@clipkit/runtime 1.0.0

package/dist/audio/limiter.js

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+// Master output limiter — a fixed, knob-less safety stage on the master bus,
+// identical in preview (playback) and export (runtime mixer) so what you hear
+// is what you render.
+//
+// It is TRANSPARENT below the ceiling: signal under the threshold passes at
+// unity (Web Audio's DynamicsCompressorNode applies no makeup gain), so it
+// only acts when the summed mix would otherwise clip past 0 dBFS — replacing
+// ugly digital clipping with controlled limiting. This is an output-stage
+// guarantee, NOT a document field: nothing here is serialized into the Source,
+// and there are no per-element side effects (it sees only the final sum).
+//
+// Tradeoff: a DynamicsCompressorNode at ratio 20 is not a perfectly infinite
+// brick-wall, but with -1 dB of headroom it keeps typical material comfortably
+// under 0 dBFS without the lookahead cost of a custom limiter. Lean by design.
+/** Threshold just below 0 dBFS — a hair of headroom before the ceiling. */
+const THRESHOLD_DB = -1;
+/** Hard knee — no soft transition, so quiet content stays untouched. */
+const KNEE_DB = 0;
+/** Steepest ratio Web Audio allows; effectively a limiter. */
+const RATIO = 20;
+/** Fast enough to catch transients without audible pumping. */
+const ATTACK_S = 0.003;
+const RELEASE_S = 0.1;
+/**
+ * Create the master limiter node. Works on both AudioContext (preview) and
+ * OfflineAudioContext (export). Route the summed master gain through this,
+ * then to the destination — and tap meters POST-limiter so they show what
+ * you actually hear.
+ */
+export function createMasterLimiter(ctx) {
+    const limiter = ctx.createDynamicsCompressor();
+    limiter.threshold.value = THRESHOLD_DB;
+    limiter.knee.value = KNEE_DB;
+    limiter.ratio.value = RATIO;
+    limiter.attack.value = ATTACK_S;
+    limiter.release.value = RELEASE_S;
+    return limiter;
+}
+//# sourceMappingURL=limiter.js.map

package/dist/audio/limiter.js.map

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package/dist/audio/loader.d.ts

@@ -0,0 +1,6 @@
+/**
+ * Fetch + decode a URL into an AudioBuffer. Throws on network or decode
+ * errors; callers should let the error propagate through preload().
+ */
+export declare function loadAudio(url: string): Promise<AudioBuffer>;
+//# sourceMappingURL=loader.d.ts.map

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

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

@@ -0,0 +1,42 @@
+// Audio loader. Fetches a URL and decodes it into an AudioBuffer that we
+// can later play (preview) or mix offline (export).
+//
+// Uses the regular (non-offline) AudioContext to decode because
+// decodeAudioData isn't exposed on OfflineAudioContext. The resulting
+// AudioBuffer is detached from any context and can be used in either.
+let sharedDecodeContext = null;
+function getDecodeContext() {
+    if (!sharedDecodeContext) {
+        sharedDecodeContext = new AudioContext();
+        // Most browsers auto-suspend until a user gesture; suspend explicitly
+        // to avoid burning cycles during decode (we don't play through this one).
+        void sharedDecodeContext.suspend();
+    }
+    return sharedDecodeContext;
+}
+/**
+ * Fetch + decode a URL into an AudioBuffer. Throws on network or decode
+ * errors; callers should let the error propagate through preload().
+ */
+export async function loadAudio(url) {
+    // 10s fetch timeout (mirrors loadImage): a dead/stalled URL (e.g. a retired
+    // host) otherwise dangles a connection and leans entirely on preload()'s
+    // per-asset guard. AbortController cancels the fetch itself.
+    const ac = new AbortController();
+    const timer = setTimeout(() => ac.abort(), 10000);
+    let response;
+    try {
+        response = await fetch(url, { mode: 'cors', signal: ac.signal });
+    }
+    finally {
+        clearTimeout(timer);
+    }
+    if (!response.ok) {
+        throw new Error(`Failed to fetch audio ${url}: HTTP ${response.status}`);
+    }
+    const arrayBuffer = await response.arrayBuffer();
+    const ctx = getDecodeContext();
+    // decodeAudioData returns a Promise in the modern API.
+    return ctx.decodeAudioData(arrayBuffer);
+}
+//# sourceMappingURL=loader.js.map

package/dist/audio/loader.js.map

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package/dist/audio/mixer.d.ts

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+import type { Source } from '@clipkit/protocol';
+export interface MixOptions {
+    /** Sample rate of the output buffer. Common: 48000 (matches AAC defaults). */
+    sampleRate?: number;
+    /** Number of channels. 2 = stereo. */
+    numberOfChannels?: number;
+}
+/**
+ * Render all `audio` elements in a Source to a single mixed AudioBuffer.
+ *
+ * @param source         The Source being exported.
+ * @param audioBuffers   Map of element.source URL → decoded AudioBuffer (preloaded).
+ * @param totalDuration  Duration of the export in seconds.
+ * @param options        Sample rate / channels override.
+ */
+export declare function mixSourceAudio(source: Source, audioBuffers: Map<string, AudioBuffer>, totalDuration: number, options?: MixOptions): Promise<AudioBuffer | null>;
+//# sourceMappingURL=mixer.d.ts.map

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

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package/dist/audio/mixer.js

@@ -0,0 +1,204 @@
+// Offline audio mixing for export.
+//
+// Schedules each audio element — AND each video element's embedded
+// audio track — on an OfflineAudioContext at its `time`, trimmed by
+// `trim_start`/`trim_duration`, rate-adjusted by `playback_rate`
+// (videos), capped at `duration`, with gain set from `volume`. The
+// rendered AudioBuffer goes to the encoder.
+//
+// Volume keyframe animations are NOT applied in v1 — the static `volume`
+// is used. Animating volume requires running a separate Web Audio graph
+// per element with a GainNode driven by automation; small refactor when
+// we want it.
+import { interpolateKeyframes } from '../animation/keyframes.js';
+import { mapToMediaTime, rateOf, timeRemapOf, trimDurationOf, trimWindow } from '../assets/media-time.js';
+import { fadeBreakpoints } from './fades.js';
+import { createMasterLimiter } from './limiter.js';
+import { scheduleVarispeedRun, varispeedRuns } from './varispeed.js';
+import { getLogger } from '../logger.js';
+/** Recursively collect sound elements with their ancestor time scopes. */
+function collectSound(elements, chain, parentDur, out) {
+    for (const el of elements) {
+        const start = numberOr(el.time, 0);
+        const dur = parseDuration(el.duration, parentDur - start);
+        if (el.type === 'audio' || el.type === 'video') {
+            out.push({ el: el, chain });
+        }
+        else if (el.type === 'group') {
+            const kids = el.elements;
+            if (Array.isArray(kids)) {
+                collectSound(kids, [...chain, { start, dur, remap: timeRemapOf(el.time_remap) }], dur, out);
+            }
+        }
+    }
+}
+const DEFAULT_SAMPLE_RATE = 48000;
+const DEFAULT_CHANNELS = 2;
+/**
+ * Render all `audio` elements in a Source to a single mixed AudioBuffer.
+ *
+ * @param source         The Source being exported.
+ * @param audioBuffers   Map of element.source URL → decoded AudioBuffer (preloaded).
+ * @param totalDuration  Duration of the export in seconds.
+ * @param options        Sample rate / channels override.
+ */
+export async function mixSourceAudio(source, audioBuffers, totalDuration, options = {}) {
+    const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE;
+    const channels = options.numberOfChannels ?? DEFAULT_CHANNELS;
+    const totalFrames = Math.ceil(totalDuration * sampleRate);
+    const soundEntries = [];
+    collectSound(source.elements, [], totalDuration, soundEntries);
+    if (soundEntries.length === 0)
+        return null;
+    // OfflineAudioContext renders deterministically with no realtime constraint.
+    const ctx = new OfflineAudioContext({
+        numberOfChannels: channels,
+        length: totalFrames,
+        sampleRate,
+    });
+    // Master bus: every source sums into masterGain, then passes through a fixed
+    // safety limiter to the destination. The limiter is transparent below 0 dBFS
+    // and only engages when the sum would otherwise clip — identical to the
+    // preview scheduler's master bus, so export matches what you hear.
+    const masterGain = ctx.createGain();
+    const limiter = createMasterLimiter(ctx);
+    masterGain.connect(limiter).connect(ctx.destination);
+    let scheduledCount = 0;
+    for (const { el: element, chain } of soundEntries) {
+        const url = String(element.source ?? '');
+        const buffer = audioBuffers.get(url);
+        if (!buffer) {
+            // Expected for silent videos; a real miss for audio elements.
+            if (element.type === 'audio') {
+                getLogger().warn(`Audio element ${element.id ?? '(no id)'} skipped — buffer not preloaded for ${url}`);
+            }
+            continue;
+        }
+        const ownRemap = element.type === 'video' ? timeRemapOf(element.time_remap) : null;
+        const warped = ownRemap !== null || chain.some((c) => c.remap !== null);
+        if (warped) {
+            // Varispeed (§5.3.2/§5.8.3): sample the effective media-time
+            // function through the warp chain, split into monotonic runs,
+            // schedule each (reversed buffer for backward runs). Fades are
+            // not applied under warps in v1; volume is.
+            const elStart = numberOr(element.time, 0);
+            const elDur = parseDuration(element.duration, totalDuration);
+            const rate0 = element.type === 'video' ? rateOf(element.playback_rate) : 1;
+            const mediaAt = (t) => {
+                let c = t;
+                for (const scope of chain) {
+                    if (c < scope.start || c > scope.start + scope.dur)
+                        return null;
+                    c -= scope.start;
+                    if (scope.remap)
+                        c = Math.max(0, interpolateKeyframes(scope.remap, c));
+                }
+                if (c < elStart || c > elStart + elDur)
+                    return null;
+                const m = mapToMediaTime(c, {
+                    elementStart: elStart,
+                    trimStart: clampNonNeg(numberOr(element.trim_start, 0)),
+                    trimDuration: trimDurationOf(element.trim_duration),
+                    rate: rate0,
+                    loop: element.loop === true,
+                    timeRemap: ownRemap,
+                }, buffer.duration);
+                return Math.max(0, Math.min(m, buffer.duration));
+            };
+            const volumePct0 = numberOr(element.volume, 100);
+            const gain0 = Math.max(0, volumePct0 / 100);
+            for (const run of varispeedRuns(mediaAt, 0, totalDuration)) {
+                scheduleVarispeedRun(ctx, buffer, run, masterGain, run.t0, gain0);
+                scheduledCount++;
+            }
+            continue;
+        }
+        // Static path — translate-only chains: the audible window is the
+        // element's own, offset by its ancestors' starts.
+        const chainOffset = chain.reduce((a, c) => a + c.start, 0);
+        const startTime = clampNonNeg(chainOffset + numberOr(element.time, 0));
+        // playback_rate only exists on video; audio elements run at 1.
+        const rate = element.type === 'video' ? rateOf(element.playback_rate) : 1;
+        const window = trimWindow({
+            trimStart: clampNonNeg(numberOr(element.trim_start, 0)),
+            trimDuration: trimDurationOf(element.trim_duration),
+        }, buffer.duration);
+        // Bail if scheduling is past the end of the export.
+        if (startTime >= totalDuration)
+            continue;
+        if (window.length <= 0)
+            continue;
+        // Timeline seconds the element occupies; media plays at `rate`.
+        const elementDuration = parseDuration(element.duration, window.length / rate);
+        const timelineCap = Math.min(elementDuration, totalDuration - startTime);
+        // start()'s duration argument is in BUFFER seconds.
+        const bufferPlayDuration = Math.min(window.length, timelineCap * rate);
+        if (bufferPlayDuration <= 0)
+            continue;
+        const sourceNode = ctx.createBufferSource();
+        sourceNode.buffer = buffer;
+        sourceNode.playbackRate.value = rate;
+        const gainNode = ctx.createGain();
+        const volumePct = numberOr(element.volume, 100);
+        const baseGain = Math.max(0, volumePct / 100);
+        gainNode.gain.value = baseGain;
+        // audio_fade_in / audio_fade_out — piecewise-linear gain envelope
+        // over the element's timeline window (shared with the preview
+        // scheduler via fades.ts).
+        const fadeIn = Math.max(0, numberOr(element.audio_fade_in, 0));
+        const fadeOut = Math.max(0, numberOr(element.audio_fade_out, 0));
+        if (fadeIn > 0 || fadeOut > 0) {
+            const points = fadeBreakpoints(0, timelineCap, fadeIn, fadeOut);
+            gainNode.gain.setValueAtTime(baseGain * points[0].gain, startTime);
+            for (const p of points.slice(1)) {
+                gainNode.gain.linearRampToValueAtTime(baseGain * p.gain, startTime + p.tau);
+            }
+        }
+        sourceNode.connect(gainNode).connect(masterGain);
+        if (element.loop) {
+            sourceNode.loop = true;
+            sourceNode.loopStart = window.start;
+            sourceNode.loopEnd = window.start + window.length;
+            // Looping nodes ignore start()'s duration in some implementations;
+            // stop explicitly at the element's timeline end.
+            sourceNode.start(startTime, window.start);
+            sourceNode.stop(startTime + timelineCap);
+        }
+        else {
+            sourceNode.start(startTime, window.start, bufferPlayDuration);
+        }
+        scheduledCount++;
+    }
+    if (scheduledCount === 0)
+        return null;
+    return ctx.startRendering();
+}
+function numberOr(v, fallback) {
+    if (typeof v === 'number' && Number.isFinite(v))
+        return v;
+    if (typeof v === 'string') {
+        const n = parseFloat(v);
+        if (Number.isFinite(n))
+            return n;
+    }
+    return fallback;
+}
+function numberOrNull(v) {
+    if (typeof v === 'number' && Number.isFinite(v))
+        return v;
+    if (typeof v === 'string') {
+        const n = parseFloat(v);
+        if (Number.isFinite(n))
+            return n;
+    }
+    return null;
+}
+function parseDuration(v, fallback) {
+    if (v === 'auto' || v === 'end' || v == null)
+        return fallback;
+    return numberOr(v, fallback);
+}
+function clampNonNeg(n) {
+    return n < 0 ? 0 : n;
+}
+//# sourceMappingURL=mixer.js.map

package/dist/audio/mixer.js.map

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

@@ -0,0 +1,24 @@
+/** Sampling interval for the media-time function, seconds. */
+export declare const VARISPEED_STEP = 0.01;
+export interface VarispeedRun {
+    /** Composition-time window of the run. */
+    t0: number;
+    t1: number;
+    /** Media position at each 10 ms sample inside [t0, t1]. */
+    media: number[];
+    direction: 'forward' | 'backward';
+}
+/**
+ * Split a sampled media-time function into monotonic runs. `mediaAt`
+ * returns the media position at a real composition time, or null when
+ * the element is inactive (outside its own or an ancestor's window).
+ */
+export declare function varispeedRuns(mediaAt: (t: number) => number | null, start: number, end: number): VarispeedRun[];
+/**
+ * Schedule one varispeed run. Returns the created source node (caller
+ * may track it for cancellation in live contexts).
+ */
+export declare function scheduleVarispeedRun(ctx: BaseAudioContext, buffer: AudioBuffer, run: VarispeedRun, destination: AudioNode, 
+/** AudioContext time corresponding to composition time run.t0. */
+when: number, gain: number): AudioBufferSourceNode;
+//# sourceMappingURL=varispeed.d.ts.map

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

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

@@ -0,0 +1,114 @@
+// Varispeed audio under warped clocks (§5.3.2 / §5.8.3).
+//
+// When a sound element sits under group time_remap (or carries video
+// time_remap itself), its media position is a piecewise function of
+// REAL composition time. Rather than algebra over nested keyframe
+// chains, the reference implementation SAMPLES the effective
+// media-time function at 10 ms, splits it into monotonic runs, and
+// schedules each run on an AudioBufferSourceNode:
+//
+//   forward  — start(t0, offset m(t0)); playbackRate follows the
+//              sampled derivative via setValueCurveAtTime (tape-style
+//              pitch: 2× plays high, slow-mo plays low)
+//   freeze   — silence (the tape isn't moving)
+//   backward — a cached REVERSED copy of the buffer plays forward at
+//              |dm/dt|, which is reversed audio at the right speed
+//
+// Pitch-preserving time-stretch is deliberately out of scope (no
+// exactly-specifiable algorithm).
+/** Sampling interval for the media-time function, seconds. */
+export const VARISPEED_STEP = 0.01;
+/** Below this |Δmedia/Δt| a sample step counts as frozen. */
+const FREEZE_EPS = 0.02;
+/**
+ * Split a sampled media-time function into monotonic runs. `mediaAt`
+ * returns the media position at a real composition time, or null when
+ * the element is inactive (outside its own or an ancestor's window).
+ */
+export function varispeedRuns(mediaAt, start, end) {
+    const runs = [];
+    let cur = null;
+    let prev = null;
+    for (let t = start; t <= end + 1e-9; t += VARISPEED_STEP) {
+        const m = mediaAt(t);
+        const step = prev !== null && m !== null ? (m - prev) / VARISPEED_STEP : null;
+        const dir = step === null ? null : step > FREEZE_EPS ? 'forward' : step < -FREEZE_EPS ? 'backward' : 'freeze';
+        if (dir === 'forward' || dir === 'backward') {
+            if (cur && cur.direction === dir) {
+                cur.media.push(m);
+                cur.t1 = t;
+            }
+            else {
+                if (cur && cur.media.length > 1)
+                    runs.push(cur);
+                cur = { t0: t - VARISPEED_STEP, t1: t, media: [prev, m], direction: dir };
+            }
+        }
+        else {
+            if (cur && cur.media.length > 1)
+                runs.push(cur);
+            cur = null;
+        }
+        prev = m;
+    }
+    if (cur && cur.media.length > 1)
+        runs.push(cur);
+    return runs;
+}
+/** Reversed copies, keyed by the original buffer (per-mix lifetime). */
+const reversedCache = new WeakMap();
+function reversedBuffer(ctx, buffer) {
+    const hit = reversedCache.get(buffer);
+    if (hit)
+        return hit;
+    const rev = ctx.createBuffer(buffer.numberOfChannels, buffer.length, buffer.sampleRate);
+    for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
+        const src = buffer.getChannelData(ch);
+        const dst = rev.getChannelData(ch);
+        for (let i = 0, n = src.length; i < n; i++)
+            dst[i] = src[n - 1 - i];
+    }
+    reversedCache.set(buffer, rev);
+    return rev;
+}
+/**
+ * Schedule one varispeed run. Returns the created source node (caller
+ * may track it for cancellation in live contexts).
+ */
+export function scheduleVarispeedRun(ctx, buffer, run, destination, 
+/** AudioContext time corresponding to composition time run.t0. */
+when, gain) {
+    const node = ctx.createBufferSource();
+    const dur = run.t1 - run.t0;
+    const n = run.media.length;
+    // Rate curve: |Δmedia| per step, one value per sample interval.
+    const rates = new Float32Array(Math.max(2, n - 1));
+    for (let i = 0; i < n - 1; i++) {
+        rates[i] = Math.max(0.001, Math.abs(run.media[i + 1] - run.media[i]) / VARISPEED_STEP);
+    }
+    if (n - 1 === 1)
+        rates[1] = rates[0];
+    const m0 = run.media[0];
+    const mEnd = run.media[n - 1];
+    const mediaSpan = Math.abs(mEnd - m0);
+    if (run.direction === 'forward') {
+        node.buffer = buffer;
+        node.playbackRate.setValueCurveAtTime(rates, when, dur);
+        const g = ctx.createGain();
+        g.gain.value = gain;
+        node.connect(g).connect(destination);
+        node.start(when, Math.max(0, Math.min(m0, buffer.duration)), mediaSpan);
+    }
+    else {
+        const rev = reversedBuffer(ctx, buffer);
+        node.buffer = rev;
+        node.playbackRate.setValueCurveAtTime(rates, when, dur);
+        const g = ctx.createGain();
+        g.gain.value = gain;
+        node.connect(g).connect(destination);
+        // Media position p lives at reversed offset (duration − p).
+        node.start(when, Math.max(0, Math.min(buffer.duration - m0, buffer.duration)), mediaSpan);
+    }
+    return node;
+}
+//# sourceMappingURL=varispeed.js.map

package/dist/audio/varispeed.js.map

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

@@ -0,0 +1,6 @@
+/**
+ * Serialize an AudioBuffer to a 16-bit PCM (s16le) WAV file as raw bytes.
+ * Channels are interleaved; sample rate / channel count come from the buffer.
+ */
+export declare function audioBufferToWav(buffer: AudioBuffer): ArrayBuffer;
+//# sourceMappingURL=wav.d.ts.map

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

@@ -0,0 +1 @@
+{"version":3,"file":"wav.d.ts","sourceRoot":"","sources":["../../src/audio/wav.ts"],"names":[],"mappings":"AAYA;;;GAGG;AACH,wBAAgB,gBAAgB,CAAC,MAAM,EAAE,WAAW,GAAG,WAAW,CAmDjE"}

package/dist/audio/wav.js

@@ -0,0 +1,62 @@
+// Dependency-free WAV (RIFF/PCM) serializer.
+//
+// The transparent export path (ProRes 4444 / VP9-alpha) can't use the WebCodecs
+// muxer, so its mixed AudioBuffer can't ride along in an MP4. Instead we
+// serialize it to a plain 16-bit PCM WAV the server-side ffmpeg reads as a
+// second input. WAV is the lowest-common-denominator container — no codec, no
+// muxer dependency — and ffmpeg re-encodes it to the format's audio codec
+// (PCM for MOV/ProRes, Opus for WebM).
+//
+// Output is interleaved signed 16-bit little-endian PCM (s16le), the most
+// universally readable WAV flavor.
+/**
+ * Serialize an AudioBuffer to a 16-bit PCM (s16le) WAV file as raw bytes.
+ * Channels are interleaved; sample rate / channel count come from the buffer.
+ */
+export function audioBufferToWav(buffer) {
+    const numChannels = buffer.numberOfChannels;
+    const sampleRate = buffer.sampleRate;
+    const numFrames = buffer.length;
+    const bytesPerSample = 2; // s16le
+    const blockAlign = numChannels * bytesPerSample;
+    const dataSize = numFrames * blockAlign;
+    const bufferSize = 44 + dataSize; // 44-byte canonical WAV header + PCM data
+    const out = new ArrayBuffer(bufferSize);
+    const view = new DataView(out);
+    const writeAscii = (offset, text) => {
+        for (let i = 0; i < text.length; i++)
+            view.setUint8(offset + i, text.charCodeAt(i));
+    };
+    // RIFF chunk descriptor
+    writeAscii(0, 'RIFF');
+    view.setUint32(4, 36 + dataSize, true); // file size minus the first 8 bytes
+    writeAscii(8, 'WAVE');
+    // fmt subchunk (PCM)
+    writeAscii(12, 'fmt ');
+    view.setUint32(16, 16, true); // subchunk1 size (16 for PCM)
+    view.setUint16(20, 1, true); // audio format = 1 (PCM)
+    view.setUint16(22, numChannels, true);
+    view.setUint32(24, sampleRate, true);
+    view.setUint32(28, sampleRate * blockAlign, true); // byte rate
+    view.setUint16(32, blockAlign, true);
+    view.setUint16(34, bytesPerSample * 8, true); // bits per sample
+    // data subchunk
+    writeAscii(36, 'data');
+    view.setUint32(40, dataSize, true);
+    // Interleave channels and write clamped 16-bit samples.
+    const channelData = [];
+    for (let c = 0; c < numChannels; c++)
+        channelData.push(buffer.getChannelData(c));
+    let offset = 44;
+    for (let frame = 0; frame < numFrames; frame++) {
+        for (let c = 0; c < numChannels; c++) {
+            let s = channelData[c][frame];
+            // Clamp to [-1, 1] then scale to the signed 16-bit range.
+            s = s < -1 ? -1 : s > 1 ? 1 : s;
+            view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);
+            offset += 2;
+        }
+    }
+    return out;
+}
+//# sourceMappingURL=wav.js.map

package/dist/audio/wav.js.map

@@ -0,0 +1 @@
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