@motionscript/browser 0.0.0-stage → 0.1.0-alpha.0

package/dist/audio/filter-graph.js

@@ -0,0 +1,235 @@
+import { isCurve, sourceTimeAtSceneElapsed } from "@motionscript/core";
+/**
+ * How many sample points per second to bake an eased curve segment into for
+ * `setValueCurveAtTime`. 200/s (≈ every 5ms) is smooth to the ear and cheap.
+ */
+const CURVE_SAMPLES_PER_SECOND = 200;
+/**
+ * Apply a (possibly time-varying) {@link Param} to a Web Audio {@link AudioParam}.
+ *
+ * - **number** → set once (today's behavior; zero scheduling overhead).
+ * - **Curve** → resolve against `clipDuration` and schedule each segment:
+ *   - linear segments via `setValueAtTime` + `linearRampToValueAtTime`,
+ *   - eased segments baked into a `Float32Array` via `setValueCurveAtTime`.
+ *
+ * `elapsed` (scene-seconds already played when a mid-clip seek starts the source)
+ * shifts the schedule back by `elapsed` so "now" reads the curve's value at the
+ * clip's *current* position, not its start. A `clamp(v)` hook lets a param keep a
+ * valid range (e.g. echo mix 0–1) at every sampled point.
+ */
+function applyParam(ctx, audioParam, param, startTime, clipDuration, elapsed, clamp = (v) => v) {
+    // Coerce to a finite value before touching an AudioParam — the platform throws
+    // on NaN/Infinity. Falls back to the current param value (never undefined here).
+    const finite = (v) => {
+        const c = clamp(v);
+        return Number.isFinite(c) ? c : audioParam.value;
+    };
+    if (!isCurve(param)) {
+        audioParam.value = finite(param);
+        return;
+    }
+    // A non-finite clip length yields no usable schedule; hold the curve's start value.
+    if (!Number.isFinite(clipDuration) || clipDuration <= 0) {
+        audioParam.value = finite(param.staticValue(audioParam.value));
+        return;
+    }
+    const base = startTime - elapsed; // map curve t=0 to here so seeks line up
+    const resolved = param.resolve(clipDuration, audioParam.value);
+    if (resolved.length === 0)
+        return;
+    // Anchor the starting value at `startTime` (the seek point) so there is no
+    // jump from the param's default before the first scheduled segment runs.
+    const startValue = param.sampleAt(Math.max(0, elapsed), clipDuration, audioParam.value);
+    audioParam.setValueAtTime(finite(startValue), startTime);
+    for (const seg of resolved) {
+        const segStart = base + seg.startTime;
+        const segEnd = base + seg.endTime;
+        // Skip segments wholly in the past relative to the seek point, or any with a
+        // non-finite boundary (degenerate resolve) that would throw on schedule.
+        if (segEnd <= startTime || !Number.isFinite(segStart) || !Number.isFinite(segEnd))
+            continue;
+        const span = seg.endTime - seg.startTime;
+        if (span <= 0) {
+            audioParam.setValueAtTime(finite(seg.to), Math.max(segStart, startTime));
+            continue;
+        }
+        if (seg.isLinear) {
+            audioParam.setValueAtTime(finite(seg.from), Math.max(segStart, startTime));
+            audioParam.linearRampToValueAtTime(finite(seg.to), segEnd);
+        }
+        else {
+            const count = Math.max(2, Math.ceil(span * CURVE_SAMPLES_PER_SECOND));
+            const samples = new Float32Array(count);
+            for (let i = 0; i < count; i++) {
+                const local = i / (count - 1);
+                samples[i] = finite(seg.from + (seg.to - seg.from) * seg.ease(local));
+            }
+            // setValueCurveAtTime can't start in the past; clamp the start forward.
+            const curveStart = Math.max(segStart, startTime);
+            const curveDur = segEnd - curveStart;
+            if (curveDur > 0)
+                audioParam.setValueCurveAtTime(samples, curveStart, curveDur);
+        }
+    }
+}
+/**
+ * Builds the Web Audio graph for a clip's filter chain, returning the node to
+ * connect downstream. Shared by live playback ({@link WebAudioSink}) and
+ * offline export (`mixAudio`) so both paths sound identical.
+ *
+ * Filters apply in array order (index 0 closest to the source). `speed` filters
+ * are intentionally ignored here — they're realized via `source.playbackRate`
+ * by the caller (and affect timeline scheduling), not as a graph node.
+ *
+ * `startTime` is the AudioContext time the source starts; `clipDuration` is the
+ * clip's scene-time length; `elapsed` is scene-seconds already played at a mid-clip
+ * seek (0 for a clean start). These let curve-valued params schedule and align.
+ *
+ * Pass the already-created `source` so its `playbackRate` can be set by the
+ * caller; this function only wires nodes *after* the source.
+ */
+export function buildAudioFilterGraph(ctx, source, filters, startTime = ctx.currentTime, clipDuration = 0, elapsed = 0) {
+    const nodes = [];
+    const oscillators = [];
+    let cursor = source;
+    const series = (node) => {
+        cursor.connect(node);
+        nodes.push(node);
+        cursor = node;
+    };
+    const apply = (p, v, clamp) => applyParam(ctx, p, v, startTime, clipDuration, elapsed, clamp);
+    for (const filter of filters) {
+        switch (filter.type) {
+            case "gain": {
+                const gain = ctx.createGain();
+                apply(gain.gain, filter.value);
+                series(gain);
+                break;
+            }
+            case "highpass":
+            case "lowpass": {
+                const biquad = ctx.createBiquadFilter();
+                biquad.type = filter.type === "highpass" ? "highpass" : "lowpass";
+                apply(biquad.frequency, filter.frequency);
+                if (filter.q !== undefined)
+                    apply(biquad.Q, filter.q);
+                series(biquad);
+                break;
+            }
+            case "tremolo": {
+                // Carrier gain whose value is modulated by an LFO. With depth d the
+                // carrier base = (1 - d/2) and the LFO swing gain = d/2. The clamp
+                // hook applies these transforms whether depth is scalar or a curve.
+                const carrier = ctx.createGain();
+                const lfo = ctx.createOscillator();
+                const lfoDepth = ctx.createGain();
+                apply(carrier.gain, filter.depth, (d) => 1 - clamp01(d) / 2);
+                apply(lfoDepth.gain, filter.depth, (d) => clamp01(d) / 2);
+                apply(lfo.frequency, filter.rate);
+                lfo.connect(lfoDepth);
+                lfoDepth.connect(carrier.gain);
+                nodes.push(lfoDepth);
+                oscillators.push(lfo);
+                series(carrier);
+                break;
+            }
+            case "echo": {
+                // Dry + wet split. The wet path is a delay with feedback; both sum
+                // into a merge node that becomes the new cursor.
+                const split = cursor;
+                const merge = ctx.createGain();
+                const dry = ctx.createGain();
+                apply(dry.gain, filter.mix ?? 0.5, (m) => 1 - clamp01(m));
+                split.connect(dry);
+                dry.connect(merge);
+                // Delay buffer must accommodate the max delay; size from the static value.
+                const maxDelay = staticOf(filter.delay);
+                const delay = ctx.createDelay(Math.max(1, maxDelay + 1));
+                apply(delay.delayTime, filter.delay);
+                const feedbackGain = ctx.createGain();
+                apply(feedbackGain.gain, filter.feedback, (f) => Math.max(0, Math.min(0.99, f)));
+                const wet = ctx.createGain();
+                apply(wet.gain, filter.mix ?? 0.5, (m) => clamp01(m));
+                split.connect(delay);
+                delay.connect(feedbackGain);
+                feedbackGain.connect(delay); // feedback loop
+                delay.connect(wet);
+                wet.connect(merge);
+                nodes.push(dry, delay, feedbackGain, wet, merge);
+                cursor = merge;
+                break;
+            }
+            case "speed":
+                // Realized via source.playbackRate by the caller; no graph node.
+                break;
+        }
+    }
+    return { output: cursor, nodes, oscillators };
+}
+/** Clamp to [0, 1]. */
+function clamp01(v) {
+    return Math.max(0, Math.min(1, v));
+}
+/** The static/representative number of a param (number itself, or curve's t=0 value). */
+function staticOf(p) {
+    return isCurve(p) ? p.staticValue(0) : p;
+}
+/**
+ * Net *constant* playback-rate multiplier from any scalar `speed` filters in the
+ * chain (product of all), defaulting to 1. Curve-valued speeds contribute 1 here
+ * and are scheduled separately by {@link applySpeedToPlaybackRate}. Mirrors
+ * `Sound.effectiveSpeed()` so the renderer applies the same rate the core timeline
+ * reserved time for.
+ */
+export function effectiveSpeed(filters) {
+    if (!filters)
+        return 1;
+    let speed = 1;
+    for (const f of filters) {
+        if (f.type === "speed" && !isCurve(f.value) && f.value > 0)
+            speed *= f.value;
+    }
+    return speed;
+}
+/** The first curve-valued `speed` filter in the chain, if any. */
+export function speedCurveOf(filters) {
+    if (!filters)
+        return undefined;
+    for (const f of filters) {
+        if (f.type === "speed" && isCurve(f.value))
+            return f.value;
+    }
+    return undefined;
+}
+/**
+ * Schedule a time-varying speed curve onto an `AudioBufferSourceNode.playbackRate`.
+ *
+ * The curve is authored over **source time** (rate as a function of how far through
+ * the source we've played). `setValueAtTime`/`linearRampToValueAtTime` schedule in
+ * **context time**, so we walk scene-time forward in small steps, convert each to a
+ * source-time offset via the integral inverse, read the curve there, and lay down a
+ * value point at the matching context time. `scalar` folds in any constant speed
+ * filters (they scale the whole rate uniformly).
+ *
+ * `elapsed` is scene-seconds already played at a mid-clip seek, so the schedule
+ * starts at the seek point and aligns the playbackRate to the clip's true position.
+ */
+export function applySpeedToPlaybackRate(rate, curve, sourceLength, sceneDuration, scalar, startTime, elapsed) {
+    if (!curve)
+        return;
+    const step = 1 / CURVE_SAMPLES_PER_SECOND; // scene-time sampling step
+    let scene = Math.max(0, elapsed);
+    // Seed the value exactly at the seek point.
+    const seedSrc = sourceTimeAtSceneElapsed(curve, sourceLength, scene);
+    rate.setValueAtTime(Math.max(1e-4, curve.sampleAt(seedSrc, sourceLength, 1) * scalar), startTime);
+    for (let s = scene + step; s <= sceneDuration + step; s += step) {
+        const clampedScene = Math.min(s, sceneDuration);
+        const srcT = sourceTimeAtSceneElapsed(curve, sourceLength, clampedScene);
+        const value = Math.max(1e-4, curve.sampleAt(srcT, sourceLength, 1) * scalar);
+        const ctxTime = startTime + (clampedScene - scene);
+        rate.linearRampToValueAtTime(value, ctxTime);
+        if (clampedScene >= sceneDuration)
+            break;
+    }
+}
+//# sourceMappingURL=filter-graph.js.map

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

@@ -0,0 +1 @@
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{ AudioFilterItem, Param, isCurve, sourceTimeAtSceneElapsed } from \"@motionscript/core\";\n\n/**\n * Result of building an audio-filter graph for one source.\n *\n * - `output` is the last node in the chain; connect it downstream (e.g. to the\n *   per-request gain node).\n * - `nodes` are every intermediate node, so callers can `disconnect()` them on\n *   teardown.\n * - `oscillators` are LFOs (from `tremolo`) the caller must `start()` — and may\n *   `stop()` — alongside the source, since oscillators need an explicit start.\n */\nexport interface AudioFilterGraph {\n    output: AudioNode;\n    nodes: AudioNode[];\n    oscillators: OscillatorNode[];\n}\n\n/**\n * How many sample points per second to bake an eased curve segment into for\n * `setValueCurveAtTime`. 200/s (≈ every 5ms) is smooth to the ear and cheap.\n */\nconst CURVE_SAMPLES_PER_SECOND = 200;\n\n/**\n * Apply a (possibly time-varying) {@link Param} to a Web Audio {@link AudioParam}.\n *\n * - **number** → set once (today's behavior; zero scheduling overhead).\n * - **Curve** → resolve against `clipDuration` and schedule each segment:\n *   - linear segments via `setValueAtTime` + `linearRampToValueAtTime`,\n *   - eased segments baked into a `Float32Array` via `setValueCurveAtTime`.\n *\n * `elapsed` (scene-seconds already played when a mid-clip seek starts the source)\n * shifts the schedule back by `elapsed` so \"now\" reads the curve's value at the\n * clip's *current* position, not its start. A `clamp(v)` hook lets a param keep a\n * valid range (e.g. echo mix 0–1) at every sampled point.\n */\nfunction applyParam(\n    ctx: BaseAudioContext,\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    // Coerce to a finite value before touching an AudioParam — the platform throws\n    // on NaN/Infinity. Falls back to the current param value (never undefined here).\n    const finite = (v: number): number => {\n        const c = clamp(v);\n        return Number.isFinite(c) ? c : audioParam.value;\n    };\n\n    if (!isCurve(param)) {\n        audioParam.value = finite(param);\n        return;\n    }\n\n    // A non-finite clip length yields no usable schedule; hold the curve's start value.\n    if (!Number.isFinite(clipDuration) || clipDuration <= 0) {\n        audioParam.value = finite(param.staticValue(audioParam.value));\n        return;\n    }\n\n    const base = startTime - elapsed; // map curve t=0 to here so seeks line up\n    const resolved = param.resolve(clipDuration, audioParam.value);\n    if (resolved.length === 0) return;\n\n    // Anchor the starting value at `startTime` (the seek point) so there is no\n    // jump from the param's default before the first scheduled segment runs.\n    const startValue = param.sampleAt(Math.max(0, elapsed), clipDuration, audioParam.value);\n    audioParam.setValueAtTime(finite(startValue), startTime);\n\n    for (const seg of resolved) {\n        const segStart = base + seg.startTime;\n        const segEnd = base + seg.endTime;\n        // Skip segments wholly in the past relative to the seek point, or any with a\n        // non-finite boundary (degenerate resolve) that would throw on schedule.\n        if (segEnd <= startTime || !Number.isFinite(segStart) || !Number.isFinite(segEnd)) continue;\n\n        const span = seg.endTime - seg.startTime;\n\n        if (span <= 0) {\n            audioParam.setValueAtTime(finite(seg.to), Math.max(segStart, startTime));\n            continue;\n        }\n\n        if (seg.isLinear) {\n            audioParam.setValueAtTime(finite(seg.from), Math.max(segStart, startTime));\n            audioParam.linearRampToValueAtTime(finite(seg.to), segEnd);\n        } else {\n            const count = Math.max(2, Math.ceil(span * CURVE_SAMPLES_PER_SECOND));\n            const samples = new Float32Array(count);\n            for (let i = 0; i < count; i++) {\n                const local = i / (count - 1);\n                samples[i] = finite(seg.from + (seg.to - seg.from) * seg.ease(local));\n            }\n            // setValueCurveAtTime can't start in the past; clamp the start forward.\n            const curveStart = Math.max(segStart, startTime);\n            const curveDur = segEnd - curveStart;\n            if (curveDur > 0) audioParam.setValueCurveAtTime(samples, curveStart, curveDur);\n        }\n    }\n}\n\n/**\n * Builds the Web Audio graph for a clip's filter chain, returning the node to\n * connect downstream. Shared by live playback ({@link WebAudioSink}) and\n * offline export (`mixAudio`) so both paths sound identical.\n *\n * Filters apply in array order (index 0 closest to the source). `speed` filters\n * are intentionally ignored here — they're realized via `source.playbackRate`\n * by the caller (and affect timeline scheduling), not as a graph node.\n *\n * `startTime` is the AudioContext time the source starts; `clipDuration` is the\n * clip's scene-time length; `elapsed` is scene-seconds already played at a mid-clip\n * seek (0 for a clean start). These let curve-valued params schedule and align.\n *\n * Pass the already-created `source` so its `playbackRate` can be set by the\n * caller; this function only wires nodes *after* the source.\n */\nexport function buildAudioFilterGraph(\n    ctx: BaseAudioContext,\n    source: AudioNode,\n    filters: readonly AudioFilterItem[],\n    startTime: number = ctx.currentTime,\n    clipDuration: number = 0,\n    elapsed: number = 0,\n): AudioFilterGraph {\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(ctx, p, v, startTime, clipDuration, elapsed, clamp);\n\n    for (const filter of filters) {\n        switch (filter.type) {\n            case \"gain\": {\n                const gain = ctx.createGain();\n                apply(gain.gain, filter.value);\n                series(gain);\n                break;\n            }\n            case \"highpass\":\n            case \"lowpass\": {\n                const biquad = ctx.createBiquadFilter();\n                biquad.type = filter.type === \"highpass\" ? \"highpass\" : \"lowpass\";\n                apply(biquad.frequency, filter.frequency);\n                if (filter.q !== undefined) apply(biquad.Q, filter.q);\n                series(biquad);\n                break;\n            }\n            case \"tremolo\": {\n                // Carrier gain whose value is modulated by an LFO. With depth d the\n                // carrier base = (1 - d/2) and the LFO swing gain = d/2. The clamp\n                // hook applies these transforms whether depth is scalar or a curve.\n                const carrier = ctx.createGain();\n                const lfo = ctx.createOscillator();\n                const lfoDepth = ctx.createGain();\n\n                apply(carrier.gain, filter.depth, (d) => 1 - clamp01(d) / 2);\n                apply(lfoDepth.gain, filter.depth, (d) => clamp01(d) / 2);\n                apply(lfo.frequency, filter.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                // Dry + wet split. The wet path is a delay with feedback; both sum\n                // into a merge node that becomes the new cursor.\n                const split = cursor;\n                const merge = ctx.createGain();\n\n                const dry = ctx.createGain();\n                apply(dry.gain, filter.mix ?? 0.5, (m) => 1 - clamp01(m));\n                split.connect(dry);\n                dry.connect(merge);\n\n                // Delay buffer must accommodate the max delay; size from the static value.\n                const maxDelay = staticOf(filter.delay);\n                const delay = ctx.createDelay(Math.max(1, maxDelay + 1));\n                apply(delay.delayTime, filter.delay);\n                const feedbackGain = ctx.createGain();\n                apply(feedbackGain.gain, filter.feedback, (f) => Math.max(0, Math.min(0.99, f)));\n                const wet = ctx.createGain();\n                apply(wet.gain, filter.mix ?? 0.5, (m) => clamp01(m));\n\n                split.connect(delay);\n                delay.connect(feedbackGain);\n                feedbackGain.connect(delay); // feedback loop\n                delay.connect(wet);\n                wet.connect(merge);\n\n                nodes.push(dry, delay, feedbackGain, wet, merge);\n                cursor = merge;\n                break;\n            }\n            case \"speed\":\n                // Realized via source.playbackRate by the caller; no graph node.\n                break;\n        }\n    }\n\n    return { output: cursor, nodes, oscillators };\n}\n\n/** Clamp to [0, 1]. */\nfunction clamp01(v: number): number {\n    return Math.max(0, Math.min(1, v));\n}\n\n/** The static/representative number of a param (number itself, or curve's t=0 value). */\nfunction staticOf(p: Param): number {\n    return isCurve(p) ? p.staticValue(0) : p;\n}\n\n/**\n * Net *constant* playback-rate multiplier from any scalar `speed` filters in the\n * chain (product of all), defaulting to 1. Curve-valued speeds contribute 1 here\n * and are scheduled separately by {@link applySpeedToPlaybackRate}. Mirrors\n * `Sound.effectiveSpeed()` so the renderer applies the same rate the core timeline\n * reserved time for.\n */\nexport function effectiveSpeed(filters: readonly AudioFilterItem[] | undefined): number {\n    if (!filters) return 1;\n    let speed = 1;\n    for (const f of filters) {\n        if (f.type === \"speed\" && !isCurve(f.value) && f.value > 0) speed *= f.value;\n    }\n    return speed;\n}\n\n/** The first curve-valued `speed` filter in the chain, if any. */\nexport function speedCurveOf(filters: readonly AudioFilterItem[] | undefined) {\n    if (!filters) return undefined;\n    for (const f of filters) {\n        if (f.type === \"speed\" && isCurve(f.value)) return f.value;\n    }\n    return undefined;\n}\n\n/**\n * Schedule a time-varying speed curve onto an `AudioBufferSourceNode.playbackRate`.\n *\n * The curve is authored over **source time** (rate as a function of how far through\n * the source we've played). `setValueAtTime`/`linearRampToValueAtTime` schedule in\n * **context time**, so we walk scene-time forward in small steps, convert each to a\n * source-time offset via the integral inverse, read the curve there, and lay down a\n * value point at the matching context time. `scalar` folds in any constant speed\n * filters (they scale the whole rate uniformly).\n *\n * `elapsed` is scene-seconds already played at a mid-clip seek, so the schedule\n * starts at the seek point and aligns the playbackRate to the clip's true position.\n */\nexport function applySpeedToPlaybackRate(\n    rate: AudioParam,\n    curve: ReturnType<typeof speedCurveOf>,\n    sourceLength: number,\n    sceneDuration: number,\n    scalar: number,\n    startTime: number,\n    elapsed: number,\n): void {\n    if (!curve) return;\n    const step = 1 / CURVE_SAMPLES_PER_SECOND; // scene-time sampling step\n    let scene = Math.max(0, elapsed);\n    // Seed the value exactly at the seek point.\n    const seedSrc = sourceTimeAtSceneElapsed(curve, sourceLength, scene);\n    rate.setValueAtTime(Math.max(1e-4, curve.sampleAt(seedSrc, sourceLength, 1) * scalar), startTime);\n\n    for (let s = scene + step; s <= sceneDuration + step; s += step) {\n        const clampedScene = Math.min(s, sceneDuration);\n        const srcT = sourceTimeAtSceneElapsed(curve, sourceLength, clampedScene);\n        const value = Math.max(1e-4, curve.sampleAt(srcT, sourceLength, 1) * scalar);\n        const ctxTime = startTime + (clampedScene - scene);\n        rate.linearRampToValueAtTime(value, ctxTime);\n        if (clampedScene >= sceneDuration) 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

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package/dist/audio/index.js

@@ -0,0 +1,12 @@
+// @motionscript/browser/audio — stacked-audio playback and mixdown, with no
+// renderer attached.
+//
+// A separate entry point from the package barrel on purpose: that barrel loads
+// the CanvasKit/Skia backend at module scope, and nothing here draws anything.
+// A host building an audio timeline should not pay for the WASM module.
+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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package/dist/audio/mixer.d.ts

@@ -0,0 +1,59 @@
+import { type AssetCatalog } from "@motionscript/core";
+import type { AudioMixer, ScheduledAudioRequest } from "@motionscript/skia-render/export";
+/** Sample rate used when the caller doesn't pick one. */
+export declare const DEFAULT_SAMPLE_RATE = 44100;
+export interface MixAudioOptions {
+    /** Defaults to {@link DEFAULT_SAMPLE_RATE}. */
+    sampleRate?: number;
+    /**
+     * Resolves a request's `src` to the address its bytes live at, the same job
+     * `WebStorageAdapter.assetUrl` does for the live paths. Omitted, a `src` is
+     * 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

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package/dist/audio/mixer.js

@@ -0,0 +1,193 @@
+import { sourceTimeAtSceneElapsed } from "@motionscript/core";
+import { AudioBusGraph } from "./bus-graph";
+import { applySpeedToPlaybackRate, buildAudioFilterGraph, effectiveSpeed, speedCurveOf, } from "./filter-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;
+        }
+    }
+}
+/** Fetches and decodes an audio source against a scratch `OfflineAudioContext` (decoding requires a context but doesn't render through it). */
+async function fetchAudioBuffer(src, ctx) {
+    const response = await fetch(src);
+    const arrayBuffer = await response.arrayBuffer();
+    return ctx.decodeAudioData(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);
+    const decoded = new Map();
+    await Promise.all(uniqueSrcs.map(async (src) => {
+        decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));
+    }));
+    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;
+        source.loop = req.loop;
+        const scalarSpeed = effectiveSpeed(req.filters);
+        const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
+        const speedCurve = speedCurveOf(req.filters);
+        // With a curve the scalar is folded into the scheduled ramp below, so
+        // setting it here as well would apply it twice.
+        if (rate !== 1 && !speedCurve)
+            source.playbackRate.value = rate;
+        const gain = mixCtx.createGain();
+        gain.gain.value = Number.isFinite(req.volume) ? req.volume : 1;
+        // The trimmed source length the filter and speed curves are authored over.
+        const sourceLength = Math.max(0, srcBuffer.duration - req.trimStart);
+        // The clip's own scene-time length — what a curve resolves against.
+        // Deliberately *not* `playDuration`, which is that length truncated by the
+        // timeline end: resolving against the truncated value would squeeze the
+        // whole curve into the surviving part instead of cutting it off.
+        const clipDuration = Number.isFinite(req.endAt)
+            ? Math.max(0, req.endAt - req.startAt)
+            : sourceLength / rate;
+        // Filter chain sits between the source and the per-request gain, matching
+        // the live WebAudioSink graph so exports sound identical to preview.
+        // Passing the start time and clip duration is what lets curve-valued params
+        // schedule — with the defaults they collapse to a single static value.
+        // `elapsed` is 0: unlike a live seek, the mix always starts a clip at its
+        // own beginning.
+        const graph = buildAudioFilterGraph(mixCtx, source, req.filters ?? [], globalStart, clipDuration, 0);
+        graph.output.connect(gain);
+        gain.connect(buses.inputFor(req.bus, globalStart));
+        if (speedCurve) {
+            applySpeedToPlaybackRate(source.playbackRate, speedCurve, sourceLength, clipDuration, rate, globalStart, 0);
+        }
+        // start()'s 3rd arg is a source-buffer duration. A constant rate scales it
+        // linearly, but under a speed curve the buffer consumed over a span of
+        // scene time is the integral of the rate, not a multiple of it.
+        const consumed = speedCurve
+            ? sourceTimeAtSceneElapsed(speedCurve, sourceLength, playDuration)
+            : playDuration * rate;
+        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

package/dist/audio/mixer.js.map

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