@kano/stem-daw 0.1.0

package/dist/chunk-OFGZURP6.js

@@ -0,0 +1,247 @@
+// src/daw/utils/sampler-audio.ts
+var CROSSFADE_SAMPLES = 256;
+function fracBarToSample(bar, barMapping, bufLen) {
+  if (barMapping.length === 0) return 0;
+  const floorBar = Math.floor(bar);
+  const frac = bar - floorBar;
+  if (floorBar >= barMapping.length) return bufLen;
+  const entry = barMapping[Math.max(0, floorBar)];
+  return Math.round(entry.sampleStart + frac * entry.duration);
+}
+function extractRegion(source, sampleStart, sampleEnd, ctx) {
+  const len = Math.max(0, sampleEnd - sampleStart);
+  if (len === 0) return ctx.createBuffer(source.numberOfChannels, 1, source.sampleRate);
+  const out = ctx.createBuffer(source.numberOfChannels, len, source.sampleRate);
+  for (let ch = 0; ch < source.numberOfChannels; ch++) {
+    const src = source.getChannelData(ch);
+    const dst = out.getChannelData(ch);
+    dst.set(src.subarray(sampleStart, sampleStart + len));
+  }
+  return out;
+}
+function reverseBuffer(buffer) {
+  for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
+    buffer.getChannelData(ch).reverse();
+  }
+  return buffer;
+}
+function applyGain(buffer, gain) {
+  if (gain === 1) return buffer;
+  for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
+    const data = buffer.getChannelData(ch);
+    for (let i = 0; i < data.length; i++) data[i] *= gain;
+  }
+  return buffer;
+}
+function extractSliceWithTrim(sourceBuffer, slice, startOffset, endOffset, ctx) {
+  const sliceLen = slice.sampleEnd - slice.sampleStart;
+  const trimStart = slice.sampleStart + Math.round(sliceLen * startOffset);
+  const trimEnd = slice.sampleStart + Math.round(sliceLen * (1 - endOffset));
+  const len = Math.max(1, trimEnd - trimStart);
+  const out = ctx.createBuffer(sourceBuffer.numberOfChannels, len, sourceBuffer.sampleRate);
+  for (let ch = 0; ch < sourceBuffer.numberOfChannels; ch++) {
+    const src = sourceBuffer.getChannelData(ch);
+    out.getChannelData(ch).set(src.subarray(trimStart, trimStart + len));
+  }
+  return out;
+}
+async function pitchShiftClassic(buffer, semitones) {
+  if (semitones === 0) return buffer;
+  const rate = Math.pow(2, semitones / 12);
+  const newLength = Math.max(1, Math.round(buffer.length / rate));
+  const offline = new OfflineAudioContext(buffer.numberOfChannels, newLength, buffer.sampleRate);
+  const source = offline.createBufferSource();
+  source.buffer = buffer;
+  source.playbackRate.value = rate;
+  source.connect(offline.destination);
+  source.start();
+  return offline.startRendering();
+}
+async function pitchShiftFormant(buffer, semitones, _ctx) {
+  if (semitones === 0) return buffer;
+  return pitchShiftClassic(buffer, semitones);
+}
+async function renderSequenceAsync(sourceBuffer, sequence, slices, ctx, pitchMode = "classic") {
+  const sliceMap = new Map(slices.map((s) => [s.id, s]));
+  const renderedParts = [];
+  let totalSamples = 0;
+  for (const entry of sequence) {
+    const slice = sliceMap.get(entry.sliceId);
+    if (!slice) continue;
+    let part = extractSliceWithTrim(
+      sourceBuffer,
+      slice,
+      entry.startOffset ?? 0,
+      entry.endOffset ?? 0,
+      ctx
+    );
+    if (entry.reverse) part = reverseBuffer(part);
+    if (entry.pitchSemitones && entry.pitchSemitones !== 0) {
+      part = pitchMode === "formant" ? await pitchShiftFormant(part, entry.pitchSemitones) : await pitchShiftClassic(part, entry.pitchSemitones);
+    }
+    if (entry.gain !== void 0 && entry.gain !== 1) {
+      part = applyGain(part, entry.gain);
+    }
+    renderedParts.push(part);
+    totalSamples += part.length;
+  }
+  if (totalSamples === 0) {
+    return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);
+  }
+  const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);
+  let writePos = 0;
+  for (const part of renderedParts) {
+    for (let ch = 0; ch < output.numberOfChannels; ch++) {
+      const dst = output.getChannelData(ch);
+      const src = part.numberOfChannels > ch ? part.getChannelData(ch) : part.getChannelData(0);
+      dst.set(src, writePos);
+    }
+    writePos += part.length;
+  }
+  applyCrossfades(output, renderedParts);
+  return output;
+}
+function renderSequence(sourceBuffer, sequence, slices, ctx) {
+  const hasParams = sequence.some(
+    (e) => e.pitchSemitones && e.pitchSemitones !== 0 || e.gain !== void 0 && e.gain !== 1 || e.reverse || e.startOffset && e.startOffset > 0 || e.endOffset && e.endOffset > 0
+  );
+  if (hasParams) {
+    console.warn("[sampler-audio] renderSequence called with per-entry params \u2014 use renderSequenceAsync instead");
+  }
+  const sliceMap = new Map(slices.map((s) => [s.id, s]));
+  let totalSamples = 0;
+  for (const entry of sequence) {
+    const slice = sliceMap.get(entry.sliceId);
+    if (slice) totalSamples += slice.sampleEnd - slice.sampleStart;
+  }
+  if (totalSamples === 0) {
+    return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);
+  }
+  const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);
+  let writePos = 0;
+  const parts = [];
+  for (const entry of sequence) {
+    const slice = sliceMap.get(entry.sliceId);
+    if (!slice) continue;
+    const len = slice.sampleEnd - slice.sampleStart;
+    const part = ctx.createBuffer(sourceBuffer.numberOfChannels, len, sourceBuffer.sampleRate);
+    for (let ch = 0; ch < sourceBuffer.numberOfChannels; ch++) {
+      const src = sourceBuffer.getChannelData(ch);
+      const dst = output.getChannelData(ch);
+      dst.set(src.subarray(slice.sampleStart, slice.sampleStart + len), writePos);
+      part.getChannelData(ch).set(src.subarray(slice.sampleStart, slice.sampleStart + len));
+    }
+    parts.push(part);
+    writePos += len;
+  }
+  applyCrossfades(output, parts);
+  return output;
+}
+function applyCrossfades(buffer, parts) {
+  const fade = Math.min(CROSSFADE_SAMPLES, Math.floor(buffer.length / (parts.length * 4)));
+  if (fade < 2) return;
+  let pos = 0;
+  for (let i = 0; i < parts.length; i++) {
+    const len = parts[i].length;
+    if (i > 0 && pos > 0) {
+      for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
+        const data = buffer.getChannelData(ch);
+        for (let s = 0; s < fade; s++) {
+          const t = s / fade;
+          const idx = pos + s;
+          if (idx < buffer.length) data[idx] *= t;
+          const prevIdx = pos - fade + s;
+          if (prevIdx >= 0 && prevIdx < buffer.length) data[prevIdx] *= 1 - (1 - t) * 0.3;
+        }
+      }
+    }
+    pos += len;
+  }
+}
+async function renderPianoRollAsync(sourceBuffer, notes, rootPitchMidi, bpm, bars, quantizeGrid, ctx) {
+  const stepsPerBeatMap = { "1/4": 1, "1/8": 2, "1/16": 4, "1/32": 8, off: 4 };
+  const stepsPerBeat = stepsPerBeatMap[quantizeGrid] ?? 4;
+  const totalSteps = bars * 4 * stepsPerBeat;
+  const stepDurationSec = 60 / bpm / stepsPerBeat;
+  const totalDurationSec = totalSteps * stepDurationSec;
+  const totalSamples = Math.ceil(totalDurationSec * sourceBuffer.sampleRate);
+  if (notes.length === 0 || totalSamples === 0) {
+    return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);
+  }
+  const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);
+  const ATTACK_SEC = 3e-3;
+  const RELEASE_SEC = 0.02;
+  const tailSamples = Math.ceil(RELEASE_SEC * sourceBuffer.sampleRate);
+  for (const note of notes) {
+    const semitones = note.pitch - rootPitchMidi;
+    const rate = Math.pow(2, semitones / 12);
+    const noteDurationSec = note.durationSteps * stepDurationSec;
+    const noteStartSample = Math.round(note.startStep * stepDurationSec * sourceBuffer.sampleRate);
+    const pitchedSourceLen = Math.max(1, Math.round(sourceBuffer.length / rate));
+    const noteLenSamples = Math.max(1, Math.round(noteDurationSec * sourceBuffer.sampleRate));
+    const renderLen = Math.max(1, Math.min(pitchedSourceLen, noteLenSamples + tailSamples));
+    const offline = new OfflineAudioContext(sourceBuffer.numberOfChannels, renderLen, sourceBuffer.sampleRate);
+    const src = offline.createBufferSource();
+    src.buffer = sourceBuffer;
+    src.playbackRate.value = rate;
+    const gainNode = offline.createGain();
+    const velGain = note.velocity / 127;
+    const renderDurSec = renderLen / sourceBuffer.sampleRate;
+    const atk = Math.min(ATTACK_SEC, renderDurSec * 0.25);
+    const rel = Math.min(RELEASE_SEC, renderDurSec * 0.5);
+    const relStart = Math.max(atk, renderDurSec - rel);
+    gainNode.gain.setValueAtTime(0, 0);
+    gainNode.gain.linearRampToValueAtTime(velGain, atk);
+    gainNode.gain.setValueAtTime(velGain, relStart);
+    gainNode.gain.linearRampToValueAtTime(0, renderDurSec);
+    src.connect(gainNode);
+    gainNode.connect(offline.destination);
+    src.start();
+    const rendered = await offline.startRendering();
+    for (let ch = 0; ch < output.numberOfChannels; ch++) {
+      const dst = output.getChannelData(ch);
+      const srcData = rendered.numberOfChannels > ch ? rendered.getChannelData(ch) : rendered.getChannelData(0);
+      const copyLen = Math.min(srcData.length, totalSamples - noteStartSample);
+      for (let i = 0; i < copyLen; i++) {
+        dst[noteStartSample + i] += srcData[i];
+      }
+    }
+  }
+  let peak = 0;
+  for (let ch = 0; ch < output.numberOfChannels; ch++) {
+    const data = output.getChannelData(ch);
+    for (let i = 0; i < data.length; i++) {
+      const abs = Math.abs(data[i]);
+      if (abs > peak) peak = abs;
+    }
+  }
+  if (peak > 1) {
+    const scale = 0.95 / peak;
+    for (let ch = 0; ch < output.numberOfChannels; ch++) {
+      const data = output.getChannelData(ch);
+      for (let i = 0; i < data.length; i++) data[i] *= scale;
+    }
+  }
+  return output;
+}
+function spliceBuffer(original, insertAtSample, replaceLength, replacement, ctx) {
+  const newLen = original.length - replaceLength + replacement.length;
+  const out = ctx.createBuffer(original.numberOfChannels, Math.max(newLen, 1), original.sampleRate);
+  for (let ch = 0; ch < original.numberOfChannels; ch++) {
+    const src = original.getChannelData(ch);
+    const dst = out.getChannelData(ch);
+    const rep = replacement.numberOfChannels > ch ? replacement.getChannelData(ch) : replacement.getChannelData(0);
+    dst.set(src.subarray(0, insertAtSample));
+    dst.set(rep, insertAtSample);
+    const afterOriginal = insertAtSample + replaceLength;
+    const afterNew = insertAtSample + replacement.length;
+    if (afterOriginal < src.length) {
+      dst.set(src.subarray(afterOriginal), afterNew);
+    }
+  }
+  return out;
+}
+
+export { extractRegion, fracBarToSample, renderPianoRollAsync, renderSequence, renderSequenceAsync, spliceBuffer };
+//# sourceMappingURL=chunk-OFGZURP6.js.map
+//# sourceMappingURL=chunk-OFGZURP6.js.map

package/dist/chunk-OFGZURP6.js.map

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* Audio processing utilities for the Bar Sampler.\n * Handles buffer extraction, sequence rendering with per-entry pitch/gain/reverse,\n * and crossfades.\n */\n\nimport type { SamplerSlice, SamplerSequenceEntry, PianoRollNote } from '../interfaces';\nimport type { QuantizeGrid } from '../store/sampler-store';\n\nconst CROSSFADE_SAMPLES = 256;\n\n/** Convert a fractional bar position to a sample offset (for beat-level extraction). */\nexport function fracBarToSample(\n    bar: number,\n    barMapping: { sampleStart: number; duration: number }[],\n    bufLen: number,\n): number {\n    if (barMapping.length === 0) return 0;\n    const floorBar = Math.floor(bar);\n    const frac = bar - floorBar;\n    if (floorBar >= barMapping.length) return bufLen;\n    const entry = barMapping[Math.max(0, floorBar)];\n    return Math.round(entry.sampleStart + frac * entry.duration);\n}\n\n/** Extract a sample range from an AudioBuffer into a new AudioBuffer. */\nexport function extractRegion(\n    source: AudioBuffer,\n    sampleStart: number,\n    sampleEnd: number,\n    ctx: AudioContext,\n): AudioBuffer {\n    const len = Math.max(0, sampleEnd - sampleStart);\n    if (len === 0) return ctx.createBuffer(source.numberOfChannels, 1, source.sampleRate);\n    const out = ctx.createBuffer(source.numberOfChannels, len, source.sampleRate);\n    for (let ch = 0; ch < source.numberOfChannels; ch++) {\n        const src = source.getChannelData(ch);\n        const dst = out.getChannelData(ch);\n        dst.set(src.subarray(sampleStart, sampleStart + len));\n    }\n    return out;\n}\n\n/** Reverse an AudioBuffer's channel data in-place and return it. */\nfunction reverseBuffer(buffer: AudioBuffer): AudioBuffer {\n    for (let ch = 0; ch < buffer.numberOfChannels; ch++) {\n        buffer.getChannelData(ch).reverse();\n    }\n    return buffer;\n}\n\n/** Apply gain to an AudioBuffer's channel data in-place. */\nfunction applyGain(buffer: AudioBuffer, gain: number): AudioBuffer {\n    if (gain === 1) return buffer;\n    for (let ch = 0; ch < buffer.numberOfChannels; ch++) {\n        const data = buffer.getChannelData(ch);\n        for (let i = 0; i < data.length; i++) data[i] *= gain;\n    }\n    return buffer;\n}\n\n/**\n * Extract a slice region from the source buffer, applying trim offsets.\n * startOffset/endOffset are 0..1 fractions within the slice's sample range.\n */\nfunction extractSliceWithTrim(\n    sourceBuffer: AudioBuffer,\n    slice: SamplerSlice,\n    startOffset: number,\n    endOffset: number,\n    ctx: BaseAudioContext,\n): AudioBuffer {\n    const sliceLen = slice.sampleEnd - slice.sampleStart;\n    const trimStart = slice.sampleStart + Math.round(sliceLen * startOffset);\n    const trimEnd = slice.sampleStart + Math.round(sliceLen * (1 - endOffset));\n    const len = Math.max(1, trimEnd - trimStart);\n    const out = ctx.createBuffer(sourceBuffer.numberOfChannels, len, sourceBuffer.sampleRate);\n    for (let ch = 0; ch < sourceBuffer.numberOfChannels; ch++) {\n        const src = sourceBuffer.getChannelData(ch);\n        out.getChannelData(ch).set(src.subarray(trimStart, trimStart + len));\n    }\n    return out;\n}\n\n/** Pitch engine: classic = speed+pitch together (resampler), formant = preserve length */\nexport type SamplerPitchMode = 'classic' | 'formant';\n\n/**\n * Pitch-shift a buffer using playbackRate via OfflineAudioContext.\n * Classic sampler behaviour: pitch and speed change together.\n */\nasync function pitchShiftClassic(\n    buffer: AudioBuffer,\n    semitones: number,\n): Promise<AudioBuffer> {\n    if (semitones === 0) return buffer;\n    const rate = Math.pow(2, semitones / 12);\n    const newLength = Math.max(1, Math.round(buffer.length / rate));\n    const offline = new OfflineAudioContext(buffer.numberOfChannels, newLength, buffer.sampleRate);\n    const source = offline.createBufferSource();\n    source.buffer = buffer;\n    source.playbackRate.value = rate;\n    source.connect(offline.destination);\n    source.start();\n    return offline.startRendering();\n}\n\n/**\n * Pitch-shift preserving duration (formant-preserving).\n * TODO: Use phase-vocoder or Rubberband for true formant preservation.\n * For now falls back to classic for consistency.\n */\nasync function pitchShiftFormant(\n    buffer: AudioBuffer,\n    semitones: number,\n    _ctx: BaseAudioContext,\n): Promise<AudioBuffer> {\n    if (semitones === 0) return buffer;\n    // Placeholder: use classic until formant-preserving algo is implemented\n    return pitchShiftClassic(buffer, semitones);\n}\n\n/**\n * Render an arranged sequence of slices into a single AudioBuffer.\n * Supports per-entry pitch shifting, gain, reverse, and trim offsets.\n * Uses classic (playbackRate) pitch shifting for real-time sampler character.\n */\nexport async function renderSequenceAsync(\n    sourceBuffer: AudioBuffer,\n    sequence: SamplerSequenceEntry[],\n    slices: SamplerSlice[],\n    ctx: AudioContext,\n    pitchMode: SamplerPitchMode = 'classic',\n): Promise<AudioBuffer> {\n    const sliceMap = new Map(slices.map((s) => [s.id, s]));\n\n    const renderedParts: AudioBuffer[] = [];\n    let totalSamples = 0;\n\n    for (const entry of sequence) {\n        const slice = sliceMap.get(entry.sliceId);\n        if (!slice) continue;\n\n        let part = extractSliceWithTrim(\n            sourceBuffer, slice,\n            entry.startOffset ?? 0,\n            entry.endOffset ?? 0,\n            ctx,\n        );\n\n        if (entry.reverse) part = reverseBuffer(part);\n        if (entry.pitchSemitones && entry.pitchSemitones !== 0) {\n            part = pitchMode === 'formant'\n                ? await pitchShiftFormant(part, entry.pitchSemitones, ctx)\n                : await pitchShiftClassic(part, entry.pitchSemitones);\n        }\n        if (entry.gain !== undefined && entry.gain !== 1) {\n            part = applyGain(part, entry.gain);\n        }\n\n        renderedParts.push(part);\n        totalSamples += part.length;\n    }\n\n    if (totalSamples === 0) {\n        return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);\n    }\n\n    const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);\n    let writePos = 0;\n    for (const part of renderedParts) {\n        for (let ch = 0; ch < output.numberOfChannels; ch++) {\n            const dst = output.getChannelData(ch);\n            const src = part.numberOfChannels > ch ? part.getChannelData(ch) : part.getChannelData(0);\n            dst.set(src, writePos);\n        }\n        writePos += part.length;\n    }\n\n    applyCrossfades(output, renderedParts);\n    return output;\n}\n\n/**\n * Synchronous render for backwards compatibility (no pitch/gain/reverse applied).\n * Used when no entries have per-entry params set.\n */\nexport function renderSequence(\n    sourceBuffer: AudioBuffer,\n    sequence: SamplerSequenceEntry[],\n    slices: SamplerSlice[],\n    ctx: AudioContext,\n): AudioBuffer {\n    const hasParams = sequence.some((e) =>\n        (e.pitchSemitones && e.pitchSemitones !== 0) ||\n        (e.gain !== undefined && e.gain !== 1) ||\n        e.reverse ||\n        (e.startOffset && e.startOffset > 0) ||\n        (e.endOffset && e.endOffset > 0)\n    );\n\n    if (hasParams) {\n        console.warn('[sampler-audio] renderSequence called with per-entry params — use renderSequenceAsync instead');\n    }\n\n    const sliceMap = new Map(slices.map((s) => [s.id, s]));\n\n    let totalSamples = 0;\n    for (const entry of sequence) {\n        const slice = sliceMap.get(entry.sliceId);\n        if (slice) totalSamples += slice.sampleEnd - slice.sampleStart;\n    }\n\n    if (totalSamples === 0) {\n        return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);\n    }\n\n    const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);\n    let writePos = 0;\n    const parts: AudioBuffer[] = [];\n    for (const entry of sequence) {\n        const slice = sliceMap.get(entry.sliceId);\n        if (!slice) continue;\n        const len = slice.sampleEnd - slice.sampleStart;\n        const part = ctx.createBuffer(sourceBuffer.numberOfChannels, len, sourceBuffer.sampleRate);\n        for (let ch = 0; ch < sourceBuffer.numberOfChannels; ch++) {\n            const src = sourceBuffer.getChannelData(ch);\n            const dst = output.getChannelData(ch);\n            dst.set(src.subarray(slice.sampleStart, slice.sampleStart + len), writePos);\n            part.getChannelData(ch).set(src.subarray(slice.sampleStart, slice.sampleStart + len));\n        }\n        parts.push(part);\n        writePos += len;\n    }\n\n    applyCrossfades(output, parts);\n    return output;\n}\n\nfunction applyCrossfades(\n    buffer: AudioBuffer,\n    parts: AudioBuffer[],\n): void {\n    const fade = Math.min(CROSSFADE_SAMPLES, Math.floor(buffer.length / (parts.length * 4)));\n    if (fade < 2) return;\n\n    let pos = 0;\n    for (let i = 0; i < parts.length; i++) {\n        const len = parts[i].length;\n\n        if (i > 0 && pos > 0) {\n            for (let ch = 0; ch < buffer.numberOfChannels; ch++) {\n                const data = buffer.getChannelData(ch);\n                for (let s = 0; s < fade; s++) {\n                    const t = s / fade;\n                    const idx = pos + s;\n                    if (idx < buffer.length) data[idx] *= t;\n                    const prevIdx = pos - fade + s;\n                    if (prevIdx >= 0 && prevIdx < buffer.length) data[prevIdx] *= 1 - (1 - t) * 0.3;\n                }\n            }\n        }\n        pos += len;\n    }\n}\n\n/**\n * Render piano roll notes into a single AudioBuffer.\n * Each note is a pitch-shifted copy of the source sample (or first slice),\n * placed at its grid-quantized position. Polyphonic — overlapping notes are mixed.\n *\n * @param sourceBuffer   The sample audio to pitch-shift\n * @param notes          Piano roll note events\n * @param rootPitchMidi  Detected root pitch of the source (default 60 = C4)\n * @param bpm            Tempo in beats per minute\n * @param bars           Number of bars the piano roll spans\n * @param quantizeGrid   Grid resolution\n * @param ctx            AudioContext for buffer creation\n */\nexport async function renderPianoRollAsync(\n    sourceBuffer: AudioBuffer,\n    notes: PianoRollNote[],\n    rootPitchMidi: number,\n    bpm: number,\n    bars: number,\n    quantizeGrid: QuantizeGrid,\n    ctx: AudioContext,\n): Promise<AudioBuffer> {\n    const stepsPerBeatMap: Record<string, number> = { '1/4': 1, '1/8': 2, '1/16': 4, '1/32': 8, off: 4 };\n    const stepsPerBeat = stepsPerBeatMap[quantizeGrid] ?? 4;\n    const totalSteps = bars * 4 * stepsPerBeat;\n    const stepDurationSec = 60 / bpm / stepsPerBeat;\n    const totalDurationSec = totalSteps * stepDurationSec;\n    const totalSamples = Math.ceil(totalDurationSec * sourceBuffer.sampleRate);\n\n    if (notes.length === 0 || totalSamples === 0) {\n        return ctx.createBuffer(sourceBuffer.numberOfChannels, 1, sourceBuffer.sampleRate);\n    }\n\n    const output = ctx.createBuffer(sourceBuffer.numberOfChannels, totalSamples, sourceBuffer.sampleRate);\n\n    // Short fades around each note remove the clicks / pops caused by\n    // hard-cutting the pitch-shifted source mid-waveform.\n    const ATTACK_SEC = 0.003;\n    const RELEASE_SEC = 0.02;\n    const tailSamples = Math.ceil(RELEASE_SEC * sourceBuffer.sampleRate);\n\n    for (const note of notes) {\n        const semitones = note.pitch - rootPitchMidi;\n        const rate = Math.pow(2, semitones / 12);\n        const noteDurationSec = note.durationSteps * stepDurationSec;\n        const noteStartSample = Math.round(note.startStep * stepDurationSec * sourceBuffer.sampleRate);\n\n        const pitchedSourceLen = Math.max(1, Math.round(sourceBuffer.length / rate));\n        const noteLenSamples = Math.max(1, Math.round(noteDurationSec * sourceBuffer.sampleRate));\n        // Let the note ring slightly past its nominal end so the release fade\n        // doesn't eat into the audible portion of the note.\n        const renderLen = Math.max(1, Math.min(pitchedSourceLen, noteLenSamples + tailSamples));\n\n        const offline = new OfflineAudioContext(sourceBuffer.numberOfChannels, renderLen, sourceBuffer.sampleRate);\n        const src = offline.createBufferSource();\n        src.buffer = sourceBuffer;\n        src.playbackRate.value = rate;\n\n        const gainNode = offline.createGain();\n        const velGain = note.velocity / 127;\n        const renderDurSec = renderLen / sourceBuffer.sampleRate;\n        const atk = Math.min(ATTACK_SEC, renderDurSec * 0.25);\n        const rel = Math.min(RELEASE_SEC, renderDurSec * 0.5);\n        const relStart = Math.max(atk, renderDurSec - rel);\n        gainNode.gain.setValueAtTime(0, 0);\n        gainNode.gain.linearRampToValueAtTime(velGain, atk);\n        gainNode.gain.setValueAtTime(velGain, relStart);\n        gainNode.gain.linearRampToValueAtTime(0, renderDurSec);\n\n        src.connect(gainNode);\n        gainNode.connect(offline.destination);\n        src.start();\n\n        const rendered = await offline.startRendering();\n\n        // Mix into output (additive)\n        for (let ch = 0; ch < output.numberOfChannels; ch++) {\n            const dst = output.getChannelData(ch);\n            const srcData = rendered.numberOfChannels > ch ? rendered.getChannelData(ch) : rendered.getChannelData(0);\n            const copyLen = Math.min(srcData.length, totalSamples - noteStartSample);\n            for (let i = 0; i < copyLen; i++) {\n                dst[noteStartSample + i] += srcData[i];\n            }\n        }\n    }\n\n    // Normalize if any clipping\n    let peak = 0;\n    for (let ch = 0; ch < output.numberOfChannels; ch++) {\n        const data = output.getChannelData(ch);\n        for (let i = 0; i < data.length; i++) {\n            const abs = Math.abs(data[i]);\n            if (abs > peak) peak = abs;\n        }\n    }\n    if (peak > 1) {\n        const scale = 0.95 / peak;\n        for (let ch = 0; ch < output.numberOfChannels; ch++) {\n            const data = output.getChannelData(ch);\n            for (let i = 0; i < data.length; i++) data[i] *= scale;\n        }\n    }\n\n    return output;\n}\n\n/**\n * Splice a resampled buffer into a full stem buffer at a given sample position.\n * Returns a new AudioBuffer with the resampled content inserted.\n */\nexport function spliceBuffer(\n    original: AudioBuffer,\n    insertAtSample: number,\n    replaceLength: number,\n    replacement: AudioBuffer,\n    ctx: AudioContext,\n): AudioBuffer {\n    const newLen = original.length - replaceLength + replacement.length;\n    const out = ctx.createBuffer(original.numberOfChannels, Math.max(newLen, 1), original.sampleRate);\n\n    for (let ch = 0; ch < original.numberOfChannels; ch++) {\n        const src = original.getChannelData(ch);\n        const dst = out.getChannelData(ch);\n        const rep = replacement.numberOfChannels > ch ? replacement.getChannelData(ch) : replacement.getChannelData(0);\n\n        dst.set(src.subarray(0, insertAtSample));\n        dst.set(rep, insertAtSample);\n        const afterOriginal = insertAtSample + replaceLength;\n        const afterNew = insertAtSample + replacement.length;\n        if (afterOriginal < src.length) {\n            dst.set(src.subarray(afterOriginal), afterNew);\n        }\n    }\n    return out;\n}\n"]}