@motionscript/browser 0.1.0-alpha.0 → 0.1.0-alpha.5

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

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

package/dist/audio/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/audio/index.ts"],"names":[],"mappings":"AAAA,4EAA4E;AAC5E,qBAAqB;AACrB,EAAE;AACF,+EAA+E;AAC/E,+EAA+E;AAC/E,wEAAwE;AAExE,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: that barrel loads\n// the CanvasKit/Skia backend at module scope, and nothing here draws anything.\n// A host building an audio timeline should not pay for the WASM module.\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"]}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/audio/index.ts"],"names":[],"mappings":"AAAA,4EAA4E;AAC5E,qBAAqB;AACrB,EAAE;AACF,+EAA+E;AAC/E,+EAA+E;AAC/E,wEAAwE;AAExE,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\r\n// renderer attached.\r\n//\r\n// A separate entry point from the package barrel on purpose: that barrel loads\r\n// the CanvasKit/Skia backend at module scope, and nothing here draws anything.\r\n// A host building an audio timeline should not pay for the WASM module.\r\n\r\nexport { mixAudio, encodeWav, WebAudioMixer, DEFAULT_SAMPLE_RATE } from \"./mixer\";\r\nexport type { MixAudioOptions } from \"./mixer\";\r\n\r\nexport { WebAudioSink, WebAudioSink as WebAudioPlayer } from \"./player\";\r\n\r\n// The bus layer both of those share. Exported so a host writing its own device\r\n// or mixer builds the same chains, rather than a second set that drifts.\r\nexport { AudioBusGraph } from \"./bus-graph\";\r\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

@@ -0,0 +1 @@
+{"version":3,"file":"loop-region.d.ts","sourceRoot":"","sources":["../../src/audio/loop-region.ts"],"names":[],"mappings":"AAAA,OAAO,KAAK,EAAE,YAAY,EAAE,MAAM,oBAAoB,CAAC;AAEvD,iFAAiF;AACjF,MAAM,WAAW,UAAU;IACvB,KAAK,EAAE,MAAM,CAAC;IACd,GAAG,EAAE,MAAM,CAAC;CACf;AAED;;;;;;;;GAQG;AACH,wBAAgB,YAAY,CAAC,OAAO,EAAE,YAAY,EAAE,cAAc,EAAE,MAAM,GAAG,UAAU,GAAG,IAAI,CAK7F;AAED;;;;GAIG;AACH,wBAAgB,SAAS,CAAC,MAAM,EAAE,qBAAqB,EAAE,OAAO,EAAE,YAAY,GAAG,UAAU,GAAG,IAAI,CAQjG;AAED;;;;;;GAMG;AACH,wBAAgB,cAAc,CAAC,MAAM,EAAE,UAAU,GAAG,IAAI,EAAE,MAAM,EAAE,MAAM,GAAG,MAAM,CAGhF"}

package/dist/audio/loop-region.js

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

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

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

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

package/dist/audio/mixer.js

@@ -1,5 +1,7 @@
-import { sourceTimeAtSceneElapsed } from "@motionscript/core";
+import { sourceTimeAtClipElapsed } from "@motionscript/core";
+import { fetchAssetBytes, localAssetBytes } from "../asset-bytes";
 import { AudioBusGraph } from "./bus-graph";
+import { applyLoop } from "./loop-region";
 import { applySpeedToPlaybackRate, buildAudioFilterGraph, effectiveSpeed, speedCurveOf, } from "./filter-graph";
 /** Sample rate used when the caller doesn't pick one. */
 export const DEFAULT_SAMPLE_RATE = 44100;
@@ -19,11 +21,16 @@ function audioUrl(src, catalog) {
         }
     }
 }
-/** Fetches and decodes an audio source against a scratch `OfflineAudioContext` (decoding requires a context but doesn't render through it). */
+/**
+ * Reads and decodes an audio source against a scratch `OfflineAudioContext`
+ * (decoding requires a context but doesn't render through it). The host's byte
+ * source is asked first, as for every other asset read: a mix that went to the
+ * network would fail where the preview played, offline or for bytes not yet
+ * uploaded.
+ */
 async function fetchAudioBuffer(src, ctx) {
-    const response = await fetch(src);
-    const arrayBuffer = await response.arrayBuffer();
-    return ctx.decodeAudioData(arrayBuffer);
+    const blob = (await localAssetBytes(src)) ?? (await fetchAssetBytes(src));
+    return ctx.decodeAudioData(await blob.arrayBuffer());
 }
 /**
  * Render every scheduled request into one timeline-length buffer.
@@ -53,9 +60,18 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         return null;
     const uniqueSrcs = [...new Set(requests.map(r => r.request.src))];
     const scratchCtx = new OfflineAudioContext(2, frames, sampleRate);
+    // A source only a picture's own sound wants may hold no audio at all, which
+    // is silence rather than a failed mix — see `AudioRequest.optional`.
+    const asked = new Set(requests.filter(r => !r.request.optional).map(r => r.request.src));
     const decoded = new Map();
     await Promise.all(uniqueSrcs.map(async (src) => {
-        decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));
+        try {
+            decoded.set(src, await fetchAudioBuffer(audioUrl(src, options.catalog), scratchCtx));
+        }
+        catch (cause) {
+            if (asked.has(src))
+                throw cause;
+        }
     }));
     const mixCtx = new OfflineAudioContext(2, frames, sampleRate);
     // The same shared-chain rule the live device uses, for the same reason: an
@@ -74,7 +90,7 @@ export async function mixAudio(requests, totalDuration, options = {}) {
             continue;
         const source = mixCtx.createBufferSource();
         source.buffer = srcBuffer;
-        source.loop = req.loop;
+        applyLoop(source, req);
         const scalarSpeed = effectiveSpeed(req.filters);
         const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
         const speedCurve = speedCurveOf(req.filters);
@@ -86,7 +102,7 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         gain.gain.value = Number.isFinite(req.volume) ? req.volume : 1;
         // The trimmed source length the filter and speed curves are authored over.
         const sourceLength = Math.max(0, srcBuffer.duration - req.trimStart);
-        // The clip's own scene-time length — what a curve resolves against.
+        // The clip's own clip-time length — what a curve resolves against.
         // Deliberately *not* `playDuration`, which is that length truncated by the
         // timeline end: resolving against the truncated value would squeeze the
         // whole curve into the surviving part instead of cutting it off.
@@ -101,15 +117,18 @@ export async function mixAudio(requests, totalDuration, options = {}) {
         // own beginning.
         const graph = buildAudioFilterGraph(mixCtx, source, req.filters ?? [], globalStart, clipDuration, 0);
         graph.output.connect(gain);
-        gain.connect(buses.inputFor(req.bus, globalStart));
+        // 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.
+        gain.connect(buses.inputFor(req.bus, 0, -globalOffset));
         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.
+        // clip time is the integral of the rate, not a multiple of it.
         const consumed = speedCurve
-            ? sourceTimeAtSceneElapsed(speedCurve, sourceLength, playDuration)
+            ? sourceTimeAtClipElapsed(speedCurve, sourceLength, playDuration)
             : playDuration * rate;
         source.start(globalStart, req.trimStart, consumed);
         for (const osc of graph.oscillators)

package/dist/audio/mixer.js.map

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

package/dist/audio/player.d.ts

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

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

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

package/dist/audio/player.js

@@ -1,12 +1,13 @@
-import { AudioSink, sourceTimeAtSceneElapsed } from "@motionscript/core";
+import { AudioSink, sourceTimeAtClipElapsed } from "@motionscript/core";
 import { AudioBusGraph } from "./bus-graph";
+import { applyLoop, wrapIntoRegion } from "./loop-region";
 import { buildAudioFilterGraph, effectiveSpeed, speedCurveOf, applySpeedToPlaybackRate } from "./filter-graph";
 /**
  * Web Audio implementation of {@link AudioSink} — decodes and caches source
  * buffers, then schedules/plays {@link AudioRequest}s as `AudioBufferSourceNode`s
  * through a shared master gain (global mute and volume). `schedule` registers the
  * active set of requests for the current frame; `syncTo` then starts/stops
- * sources to match `sceneTime`, so playback follows scrubbing and seeks.
+ * sources to match `clipTime`, so playback follows scrubbing and seeks.
  */
 export class WebAudioSink extends AudioSink {
     context;
@@ -83,20 +84,20 @@ export class WebAudioSink extends AudioSink {
         }
         this.scheduled = requests.slice();
     }
-    /** Starts sources whose `[startAt, endAt)` window now contains `sceneTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
-    syncTo(sceneTime) {
+    /** Starts sources whose `[startAt, endAt)` window now contains `clipTime` (computing the correct buffer offset for the seek point) and stops any that have fallen outside it. */
+    syncTo(clipTime) {
         const liveIds = new Set();
         for (const req of this.scheduled) {
             // `endAt` is `Infinity` for an open clip, so the comparison needs no
             // nullish fallback — see AudioRequest.endAt.
-            const isInWindow = sceneTime >= req.startAt && sceneTime < req.endAt;
+            const isInWindow = clipTime >= req.startAt && clipTime < req.endAt;
             if (!isInWindow)
                 continue;
             liveIds.add(req.id);
             if (!this.active.has(req.id)) {
                 const buffer = this.bufferCache.get(req.src);
                 if (buffer)
-                    this.playBuffer(buffer, req, sceneTime);
+                    this.playBuffer(buffer, req, clipTime);
             }
         }
         for (const [id, active] of this.active) {
@@ -114,6 +115,9 @@ export class WebAudioSink extends AudioSink {
         for (const [id, active] of this.active) {
             this.stopSource(id, active);
         }
+        // A bus's curves are scheduled against the clock of the run that built
+        // it, so the next play, from wherever it lands, builds them again.
+        this.buses.dispose();
     }
     setMuted(muted) {
         this.muted = muted;
@@ -127,27 +131,28 @@ export class WebAudioSink extends AudioSink {
     applyMasterGain() {
         this.masterGain.gain.value = this.muted ? 0 : this.volume;
     }
-    /** Starts a buffer source mid-clip if `sceneTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
-    playBuffer(buffer, req, sceneTime) {
+    /** Starts a buffer source mid-clip if `clipTime` lands after the request's start — `audioOffset` accounts for the request's trim, the elapsed time since `startAt`, and any speed (playbackRate) change. */
+    playBuffer(buffer, req, clipTime) {
+        const startTime = this.context.currentTime;
         const gainNode = this.context.createGain();
         // Guard against a missing/NaN volume — a non-finite AudioParam value throws.
         gainNode.gain.value = Number.isFinite(req.volume) ? req.volume : 1;
         // Into this clip's bus chain, or straight to the master when it has none.
         // The chain is shared: two clips on one bus reach the same nodes, which is
         // what makes a bus effect one tail rather than one per clip.
-        gainNode.connect(this.buses.inputFor(req.bus, this.context.currentTime));
+        gainNode.connect(this.buses.inputFor(req.bus, startTime, clipTime));
         const source = this.context.createBufferSource();
         source.buffer = buffer;
-        source.loop = req.loop;
+        const loopRegion = applyLoop(source, req);
         const scalarSpeed = effectiveSpeed(req.filters);
         const rate = Number.isFinite(scalarSpeed) && scalarSpeed > 0 ? scalarSpeed : 1;
         const speedCurve = speedCurveOf(req.filters);
         if (rate !== 1 && !speedCurve)
             source.playbackRate.value = rate;
-        const elapsed = Math.max(0, sceneTime - req.startAt);
+        const elapsed = Math.max(0, clipTime - req.startAt);
         // Trimmed source length the filter curves / speed curve are authored over.
         const sourceLength = Math.max(0, buffer.duration - req.trimStart);
-        // Scene-time length of the clip. Prefer the bounded [startAt, endAt) span; if
+        // Clip-time length of the clip. Prefer the bounded [startAt, endAt) span; if
         // endAt is unbounded (an open/looping clip reaching the device unresolved),
         // fall back to the source length scaled by speed so curve params still resolve
         // (a clipDuration of 0 would otherwise silently drop every fade/sweep).
@@ -157,21 +162,20 @@ export class WebAudioSink extends AudioSink {
         // Insert the filter chain between the source and the per-request gain. Pass
         // the start time, clip duration, and elapsed so curve-valued params schedule
         // and align to a mid-clip seek.
-        const startTime = this.context.currentTime;
         const graph = buildAudioFilterGraph(this.context, source, req.filters ?? [], startTime, clipDuration, elapsed);
         graph.output.connect(gainNode);
         // Source-time offset for a mid-clip seek. Scalar speed scales linearly; a
         // speed curve uses the integral inverse to find the true source position.
         let audioOffset;
         if (speedCurve) {
-            audioOffset = req.trimStart + sourceTimeAtSceneElapsed(speedCurve, sourceLength, elapsed);
+            audioOffset = req.trimStart + sourceTimeAtClipElapsed(speedCurve, sourceLength, elapsed);
             // Schedule the time-varying playbackRate (folds in any scalar speed too).
             applySpeedToPlaybackRate(source.playbackRate, speedCurve, sourceLength, clipDuration, rate, startTime, elapsed);
         }
         else {
             audioOffset = req.trimStart + elapsed * rate;
         }
-        source.start(startTime, audioOffset);
+        source.start(startTime, wrapIntoRegion(loopRegion, audioOffset));
         for (const osc of graph.oscillators)
             osc.start(startTime);
         source.onended = () => {