    // on NaN/Infinity. Falls back to the current param value (never undefined here).\r\n    const finite = (v: number): number => {\r\n        const c = clamp(v);\r\n        return Number.isFinite(c) ? c : audioParam.value;\r\n    };\r\n\r\n    if (!isCurve(param)) {\r\n        audioParam.value = finite(param);\r\n        return;\r\n    }\r\n\r\n    // A non-finite clip length yields no usable schedule; hold the curve's start value.\r\n    if (!Number.isFinite(clipDuration) || clipDuration <= 0) {\r\n        audioParam.value = finite(param.staticValue(audioParam.value));\r\n        return;\r\n    }\r\n\r\n    const base = startTime - elapsed; // map curve t=0 to here so seeks line up\r\n    const resolved = param.resolve(clipDuration, audioParam.value);\r\n    if (resolved.length === 0) return;\r\n\r\n    // Anchor the starting value at `startTime` (the seek point) so there is no\r\n    // jump from the param's default before the first scheduled segment runs.\r\n    const startValue = param.sampleAt(Math.max(0, elapsed), clipDuration, audioParam.value);\r\n    audioParam.setValueAtTime(finite(startValue), startTime);\r\n\r\n    for (const seg of resolved) {\r\n        const segStart = base + seg.startTime;\r\n        const segEnd = base + seg.endTime;\r\n        // Skip segments wholly in the past relative to the seek point, or any with a\r\n        // non-finite boundary (degenerate resolve) that would throw on schedule.\r\n        if (segEnd <= startTime || !Number.isFinite(segStart) || !Number.isFinite(segEnd)) continue;\r\n\r\n        const span = seg.endTime - seg.startTime;\r\n\r\n        if (span <= 0) {\r\n            audioParam.setValueAtTime(finite(seg.to), Math.max(segStart, startTime));\r\n            continue;\r\n        }\r\n\r\n        // How far through this segment the seek lands. A segment already under way\r\n        // carries on from the anchor rather than starting over from its `from` —\r\n        // the case a bus hits whenever its first clip enters partway through it.\r\n        const done = Math.max(0, (startTime - segStart) / span);\r\n        const curveStart = Math.max(segStart, startTime);\r\n\r\n        if (seg.isLinear) {\r\n            if (done === 0) audioParam.setValueAtTime(finite(seg.from), curveStart);\r\n            audioParam.linearRampToValueAtTime(finite(seg.to), segEnd);\r\n        } else {\r\n            const curveDur = segEnd - curveStart;\r\n            if (curveDur <= 0) continue;\r\n            const count = Math.max(2, Math.ceil(curveDur * CURVE_SAMPLES_PER_SECOND));\r\n            const samples = new Float32Array(count);\r\n            for (let i = 0; i < count; i++) {\r\n                const local = done + (1 - done) * (i / (count - 1));\r\n                samples[i] = finite(seg.from + (seg.to - seg.from) * seg.ease(local));\r\n            }\r\n            audioParam.setValueCurveAtTime(samples, curveStart, curveDur);\r\n        }\r\n    }\r\n}\r\n\r\n/**\r\n * Builds the Web Audio graph for a clip's filter chain, returning the node to\r\n * connect downstream. Shared by live playback ({@link WebAudioSink}) and\r\n * offline export (`mixAudio`) so both paths sound identical.\r\n *\r\n * Filters apply in array order (index 0 closest to the source). `speed` filters\r\n * are intentionally ignored here — they're realized via `source.playbackRate`\r\n * by the caller (and affect timeline scheduling), not as a graph node.\r\n *\r\n * `startTime` is the AudioContext time the source starts; `clipDuration` is the\r\n * clip-time length the curves run over (a clip's, or a bus's span); `elapsed` is\r\n * clip-seconds already played at a mid-clip seek (0 for a clean start, negative\r\n * before the span begins). These let curve-valued params schedule and align.\r\n *\r\n * Pass the already-created `source` so its `playbackRate` can be set by the\r\n * caller; this function only wires nodes *after* the source.\r\n */\r\nexport function buildAudioFilterGraph(\r\n    ctx: BaseAudioContext,\r\n    source: AudioNode,\r\n    filters: readonly AudioFilterItem[],\r\n    startTime: number = ctx.currentTime,\r\n    clipDuration: number = 0,\r\n    elapsed: number = 0,\r\n): AudioFilterGraph {\r\n    const nodes: AudioNode[] = [];\r\n    const oscillators: OscillatorNode[] = [];\r\n    let cursor: AudioNode = source;\r\n\r\n    const series = (node: AudioNode): void => {\r\n        cursor.connect(node);\r\n        nodes.push(node);\r\n        cursor = node;\r\n    };\r\n\r\n    const apply = (p: AudioParam, v: Param, clamp?: (n: number) => number): void =>\r\n        applyParam(ctx, p, v, startTime, clipDuration, elapsed, clamp);\r\n\r\n    for (const filter of filters) {\r\n        switch (filter.type) {\r\n            case \"gain\": {\r\n                const gain = ctx.createGain();\r\n                apply(gain.gain, filter.value);\r\n                series(gain);\r\n                break;\r\n            }\r\n            case \"highpass\":\r\n            case \"lowpass\": {\r\n                const biquad = ctx.createBiquadFilter();\r\n                biquad.type = filter.type === \"highpass\" ? \"highpass\" : \"lowpass\";\r\n                apply(biquad.frequency, filter.frequency);\r\n                if (filter.q !== undefined) apply(biquad.Q, filter.q);\r\n                series(biquad);\r\n                break;\r\n            }\r\n            case \"tremolo\": {\r\n                // Carrier gain whose value is modulated by an LFO. With depth d the\r\n                // carrier base = (1 - d/2) and the LFO swing gain = d/2. The clamp\r\n                // hook applies these transforms whether depth is scalar or a curve.\r\n                const carrier = ctx.createGain();\r\n                const lfo = ctx.createOscillator();\r\n                const lfoDepth = ctx.createGain();\r\n\r\n                apply(carrier.gain, filter.depth, (d) => 1 - clamp01(d) / 2);\r\n                apply(lfoDepth.gain, filter.depth, (d) => clamp01(d) / 2);\r\n                apply(lfo.frequency, filter.rate);\r\n\r\n                lfo.connect(lfoDepth);\r\n                lfoDepth.connect(carrier.gain);\r\n\r\n                nodes.push(lfoDepth);\r\n                oscillators.push(lfo);\r\n                series(carrier);\r\n                break;\r\n            }\r\n            case \"echo\": {\r\n                // Dry + wet split. The wet path is a delay with feedback; both sum\r\n                // into a merge node that becomes the new cursor.\r\n                const split = cursor;\r\n                const merge = ctx.createGain();\r\n\r\n                const dry = ctx.createGain();\r\n                apply(dry.gain, filter.mix ?? 0.5, (m) => 1 - clamp01(m));\r\n                split.connect(dry);\r\n                dry.connect(merge);\r\n\r\n                // Delay buffer must accommodate the max delay; size from the static value.\r\n                const maxDelay = staticOf(filter.delay);\r\n                const delay = ctx.createDelay(Math.max(1, maxDelay + 1));\r\n                apply(delay.delayTime, filter.delay);\r\n                const feedbackGain = ctx.createGain();\r\n                apply(feedbackGain.gain, filter.feedback, (f) => Math.max(0, Math.min(0.99, f)));\r\n                const wet = ctx.createGain();\r\n                apply(wet.gain, filter.mix ?? 0.5, (m) => clamp01(m));\r\n\r\n                split.connect(delay);\r\n                delay.connect(feedbackGain);\r\n                feedbackGain.connect(delay); // feedback loop\r\n                delay.connect(wet);\r\n                wet.connect(merge);\r\n\r\n                nodes.push(dry, delay, feedbackGain, wet, merge);\r\n                cursor = merge;\r\n                break;\r\n            }\r\n            case \"speed\":\r\n                // Realized via source.playbackRate by the caller; no graph node.\r\n                break;\r\n        }\r\n    }\r\n\r\n    return { output: cursor, nodes, oscillators };\r\n}\r\n\r\n/** Clamp to [0, 1]. */\r\nfunction clamp01(v: number): number {\r\n    return Math.max(0, Math.min(1, v));\r\n}\r\n\r\n/** The static/representative number of a param (number itself, or curve's t=0 value). */\r\nfunction staticOf(p: Param): number {\r\n    return isCurve(p) ? p.staticValue(0) : p;\r\n}\r\n\r\n/**\r\n * Net *constant* playback-rate multiplier from any scalar `speed` filters in the\r\n * chain (product of all), defaulting to 1. Curve-valued speeds contribute 1 here\r\n * and are scheduled separately by {@link applySpeedToPlaybackRate}. Mirrors\r\n * `Sound.effectiveSpeed()` so the renderer applies the same rate the core timeline\r\n * reserved time for.\r\n */\r\nexport function effectiveSpeed(filters: readonly AudioFilterItem[] | undefined): number {\r\n    if (!filters) return 1;\r\n    let speed = 1;\r\n    for (const f of filters) {\r\n        if (f.type === \"speed\" && !isCurve(f.value) && f.value > 0) speed *= f.value;\r\n    }\r\n    return speed;\r\n}\r\n\r\n/** The first curve-valued `speed` filter in the chain, if any. */\r\nexport function speedCurveOf(filters: readonly AudioFilterItem[] | undefined) {\r\n    if (!filters) return undefined;\r\n    for (const f of filters) {\r\n        if (f.type === \"speed\" && isCurve(f.value)) return f.value;\r\n    }\r\n    return undefined;\r\n}\r\n\r\n/**\r\n * Schedule a time-varying speed curve onto an `AudioBufferSourceNode.playbackRate`.\r\n *\r\n * The curve is authored over **source time** (rate as a function of how far through\r\n * the source we've played). `setValueAtTime`/`linearRampToValueAtTime` schedule in\r\n * **context time**, so we walk clip-time forward in small steps, convert each to a\r\n * source-time offset via the integral inverse, read the curve there, and lay down a\r\n * value point at the matching context time. `scalar` folds in any constant speed\r\n * filters (they scale the whole rate uniformly).\r\n *\r\n * `elapsed` is clip-seconds already played at a mid-clip seek, so the schedule\r\n * starts at the seek point and aligns the playbackRate to the clip's true position.\r\n */\r\nexport function applySpeedToPlaybackRate(\r\n    rate: AudioParam,\r\n    curve: ReturnType<typeof speedCurveOf>,\r\n    sourceLength: number,\r\n    clipDuration: number,\r\n    scalar: number,\r\n    startTime: number,\r\n    elapsed: number,\r\n): void {\r\n    if (!curve) return;\r\n    const step = 1 / CURVE_SAMPLES_PER_SECOND; // clip-time sampling step\r\n    const seekAt = Math.max(0, elapsed);\r\n    // Seed the value exactly at the seek point.\r\n    const seedSrc = sourceTimeAtClipElapsed(curve, sourceLength, seekAt);\r\n    rate.setValueAtTime(Math.max(1e-4, curve.sampleAt(seedSrc, sourceLength, 1) * scalar), startTime);\r\n\r\n    for (let s = seekAt + step; s <= clipDuration + step; s += step) {\r\n        const clampedElapsed = Math.min(s, clipDuration);\r\n        const srcT = sourceTimeAtClipElapsed(curve, sourceLength, clampedElapsed);\r\n        const value = Math.max(1e-4, curve.sampleAt(srcT, sourceLength, 1) * scalar);\r\n        const ctxTime = startTime + (clampedElapsed - seekAt);\r\n        rate.linearRampToValueAtTime(value, ctxTime);\r\n        if (clampedElapsed >= clipDuration) break;\r\n    }\r\n}\r\n", "import { 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", "import type { AudioRequest } from \"@motionscript/core\";\r\n\r\n/** The span of a decoded buffer a looping request repeats, in source seconds. */\r\nexport interface LoopRegion {\r\n    start: number;\r\n    end: number;\r\n}\r\n\r\n/**\r\n * The region a looping request repeats: from its in-point to its out-point, or to\r\n * the end of the buffer when it has none. `null` for a request that does not loop,\r\n * and for a degenerate region (an out-point at or before the in-point), where the\r\n * device falls back to wrapping the whole buffer.\r\n *\r\n * Shared by the live device and the offline mixer so a preview and an export loop\r\n * the same samples.\r\n */\r\nexport function loopRegionOf(request: AudioRequest, bufferDuration: number): LoopRegion | null {\r\n    if (!request.loop) return null;\r\n    const start = Math.min(Math.max(0, request.trimStart), bufferDuration);\r\n    const end = Math.min(request.trimEnd ?? bufferDuration, bufferDuration);\r\n    return end > start ? { start, end } : null;\r\n}\r\n\r\n/**\r\n * Configure `source` to play `request`'s loop: on or off, and over the trimmed\r\n * region rather than the whole buffer (`loopStart`/`loopEnd` default to the\r\n * whole buffer, which is how a trim used to be lost).\r\n */\r\nexport function applyLoop(source: AudioBufferSourceNode, request: AudioRequest): LoopRegion | null {\r\n    source.loop = request.loop;\r\n    const region = source.buffer ? loopRegionOf(request, source.buffer.duration) : null;\r\n    if (region) {\r\n        source.loopStart = region.start;\r\n        source.loopEnd = region.end;\r\n    }\r\n    return region;\r\n}\r\n\r\n/**\r\n * Fold a source offset into a loop region.\r\n *\r\n * A seek into the middle of a looping clip lands at `trimStart + elapsed`, which\r\n * can be many passes past the region's end; the device would not wrap an offset\r\n * that starts outside the region, so it is wrapped here.\r\n */\r\nexport function wrapIntoRegion(region: LoopRegion | null, offset: number): number {\r\n    if (!region || offset < region.end) return offset;\r\n    return region.start + ((offset - region.start) % (region.end - region.start));\r\n}\r\n", "import { 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"],
-  "mappings": 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-}