@motionscript/browser 0.0.0-stage → 0.1.0-alpha.10

package/dist/audio/effect-graph.js.map

@@ -0,0 +1 @@
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{\n    clipTimeAtSource,\n    isRamp,\n    resolveRamp,\n    sampleRamp,\n    sourceTimeAtClip,\n    staticValue,\n    transitionGainAt,\n    transitionSpans,\n    type AudioEffect,\n    type AudioRequest,\n    type AudioTransition,\n    type Param,\n    type Ramp,\n} from \"@motionscript/core\";\n\n/**\n * The Web Audio graph for one source's effect chain.\n *\n * `output` is the last node, to connect downstream; `nodes` are every node to\n * `disconnect()` on teardown; `oscillators` are tremolo LFOs the caller starts\n * and stops alongside the source.\n */\nexport interface AudioEffectGraph {\n    output: AudioNode;\n    nodes: AudioNode[];\n    oscillators: OscillatorNode[];\n}\n\n/** Sample points per second an eased segment or a transition is baked into: about every 5 ms. */\nconst CURVE_SAMPLES_PER_SECOND = 200;\n\n/**\n * Apply a {@link Param} to a Web Audio {@link AudioParam}: a number once, a\n * ramp as a schedule — linear segments as `linearRampToValueAtTime`, eased\n * ones baked into `setValueCurveAtTime`.\n *\n * `elapsed` is the clip seconds already played when a mid-clip seek starts the\n * source, so \"now\" reads the ramp at the clip's current position. `clamp` keeps\n * every scheduled value in a param's valid range.\n */\nfunction applyParam(\n    audioParam: AudioParam,\n    param: Param,\n    startTime: number,\n    clipDuration: number,\n    elapsed: number,\n    clamp: (v: number) => number = (v) => v,\n): void {\n    // The platform throws on a non-finite value.\n    const finite = (v: number): number => {\n        const c = clamp(v);\n        return Number.isFinite(c) ? c : audioParam.value;\n    };\n\n    if (!isRamp(param)) {\n        audioParam.value = finite(param);\n        return;\n    }\n    if (!Number.isFinite(clipDuration) || clipDuration <= 0) {\n        audioParam.value = finite(staticValue(param, audioParam.value));\n        return;\n    }\n\n    const keys = resolveRamp(param, clipDuration);\n    if (keys.length === 0) return;\n    const base = startTime - elapsed;\n    audioParam.setValueAtTime(finite(sampleRamp(param, Math.max(0, elapsed), clipDuration, audioParam.value)), startTime);\n\n    for (let i = 1; i < keys.length; i++) {\n        const previous = keys[i - 1];\n        const key = keys[i];\n        const segStart = base + previous.time;\n        const segEnd = base + key.time;\n        if (segEnd <= startTime) continue;\n\n        const span = key.time - previous.time;\n        if (span <= 0) {\n            audioParam.setValueAtTime(finite(key.value), Math.max(segEnd, startTime));\n            continue;\n        }\n\n        // A segment already under way when the source starts carries on from\n        // the value set at `startTime` rather than restarting from its own.\n        const done = Math.max(0, (startTime - segStart) / span);\n        const curveStart = Math.max(segStart, startTime);\n        if (key.linear) {\n            if (done === 0) audioParam.setValueAtTime(finite(previous.value), curveStart);\n            audioParam.linearRampToValueAtTime(finite(key.value), segEnd);\n        } else {\n            const curveDur = segEnd - curveStart;\n            const count = Math.max(2, Math.ceil(curveDur * CURVE_SAMPLES_PER_SECOND));\n            const samples = new Float32Array(count);\n            for (let n = 0; n < count; n++) {\n                const local = done + (1 - done) * (n / (count - 1));\n                samples[n] = finite(previous.value + (key.value - previous.value) * key.ease(local));\n            }\n            audioParam.setValueCurveAtTime(samples, curveStart, curveDur);\n        }\n    }\n}\n\n/**\n * Builds the Web Audio graph for a clip's effect chain, after an\n * already-created `source`. Shared by live playback ({@link WebAudioSink}) and\n * offline export (`mixAudio`) so both sound identical.\n *\n * `startTime` is the context time the source starts; `clipDuration` the clip\n * seconds ramps run over (a clip's, or a bus's span); `elapsed` the clip seconds\n * already played at a mid-clip seek (0 for a clean start, negative before the\n * span begins).\n */\nexport function buildAudioEffectGraph(\n    ctx: BaseAudioContext,\n    source: AudioNode,\n    effects: readonly AudioEffect[],\n    startTime: number = ctx.currentTime,\n    clipDuration: number = 0,\n    elapsed: number = 0,\n): AudioEffectGraph {\n    const nodes: AudioNode[] = [];\n    const oscillators: OscillatorNode[] = [];\n    let cursor: AudioNode = source;\n\n    const series = (node: AudioNode): void => {\n        cursor.connect(node);\n        nodes.push(node);\n        cursor = node;\n    };\n\n    const apply = (p: AudioParam, v: Param, clamp?: (n: number) => number): void =>\n        applyParam(p, v, startTime, clipDuration, elapsed, clamp);\n\n    for (const effect of effects) {\n        switch (effect.type) {\n            case \"gain\": {\n                const gain = ctx.createGain();\n                apply(gain.gain, effect.value);\n                series(gain);\n                break;\n            }\n            case \"highpass\":\n            case \"lowpass\": {\n                const biquad = ctx.createBiquadFilter();\n                biquad.type = effect.type;\n                apply(biquad.frequency, effect.frequency);\n                if (effect.q !== undefined) apply(biquad.Q, effect.q);\n                series(biquad);\n                break;\n            }\n            case \"tremolo\": {\n                // A carrier gain an LFO swings: with depth d the carrier sits at\n                // 1 - d/2 and the swing is d/2.\n                const carrier = ctx.createGain();\n                const lfo = ctx.createOscillator();\n                const lfoDepth = ctx.createGain();\n\n                apply(carrier.gain, effect.depth, (d) => 1 - clamp01(d) / 2);\n                apply(lfoDepth.gain, effect.depth, (d) => clamp01(d) / 2);\n                apply(lfo.frequency, effect.rate);\n\n                lfo.connect(lfoDepth);\n                lfoDepth.connect(carrier.gain);\n\n                nodes.push(lfoDepth);\n                oscillators.push(lfo);\n                series(carrier);\n                break;\n            }\n            case \"echo\": {\n                const split = cursor;\n                const merge = ctx.createGain();\n\n                const dry = ctx.createGain();\n                apply(dry.gain, effect.mix ?? 0.5, (m) => 1 - clamp01(m));\n                split.connect(dry);\n                dry.connect(merge);\n\n                // The delay line is sized once, so it has to hold the ramp's longest delay.\n                const delay = ctx.createDelay(Math.max(1, maxOf(effect.delay) + 1));\n                apply(delay.delayTime, effect.delay);\n                const feedbackGain = ctx.createGain();\n                apply(feedbackGain.gain, effect.feedback, (f) => Math.max(0, Math.min(0.99, f)));\n                const wet = ctx.createGain();\n                apply(wet.gain, effect.mix ?? 0.5, (m) => clamp01(m));\n\n                split.connect(delay);\n                delay.connect(feedbackGain);\n                feedbackGain.connect(delay);\n                delay.connect(wet);\n                wet.connect(merge);\n\n                nodes.push(dry, delay, feedbackGain, wet, merge);\n                cursor = merge;\n                break;\n            }\n        }\n    }\n\n    return { output: cursor, nodes, oscillators };\n}\n\nfunction clamp01(v: number): number {\n    return Math.max(0, Math.min(1, v));\n}\n\nfunction maxOf(p: Param): number {\n    return isRamp(p) ? Math.max(0, ...p.keys.map((k) => k.value)) : p;\n}\n\n/**\n * Schedule `transitionIn`/`transitionOut` onto a gain over a span `length`\n * seconds long, baked like an eased ramp. Between the two it holds at 1.\n */\nexport function applyTransitions(\n    gain: AudioParam,\n    length: number,\n    transitionIn: AudioTransition | undefined,\n    transitionOut: AudioTransition | undefined,\n    startTime: number,\n    elapsed: number,\n): void {\n    if (!transitionIn && !transitionOut) return;\n    const seek = Math.max(0, elapsed);\n    const at = (t: number): number => transitionGainAt(t, length, transitionIn, transitionOut);\n    gain.setValueAtTime(at(seek), startTime);\n\n    const spans = transitionSpans(length, transitionIn, transitionOut);\n    const windows: [number, number][] = [];\n    if (spans.in > 0) windows.push([0, spans.in]);\n    if (spans.out > 0 && Number.isFinite(length)) windows.push([length - spans.out, length]);\n    const base = startTime - elapsed;\n    for (const [from, to] of windows) {\n        const begin = Math.max(from, seek);\n        const duration = to - begin;\n        if (duration <= 0) continue;\n        const count = Math.max(2, Math.ceil(duration * CURVE_SAMPLES_PER_SECOND));\n        const samples = new Float32Array(count);\n        for (let n = 0; n < count; n++) samples[n] = at(begin + duration * (n / (count - 1)));\n        gain.setValueCurveAtTime(samples, base + begin, duration);\n    }\n}\n\n/** How one request plays its source: the rate, and the lengths its ramps and transitions run over. */\nexport interface Playback {\n    /** The constant `playbackRate`; 1 under a ramp, which is scheduled instead. */\n    rate: number;\n    speedRamp?: Ramp;\n    /** Source seconds a speed ramp is placed on. */\n    sourceLength: number;\n    /** Clip seconds the volume and effect ramps run over. */\n    clipDuration: number;\n    /** What the transitions run over: the clip's length, unknown for an open one. */\n    transitionLength: number;\n}\n\nexport function playbackOf(req: AudioRequest, bufferDuration: number): Playback {\n    const speed = req.speed ?? 1;\n    const speedRamp = isRamp(speed) ? speed : undefined;\n    const rate = speedRamp || !(Number.isFinite(speed) && (speed as number) > 0) ? 1 : (speed as number);\n    // Placed on the same length core sized the request with, not the decoded buffer.\n    const sourceLength = req.sourceLength ?? Math.max(0, bufferDuration - req.trimStart);\n    const bounded = Number.isFinite(req.endAt);\n    const clipDuration = bounded ? Math.max(0, req.endAt - req.startAt) : clipTimeAtSource(speed, sourceLength);\n    return { rate, speedRamp, sourceLength, clipDuration, transitionLength: bounded ? clipDuration : Infinity };\n}\n\n/** Source seconds after `trimStart` that `clipElapsed` clip seconds have played. */\nexport function sourceOffsetOf(playback: Playback, clipElapsed: number): number {\n    return playback.speedRamp\n        ? sourceTimeAtClip(playback.speedRamp, playback.sourceLength, clipElapsed)\n        : clipElapsed * playback.rate;\n}\n\n/**\n * A request's level, then its transitions, after `from`: the gains every\n * request plays through on its way to its bus.\n */\nexport function buildLevel(\n    ctx: BaseAudioContext,\n    from: AudioNode,\n    req: AudioRequest,\n    playback: Playback,\n    startTime: number,\n    elapsed: number,\n): { output: AudioNode; nodes: AudioNode[] } {\n    const level = ctx.createGain();\n    applyParam(level.gain, req.volume ?? 1, startTime, playback.clipDuration, elapsed);\n    from.connect(level);\n    if (!req.transitionIn && !req.transitionOut) return { output: level, nodes: [level] };\n    const transition = ctx.createGain();\n    applyTransitions(transition.gain, playback.transitionLength, req.transitionIn, req.transitionOut, startTime, elapsed);\n    level.connect(transition);\n    return { output: transition, nodes: [level, transition] };\n}\n\n/**\n * Schedule a speed ramp onto an `AudioBufferSourceNode.playbackRate`.\n *\n * The ramp runs over **source** seconds and Web Audio schedules in context\n * seconds, so this walks clip time in small steps, converts each to a source\n * offset through the integral inverse, and lays down the rate there. `elapsed`\n * is clip seconds already played at a mid-clip seek.\n */\nexport function applySpeedToPlaybackRate(\n    rate: AudioParam,\n    speed: Ramp | undefined,\n    sourceLength: number,\n    clipDuration: number,\n    startTime: number,\n    elapsed: number,\n): void {\n    if (!speed) return;\n    const step = 1 / CURVE_SAMPLES_PER_SECOND;\n    const seekAt = Math.max(0, elapsed);\n    const valueAt = (clipElapsed: number): number =>\n        Math.max(1e-4, sampleRamp(speed, sourceTimeAtClip(speed, sourceLength, clipElapsed), sourceLength, 1));\n    rate.setValueAtTime(valueAt(seekAt), startTime);\n\n    for (let s = seekAt + step; s <= clipDuration + step; s += step) {\n        const clamped = Math.min(s, clipDuration);\n        rate.linearRampToValueAtTime(valueAt(clamped), startTime + (clamped - seekAt));\n        if (clamped >= clipDuration) break;\n    }\n}\n"]}

package/dist/audio/index.d.ts

@@ -0,0 +1,5 @@
+export { mixAudio, encodeWav, WebAudioMixer, DEFAULT_SAMPLE_RATE } from "./mixer";
+export type { MixAudioOptions } from "./mixer";
+export { WebAudioSink, WebAudioSink as WebAudioPlayer } from "./player";
+export { AudioBusGraph } from "./bus-graph";
+//# sourceMappingURL=index.d.ts.map

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

@@ -0,0 +1 @@
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package/dist/audio/index.js

@@ -0,0 +1,11 @@
+// @motionscript/browser/audio — stacked-audio playback and mixdown, with no
+// renderer attached.
+//
+// A separate entry point from the package barrel on purpose: nothing here draws,
+// so a host building an audio timeline should not load the renderer.
+export { mixAudio, encodeWav, WebAudioMixer, DEFAULT_SAMPLE_RATE } from "./mixer";
+export { WebAudioSink, WebAudioSink as WebAudioPlayer } from "./player";
+// The bus layer both of those share. Exported so a host writing its own device
+// or mixer builds the same chains, rather than a second set that drifts.
+export { AudioBusGraph } from "./bus-graph";
+//# sourceMappingURL=index.js.map

package/dist/audio/index.js.map

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+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/audio/index.ts"],"names":[],"mappings":"AAAA,4EAA4E;AAC5E,qBAAqB;AACrB,EAAE;AACF,iFAAiF;AACjF,qEAAqE;AAErE,OAAO,EAAE,QAAQ,EAAE,SAAS,EAAE,aAAa,EAAE,mBAAmB,EAAE,MAAM,SAAS,CAAC;AAGlF,OAAO,EAAE,YAAY,EAAE,YAAY,IAAI,cAAc,EAAE,MAAM,UAAU,CAAC;AAExE,+EAA+E;AAC/E,yEAAyE;AACzE,OAAO,EAAE,aAAa,EAAE,MAAM,aAAa,CAAC","sourcesContent":["// @motionscript/browser/audio — stacked-audio playback and mixdown, with no\n// renderer attached.\n//\n// A separate entry point from the package barrel on purpose: nothing here draws,\n// so a host building an audio timeline should not load the renderer.\n\nexport { mixAudio, encodeWav, WebAudioMixer, DEFAULT_SAMPLE_RATE } from \"./mixer\";\nexport type { MixAudioOptions } from \"./mixer\";\n\nexport { WebAudioSink, WebAudioSink as WebAudioPlayer } from \"./player\";\n\n// The bus layer both of those share. Exported so a host writing its own device\n// or mixer builds the same chains, rather than a second set that drifts.\nexport { AudioBusGraph } from \"./bus-graph\";\n"]}

package/dist/audio/loop-region.d.ts

@@ -0,0 +1,31 @@
+import type { AudioRequest } from "@motionscript/core";
+/** The span of a decoded buffer a looping request repeats, in source seconds. */
+export interface LoopRegion {
+    start: number;
+    end: number;
+}
+/**
+ * The region a looping request repeats: from its in-point to its out-point, or to
+ * the end of the buffer when it has none. `null` for a request that does not loop,
+ * and for a degenerate region (an out-point at or before the in-point), where the
+ * device falls back to wrapping the whole buffer.
+ *
+ * Shared by the live device and the offline mixer so a preview and an export loop
+ * the same samples.
+ */
+export declare function loopRegionOf(request: AudioRequest, bufferDuration: number): LoopRegion | null;
+/**
+ * Configure `source` to play `request`'s loop: on or off, and over the trimmed
+ * region rather than the whole buffer (`loopStart`/`loopEnd` default to the
+ * whole buffer, which is how a trim used to be lost).
+ */
+export declare function applyLoop(source: AudioBufferSourceNode, request: AudioRequest): LoopRegion | null;
+/**
+ * Fold a source offset into a loop region.
+ *
+ * A seek into the middle of a looping clip lands at `trimStart + elapsed`, which
+ * can be many passes past the region's end; the device would not wrap an offset
+ * that starts outside the region, so it is wrapped here.
+ */
+export declare function wrapIntoRegion(region: LoopRegion | null, offset: number): number;
+//# sourceMappingURL=loop-region.d.ts.map

package/dist/audio/loop-region.d.ts.map

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package/dist/audio/loop-region.js

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

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

@@ -0,0 +1 @@
+{"version":3,"file":"loop-region.js","sourceRoot":"","sources":["../../src/audio/loop-region.ts"],"names":[],"mappings":"AAQA;;;;;;;;GAQG;AACH,MAAM,UAAU,YAAY,CAAC,OAAqB,EAAE,cAAsB;IACtE,IAAI,CAAC,OAAO,CAAC,IAAI;QAAE,OAAO,IAAI,CAAC;IAC/B,MAAM,KAAK,GAAG,IAAI,CAAC,GAAG,CAAC,IAAI,CAAC,GAAG,CAAC,CAAC,EAAE,OAAO,CAAC,SAAS,CAAC,EAAE,cAAc,CAAC,CAAC;IACvE,MAAM,GAAG,GAAG,IAAI,CAAC,GAAG,CAAC,OAAO,CAAC,OAAO,IAAI,cAAc,EAAE,cAAc,CAAC,CAAC;IACxE,OAAO,GAAG,GAAG,KAAK,CAAC,CAAC,CAAC,EAAE,KAAK,EAAE,GAAG,EAAE,CAAC,CAAC,CAAC,IAAI,CAAC;AAC/C,CAAC;AAED;;;;GAIG;AACH,MAAM,UAAU,SAAS,CAAC,MAA6B,EAAE,OAAqB;IAC1E,MAAM,CAAC,IAAI,GAAG,OAAO,CAAC,IAAI,CAAC;IAC3B,MAAM,MAAM,GAAG,MAAM,CAAC,MAAM,CAAC,CAAC,CAAC,YAAY,CAAC,OAAO,EAAE,MAAM,CAAC,MAAM,CAAC,QAAQ,CAAC,CAAC,CAAC,CAAC,IAAI,CAAC;IACpF,IAAI,MAAM,EAAE,CAAC;QACT,MAAM,CAAC,SAAS,GAAG,MAAM,CAAC,KAAK,CAAC;QAChC,MAAM,CAAC,OAAO,GAAG,MAAM,CAAC,GAAG,CAAC;IAChC,CAAC;IACD,OAAO,MAAM,CAAC;AAClB,CAAC;AAED;;;;;;GAMG;AACH,MAAM,UAAU,cAAc,CAAC,MAAyB,EAAE,MAAc;IACpE,IAAI,CAAC,MAAM,IAAI,MAAM,GAAG,MAAM,CAAC,GAAG;QAAE,OAAO,MAAM,CAAC;IAClD,OAAO,MAAM,CAAC,KAAK,GAAG,CAAC,CAAC,MAAM,GAAG,MAAM,CAAC,KAAK,CAAC,GAAG,CAAC,MAAM,CAAC,GAAG,GAAG,MAAM,CAAC,KAAK,CAAC,CAAC,CAAC;AAClF,CAAC","sourcesContent":["import type { AudioRequest } from \"@motionscript/core\";\n\n/** The span of a decoded buffer a looping request repeats, in source seconds. */\nexport interface LoopRegion {\n    start: number;\n    end: number;\n}\n\n/**\n * The region a looping request repeats: from its in-point to its out-point, or to\n * the end of the buffer when it has none. `null` for a request that does not loop,\n * and for a degenerate region (an out-point at or before the in-point), where the\n * device falls back to wrapping the whole buffer.\n *\n * Shared by the live device and the offline mixer so a preview and an export loop\n * the same samples.\n */\nexport function loopRegionOf(request: AudioRequest, bufferDuration: number): LoopRegion | null {\n    if (!request.loop) return null;\n    const start = Math.min(Math.max(0, request.trimStart), bufferDuration);\n    const end = Math.min(request.trimEnd ?? bufferDuration, bufferDuration);\n    return end > start ? { start, end } : null;\n}\n\n/**\n * Configure `source` to play `request`'s loop: on or off, and over the trimmed\n * region rather than the whole buffer (`loopStart`/`loopEnd` default to the\n * whole buffer, which is how a trim used to be lost).\n */\nexport function applyLoop(source: AudioBufferSourceNode, request: AudioRequest): LoopRegion | null {\n    source.loop = request.loop;\n    const region = source.buffer ? loopRegionOf(request, source.buffer.duration) : null;\n    if (region) {\n        source.loopStart = region.start;\n        source.loopEnd = region.end;\n    }\n    return region;\n}\n\n/**\n * Fold a source offset into a loop region.\n *\n * A seek into the middle of a looping clip lands at `trimStart + elapsed`, which\n * can be many passes past the region's end; the device would not wrap an offset\n * that starts outside the region, so it is wrapped here.\n */\nexport function wrapIntoRegion(region: LoopRegion | null, offset: number): number {\n    if (!region || offset < region.end) return offset;\n    return region.start + ((offset - region.start) % (region.end - region.start));\n}\n"]}

package/dist/audio/mixer.d.ts

@@ -0,0 +1,59 @@
+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, 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;
+}
+/**
+ * Render every scheduled request into one timeline-length buffer.
+ *
+ * An `OfflineAudioContext` graph — gain per request, looping/trimming, and the
+ * same filter chain the live {@link WebAudioSink} builds — rendered in one
+ * pass rather than scheduled per frame.
+ *
+ * Shared by the video exporter and by `mixTimelineAudio`, deliberately: a split
+ * export and a whole one have to be the same mix, and two implementations of
+ * "stack these clips" would eventually disagree.
+ *
+ * Curve-valued filter params and `speed` curves schedule here exactly as they do
+ * in {@link WebAudioSink.playBuffer}, which is the reference implementation —
+ * the two must be read side by side when either changes.
+ *
+ * One deliberate difference: this schedules at absolute context times, so clip
+ * starts are sample-accurate here and quantized to `1/fps` in live playback,
+ * which starts sources at `currentTime` on a clock tick.
+ */
+export declare function mixAudio(requests: readonly ScheduledAudioRequest[], totalDuration: number, options?: MixAudioOptions): Promise<AudioBuffer | null>;
+/**
+ * Adapter presenting {@link mixAudio} as the exporter's injected mixer seam.
+ *
+ * The sample rate is a constructor argument rather than a third `mix` parameter
+ * because {@link AudioMixer} — the seam `renderTimeline` calls through — takes
+ * exactly two, so anything passed as a third would be silently dropped. Fixing
+ * it here rather than widening the interface keeps the rate a property of *this*
+ * mixer, which is whose business it is.
+ */
+export declare class WebAudioMixer implements AudioMixer<AudioBuffer> {
+    private readonly sampleRate;
+    private readonly catalog?;
+    constructor(sampleRate?: number, catalog?: AssetCatalog | undefined);
+    mix(requests: readonly ScheduledAudioRequest[], totalDuration: number): Promise<AudioBuffer | null>;
+}
+/**
+ * Encode an `AudioBuffer` as a 16-bit PCM WAV.
+ *
+ * The only other encoder in this package is the AAC track inside the mp4 muxer,
+ * which is reachable solely by exporting a video — so without this a mixdown has
+ * no file representation at all. WAV because it needs no codec support and is
+ * lossless; a host wanting mp3/aac can re-encode from here.
+ */
+export declare function encodeWav(buffer: AudioBuffer): Uint8Array<ArrayBuffer>;
+//# sourceMappingURL=mixer.d.ts.map

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

@@ -0,0 +1 @@
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package/dist/audio/mixer.js

@@ -0,0 +1,192 @@
+import { fetchAssetBytes, localAssetBytes } from "../asset-bytes";
+import { AudioBusGraph } from "./bus-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. */
+function audioUrl(src, catalog) {
+    if (!catalog)
+        return src;
+    try {
+        return catalog.getAudioMeta(src).src || src;
+    }
+    catch {
+        try {
+            return catalog.getVideoMeta(src).src || src;
+        }
+        catch {
+            return src;
+        }
+    }
+}
+/**
+ * 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 blob = (await localAssetBytes(src)) ?? (await fetchAssetBytes(src));
+    return ctx.decodeAudioData(await blob.arrayBuffer());
+}
+/**
+ * Render every scheduled request into one timeline-length buffer.
+ *
+ * An `OfflineAudioContext` graph — gain per request, looping/trimming, and the
+ * same filter chain the live {@link WebAudioSink} builds — rendered in one
+ * pass rather than scheduled per frame.
+ *
+ * Shared by the video exporter and by `mixTimelineAudio`, deliberately: a split
+ * export and a whole one have to be the same mix, and two implementations of
+ * "stack these clips" would eventually disagree.
+ *
+ * Curve-valued filter params and `speed` curves schedule here exactly as they do
+ * in {@link WebAudioSink.playBuffer}, which is the reference implementation —
+ * the two must be read side by side when either changes.
+ *
+ * One deliberate difference: this schedules at absolute context times, so clip
+ * starts are sample-accurate here and quantized to `1/fps` in live playback,
+ * which starts sources at `currentTime` on a clock tick.
+ */
+export async function mixAudio(requests, totalDuration, options = {}) {
+    if (requests.length === 0)
+        return null;
+    const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE;
+    const frames = Math.ceil(totalDuration * sampleRate);
+    if (frames <= 0)
+        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) => {
+        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
+    // export that flattened buses into per-clip processing would not match what
+    // the preview sounded like.
+    const buses = new AudioBusGraph(mixCtx, mixCtx.destination);
+    for (const { request: req, globalOffset } of requests) {
+        const srcBuffer = decoded.get(req.src);
+        if (!srcBuffer)
+            continue;
+        const globalStart = globalOffset + req.startAt;
+        // `endAt` is `Infinity` for an open clip; either way the timeline bounds it.
+        const globalEnd = Math.min(globalOffset + req.endAt, totalDuration);
+        const playDuration = globalEnd - globalStart;
+        if (playDuration <= 0)
+            continue;
+        const source = mixCtx.createBufferSource();
+        source.buffer = srcBuffer;
+        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);
+    }
+    return mixCtx.startRendering();
+}
+/**
+ * Adapter presenting {@link mixAudio} as the exporter's injected mixer seam.
+ *
+ * The sample rate is a constructor argument rather than a third `mix` parameter
+ * because {@link AudioMixer} — the seam `renderTimeline` calls through — takes
+ * exactly two, so anything passed as a third would be silently dropped. Fixing
+ * it here rather than widening the interface keeps the rate a property of *this*
+ * mixer, which is whose business it is.
+ */
+export class WebAudioMixer {
+    sampleRate;
+    catalog;
+    constructor(sampleRate = DEFAULT_SAMPLE_RATE, catalog) {
+        this.sampleRate = sampleRate;
+        this.catalog = catalog;
+    }
+    mix(requests, totalDuration) {
+        return mixAudio(requests, totalDuration, {
+            sampleRate: this.sampleRate,
+            catalog: this.catalog,
+        });
+    }
+}
+/**
+ * Encode an `AudioBuffer` as a 16-bit PCM WAV.
+ *
+ * The only other encoder in this package is the AAC track inside the mp4 muxer,
+ * which is reachable solely by exporting a video — so without this a mixdown has
+ * no file representation at all. WAV because it needs no codec support and is
+ * lossless; a host wanting mp3/aac can re-encode from here.
+ */
+export function encodeWav(buffer) {
+    const channelCount = buffer.numberOfChannels;
+    const frameCount = buffer.length;
+    const bytesPerSample = 2;
+    const blockAlign = channelCount * bytesPerSample;
+    const dataBytes = frameCount * blockAlign;
+    const out = new ArrayBuffer(44 + dataBytes);
+    const view = new DataView(out);
+    const ascii = (offset, text) => {
+        for (let i = 0; i < text.length; i++)
+            view.setUint8(offset + i, text.charCodeAt(i));
+    };
+    ascii(0, "RIFF");
+    view.setUint32(4, 36 + dataBytes, true);
+    ascii(8, "WAVE");
+    ascii(12, "fmt ");
+    view.setUint32(16, 16, true); // PCM header length
+    view.setUint16(20, 1, true); // format: PCM
+    view.setUint16(22, channelCount, true);
+    view.setUint32(24, buffer.sampleRate, true);
+    view.setUint32(28, buffer.sampleRate * blockAlign, true);
+    view.setUint16(32, blockAlign, true);
+    view.setUint16(34, 8 * bytesPerSample, true);
+    ascii(36, "data");
+    view.setUint32(40, dataBytes, true);
+    // Pull each channel out once — getChannelData is a live view, but calling it
+    // per sample inside the interleave loop is measurably slower.
+    const channels = [];
+    for (let c = 0; c < channelCount; c++)
+        channels.push(buffer.getChannelData(c));
+    let offset = 44;
+    for (let frame = 0; frame < frameCount; frame++) {
+        for (let c = 0; c < channelCount; c++) {
+            // Clamp before scaling: the mix bus can exceed [-1,1] where clips
+            // overlap, and letting that wrap turns a loud passage into noise.
+            const sample = Math.max(-1, Math.min(1, channels[c][frame]));
+            view.setInt16(offset, sample < 0 ? sample * 0x8000 : sample * 0x7fff, true);
+            offset += bytesPerSample;
+        }
+    }
+    return new Uint8Array(out);
+}
+//# sourceMappingURL=mixer.js.map

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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

@@ -0,0 +1,55 @@
+import { AudioSink, AudioRequest } from "@motionscript/core";
+/**
+ * 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 `clipTime`, so playback follows scrubbing and seeks.
+ */
+export declare class WebAudioSink extends AudioSink {
+    private readonly borrowed;
+    private _context;
+    private bufferCache;
+    private decoding;
+    private active;
+    private scheduled;
+    private disposed;
+    /** 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
+     * suspended and no audio is heard.
+     */
+    unlock(): void;
+    has(src: string): boolean;
+    append(src: string, data: ArrayBuffer): Promise<void>;
+    /** 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). */
+    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 `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 `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;
+}
+//# sourceMappingURL=player.d.ts.map

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

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