@kano/stem-daw 0.1.0

package/dist/chunk-OYNES5W3.js.map

@@ -0,0 +1 @@
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* Minimal browser-compatible EventEmitter replacement.\n * Avoids importing Node.js \"events\" module which Vite externalises.\n */\n\ntype Listener = (...args: any[]) => void;\n\nexport class BrowserEventEmitter {\n    private _listeners: Map<string, Set<Listener>> = new Map();\n\n    on(event: string, listener: Listener): this {\n        if (!this._listeners.has(event)) {\n            this._listeners.set(event, new Set());\n        }\n        this._listeners.get(event)!.add(listener);\n        return this;\n    }\n\n    off(event: string, listener: Listener): this {\n        this._listeners.get(event)?.delete(listener);\n        return this;\n    }\n\n    once(event: string, listener: Listener): this {\n        const wrapper: Listener = (...args) => {\n            this.off(event, wrapper);\n            listener(...args);\n        };\n        return this.on(event, wrapper);\n    }\n\n    emit(event: string, ...args: any[]): boolean {\n        const listeners = this._listeners.get(event);\n        if (!listeners || listeners.size === 0) return false;\n        for (const listener of listeners) {\n            try { listener(...args); } catch (e) { console.error(`EventEmitter error on \"${event}\":`, e); }\n        }\n        return true;\n    }\n\n    removeAllListeners(event?: string): this {\n        if (event) {\n            this._listeners.delete(event);\n        } else {\n            this._listeners.clear();\n        }\n        return this;\n    }\n}\n","/**\n * Progressive Track Loader — streams fMP4 stems segment-by-segment so a\n * newly added track can appear on the timeline instantly (bars/sections/beats\n * come from trackData metadata) while the audio fills in in the background.\n *\n * Pipeline per stem:\n *   TFRA index → batched HTTP Range fetch → MP4MoofExtractor (per segment)\n *     → AACAdtsPackager (per segment) → WebCodecs AudioDecoder (one per stem,\n *     fed in order so cross-batch alignment stays sample-accurate)\n *     → StemPatch emitted to subscribers\n *\n * Encoder priming: FDK-AAC prepends 2112 priming samples. We skip those once\n * per stem decoder so sample positions match the original source (matching\n * the trimEncoderDelay behaviour in daw-track-loader.ts).\n *\n * If WebCodecs AudioDecoder is unavailable (Safari < 16.4), the caller is\n * expected to fall back to the legacy one-shot path.\n */\n\nimport { extractTfraData, type TfraExtractResult } from '@/stem-audio-engine-v3/utils/mp4';\nimport { MP4MoofExtractor } from '@/stem-audio-engine-v3/utils/mp4-parser/mp4-moof-extractor';\nimport { AACAdtsPackager } from '@/stem-audio-engine-v3/utils/mp4-parser/aac-adts-packager';\nimport { AEV3Config } from '@/stem-audio-engine-v3/aev3-env-config';\nimport { cacheAudioBuffer, getCachedAudioBuffer } from './daw-cache';\n\nconst SAMPLE_RATE = 48000;\n/** FDK-AAC encoder priming samples (2048 look-ahead + 64 SBR overlap) */\nconst AAC_ENCODER_DELAY_SAMPLES = 2112;\n/** Segments per batch — ~1-2s of audio per segment so 8 ≈ 8-16s per batch */\nconst BATCH_SEGMENTS = 8;\n/** Threshold for resolving the `playable` promise — first 8s decoded */\nconst PLAYABLE_THRESHOLD_SAMPLES = 8 * SAMPLE_RATE;\n/** One AAC frame = 1024 PCM samples at 48kHz */\nconst AAC_FRAME_SAMPLES = 1024;\n\nexport type StemLabel = 'bass' | 'drums' | 'other' | 'vocals';\nconst STEM_LABELS: readonly StemLabel[] = ['bass', 'drums', 'other', 'vocals'] as const;\n/** TFRA stream index → stem label (matches daw-track-loader TFRA_STEM_MAP) */\nconst TFRA_STEM_MAP: Record<number, StemLabel> = {\n    0: 'bass',\n    1: 'drums',\n    2: 'other',\n    3: 'vocals',\n};\n\nexport type LoadProgress = {\n    phase: 'fetching' | 'parsing' | 'decoding';\n    percent: number;\n    detail?: string;\n};\n\nexport interface StemPatch {\n    stem: StemLabel;\n    /** First sample index of this patch in the pre-allocated stem buffer */\n    sampleOffset: number;\n    /** [L, R] PCM data for this slot */\n    samples: Float32Array[];\n}\n\nexport interface ProgressiveLoadHandle {\n    /** Per-stem total samples in the pre-allocated buffer */\n    totalSamples: number;\n    /** Resolves after every stem has decoded at least PLAYABLE_THRESHOLD_SAMPLES */\n    playable: Promise<void>;\n    /** Resolves when every batch has been decoded */\n    complete: Promise<void>;\n    /** Cancel pending fetches/decodes (called from removeTrack) */\n    abort(): void;\n    /** Subscribe to per-stem patches as they arrive. Returns unsubscribe fn. */\n    onPatch(cb: (p: StemPatch) => void): () => void;\n    /** Subscribe to overall progress (0-100 percent). Returns unsubscribe fn. */\n    onProgress(cb: (p: LoadProgress) => void): () => void;\n}\n\n/**\n * Feature detect WebCodecs AudioDecoder. Callers should fall back to the\n * legacy non-progressive loader when this returns false.\n */\nexport function isProgressiveDecodeSupported(): boolean {\n    return typeof (globalThis as { AudioDecoder?: unknown }).AudioDecoder !== 'undefined';\n}\n\n/**\n * Derive the total expected sample count per stem from TFRA.\n *\n * Each TFRA entry's `time` field is in MP4 timescale (which matches the\n * 48 kHz sample rate of these encodes — see SAMPLE_RATE_OF_TIME_PROP in\n * stem-audio-engine-v3/utils/mp4.ts). We decode segments 0..N-2 (the last\n * entry has no \"next\" entry to bound its moof+mdat), so the cumulative\n * sample length is the last decoded segment's end time, which equals\n * `entries[N-1].time`.\n *\n * After the FDK-AAC priming trim (AAC_ENCODER_DELAY_SAMPLES), the audible\n * length is approximately `entries[N-1].time - AAC_ENCODER_DELAY_SAMPLES`.\n * We size the buffer to the LARGER of TFRA-derived and duration-derived\n * (plus a small safety margin) so we never silently truncate the tail.\n */\nexport function deriveTotalSamplesFromTfra(\n    tfraData: TfraExtractResult,\n    expectedDurationSec: number,\n): number {\n    const entries = tfraData[0]?.entries ?? [];\n    const lastIdx = entries.length - 1;\n    const tfraSamples = lastIdx >= 0 ? entries[lastIdx].time : 0;\n    const durationSamples = Math.ceil(expectedDurationSec * SAMPLE_RATE);\n    // 1s safety margin handles small rounding / priming-trim mismatches\n    return Math.max(tfraSamples, durationSamples) + SAMPLE_RATE;\n}\n\ninterface BatchPlan {\n    /** Inclusive segment index in tfraData.entries */\n    startSeg: number;\n    /** Exclusive segment index */\n    endSeg: number;\n    /** Byte range of the contiguous chunk that covers all 4 stems for these segments */\n    byteStart: number;\n    byteEnd: number;\n}\n\ninterface StemDecodeState {\n    decoder: AudioDecoder;\n    /** Where the next decoded chunk should be written in the output buffer */\n    writeOffset: number;\n    /** Remaining encoder-priming samples to skip (initially AAC_ENCODER_DELAY_SAMPLES) */\n    samplesToSkip: number;\n    /** Pending samples for the in-flight batch (filled by output callback) */\n    pendingChunks: Float32Array[][];\n    /** Sample offset where the current batch began writing */\n    batchSampleStart: number;\n    /** Total samples emitted via patches (decoded watermark) */\n    decodedSamples: number;\n    /** Monotonic timestamp counter for EncodedAudioChunk timestamps (microseconds) */\n    timestampUs: number;\n}\n\n/**\n * Group segment indices into BATCH_SEGMENTS-sized batches and pre-compute\n * the byte range that covers all 4 stems' moof+mdat for those segments.\n *\n * fMP4 layout per segment (with separate_moof):\n *   [bass moof+mdat][drums moof+mdat][other moof+mdat][vocals moof+mdat]\n *\n * The combined chunk for segments [s, s+N) spans:\n *   tfraData[0].entries[s].moof_offset  ..  tfraData[0].entries[s+N].moof_offset\n */\nfunction planBatches(tfraData: TfraExtractResult, totalFileBytes: number | null): BatchPlan[] {\n    const entryCount = tfraData[0].entries.length;\n    if (entryCount < 2) return [];\n\n    const usableEntries = entryCount - 1;\n    const batches: BatchPlan[] = [];\n    for (let s = 0; s < usableEntries; s += BATCH_SEGMENTS) {\n        const startSeg = s;\n        const endSeg = Math.min(s + BATCH_SEGMENTS, usableEntries);\n\n        const byteStart = tfraData[0].entries[startSeg].moof_offset;\n        const nextSegMoofOffset = endSeg < entryCount\n            ? tfraData[0].entries[endSeg].moof_offset\n            : totalFileBytes ?? tfraData[0].entries[endSeg - 1].moof_offset;\n        const byteEnd = nextSegMoofOffset;\n\n        batches.push({ startSeg, endSeg, byteStart, byteEnd });\n    }\n    return batches;\n}\n\n/**\n * Slice the per-stem moof+mdat range out of the batch chunk for a single\n * (segment, stem) pair. Mirrors the offset math in loadTrackStems().\n */\nfunction getStemSliceWithinSegment(\n    tfraData: TfraExtractResult,\n    segIndex: number,\n    stemTfraIdx: number,\n): { moofOffsetInFile: number; moofEndInFile: number } {\n    const moofOffset = tfraData[stemTfraIdx].entries[segIndex].moof_offset;\n    let moofEnd: number;\n    if (stemTfraIdx < 3) {\n        // Not the last stream: ends where the next stem's moof starts in same segment\n        moofEnd = tfraData[stemTfraIdx + 1].entries[segIndex].moof_offset;\n    } else {\n        // Last stream (vocals): ends where bass's moof starts in next segment\n        moofEnd = tfraData[0].entries[segIndex + 1].moof_offset;\n    }\n    return { moofOffsetInFile: moofOffset, moofEndInFile: moofEnd };\n}\n\n/**\n * Issue an HTTP Range request for [byteStart, byteEnd). Returns the ArrayBuffer\n * with that range. Throws if the server doesn't honor the range header.\n */\nasync function fetchRange(\n    url: string,\n    byteStart: number,\n    byteEnd: number,\n    token: string,\n    signal: AbortSignal,\n): Promise<ArrayBuffer> {\n    const headers: Record<string, string> = {\n        Range: `bytes=${byteStart}-${byteEnd - 1}`,\n    };\n    if (token) headers['Authorization'] = `Bearer ${token}`;\n\n    const res = await fetch(url, { headers, cache: 'no-cache', signal });\n    if (!res.ok && res.status !== 206) {\n        throw new Error(`Range fetch failed: ${res.status} for bytes=${byteStart}-${byteEnd - 1}`);\n    }\n    return res.arrayBuffer();\n}\n\nfunction createStemDecoder(\n    onChunk: (left: Float32Array, right: Float32Array, frames: number) => void,\n    onError: (e: Error) => void,\n): AudioDecoder {\n    // Surface the global type. We've already feature-detected.\n    const Ctor = (globalThis as unknown as { AudioDecoder: typeof AudioDecoder }).AudioDecoder;\n    const decoder = new Ctor({\n        output: (audioData: AudioData) => {\n            const frames = audioData.numberOfFrames;\n            const channels = audioData.numberOfChannels;\n            const left = new Float32Array(frames);\n            const right = new Float32Array(frames);\n            try {\n                audioData.copyTo(left, { planeIndex: 0, format: 'f32-planar' });\n                if (channels > 1) {\n                    audioData.copyTo(right, { planeIndex: 1, format: 'f32-planar' });\n                } else {\n                    right.set(left);\n                }\n            } catch (e) {\n                onError(e as Error);\n                audioData.close();\n                return;\n            }\n            audioData.close();\n            onChunk(left, right, frames);\n        },\n        error: (e) => onError(e as Error),\n    });\n    decoder.configure({\n        codec: 'mp4a.40.2',\n        sampleRate: SAMPLE_RATE,\n        numberOfChannels: 2,\n    });\n    return decoder;\n}\n\ninterface BeginOptions {\n    /** Pre-fetched TFRA data (avoids the extra HEAD/tail roundtrip on cache hits) */\n    tfraData?: TfraExtractResult;\n    /** When provided, batches are sourced from this ArrayBuffer instead of the network */\n    sourceBuffer?: ArrayBuffer;\n    /**\n     * Explicit per-stem buffer length. When omitted we derive it from\n     * `expectedDurationSec` — but that can under-estimate the real decoded\n     * length and cause the tail of the track to be silently dropped (see\n     * deriveTotalSamplesFromTfra). Prefer passing this explicitly.\n     */\n    totalSamples?: number;\n}\n\nexport function beginProgressiveLoad(\n    mp4Url: string,\n    expectedDurationSec: number,\n    options: BeginOptions = {},\n): ProgressiveLoadHandle {\n    // Use the explicit hint if provided; otherwise fall back to a duration-based\n    // estimate plus a 1s margin so a small under-estimate doesn't cap the tail.\n    const totalSamples = options.totalSamples\n        ?? (Math.ceil(expectedDurationSec * SAMPLE_RATE) + SAMPLE_RATE);\n    const abortController = new AbortController();\n    const patchListeners = new Set<(p: StemPatch) => void>();\n    const progressListeners = new Set<(p: LoadProgress) => void>();\n\n    let playableResolve!: () => void;\n    let playableReject!: (e: Error) => void;\n    const playable = new Promise<void>((resolve, reject) => {\n        playableResolve = resolve;\n        playableReject = reject;\n    });\n    let completeResolve!: () => void;\n    let completeReject!: (e: Error) => void;\n    const complete = new Promise<void>((resolve, reject) => {\n        completeResolve = resolve;\n        completeReject = reject;\n    });\n\n    let playableResolved = false;\n    let completeResolved = false;\n    let aborted = false;\n\n    const emitPatch = (p: StemPatch) => {\n        if (aborted) return;\n        for (const cb of patchListeners) {\n            try { cb(p); } catch (e) { console.warn('[ProgressiveLoader] patch listener threw', e); }\n        }\n    };\n    const emitProgress = (p: LoadProgress) => {\n        if (aborted) return;\n        for (const cb of progressListeners) {\n            try { cb(p); } catch (e) { console.warn('[ProgressiveLoader] progress listener threw', e); }\n        }\n    };\n\n    const run = async () => {\n        try {\n            const token = AEV3Config.getInstance()?.getAccessToken?.() ?? '';\n\n            emitProgress({ phase: 'fetching', percent: 0, detail: 'Reading MP4 index…' });\n            const tfraData = options.tfraData ?? await extractTfraData(mp4Url, token, false);\n            if (aborted) return;\n            if (!tfraData || tfraData.length < 4) {\n                throw new Error('TFRA data missing or incomplete – expected 4 tracks');\n            }\n\n            // We don't know the total file size from TFRA alone. The byte range\n            // for the LAST batch ends at the start of the trailing moov/mfra\n            // boxes; for simplicity we drop the final (incomplete) segment from\n            // the loop in loadTrackStems and we do the same here.\n            const batches = planBatches(tfraData, null);\n            if (batches.length === 0) {\n                throw new Error('No decodable segments found in TFRA index');\n            }\n\n            const moofExtractor = new MP4MoofExtractor();\n            const adtsPackager = new AACAdtsPackager();\n\n            // Build one decoder per stem. Each decoder's output writes into\n            // the running pending list for the current batch.\n            const stemStates: Record<StemLabel, StemDecodeState> = {} as Record<StemLabel, StemDecodeState>;\n            for (const label of STEM_LABELS) {\n                const state: StemDecodeState = {\n                    decoder: null as unknown as AudioDecoder,\n                    writeOffset: 0,\n                    samplesToSkip: AAC_ENCODER_DELAY_SAMPLES,\n                    pendingChunks: [],\n                    batchSampleStart: 0,\n                    decodedSamples: 0,\n                    timestampUs: 0,\n                };\n                state.decoder = createStemDecoder(\n                    (left, right) => {\n                        // Skip encoder priming on the first frames\n                        let l = left;\n                        let r = right;\n                        if (state.samplesToSkip > 0) {\n                            const drop = Math.min(state.samplesToSkip, l.length);\n                            l = l.subarray(drop);\n                            r = r.subarray(drop);\n                            state.samplesToSkip -= drop;\n                            if (l.length === 0) return;\n                        }\n                        state.pendingChunks.push([l, r]);\n                    },\n                    (e) => {\n                        console.error(`[ProgressiveLoader] decoder error for ${label}:`, e);\n                    },\n                );\n                stemStates[label] = state;\n            }\n\n            const checkPlayable = () => {\n                if (playableResolved) return;\n                let minDecoded = Infinity;\n                for (const label of STEM_LABELS) {\n                    if (stemStates[label].decodedSamples < minDecoded) {\n                        minDecoded = stemStates[label].decodedSamples;\n                    }\n                }\n                if (minDecoded >= PLAYABLE_THRESHOLD_SAMPLES) {\n                    playableResolved = true;\n                    playableResolve();\n                }\n            };\n\n            for (let batchIdx = 0; batchIdx < batches.length; batchIdx++) {\n                if (aborted) return;\n                const batch = batches[batchIdx];\n\n                emitProgress({\n                    phase: 'fetching',\n                    percent: Math.round((batchIdx / batches.length) * 100),\n                    detail: `Batch ${batchIdx + 1}/${batches.length}`,\n                });\n\n                let chunkBuf: ArrayBuffer;\n                if (options.sourceBuffer) {\n                    // Source from a fully-cached ArrayBuffer (offline reopen)\n                    chunkBuf = options.sourceBuffer.slice(batch.byteStart, batch.byteEnd);\n                } else {\n                    chunkBuf = await fetchRange(mp4Url, batch.byteStart, batch.byteEnd, token, abortController.signal);\n                }\n                if (aborted) return;\n\n                // For each stem, slice out per-segment moof+mdat ranges from\n                // the batch chunk and feed ADTS frames to that stem's decoder.\n                for (let tfraIdx = 0; tfraIdx < 4; tfraIdx++) {\n                    const stemLabel = TFRA_STEM_MAP[tfraIdx];\n                    const state = stemStates[stemLabel];\n                    state.pendingChunks = [];\n                    state.batchSampleStart = state.writeOffset;\n                    const mp4TrackId = tfraIdx + 1;\n\n                    for (let segIdx = batch.startSeg; segIdx < batch.endSeg; segIdx++) {\n                        const { moofOffsetInFile, moofEndInFile } = getStemSliceWithinSegment(tfraData, segIdx, tfraIdx);\n                        const relStart = moofOffsetInFile - batch.byteStart;\n                        const relEnd = moofEndInFile - batch.byteStart;\n                        if (relStart < 0 || relEnd > chunkBuf.byteLength || relEnd <= relStart) {\n                            // Out-of-bounds within this chunk (shouldn't happen with a valid plan)\n                            continue;\n                        }\n                        const segBuf = chunkBuf.slice(relStart, relEnd);\n\n                        let adtsFrames: Uint8Array[];\n                        try {\n                            const { trunData, mdatData } = moofExtractor.extractMoofDataForAEV3(\n                                segBuf, mp4TrackId, segIdx, '', '', `stem-${stemLabel}`,\n                            );\n                            const { buffers } = adtsPackager.packageAacFramesForAEV3(\n                                mdatData, trunData, `stem-${stemLabel}`,\n                            );\n                            adtsFrames = buffers;\n                        } catch (e) {\n                            // Tail-end MOOF extraction failures are expected near EOF\n                            console.warn(`[ProgressiveLoader] failed to parse seg ${segIdx} stem ${stemLabel}:`, e);\n                            continue;\n                        }\n\n                        for (const frame of adtsFrames) {\n                            if (aborted) return;\n                            try {\n                                state.decoder.decode(new EncodedAudioChunk({\n                                    type: 'key',\n                                    timestamp: state.timestampUs,\n                                    duration: Math.round(AAC_FRAME_SAMPLES * 1e6 / SAMPLE_RATE),\n                                    data: frame,\n                                }));\n                                state.timestampUs += Math.round(AAC_FRAME_SAMPLES * 1e6 / SAMPLE_RATE);\n                            } catch (e) {\n                                console.warn(`[ProgressiveLoader] decode() threw for ${stemLabel}:`, e);\n                            }\n                        }\n                    }\n                }\n\n                // Wait for all 4 decoders to finish this batch's queued frames.\n                // flush() does NOT reset state — it just drains the queue.\n                await Promise.all(STEM_LABELS.map(async (label) => {\n                    const state = stemStates[label];\n                    try {\n                        await state.decoder.flush();\n                    } catch (e) {\n                        console.warn(`[ProgressiveLoader] flush failed for ${label}:`, e);\n                    }\n                }));\n\n                if (aborted) return;\n\n                // Coalesce each stem's pending chunks into a single contiguous\n                // patch and emit. Slices stay inside the pre-allocated total\n                // buffer (cap at totalSamples to guard against duration drift).\n                for (const label of STEM_LABELS) {\n                    const state = stemStates[label];\n                    if (state.pendingChunks.length === 0) continue;\n\n                    let total = 0;\n                    for (const [l] of state.pendingChunks) total += l.length;\n                    if (total === 0) continue;\n\n                    const writeStart = state.batchSampleStart;\n                    const maxWritable = Math.max(0, totalSamples - writeStart);\n                    const writeLen = Math.min(total, maxWritable);\n                    if (writeLen <= 0) {\n                        // Pre-allocated buffer is too small for the actual\n                        // decoded length — every subsequent batch will hit\n                        // this and be silently dropped. Surface it loudly\n                        // so we can adjust totalSamples derivation.\n                        console.warn(\n                            `[ProgressiveLoader] buffer cap hit for ${label}: ` +\n                            `writeStart=${writeStart} >= totalSamples=${totalSamples} ` +\n                            `(have ${total} decoded samples queued, batch ${batchIdx + 1}/${batches.length})`,\n                        );\n                        state.pendingChunks = [];\n                        continue;\n                    }\n                    if (writeLen < total) {\n                        console.warn(\n                            `[ProgressiveLoader] truncating ${label} batch ${batchIdx + 1}: ` +\n                            `wrote ${writeLen} of ${total} samples (buffer ends at ${totalSamples})`,\n                        );\n                    }\n\n                    const mergedL = new Float32Array(writeLen);\n                    const mergedR = new Float32Array(writeLen);\n                    let off = 0;\n                    for (const [l, r] of state.pendingChunks) {\n                        const remaining = writeLen - off;\n                        if (remaining <= 0) break;\n                        const take = Math.min(l.length, remaining);\n                        mergedL.set(l.subarray(0, take), off);\n                        mergedR.set(r.subarray(0, take), off);\n                        off += take;\n                    }\n                    state.pendingChunks = [];\n                    state.writeOffset = writeStart + writeLen;\n                    state.decodedSamples = state.writeOffset;\n\n                    emitPatch({\n                        stem: label,\n                        sampleOffset: writeStart,\n                        samples: [mergedL, mergedR],\n                    });\n                }\n\n                checkPlayable();\n\n                emitProgress({\n                    phase: 'decoding',\n                    percent: Math.round(((batchIdx + 1) / batches.length) * 100),\n                    detail: `Decoded batch ${batchIdx + 1}/${batches.length}`,\n                });\n            }\n\n            // Final flush + decoder cleanup\n            for (const label of STEM_LABELS) {\n                const state = stemStates[label];\n                try {\n                    await state.decoder.flush();\n                    state.decoder.close();\n                } catch { /* */ }\n            }\n\n            // In case we never crossed the 8s threshold (very short tracks)\n            if (!playableResolved) {\n                playableResolved = true;\n                playableResolve();\n            }\n            if (!completeResolved) {\n                completeResolved = true;\n                completeResolve();\n            }\n            emitProgress({ phase: 'decoding', percent: 100, detail: 'Done' });\n        } catch (e) {\n            if (aborted) return;\n            const err = e as Error;\n            console.error('[ProgressiveLoader] run failed:', err);\n            if (!playableResolved) {\n                playableResolved = true;\n                playableReject(err);\n            }\n            if (!completeResolved) {\n                completeResolved = true;\n                completeReject(err);\n            }\n        }\n    };\n\n    // Kick off — fire and forget; consumers track via promises/listeners.\n    void run();\n\n    return {\n        totalSamples,\n        playable,\n        complete,\n        abort: () => {\n            aborted = true;\n            abortController.abort();\n        },\n        onPatch: (cb) => {\n            patchListeners.add(cb);\n            return () => patchListeners.delete(cb);\n        },\n        onProgress: (cb) => {\n            progressListeners.add(cb);\n            return () => progressListeners.delete(cb);\n        },\n    };\n}\n\n/**\n * After the progressive load completes, opportunistically full-fetch the file\n * and stash it in IndexedDB so offline reopens stay fast. No-ops if cache hit.\n *\n * Doesn't block — invoke from a microtask off the `complete` promise.\n */\nexport async function cacheFullFileForOffline(url: string): Promise<void> {\n    try {\n        const existing = await getCachedAudioBuffer(url);\n        if (existing) return;\n        const token = AEV3Config.getInstance()?.getAccessToken?.() ?? '';\n        const headers: Record<string, string> = {};\n        if (token) headers['Authorization'] = `Bearer ${token}`;\n        const res = await fetch(url, { headers, cache: 'force-cache' });\n        if (!res.ok) return;\n        const ab = await res.arrayBuffer();\n        await cacheAudioBuffer(url, ab);\n    } catch { /* offline cache is best-effort */ }\n}\n","/**\n * DAWTrackLoader – Downloads a full fMP4 file and decodes all 4 stems\n * into separate AudioBuffers for offline/DAW-style manipulation.\n *\n * This intentionally avoids the streaming AudioSourceSequence pipeline.\n * The trade-off is a longer initial load, but after that every stem is\n * instantly accessible at any sample position.\n */\n\nimport { extractTfraData } from '@/stem-audio-engine-v3/utils/mp4';\nimport { MP4MoofExtractor } from '@/stem-audio-engine-v3/utils/mp4-parser/mp4-moof-extractor';\nimport { AACAdtsPackager } from '@/stem-audio-engine-v3/utils/mp4-parser/aac-adts-packager';\nimport { AEV3Config } from '@/stem-audio-engine-v3/aev3-env-config';\nimport { cacheAudioBuffer, getCachedAudioBuffer } from './daw-cache';\nimport {\n    beginProgressiveLoad,\n    isProgressiveDecodeSupported,\n    deriveTotalSamplesFromTfra,\n    type ProgressiveLoadHandle,\n} from './progressive-track-loader';\n\nconst SAMPLE_RATE = 48000;\n\nlet _decodingCtx: AudioContext | null = null;\nfunction getDecodingContext(): AudioContext {\n    if (!_decodingCtx || _decodingCtx.state === 'closed') {\n        _decodingCtx = new AudioContext({ sampleRate: SAMPLE_RATE });\n    }\n    return _decodingCtx;\n}\n\nexport interface StemAudioBuffers {\n    other: AudioBuffer;\n    vocals: AudioBuffer;\n    bass: AudioBuffer;\n    drums: AudioBuffer;\n}\n\nexport type LoadProgress = {\n    phase: 'fetching' | 'parsing' | 'decoding';\n    percent: number;\n    detail?: string;\n};\n\n// TFRA index → stem label\nconst TFRA_STEM_MAP: Record<number, keyof StemAudioBuffers> = {\n    0: 'bass',\n    1: 'drums',\n    2: 'other',\n    3: 'vocals',\n};\n\n/** Stem URLs from REST API (e.g. newly uploaded tracks before fMP4 is ready) */\nexport interface StemUrls {\n    bass: string;\n    drums: string;\n    other: string;\n    vocals: string;\n}\n\n/**\n * Load stems from 4 individual MP3 URLs (REST API format).\n * Used when fMP4 is not yet available for newly uploaded tracks.\n */\nexport async function loadTrackStemsFromMp3Urls(\n    urls: StemUrls,\n    onProgress?: (p: LoadProgress) => void,\n): Promise<StemAudioBuffers> {\n    const ctx = getDecodingContext();\n    const result: Partial<StemAudioBuffers> = {};\n    const labels: (keyof StemAudioBuffers)[] = ['bass', 'drums', 'other', 'vocals'];\n\n    for (let i = 0; i < labels.length; i++) {\n        const label = labels[i];\n        const url = urls[label];\n        const pct = Math.round(((i + 1) / 4) * 100);\n        onProgress?.({ phase: 'fetching', percent: pct, detail: `Loading ${label}…` });\n\n        const res = await fetch(url, { cache: 'no-cache' });\n        if (!res.ok) throw new Error(`Failed to fetch ${label}: ${res.status}`);\n        const ab = await res.arrayBuffer();\n        const buf = await ctx.decodeAudioData(ab);\n\n        // Ensure stereo (DAW expects 2 channels)\n        if (buf.numberOfChannels === 1) {\n            const stereo = ctx.createBuffer(2, buf.length, buf.sampleRate);\n            const mono = buf.getChannelData(0);\n            stereo.copyToChannel(mono, 0);\n            stereo.copyToChannel(mono, 1);\n            result[label] = stereo;\n        } else {\n            result[label] = buf;\n        }\n    }\n\n    onProgress?.({ phase: 'decoding', percent: 100, detail: 'Done' });\n    return result as StemAudioBuffers;\n}\n\n/**\n * Download an fMP4 track and decode each stem into a standalone AudioBuffer.\n */\nexport async function loadTrackStems(\n    mp4Url: string,\n    onProgress?: (p: LoadProgress) => void,\n): Promise<StemAudioBuffers> {\n    const token = AEV3Config.getInstance()?.getAccessToken?.() ?? '';\n\n    onProgress?.({ phase: 'fetching', percent: 0, detail: 'Reading MP4 index…' });\n    const tfraData = await extractTfraData(mp4Url, token, false);\n    if (!tfraData || tfraData.length < 4) {\n        throw new Error('TFRA data missing or incomplete – expected 4 tracks');\n    }\n\n    let fullBuffer: ArrayBuffer;\n    const cached = await getCachedAudioBuffer(mp4Url);\n    if (cached) {\n        onProgress?.({ phase: 'fetching', percent: 35, detail: 'Loaded from cache' });\n        fullBuffer = cached;\n    } else {\n        onProgress?.({ phase: 'fetching', percent: 10, detail: 'Downloading audio…' });\n        const headers: Record<string, string> = {};\n        if (token) headers['Authorization'] = `Bearer ${token}`;\n\n        const response = await fetch(mp4Url, { headers, cache: 'no-cache' });\n        if (!response.ok) throw new Error(`MP4 fetch failed: ${response.status}`);\n        fullBuffer = await response.arrayBuffer();\n\n        cacheAudioBuffer(mp4Url, fullBuffer.slice(0));\n    }\n    onProgress?.({ phase: 'parsing', percent: 40, detail: 'Extracting stems…' });\n\n    // ── 3. Extract per-stem AAC frames ────────────────────────────────\n    const moofExtractor = new MP4MoofExtractor();\n    const adtsPackager = new AACAdtsPackager();\n\n    const stemFrames: Record<keyof StemAudioBuffers, Uint8Array[]> = {\n        bass: [], drums: [], other: [], vocals: [],\n    };\n\n    const bufLen = fullBuffer.byteLength;\n\n    // With separate_moof, the per-segment layout is:\n    //   [bass MOOF+MDAT] [drums MOOF+MDAT] [other MOOF+MDAT] [vocals MOOF+MDAT]\n    // To get the byte-length of one track's MOOF+MDAT we use the offset of the\n    // next track's MOOF in the same segment (or the next segment's first track).\n    const entryCount = tfraData[0].entries.length;\n\n    for (let tfraIdx = 0; tfraIdx < 4; tfraIdx++) {\n        const stemLabel = TFRA_STEM_MAP[tfraIdx];\n        const entries = tfraData[tfraIdx].entries;\n        const mp4TrackId = tfraIdx + 1;\n\n        for (let i = 0; i < entries.length - 1; i++) {\n            const moofOffset = entries[i].moof_offset;\n\n            // Calculate the end of this track's MOOF+MDAT using the next\n            // track's MOOF offset in the same segment index\n            let moofEnd: number;\n            if (tfraIdx < 3) {\n                // Not the last track: ends where the next track's MOOF starts\n                moofEnd = tfraData[tfraIdx + 1].entries[i].moof_offset;\n            } else {\n                // Last track (vocals): ends where bass's MOOF starts in the NEXT segment\n                moofEnd = tfraData[0].entries[i + 1].moof_offset;\n            }\n\n            const byteLength = moofEnd - moofOffset;\n            if (moofOffset + byteLength > bufLen || byteLength <= 0) {\n                continue; // skip out-of-bounds or nonsensical ranges\n            }\n\n            const trackBuffer = fullBuffer.slice(moofOffset, moofOffset + byteLength);\n\n            try {\n                const { trunData, mdatData } = moofExtractor.extractMoofDataForAEV3(\n                    trackBuffer, mp4TrackId, i, '', '', `stem-${stemLabel}`,\n                );\n                const { buffers: adtsFrames } = adtsPackager.packageAacFramesForAEV3(\n                    mdatData, trunData, `stem-${stemLabel}`,\n                );\n                stemFrames[stemLabel].push(...adtsFrames);\n            } catch {\n                // Tail-end MOOF extraction failures are expected near EOF\n            }\n        }\n\n        const pct = 40 + Math.round(((tfraIdx + 1) / 4) * 30);\n        onProgress?.({ phase: 'parsing', percent: pct, detail: `Extracted ${stemLabel}` });\n    }\n\n    // ── 4. Decode each stem ───────────────────────────────────────────\n    onProgress?.({ phase: 'decoding', percent: 70, detail: 'Decoding audio…' });\n\n    const ctx = getDecodingContext();\n    const result: Partial<StemAudioBuffers> = {};\n\n    const stemLabels = Object.keys(stemFrames) as (keyof StemAudioBuffers)[];\n    for (let si = 0; si < stemLabels.length; si++) {\n        const label = stemLabels[si];\n        const frames = stemFrames[label];\n\n        if (frames.length === 0) {\n            // Create silent buffer as fallback\n            result[label] = ctx.createBuffer(2, SAMPLE_RATE, SAMPLE_RATE);\n            continue;\n        }\n\n        const merged = mergeFrames(frames);\n        let decoded: AudioBuffer;\n        try {\n            decoded = await ctx.decodeAudioData(merged.buffer);\n        } catch {\n            // Fallback: go through Blob to get a fresh ArrayBuffer\n            const blob = new Blob([merged], { type: 'audio/aac' });\n            const ab = await blob.arrayBuffer();\n            decoded = await ctx.decodeAudioData(ab);\n        }\n\n        result[label] = trimEncoderDelay(decoded, ctx);\n\n        const pct = 70 + Math.round(((si + 1) / 4) * 28);\n        onProgress?.({ phase: 'decoding', percent: pct, detail: `Decoded ${label}` });\n    }\n\n    onProgress?.({ phase: 'decoding', percent: 100, detail: 'Done' });\n\n    return result as StemAudioBuffers;\n}\n\n// ── helpers ───────────────────────────────────────────────────────────\n\n/**\n * AAC encoder priming delay: the FDK-AAC encoder prepends 2048 samples of\n * look-ahead plus a 64-sample SBR overlap = 2112 samples at 48 kHz.\n * When decoding raw ADTS frames (no MP4 edit list), decodeAudioData returns\n * these priming samples at the start of the buffer. We strip them so that\n * sample positions match the original source used for beat detection.\n */\nconst AAC_ENCODER_DELAY_SAMPLES = 2112;\n\nfunction trimEncoderDelay(buf: AudioBuffer, ctx: AudioContext): AudioBuffer {\n    if (buf.length <= AAC_ENCODER_DELAY_SAMPLES) return buf;\n    const trimmedLength = buf.length - AAC_ENCODER_DELAY_SAMPLES;\n    const trimmed = ctx.createBuffer(buf.numberOfChannels, trimmedLength, buf.sampleRate);\n    for (let ch = 0; ch < buf.numberOfChannels; ch++) {\n        const src = buf.getChannelData(ch);\n        const dst = trimmed.getChannelData(ch);\n        dst.set(src.subarray(AAC_ENCODER_DELAY_SAMPLES));\n    }\n    return trimmed;\n}\n\nfunction mergeFrames(frames: Uint8Array[]): Uint8Array {\n    let totalLength = 0;\n    for (const f of frames) totalLength += f.byteLength;\n    const out = new Uint8Array(totalLength);\n    let offset = 0;\n    for (const f of frames) {\n        out.set(f, offset);\n        offset += f.byteLength;\n    }\n    return out;\n}\n\n/**\n * Allocate four zero-filled stereo AudioBuffers of the given length. Used as\n * the placeholder audio for a track that is still streaming in.\n */\nfunction allocateEmptyStemBuffers(totalSamples: number, ctx: AudioContext): StemAudioBuffers {\n    const len = Math.max(1, totalSamples);\n    return {\n        bass: ctx.createBuffer(2, len, SAMPLE_RATE),\n        drums: ctx.createBuffer(2, len, SAMPLE_RATE),\n        other: ctx.createBuffer(2, len, SAMPLE_RATE),\n        vocals: ctx.createBuffer(2, len, SAMPLE_RATE),\n    };\n}\n\nexport interface ProgressiveLoadResult {\n    /** Zero-filled stems ready to hand to the worklets immediately */\n    initial: StemAudioBuffers;\n    /** Stream handle — caller wires onPatch to update buffers + waveform cache */\n    handle: ProgressiveLoadHandle;\n}\n\n/**\n * Progressive variant of loadTrackStems. Returns immediately with zero-filled\n * AudioBuffers (sized by expectedDurationSec) plus a handle whose onPatch\n * callback fires as each batch of samples is decoded.\n *\n * If WebCodecs AudioDecoder isn't available (Safari < 16.4), this falls back\n * to the legacy full-download loadTrackStems and emits a single full-length\n * patch once decoding completes — preserves correctness, loses progressivity.\n */\nexport async function loadTrackStemsProgressively(\n    mp4Url: string,\n    expectedDurationSec: number,\n    onProgress?: (p: LoadProgress) => void,\n): Promise<ProgressiveLoadResult> {\n    const ctx = getDecodingContext();\n\n    if (!isProgressiveDecodeSupported()) {\n        // Fallback path: decode the whole track up-front, then emit one patch\n        // at offset 0 once it's done. The handle stays alive so the controller\n        // wiring is identical. Use a duration-based estimate since we don't\n        // have TFRA in this branch.\n        const fallbackTotalSamples = Math.max(1, Math.ceil(expectedDurationSec * SAMPLE_RATE) + SAMPLE_RATE);\n        const initial = allocateEmptyStemBuffers(fallbackTotalSamples, ctx);\n        return fallbackToOneShotPatch(mp4Url, initial, fallbackTotalSamples, onProgress);\n    }\n\n    // Fetch TFRA first so we can size the stem buffers from the actual\n    // sample count, not just `trackData.duration` which can under-estimate\n    // (e.g. when duration represents musical length only). Without this\n    // every batch past the cap is silently dropped — that's the \"doesn't\n    // decode the whole song\" bug.\n    const token = AEV3Config.getInstance()?.getAccessToken?.() ?? '';\n    let tfraData;\n    try {\n        tfraData = await extractTfraData(mp4Url, token, false);\n    } catch (err) {\n        console.warn('[loadTrackStemsProgressively] TFRA fetch failed, falling back to duration-only sizing:', err);\n        tfraData = null;\n    }\n\n    const totalSamples = (tfraData && tfraData.length >= 4)\n        ? deriveTotalSamplesFromTfra(tfraData, expectedDurationSec)\n        : Math.max(1, Math.ceil(expectedDurationSec * SAMPLE_RATE) + SAMPLE_RATE);\n\n    const initial = allocateEmptyStemBuffers(totalSamples, ctx);\n\n    // If the file is already in IndexedDB, drive the progressive decoder\n    // from that cached ArrayBuffer instead of the network — same UX, no\n    // network roundtrips.\n    const cached = await getCachedAudioBuffer(mp4Url);\n    if (cached && tfraData && tfraData.length >= 4) {\n        const handle = beginProgressiveLoad(mp4Url, expectedDurationSec, {\n            tfraData,\n            sourceBuffer: cached,\n            totalSamples,\n        });\n        if (onProgress) handle.onProgress(onProgress);\n        return { initial, handle };\n    }\n\n    const handle = beginProgressiveLoad(mp4Url, expectedDurationSec, {\n        tfraData: tfraData ?? undefined,\n        totalSamples,\n    });\n    if (onProgress) handle.onProgress(onProgress);\n    return { initial, handle };\n}\n\n/** Adapter that runs the legacy loader and replays the result as a single patch */\nasync function fallbackToOneShotPatch(\n    mp4Url: string,\n    initial: StemAudioBuffers,\n    totalSamples: number,\n    onProgress?: (p: LoadProgress) => void,\n): Promise<ProgressiveLoadResult> {\n    const patchListeners = new Set<(p: import('./progressive-track-loader').StemPatch) => void>();\n    const progressListeners = new Set<(p: LoadProgress) => void>();\n    if (onProgress) progressListeners.add(onProgress);\n\n    let completeResolve!: () => void;\n    let completeReject!: (e: Error) => void;\n    const complete = new Promise<void>((resolve, reject) => {\n        completeResolve = resolve;\n        completeReject = reject;\n    });\n\n    let aborted = false;\n\n    (async () => {\n        try {\n            const stems = await loadTrackStems(mp4Url, (p) => {\n                for (const cb of progressListeners) {\n                    try { cb(p); } catch { /* */ }\n                }\n            });\n            if (aborted) return;\n\n            const stemKeys: (keyof StemAudioBuffers)[] = ['bass', 'drums', 'other', 'vocals'];\n            for (const k of stemKeys) {\n                const buf = stems[k];\n                const lenToCopy = Math.min(buf.length, totalSamples);\n                const left = new Float32Array(lenToCopy);\n                const right = new Float32Array(lenToCopy);\n                left.set(buf.getChannelData(0).subarray(0, lenToCopy));\n                if (buf.numberOfChannels > 1) {\n                    right.set(buf.getChannelData(1).subarray(0, lenToCopy));\n                } else {\n                    right.set(left);\n                }\n                for (const cb of patchListeners) {\n                    try { cb({ stem: k, sampleOffset: 0, samples: [left, right] }); } catch { /* */ }\n                }\n            }\n            completeResolve();\n        } catch (e) {\n            completeReject(e as Error);\n        }\n    })();\n\n    const handle: ProgressiveLoadHandle = {\n        totalSamples,\n        playable: complete,\n        complete,\n        abort: () => { aborted = true; },\n        onPatch: (cb) => { patchListeners.add(cb); return () => patchListeners.delete(cb); },\n        onProgress: (cb) => { progressListeners.add(cb); return () => progressListeners.delete(cb); },\n    };\n\n    return { initial, handle };\n}\n","/**\n * Shared waveform mipmap cache store.\n * Separated from DAWTimeline to avoid breaking React Fast Refresh\n * when non-component functions are exported from component files.\n *\n * Progressive support: a track's audio may arrive in batches via the\n * progressive-track-loader. We pre-allocate the mipmap arrays from the\n * track's expected total samples and patch them in place as decoded slots\n * land. The `version` counter is bumped on every patch so the timeline\n * canvas can invalidate cleanly.\n */\n\n/**\n * Power-of-2 bin counts used for the per-track waveform mipmap. Kept here\n * as the single source of truth — DAWTimeline.tsx imports this constant.\n *\n * The finest level (`65536`) is what supports the high-zoom \"see individual\n * transients\" look in the filled-envelope renderer. Memory cost per stem:\n *   65536 × 4 bytes ≈ 256 KB.  ×4 stems ×N tracks → ~1 MB per track.\n * For a 4-minute song at 48 kHz that's ~175 samples per bin (~3.6 ms),\n * which preserves every drum hit and sharp transient when zoomed in.\n */\nexport const MIPMAP_LEVELS = [64, 256, 1024, 4096, 16384, 65536] as const;\n\nexport type MipmapCache = Map<number, Float32Array>;\n\nexport interface MipmapState {\n    /** Power-of-2 mipmap arrays, all sized from the track's total samples */\n    levels: MipmapCache;\n    /** Highest sample index that has been written to (decoded watermark) */\n    decodedSampleHigh: number;\n    /** Fixed total length of the underlying stem buffer (samples) */\n    totalSamples: number;\n    /** Bumped on every appendPeaks; canvases compare to detect changes */\n    version: number;\n}\n\n/** key = `${trackId}:${stemKey}` */\nexport const mipmapStore = new Map<string, MipmapState>();\n\nconst STEM_KEYS = ['other', 'vocals', 'bass', 'drums'];\n\n// ── Change subscriptions ─────────────────────────────────────────────\n// The timeline canvas subscribes here so it can redraw as peaks fill in\n// (the canvas animation loop only ticks during playback; progressive\n// patches arrive while paused, hence the explicit notification).\n\nconst changeListeners = new Set<() => void>();\n\nexport function subscribeMipmapChanges(cb: () => void): () => void {\n    changeListeners.add(cb);\n    return () => changeListeners.delete(cb);\n}\n\nfunction notifyChange(): void {\n    for (const cb of changeListeners) {\n        try { cb(); } catch { /* */ }\n    }\n}\n\nfunction makeKey(trackId: string, stemKey: string): string {\n    return `${trackId}:${stemKey}`;\n}\n\n/**\n * Allocate zero-filled mipmap arrays for a track's stem. The base-level\n * peaks have one entry per sample, but to keep memory sane we size each\n * level relative to MIPMAP_LEVELS as a target bin count per stem. Sample\n * positions are mapped to bin indices via the standard fractional formula\n * (no accumulating drift — see waveform-utils.extractPeaks).\n */\nexport function initMipmapsForTrack(\n    trackId: string,\n    stemKey: string,\n    totalSamples: number,\n): MipmapState {\n    const key = makeKey(trackId, stemKey);\n    const existing = mipmapStore.get(key);\n    if (existing && existing.totalSamples === totalSamples) {\n        // Reset to zeros so a re-init (e.g. re-add) starts fresh\n        for (const arr of existing.levels.values()) arr.fill(0);\n        existing.decodedSampleHigh = 0;\n        existing.version++;\n        return existing;\n    }\n\n    const levels: MipmapCache = new Map();\n    for (const numBins of MIPMAP_LEVELS) {\n        if (totalSamples < numBins) continue;\n        levels.set(numBins, new Float32Array(numBins));\n    }\n\n    const state: MipmapState = {\n        levels,\n        decodedSampleHigh: 0,\n        totalSamples,\n        version: 1,\n    };\n    mipmapStore.set(key, state);\n    return state;\n}\n\n/**\n * Patch peaks into the mipmap for samples [sampleOffset, sampleOffset+samplesL).\n * Recomputes the affected slice of each mipmap level from the raw stereo\n * patch (left + right channels, MAX'd to one mono peaks line). Bumps the\n * version counter so the timeline canvas can detect the change.\n *\n * If the track hasn't been init'd yet, this is a no-op (the controller\n * should call initMipmapsForTrack before wiring patches).\n */\nexport function appendPeaksToMipmapCache(\n    trackId: string,\n    stemKey: string,\n    sampleOffset: number,\n    leftChannel: Float32Array,\n    rightChannel: Float32Array | null,\n): void {\n    const state = mipmapStore.get(makeKey(trackId, stemKey));\n    if (!state) return;\n\n    const total = state.totalSamples;\n    const startSample = Math.max(0, sampleOffset);\n    const endSample = Math.min(total, sampleOffset + leftChannel.length);\n    if (endSample <= startSample) return;\n\n    for (const [numBins, peaks] of state.levels) {\n        // Convert sample range to bin range using the same fractional binning\n        // formula used in waveform-utils.extractPeaks — guarantees alignment\n        // with downstream sample-to-time math.\n        // sampleIndex in [floor(b * total/numBins), floor((b+1) * total/numBins))\n        // → bin b covers samples in that range.\n        // Find first and last bin touched by [startSample, endSample):\n        const firstBin = Math.floor((startSample * numBins) / total);\n        const lastBinExcl = Math.min(numBins, Math.ceil((endSample * numBins) / total));\n\n        for (let b = firstBin; b < lastBinExcl; b++) {\n            const binStart = Math.floor((b * total) / numBins);\n            const binEnd = Math.floor(((b + 1) * total) / numBins);\n            // Only sample the portion of this bin that lies inside our patch.\n            const sFrom = Math.max(binStart, startSample);\n            const sToExcl = Math.min(binEnd, endSample);\n            if (sToExcl <= sFrom) continue;\n\n            let max = peaks[b]; // preserve any previous patch's value in this bin\n            for (let i = sFrom; i < sToExcl; i++) {\n                const localI = i - sampleOffset;\n                const l = leftChannel[localI];\n                const r = rightChannel ? rightChannel[localI] : l;\n                const al = l < 0 ? -l : l;\n                const ar = r < 0 ? -r : r;\n                const v = al > ar ? al : ar;\n                if (v > max) max = v;\n            }\n            peaks[b] = max;\n        }\n    }\n\n    if (endSample > state.decodedSampleHigh) state.decodedSampleHigh = endSample;\n    state.version++;\n    notifyChange();\n}\n\nexport function invalidateWaveformCache(trackId: string, stemIndex: number): void {\n    const key = makeKey(trackId, STEM_KEYS[stemIndex]);\n    const had = mipmapStore.has(key);\n    mipmapStore.delete(key);\n    console.log('[DAW Waveform] invalidateWaveformCache:', key, had ? '(cache cleared)' : '(was not cached)');\n}\n\n/** Remove every cached stem for a track (called on removeTrack). */\nexport function clearWaveformCacheForTrack(trackId: string): void {\n    for (const k of STEM_KEYS) {\n        mipmapStore.delete(makeKey(trackId, k));\n    }\n}\n\n/** Read-only accessor for the timeline canvas. */\nexport function getMipmapState(trackId: string, stemKey: string): MipmapState | undefined {\n    return mipmapStore.get(makeKey(trackId, stemKey));\n}\n","/**\n * StemEffectChain — Per-stem real-time effect chain using Web Audio API.\n *\n * Signal path:  input → EQ → Compressor → Reverb → Delay → output\n *\n * Each effect can be bypassed independently. When bypassed, the signal\n * routes through a dry path. All nodes are pre-created so toggling\n * is glitch-free (just reconnect, no allocation).\n */\n\n// ── Effect parameter interfaces ──────────────────────────────────────\n\nexport interface EQParams {\n    enabled: boolean;\n    lowGain: number;    // dB, -12 to +12\n    midGain: number;    // dB, -12 to +12\n    highGain: number;   // dB, -12 to +12\n    lowFreq: number;    // Hz, default 320\n    highFreq: number;   // Hz, default 3200\n}\n\nexport interface CompressorParams {\n    enabled: boolean;\n    threshold: number;  // dB, -60 to 0\n    ratio: number;      // 1 to 20\n    attack: number;     // seconds, 0.001 to 1\n    release: number;    // seconds, 0.01 to 1\n    knee: number;       // dB, 0 to 40\n}\n\nexport interface ReverbParams {\n    enabled: boolean;\n    mix: number;        // 0 to 1 (dry/wet)\n    decay: number;      // seconds, 0.1 to 10\n}\n\nexport interface DelayParams {\n    enabled: boolean;\n    time: number;       // seconds, 0.01 to 2\n    feedback: number;   // 0 to 0.95\n    mix: number;        // 0 to 1 (dry/wet)\n}\n\nexport interface StemEffects {\n    eq: EQParams;\n    compressor: CompressorParams;\n    reverb: ReverbParams;\n    delay: DelayParams;\n}\n\nexport const DEFAULT_EFFECTS: StemEffects = {\n    eq: { enabled: false, lowGain: 0, midGain: 0, highGain: 0, lowFreq: 320, highFreq: 3200 },\n    compressor: { enabled: false, threshold: -24, ratio: 4, attack: 0.003, release: 0.25, knee: 30 },\n    reverb: { enabled: false, mix: 0.3, decay: 2.0 },\n    delay: { enabled: false, time: 0.375, feedback: 0.35, mix: 0.25 },\n};\n\nexport function hasActiveEffects(fx: StemEffects): boolean {\n    return fx.eq.enabled || fx.compressor.enabled || fx.reverb.enabled || fx.delay.enabled;\n}\n\n// ── Effect chain ─────────────────────────────────────────────────────\n\nexport class StemEffectChain {\n    private ctx: AudioContext;\n\n    // Input/output anchors — these never change\n    readonly input: GainNode;\n    readonly output: GainNode;\n\n    // EQ: 3-band (lowshelf + peaking + highshelf)\n    private eqLow: BiquadFilterNode;\n    private eqMid: BiquadFilterNode;\n    private eqHigh: BiquadFilterNode;\n\n    // Compressor\n    private compressor: DynamicsCompressorNode;\n\n    // Reverb (algorithmic via ConvolverNode with generated impulse)\n    private reverbDry: GainNode;\n    private reverbWet: GainNode;\n    private convolver: ConvolverNode;\n\n    // Delay\n    private delayNode: DelayNode;\n    private delayFeedback: GainNode;\n    private delayDry: GainNode;\n    private delayWet: GainNode;\n\n    // Per-slot bypass routing\n    private eqBypass: GainNode;\n    private compBypass: GainNode;\n    private reverbBypass: GainNode;\n    private delayBypass: GainNode;\n\n    // Junction nodes between stages\n    private postEq: GainNode;\n    private postComp: GainNode;\n    private postReverb: GainNode;\n\n    private _params: StemEffects;\n\n    constructor(ctx: AudioContext) {\n        this.ctx = ctx;\n        this._params = structuredClone(DEFAULT_EFFECTS);\n\n        this.input = ctx.createGain();\n        this.output = ctx.createGain();\n\n        // ── EQ ──\n        this.eqLow = ctx.createBiquadFilter();\n        this.eqLow.type = 'lowshelf';\n        this.eqLow.frequency.value = 320;\n        this.eqLow.gain.value = 0;\n\n        this.eqMid = ctx.createBiquadFilter();\n        this.eqMid.type = 'peaking';\n        this.eqMid.frequency.value = 1000;\n        this.eqMid.Q.value = 0.7;\n        this.eqMid.gain.value = 0;\n\n        this.eqHigh = ctx.createBiquadFilter();\n        this.eqHigh.type = 'highshelf';\n        this.eqHigh.frequency.value = 3200;\n        this.eqHigh.gain.value = 0;\n\n        this.eqBypass = ctx.createGain();\n\n        // ── Compressor ──\n        this.compressor = ctx.createDynamicsCompressor();\n        this.compressor.threshold.value = -24;\n        this.compressor.ratio.value = 4;\n        this.compressor.attack.value = 0.003;\n        this.compressor.release.value = 0.25;\n        this.compressor.knee.value = 30;\n\n        this.compBypass = ctx.createGain();\n\n        // ── Reverb ──\n        this.convolver = ctx.createConvolver();\n        this.convolver.buffer = this.generateImpulse(2.0);\n        this.reverbDry = ctx.createGain();\n        this.reverbDry.gain.value = 1;\n        this.reverbWet = ctx.createGain();\n        this.reverbWet.gain.value = 0;\n        this.reverbBypass = ctx.createGain();\n\n        // ── Delay ──\n        this.delayNode = ctx.createDelay(5);\n        this.delayNode.delayTime.value = 0.375;\n        this.delayFeedback = ctx.createGain();\n        this.delayFeedback.gain.value = 0.35;\n        this.delayDry = ctx.createGain();\n        this.delayDry.gain.value = 1;\n        this.delayWet = ctx.createGain();\n        this.delayWet.gain.value = 0;\n        this.delayBypass = ctx.createGain();\n\n        // Delay feedback loop\n        this.delayNode.connect(this.delayFeedback);\n        this.delayFeedback.connect(this.delayNode);\n\n        // ── Junction nodes ──\n        this.postEq = ctx.createGain();\n        this.postComp = ctx.createGain();\n        this.postReverb = ctx.createGain();\n\n        // Initial routing (all bypassed)\n        this.rebuildRouting();\n    }\n\n    /**\n     * Rebuild the full routing graph based on current bypass states.\n     * Uses disconnect-all then reconnect to avoid orphaned connections.\n     */\n    private rebuildRouting(): void {\n        const p = this._params;\n\n        // Disconnect everything from the chain anchors\n        this.input.disconnect();\n        this.eqLow.disconnect();\n        this.eqMid.disconnect();\n        this.eqHigh.disconnect();\n        this.eqBypass.disconnect();\n        this.postEq.disconnect();\n        this.compressor.disconnect();\n        this.compBypass.disconnect();\n        this.postComp.disconnect();\n        this.reverbDry.disconnect();\n        this.reverbWet.disconnect();\n        this.convolver.disconnect();\n        this.reverbBypass.disconnect();\n        this.postReverb.disconnect();\n        this.delayDry.disconnect();\n        this.delayWet.disconnect();\n        this.delayNode.disconnect();\n        this.delayFeedback.disconnect();\n        this.delayBypass.disconnect();\n\n        // Re-establish delay feedback loop\n        this.delayNode.connect(this.delayFeedback);\n        this.delayFeedback.connect(this.delayNode);\n\n        // Stage 1: EQ\n        if (p.eq.enabled) {\n            this.input.connect(this.eqLow);\n            this.eqLow.connect(this.eqMid);\n            this.eqMid.connect(this.eqHigh);\n            this.eqHigh.connect(this.postEq);\n        } else {\n            this.input.connect(this.postEq);\n        }\n\n        // Stage 2: Compressor\n        if (p.compressor.enabled) {\n            this.postEq.connect(this.compressor);\n            this.compressor.connect(this.postComp);\n        } else {\n            this.postEq.connect(this.postComp);\n        }\n\n        // Stage 3: Reverb (parallel dry/wet)\n        if (p.reverb.enabled) {\n            this.postComp.connect(this.reverbDry);\n            this.postComp.connect(this.convolver);\n            this.convolver.connect(this.reverbWet);\n            this.reverbDry.connect(this.postReverb);\n            this.reverbWet.connect(this.postReverb);\n        } else {\n            this.postComp.connect(this.postReverb);\n        }\n\n        // Stage 4: Delay (parallel dry/wet)\n        if (p.delay.enabled) {\n            this.postReverb.connect(this.delayDry);\n            this.postReverb.connect(this.delayNode);\n            this.delayNode.connect(this.delayWet);\n            this.delayDry.connect(this.output);\n            this.delayWet.connect(this.output);\n        } else {\n            this.postReverb.connect(this.output);\n        }\n    }\n\n    /** Generate a synthetic impulse response for reverb */\n    private generateImpulse(decay: number): AudioBuffer {\n        const rate = this.ctx.sampleRate;\n        const length = Math.floor(rate * Math.min(decay, 6));\n        const buf = this.ctx.createBuffer(2, length, rate);\n        for (let ch = 0; ch < 2; ch++) {\n            const data = buf.getChannelData(ch);\n            for (let i = 0; i < length; i++) {\n                data[i] = (Math.random() * 2 - 1) * Math.pow(1 - i / length, decay * 0.8);\n            }\n        }\n        return buf;\n    }\n\n    // ── Public API ───────────────────────────────────────────────────\n\n    get params(): StemEffects { return this._params; }\n\n    updateEQ(p: Partial<EQParams>): void {\n        Object.assign(this._params.eq, p);\n        const eq = this._params.eq;\n        this.eqLow.frequency.value = eq.lowFreq;\n        this.eqLow.gain.value = eq.lowGain;\n        this.eqMid.gain.value = eq.midGain;\n        this.eqHigh.frequency.value = eq.highFreq;\n        this.eqHigh.gain.value = eq.highGain;\n        if ('enabled' in p) this.rebuildRouting();\n    }\n\n    updateCompressor(p: Partial<CompressorParams>): void {\n        Object.assign(this._params.compressor, p);\n        const c = this._params.compressor;\n        const t = this.ctx.currentTime;\n        this.compressor.threshold.setValueAtTime(c.threshold, t);\n        this.compressor.ratio.setValueAtTime(c.ratio, t);\n        this.compressor.attack.setValueAtTime(c.attack, t);\n        this.compressor.release.setValueAtTime(c.release, t);\n        this.compressor.knee.setValueAtTime(c.knee, t);\n        if ('enabled' in p) this.rebuildRouting();\n    }\n\n    updateReverb(p: Partial<ReverbParams>): void {\n        Object.assign(this._params.reverb, p);\n        const r = this._params.reverb;\n        const t = this.ctx.currentTime;\n        this.reverbDry.gain.setValueAtTime(1 - r.mix, t);\n        this.reverbWet.gain.setValueAtTime(r.mix, t);\n        if ('decay' in p) {\n            this.convolver.buffer = this.generateImpulse(r.decay);\n        }\n        if ('enabled' in p) this.rebuildRouting();\n    }\n\n    updateDelay(p: Partial<DelayParams>): void {\n        Object.assign(this._params.delay, p);\n        const d = this._params.delay;\n        const t = this.ctx.currentTime;\n        this.delayNode.delayTime.setValueAtTime(d.time, t);\n        this.delayFeedback.gain.setValueAtTime(d.feedback, t);\n        this.delayDry.gain.setValueAtTime(1 - d.mix, t);\n        this.delayWet.gain.setValueAtTime(d.mix, t);\n        if ('enabled' in p) this.rebuildRouting();\n    }\n\n    applyAll(fx: StemEffects): void {\n        this.updateEQ(fx.eq);\n        this.updateCompressor(fx.compressor);\n        this.updateReverb(fx.reverb);\n        this.updateDelay(fx.delay);\n    }\n\n    dispose(): void {\n        try {\n            this.input.disconnect();\n            this.output.disconnect();\n        } catch { /* already disconnected */ }\n    }\n}\n","/**\n * DAWController – Per-bar quantized worklet playback with independent stem positioning.\n *\n * Each track's fMP4 is downloaded and decoded into 4 AudioBuffers (one per stem).\n * Playback uses persistent daw-stem-processor AudioWorklets that hold the full\n * buffer and use a per-bar speed map for sample-accurate tempo quantization.\n *\n * Each bar of each track is played at the exact speed needed to fit the master\n * bar grid, so bars align perfectly across tracks regardless of tempo drift in\n * the original recordings.\n */\n\nimport { BrowserEventEmitter } from './event-emitter';\nimport type { ITrackForSequence } from '@/stem-audio-engine-v3/interface';\nimport { DEFAULT_PITCH_SETTINGS } from '../interfaces';\nimport type { DAWTrack, DAWChannel, PitchEngineSettings } from '../interfaces';\nimport { barToSample, type IBarMapping } from '../primitives/bar-utils';\nimport { useDAWSessionStore } from '../store/daw-session-store';\nimport {\n    loadTrackStemsFromMp3Urls,\n    loadTrackStemsProgressively,\n    type StemAudioBuffers,\n    type LoadProgress,\n    type StemUrls,\n} from './daw-track-loader';\nimport {\n    cacheFullFileForOffline,\n    type ProgressiveLoadHandle,\n    type StemPatch,\n} from './progressive-track-loader';\nimport {\n    initMipmapsForTrack,\n    appendPeaksToMipmapCache,\n    clearWaveformCacheForTrack,\n} from '../components/waveform-cache';\nimport { calculateTempoMatch, semitonesToPitchFactor } from '../primitives/tempo-utils';\nimport { StemEffectChain, type StemEffects } from './stem-effect-chain';\nimport isMobile from 'is-mobile';\nimport { getWorkletPath_RubberBand, getWorkletPath_Phaze } from '@/utils/platform-env';\n\nconst SAMPLE_RATE = 48000;\nconst STEM_LABELS: (keyof StemAudioBuffers)[] = ['other', 'vocals', 'bass', 'drums'];\nconst DAW_WORKLET_PATH = '/workers/daw-stem-processor.js';\nconst USE_PHAZE = isMobile();\nconst PITCH_WORKLET_NAME = USE_PHAZE ? 'phase-vocoder-processor' : 'realtime-pitch-shift-processor';\n\nconst METRO_LOOK_AHEAD_SEC = 0.1;\nconst METRO_SCHEDULE_INTERVAL_MS = 25;\n\n/**\n * Soft cap on how many tracks may hold their decoded PCM in memory at once.\n *\n * A fully decoded 4-min stereo track at 48 kHz costs ~4 stems × ~46M samples\n * × 4 bytes ≈ 180 MB before the worklet+mirror+mipmap triplication. Without\n * a cap, mass imports (stem.fm sessions) or long sessions on low-RAM\n * devices eventually exhaust browser memory and new `decodeAudioData` /\n * `AudioBuffer` allocations start failing silently — the symptom the user\n * sees is \"tracks won't load after the Nth one\".\n *\n * When the cap is reached we DUMP the LRU undumped tracks' PCM (worklet\n * channels + main-thread mirror + mipmap), keep their store metadata +\n * timeline placement intact, and re-decode from IndexedDB on demand when\n * the user plays them again. The encoded fMP4 (~3-5 MB / track) stays in\n * IndexedDB regardless, so re-hydration is local and fast.\n *\n * Mobile devices get a tighter cap by default. Override at runtime via\n * `DAWController.getInstance().setMaxDecodedTracks(n)`.\n */\nconst DEFAULT_MAX_DECODED_TRACKS = isMobile() ? 6 : 12;\n\n// ── Bar map entry (sent to worklet) ─────────────────────────────────\n\ninterface BarMapEntry {\n    sampleStart: number;\n    speed: number;\n}\n\n/** Per-bar pitch correction data (kept on main thread for Rubberband updates) */\ninterface BarPitchEntry {\n    /** Pitch correction in semitones to compensate this bar's speed change */\n    pitchCorrectionSemitones: number;\n    /** The speed ratio for this bar (for logging) */\n    speed: number;\n}\n\n/**\n * Build a per-bar speed map for a track.\n *\n * When subdivisions > 1, speed is linearly interpolated between parent bar\n * midpoints, then rescaled so the average speed across each parent bar's\n * sub-bars equals the original parent speed. This preserves total timing\n * while giving a smooth speed ramp at bar boundaries.\n */\nfunction buildBarMap(\n    barMapping: IBarMapping[],\n    masterBarLength: number,\n    nativeBarLength: number,\n    subdivisions: number = 1,\n): BarMapEntry[] {\n    const tempoMatch = calculateTempoMatch(masterBarLength, nativeBarLength);\n    const targetBarLength = tempoMatch.targetBarLength;\n    const subs = Math.max(1, subdivisions);\n\n    const parentSpeeds = barMapping.map((bar) => bar.duration / targetBarLength);\n\n    if (subs <= 1) {\n        return barMapping.map((bar, i) => ({\n            sampleStart: bar.sampleStart,\n            speed: parentSpeeds[i],\n        }));\n    }\n\n    // Compute globally continuous interpolated speeds — no per-bar rescaling.\n    // Each sub-bar's speed is linearly interpolated between parent bar midpoints,\n    // giving a perfectly smooth ramp with zero boundary discontinuity.\n    //\n    // Timing is preserved by computing each sub-bar's sampleStart from the\n    // cumulative native samples consumed at the interpolated speed, anchored to\n    // each parent bar's known sampleStart.\n    const totalSubs = barMapping.length * subs;\n    const speeds: number[] = new Array(totalSubs);\n\n    for (let i = 0; i < totalSubs; i++) {\n        const parentPos = (i + 0.5) / subs;\n        const lo = Math.floor(parentPos - 0.5);\n        const hi = lo + 1;\n        const t = parentPos - 0.5 - lo;\n        const loSpeed = parentSpeeds[Math.max(0, Math.min(lo, parentSpeeds.length - 1))];\n        const hiSpeed = parentSpeeds[Math.max(0, Math.min(hi, parentSpeeds.length - 1))];\n        speeds[i] = loSpeed + (hiSpeed - loSpeed) * t;\n    }\n\n    // Build result with sampleStart positions derived from within each parent\n    // bar. Sub-bar durations are proportional to their speed relative to the\n    // bar's average speed, so the total duration stays correct.\n    const result: BarMapEntry[] = [];\n    for (let b = 0; b < barMapping.length; b++) {\n        const bar = barMapping[b];\n        const offset = b * subs;\n\n        // Sum of speeds within this parent bar (for proportional distribution)\n        let speedSum = 0;\n        for (let s = 0; s < subs; s++) speedSum += speeds[offset + s];\n\n        // Distribute the parent bar's native sample duration proportionally\n        let cumSamples = 0;\n        for (let s = 0; s < subs; s++) {\n            result.push({\n                sampleStart: Math.round(bar.sampleStart + cumSamples),\n                speed: speeds[offset + s],\n            });\n            // Samples consumed by this sub-bar = (its share of the total speed) * total duration\n            cumSamples += (speeds[offset + s] / speedSum) * bar.duration;\n        }\n    }\n    return result;\n}\n\n/**\n * Build per-bar pitch correction values.\n * Each entry's pitch correction exactly cancels the speed artifact for that\n * entry's speed: -12 * log2(speed). Speed and pitch are always paired —\n * when speed is interpolated across sub-bars, the pitch correction follows.\n */\nfunction buildBarPitchMap(barMap: BarMapEntry[]): BarPitchEntry[] {\n    return barMap.map((entry) => ({\n        pitchCorrectionSemitones: -12 * Math.log2(entry.speed),\n        speed: entry.speed,\n    }));\n}\n\n// `barToSample` is the canonical bar→sample function, defined alongside\n// `IBarMapping` in primitives/bar-utils.ts and imported above. Keeping a\n// single implementation is critical: the timeline waveform renderer uses\n// the same function to slice the mipmap, which is the only way to keep\n// the drawn peaks visually aligned with the audio that's actually played\n// when bars have non-uniform durations (per-bar tempo variation).\n\n// ── Per-track runtime state ──────────────────────────────────────────\n\ninterface TrackRuntime {\n    stems: StemAudioBuffers;\n    worklets: AudioWorkletNode[];   // [other, vocals, bass, drums]\n    stemGains: GainNode[];\n    stemEffects: StemEffectChain[];\n    trackGain: GainNode;\n    pitchWorklet: AudioWorkletNode | null;\n    barPitchMap: BarPitchEntry[];\n    /** Track which bar's pitch correction is currently applied (-1 = none) */\n    currentPitchBar: number;\n    /** Current autotune correction in semitones (set by the autotune panel, summed into all pitch updates) */\n    autotuneOffset: number;\n    /** Last pitch string sent to the pitch worklet — used to skip redundant commands */\n    lastPitchStr: string;\n    /** Last correction in semitones — used for delta logging */\n    lastCorrectionSt?: number;\n    /** Progressive load handle (fMP4 streaming) — shared via `sourceCache`\n     *  when multiple track instances reference the same source URL. */\n    loadHandle?: ProgressiveLoadHandle | null;\n    /** Cache key (source URL) so `removeTrack` can decrement refcount on\n     *  the shared `sourceCache` entry. `null` for legacy MP3 / stem-URL\n     *  loads which aren't cached (always one-shot). */\n    sourceCacheKey?: string | null;\n    /** Unsubscribe handle for the shared `loadHandle.onPatch` listener\n     *  registered by this track instance. Called from `removeTrack` so\n     *  the cache stops fanning patches at a disposed worklet. */\n    unsubscribePatch?: (() => void) | null;\n    /** True once the playable threshold has been crossed (first ~8s decoded) */\n    playable?: boolean;\n    /** True once every byte has decoded */\n    loaded?: boolean;\n    /**\n     * True when the audio PCM has been dumped to free memory. The worklets,\n     * audio graph and store track stay intact — the track still exists on\n     * the timeline, just silent — until `rehydrateTrackAudio` re-decodes\n     * it from IndexedDB.\n     */\n    dumped?: boolean;\n    /**\n     * `performance.now()` of the last user-facing interaction with this\n     * track (play, seek into its range, patch landing). Drives LRU eviction\n     * when the decoded-tracks cap is exceeded.\n     */\n    lastAccessAt: number;\n    /**\n     * Snapshot of the original `ITrackForSequence` used to add this track.\n     * Needed so `rehydrateTrackAudio` can re-run the progressive loader\n     * without depending on the (mutated) store state.\n     */\n    trackData: ITrackForSequence;\n    /**\n     * True while a dump or rehydrate is in flight. Prevents reentrancy\n     * (e.g. the eviction sweep racing with auto-rehydration on play).\n     */\n    transitioning?: boolean;\n}\n\n// ── Shared source cache (Phase 1: dedup decode work across clips) ────\n\n/**\n * One-per-unique-source-URL entry. Lets multiple `DAWTrack` instances\n * (e.g. the N clips a stem.fm session emits for the same source track)\n * share a single fMP4 fetch + WebCodecs decode pipeline.\n *\n * Each subscriber still owns its own worklets, mute regions and\n * timeline placement — only the expensive bits (network + decode) are\n * shared. Patches arrive from `loadHandle.onPatch` and are fanned out\n * to every subscriber's per-track listener via `loadHandle.onPatch`'s\n * native multi-subscribe (we just clone the patch arrays before each\n * worklet `postMessage` so the transfer-to-worklet on subscriber #1\n * doesn't detach the buffer for subscribers #2…N).\n */\ninterface SourceCacheEntry {\n    /** Source URL — also the cache key. */\n    url: string;\n    /** How many `DAWTrack` instances currently reference this source. */\n    refCount: number;\n    /**\n     * Initial stem buffers returned by `loadTrackStemsProgressively`.\n     * Filled in by the patch listener as decoding progresses; kept for\n     * the lifetime of the cache entry so newly-joining tracks can\n     * snapshot the already-decoded portion when binding their worklets.\n     */\n    stems: StemAudioBuffers;\n    /** Underlying fMP4 progressive decode handle (shared). */\n    loadHandle: ProgressiveLoadHandle | null;\n}\n\n// ═════════════════════════════════════════════════════════════════════\n\nexport class DAWController extends BrowserEventEmitter {\n    private ctx: AudioContext | null = null;\n    private masterGain: GainNode | null = null;\n    private tracks: Map<string, TrackRuntime> = new Map();\n    /** URL → shared decode entry. See `SourceCacheEntry`. */\n    private sourceCache: Map<string, SourceCacheEntry> = new Map();\n    private isInitialised = false;\n    private _pitchWorkletAvailable = false;\n\n    private _playing = false;\n    private _playStartCtxTime = 0;\n    private _playStartOffset = 0;\n    private playheadAnimationId: number | null = null;\n    private _seekGen = 0;\n\n    /**\n     * Max number of tracks allowed to hold their decoded PCM concurrently.\n     * Beyond this we evict LRU tracks via `dumpTrackAudio`. See\n     * `DEFAULT_MAX_DECODED_TRACKS` for the rationale.\n     */\n    private _maxDecodedTracks = DEFAULT_MAX_DECODED_TRACKS;\n\n    // Track which stems are currently playing so we can start/stop\n    // them as the playhead crosses their timeline boundaries.\n    private _stemActive = new Map<string, boolean>();\n\n    // Metronome\n    private _metronomeOn = false;\n    private _metronomeDownbeatOnly = true;\n    private _metronomeGain: GainNode | null = null;\n    private _metronomeScheduleId: ReturnType<typeof setInterval> | null = null;\n    private _metronomeNextBeat = 0;\n\n    // CPU monitoring\n    private _cpuPollId: ReturnType<typeof setInterval> | null = null;\n    private _cpuUsage = 0;\n\n    // Native playback mode — disables tempo quantization for a single track\n    // so the beat lab can play audio at original speed (beat markers align 1:1).\n    private _nativeModeTrackId: string | null = null;\n    private _nativeModeOriginalBarMap: BarMapEntry[] | null = null;\n    /** Native-time offset (samples) at the moment play/seek was last called in native mode */\n    private _nativeStartSample = 0;\n\n    private static instance: DAWController | null = null;\n    static getInstance(): DAWController {\n        if (!DAWController.instance) DAWController.instance = new DAWController();\n        return DAWController.instance;\n    }\n    private constructor() { super(); }\n\n    // ── Init ──────────────────────────────────────────────────────────\n\n    async init(): Promise<void> {\n        if (this.isInitialised) return;\n        this.ctx = new AudioContext({ sampleRate: SAMPLE_RATE });\n        console.log('[DAWController.init] AudioContext created, state:', this.ctx.state);\n        this.ctx.addEventListener('statechange', () => {\n            console.log('[DAWController] AudioContext statechange →', this.ctx!.state);\n            if (this.ctx!.state === 'suspended' && this._playing) {\n                console.warn('[DAWController] Context suspended while playing — auto-resuming');\n                this.ctx!.resume().catch(() => {});\n            }\n        });\n        await this.ctx.audioWorklet.addModule(DAW_WORKLET_PATH);\n        const pitchWorkletPath = USE_PHAZE ? getWorkletPath_Phaze() : getWorkletPath_RubberBand();\n        try {\n            await this.ctx.audioWorklet.addModule(pitchWorkletPath);\n            this._pitchWorkletAvailable = true;\n        } catch (err) {\n            console.warn('[DAWController] Pitch worklet failed to load, transpose unavailable:', err);\n            this._pitchWorkletAvailable = false;\n        }\n        this.masterGain = this.ctx.createGain();\n        this.masterGain.connect(this.ctx.destination);\n        this.isInitialised = true;\n        this.emit('initialised');\n    }\n\n    isReady() { return this.isInitialised; }\n\n    /**\n     * Resolve when the named track's audio has fully decoded. If the\n     * decode is already complete (common with the source-cache when\n     * multiple clips share one source), resolve immediately on the next\n     * microtask. Rejects if the track is removed or no such track exists.\n     *\n     * Use this instead of `controller.on('trackLoaded', …)` from\n     * importers / batch loaders — the event fires once and a late\n     * subscriber would miss it. This helper closes that race.\n     */\n    whenTrackLoaded(trackId: string): Promise<void> {\n        const rt = this.tracks.get(trackId);\n        if (!rt) return Promise.reject(new Error(`Unknown track ${trackId}`));\n        if (rt.loaded) return Promise.resolve();\n        return new Promise<void>((resolve, reject) => {\n            const onLoaded = (loadedId: string) => {\n                if (loadedId !== trackId) return;\n                cleanup();\n                resolve();\n            };\n            const onRemoved = (removedId: string) => {\n                if (removedId !== trackId) return;\n                cleanup();\n                reject(new Error(`Track ${trackId} removed before load completed`));\n            };\n            const cleanup = () => {\n                this.off('trackLoaded', onLoaded);\n                this.off('trackRemoved', onRemoved);\n            };\n            this.on('trackLoaded', onLoaded);\n            this.on('trackRemoved', onRemoved);\n        });\n    }\n    getAudioContext() { return this.ctx; }\n    getMasterGainNode() { return this.masterGain; }\n\n    // ── Track lifecycle ──────────────────────────────────────────────\n\n    async addTrack(\n        trackData: ITrackForSequence,\n        onProgress?: (p: LoadProgress) => void,\n    ): Promise<DAWTrack | null> {\n        if (!this.isInitialised) await this.init();\n\n        // Before kicking off a new decode, free PCM from the LRU tracks if\n        // we're at or above the cap. We free down to `cap - 1` so the new\n        // track has headroom. Dumped tracks keep their store entry and\n        // timeline placement — they can be rehydrated on play/seek from\n        // IndexedDB. Without this gate, mass imports eventually OOM the\n        // browser and subsequent `addTrack`s fail silently inside the\n        // progressive decoder.\n        this.evictLRUDecodedTracks(this._maxDecodedTracks - 1);\n\n        let stems: StemAudioBuffers;\n        let loadHandle: ProgressiveLoadHandle | null = null;\n        let sourceCacheKey: string | null = null;\n        const stemUrls = (trackData as { stemUrls?: StemUrls }).stemUrls;\n        try {\n            if (stemUrls) {\n                // MP3 stem-URL path stays as the legacy full-download flow.\n                // Not source-cached — it's the legacy path used only by\n                // user-uploaded stems, which are unique per upload anyway.\n                // TODO: progressive variant requires a WebCodecs AudioDecoder\n                // with mp3 codec + a custom MPEG frame splitter.\n                stems = await loadTrackStemsFromMp3Urls(stemUrls, onProgress);\n            } else {\n                // Progressive fMP4 path with shared-source cache.\n                //\n                // If another `DAWTrack` is already streaming the same URL\n                // (e.g. a previous clip from the same stem.fm session\n                // import), reuse its in-flight decode pipeline — we get\n                // back the same `stems` buffers (which fill in as decoding\n                // progresses) and the same `loadHandle` (which fans patches\n                // to every subscriber). On a fresh URL we kick off a new\n                // progressive load and stash it for any future siblings.\n                sourceCacheKey = trackData.url;\n                const cached = this.sourceCache.get(sourceCacheKey);\n                if (cached) {\n                    cached.refCount++;\n                    stems = cached.stems;\n                    loadHandle = cached.loadHandle;\n                    // No fresh `onProgress` events will fire for a cache\n                    // hit (the loadHandle was created earlier), so synth a\n                    // best-effort \"instant\" progress to the caller.\n                    onProgress?.({ phase: 'decoding', percent: 100, detail: 'Using shared source decode' });\n                } else {\n                    const result = await loadTrackStemsProgressively(\n                        trackData.url,\n                        trackData.duration,\n                        onProgress,\n                    );\n                    stems = result.initial;\n                    loadHandle = result.handle;\n                    this.sourceCache.set(sourceCacheKey, {\n                        url: sourceCacheKey,\n                        refCount: 1,\n                        stems,\n                        loadHandle,\n                    });\n                }\n            }\n        } catch (err) {\n            console.error('[DAWController] Failed to load stems:', err);\n            return null;\n        }\n\n        const store = useDAWSessionStore.getState();\n        const dawTrack = store.addTrack(trackData);\n        const trackId = dawTrack.id;\n\n        const trackGain = this.ctx!.createGain();\n        trackGain.connect(this.masterGain!);\n\n        // Create pitch worklet (one per track, between stem gains and trackGain)\n        let pitchWorklet: AudioWorkletNode | null = null;\n        if (this._pitchWorkletAvailable) {\n            try {\n                const settings = dawTrack.pitchSettings;\n                if (USE_PHAZE) {\n                    pitchWorklet = new AudioWorkletNode(this.ctx!, PITCH_WORKLET_NAME, {\n                        channelCount: 2,\n                        channelCountMode: 'explicit',\n                        outputChannelCount: [2],\n                    });\n                } else {\n                    pitchWorklet = new AudioWorkletNode(this.ctx!, PITCH_WORKLET_NAME, {\n                        numberOfInputs: 1,\n                        numberOfOutputs: 1,\n                        channelCount: 2,\n                        channelCountMode: 'explicit',\n                        outputChannelCount: [2],\n                        processorOptions: {\n                            pitchTransitionStep: 1.0,\n                            formantOption: settings.formantOption,\n                            pitchOption: settings.pitchOption,\n                            engineOption: settings.engineOption,\n                            quality: true,\n                        },\n                    });\n                }\n                pitchWorklet.connect(trackGain);\n                this.configurePitchWorklet(pitchWorklet, settings);\n            } catch (err) {\n                console.warn('[DAWController] Failed to create pitch worklet for track:', err);\n                pitchWorklet = null;\n            }\n        }\n\n        // Stem signals connect to pitchWorklet if available, otherwise directly to trackGain\n        const stemTarget = pitchWorklet ?? trackGain;\n\n        const worklets: AudioWorkletNode[] = [];\n        const stemGains: GainNode[] = [];\n        const stemEffects: StemEffectChain[] = [];\n\n        // Build the per-bar speed map for this track (subdivided if configured)\n        const freshStore = useDAWSessionStore.getState();\n        const freshTrack = freshStore.getTrackById(trackId)!;\n        const barMap = buildBarMap(\n            freshTrack.barMapping,\n            freshStore.masterBarLength48000,\n            freshTrack.nativeBarLength,\n            freshStore.barSubdivisions,\n        );\n\n        STEM_LABELS.forEach((label, stemIdx) => {\n            const stemGain = this.ctx!.createGain();\n            stemGain.connect(stemTarget);\n\n            // Per-stem effect chain: worklet → effectChain → stemGain\n            const fxChain = new StemEffectChain(this.ctx!);\n            fxChain.output.connect(stemGain);\n\n            const worklet = new AudioWorkletNode(this.ctx!, 'daw-stem-processor', {\n                numberOfInputs: 0,\n                numberOfOutputs: 1,\n                outputChannelCount: [2],\n            });\n            worklet.connect(fxChain.input);\n\n            const buf = stems[label];\n            if (buf) {\n                const channels: Float32Array[] = [];\n                for (let ch = 0; ch < buf.numberOfChannels; ch++) {\n                    channels.push(new Float32Array(buf.getChannelData(ch)));\n                }\n                worklet.port.postMessage({ type: 'load-buffer', channels }, channels.map(c => c.buffer));\n            }\n\n            // Send bar map so the worklet can do per-bar speed adjustment\n            worklet.port.postMessage({ type: 'load-bar-map', entries: barMap });\n            worklet.port.postMessage({ type: 'set-global-rate', rate: freshStore.globalPlaybackRate });\n\n            // Listen for bar-boundary crossings from the first stem worklet only\n            if (stemIdx === 0) {\n                worklet.port.onmessage = (msg: MessageEvent) => {\n                    const d = msg.data;\n                    if (d?.type === 'bar-changed') {\n                        this.handleWorkletBarChange(trackId, d.barIndex);\n                    }\n                };\n            }\n\n            worklets.push(worklet);\n            stemGains.push(stemGain);\n            stemEffects.push(fxChain);\n        });\n\n        const barPitchMap = buildBarPitchMap(barMap);\n        const rt: TrackRuntime = {\n            stems,\n            worklets,\n            stemGains,\n            stemEffects,\n            trackGain,\n            pitchWorklet,\n            barPitchMap,\n            currentPitchBar: -1,\n            autotuneOffset: 0,\n            lastPitchStr: '',\n            loadHandle,\n            sourceCacheKey,\n            unsubscribePatch: null,\n            playable: loadHandle ? false : true,\n            loaded: loadHandle ? false : true,\n            dumped: false,\n            lastAccessAt: performance.now(),\n            trackData,\n        };\n        this.tracks.set(trackId, rt);\n\n        // Wire per-bar pitch data to the first stem worklet for sample-accurate\n        // pitch changes synced with speed changes (no main-thread round-trip)\n        this.syncWorkletPitchMap(trackId);\n\n        // Apply initial pitch correction (compensate tempo-induced drift)\n        this.syncPitchForTrack(trackId);\n        this.syncGainsForTrack(trackId);\n\n        // ── Progressive decode wiring ─────────────────────────────────\n        // Init waveform mipmaps for all 4 stems at the track's total length.\n        // Patches will fill in the mipmap incrementally; un-decoded regions\n        // render as a flat centerline until they land.\n        if (loadHandle) {\n            const totalSamples = loadHandle.totalSamples;\n            for (const stemKey of ['bass', 'drums', 'other', 'vocals'] as const) {\n                initMipmapsForTrack(trackId, stemKey, totalSamples);\n            }\n\n            // Patch listener: route incoming PCM slots to (a) the worklet,\n            // (b) the main-thread AudioBuffer mirror used by non-worklet\n            // consumers (BarSampler, copy/paste), (c) the waveform mipmap.\n            //\n            // IMPORTANT: With shared-source caching, multiple `DAWTrack`\n            // instances may register patch listeners on the same handle.\n            // The `loadHandle` calls each listener synchronously with the\n            // SAME `patch.samples` reference. If we transfer those buffers\n            // to a worklet, subscribers further down the list see detached\n            // (length-0) Float32Arrays. So we ALWAYS clone for the worklet\n            // post — losing the zero-copy optimisation, but keeping the\n            // patch arrays usable for sibling subscribers. Clone cost is\n            // ~30ms per multi-stem batch, negligible vs. the decode work\n            // it lets us avoid re-doing.\n            const unsubscribe = loadHandle.onPatch((patch: StemPatch) => {\n                const rtNow = this.tracks.get(trackId);\n                if (!rtNow) return; // track removed while loading\n                if (rtNow.dumped) return; // patches arriving for a dumped track are stale\n\n                // Decoding patches are user-relevant activity — bump LRU so\n                // a still-loading track isn't picked for eviction.\n                rtNow.lastAccessAt = performance.now();\n\n                // STEM_LABELS = ['other', 'vocals', 'bass', 'drums']\n                const stemIdx = STEM_LABELS.indexOf(patch.stem as keyof StemAudioBuffers);\n                if (stemIdx < 0) return;\n\n                const w = rtNow.worklets[stemIdx];\n                const [left, right] = patch.samples;\n\n                // Clone three times (worklet, main-thread mirror, mipmap)\n                // so this listener never observes a buffer that's been\n                // transferred elsewhere by a sibling subscriber.\n                const workletL = new Float32Array(left);\n                const workletR = new Float32Array(right);\n                const mirrorL = new Float32Array(left);\n                const mirrorR = new Float32Array(right);\n\n                // (a) Hand the patch to the worklet. The clones are local\n                //     to this subscriber so transferring is safe.\n                w.port.postMessage(\n                    { type: 'patch-buffer', sampleOffset: patch.sampleOffset, channels: [workletL, workletR] },\n                    [workletL.buffer, workletR.buffer],\n                );\n\n                // (b) Main-thread AudioBuffer mirror — keeps getStemBuffers()\n                //     and bar-sampler/paste-audio operations in sync with\n                //     what the worklet hears. With shared sources every\n                //     subscriber writes the same data to the same buffer\n                //     (the cache entry's stems object IS rtNow.stems);\n                //     it's idempotent and cheap.\n                const buf = rtNow.stems[patch.stem];\n                if (buf) {\n                    const writeStart = patch.sampleOffset;\n                    const remaining = buf.length - writeStart;\n                    const writeLen = Math.max(0, Math.min(mirrorL.length, remaining));\n                    if (writeLen > 0) {\n                        const ch0 = buf.getChannelData(0);\n                        ch0.set(mirrorL.subarray(0, writeLen), writeStart);\n                        if (buf.numberOfChannels > 1) {\n                            const ch1 = buf.getChannelData(1);\n                            ch1.set(mirrorR.subarray(0, writeLen), writeStart);\n                        }\n                    }\n                }\n\n                // (c) Waveform mipmap update — bumps the version counter so\n                //     the timeline canvas will redraw on its next frame.\n                //     Per-track because mipmaps are keyed by trackId.\n                appendPeaksToMipmapCache(\n                    trackId,\n                    patch.stem,\n                    patch.sampleOffset,\n                    mirrorL,\n                    mirrorR,\n                );\n            });\n            rt.unsubscribePatch = unsubscribe;\n\n            // Cache hit: the source has likely already received some\n            // patches before this track joined. The fresh worklet only\n            // got the load-buffer snapshot of `stems` AS IT WAS at that\n            // moment, but the mipmap cache for this trackId is empty —\n            // backfill it from the snapshot we already loaded into the\n            // worklet so the timeline waveform renders without waiting\n            // for the next patch.\n            //\n            // For the fresh-load case the snapshot is all zeros, so\n            // backfilling is a no-op (mipmaps stay flat until patches\n            // arrive). For cache-hit it produces an instant waveform.\n            if (sourceCacheKey && this.sourceCache.get(sourceCacheKey)?.refCount && this.sourceCache.get(sourceCacheKey)!.refCount > 1) {\n                for (const stemKey of ['bass', 'drums', 'other', 'vocals'] as const) {\n                    const buf = stems[stemKey];\n                    if (!buf) continue;\n                    const ch0 = new Float32Array(buf.getChannelData(0));\n                    const ch1 = buf.numberOfChannels > 1\n                        ? new Float32Array(buf.getChannelData(1))\n                        : ch0;\n                    appendPeaksToMipmapCache(trackId, stemKey, 0, ch0, ch1);\n                }\n            }\n\n            loadHandle.playable\n                .then(() => {\n                    const rtNow = this.tracks.get(trackId);\n                    if (!rtNow) return;\n                    rtNow.playable = true;\n                    this.emit('trackPlayable', trackId);\n                })\n                .catch((err) => {\n                    console.warn('[DAWController] progressive load failed before playable threshold:', err);\n                });\n\n            loadHandle.complete\n                .then(() => {\n                    const rtNow = this.tracks.get(trackId);\n                    if (!rtNow) return;\n                    rtNow.loaded = true;\n                    this.emit('trackLoaded', trackId);\n                    // Best-effort cache full file for offline reopen\n                    void cacheFullFileForOffline(trackData.url);\n                })\n                .catch((err) => {\n                    console.warn('[DAWController] progressive load failed:', err);\n                });\n        }\n\n        this.emit('trackAdded', trackId);\n        return dawTrack;\n    }\n\n    // ── Decoded-tracks memory cap (dump / rehydrate) ─────────────────\n\n    /**\n     * Read the current cap. See `DEFAULT_MAX_DECODED_TRACKS` for context.\n     */\n    get maxDecodedTracks(): number { return this._maxDecodedTracks; }\n\n    /**\n     * Update the soft cap on simultaneously-decoded tracks. If the new cap\n     * is lower than the current decoded count, the surplus LRU tracks are\n     * dumped immediately.\n     */\n    setMaxDecodedTracks(n: number): void {\n        const clamped = Math.max(1, Math.floor(n));\n        if (clamped === this._maxDecodedTracks) return;\n        this._maxDecodedTracks = clamped;\n        this.evictLRUDecodedTracks(clamped);\n        this.emit('maxDecodedTracksChanged', clamped);\n    }\n\n    /** Tracks currently holding their decoded PCM in memory. */\n    private decodedTrackCount(): number {\n        let n = 0;\n        for (const rt of this.tracks.values()) if (!rt.dumped) n++;\n        return n;\n    }\n\n    /** True if the named track is currently audible (playing into its range). */\n    private isTrackAudible(trackId: string): boolean {\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!track) return false;\n        const bar = store.playheadBar;\n        for (const ch of track.channels) {\n            const visStart = ch.timelineStartBar + ch.trimStartBar;\n            const visEnd = ch.timelineStartBar + ch.trimEndBar;\n            if (bar >= visStart && bar <= visEnd) return true;\n        }\n        return false;\n    }\n\n    /**\n     * Free PCM from the least-recently-used undumped tracks until at most\n     * `keep` tracks remain decoded. Audible tracks (containing the playhead)\n     * and currently-transitioning tracks are excluded from eviction so\n     * playback never goes silent unexpectedly.\n     */\n    private evictLRUDecodedTracks(keep: number): void {\n        const max = Math.max(0, Math.floor(keep));\n        let decoded = this.decodedTrackCount();\n        if (decoded <= max) return;\n\n        const candidates: Array<{ id: string; rt: TrackRuntime }> = [];\n        for (const [id, rt] of this.tracks) {\n            if (rt.dumped) continue;\n            if (rt.transitioning) continue;\n            if (this.isTrackAudible(id)) continue;\n            candidates.push({ id, rt });\n        }\n        candidates.sort((a, b) => a.rt.lastAccessAt - b.rt.lastAccessAt);\n\n        while (decoded > max && candidates.length > 0) {\n            const victim = candidates.shift();\n            if (!victim) break;\n            this.dumpTrackAudio(victim.id);\n            decoded--;\n        }\n\n        if (decoded > max) {\n            // Every candidate was audible / transitioning — log and move on\n            // rather than silencing the playhead. The cap will be respected\n            // again on the next eviction sweep.\n            console.warn(\n                `[DAWController] decoded-tracks cap exceeded: have ${decoded}, ` +\n                `want ${max}, but all extras are audible/transitioning.`,\n            );\n        }\n    }\n\n    /**\n     * Release the decoded PCM held by a track to free memory without\n     * removing the track from the timeline. The worklets, gains, and\n     * effect chain stay wired so rehydration is a single message away.\n     *\n     *   - Worklet stem buffers: `dump-buffer` clears the Float32Arrays.\n     *   - Main-thread mirror (`rt.stems`): replaced with 1-sample silent\n     *     AudioBuffers so legacy consumers (BarSampler, paste-audio) don't\n     *     null-deref. Re-allocated to full length on rehydrate.\n     *   - Waveform mipmaps: cleared so the timeline shows a flat\n     *     centerline until the track is rehydrated.\n     *   - In-flight decode: aborted (refcount on shared source decremented).\n     */\n    dumpTrackAudio(trackId: string): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt || rt.dumped) return;\n\n        rt.transitioning = true;\n        try {\n            try { rt.unsubscribePatch?.(); } catch { /* */ }\n            rt.unsubscribePatch = null;\n\n            if (rt.sourceCacheKey) {\n                const entry = this.sourceCache.get(rt.sourceCacheKey);\n                if (entry) {\n                    entry.refCount--;\n                    if (entry.refCount <= 0) {\n                        try { entry.loadHandle?.abort(); } catch { /* */ }\n                        this.sourceCache.delete(rt.sourceCacheKey);\n                    }\n                }\n            } else {\n                try { rt.loadHandle?.abort(); } catch { /* */ }\n            }\n            rt.loadHandle = null;\n            // Clear so a later `removeTrack` doesn't double-decrement the\n            // shared refcount. `rehydrateTrackAudio` re-acquires the slot.\n            rt.sourceCacheKey = null;\n\n            for (const w of rt.worklets) {\n                try { w.port.postMessage({ type: 'dump-buffer' }); } catch { /* */ }\n            }\n            for (let i = 0; i < rt.worklets.length; i++) {\n                const key = `${trackId}:${i}`;\n                this._stemActive.set(key, false);\n            }\n\n            // Replace the 4 AudioBuffers with silent stubs. We can't `null`\n            // them because the type contract is non-nullable and several\n            // call sites assume the keys exist (waveform renderer, paste\n            // audio, native-mode seek). 1-sample stereo buffers are\n            // ~triple-pointer overhead each — negligible vs. the multi-MB\n            // they replace.\n            if (this.ctx) {\n                const ctx = this.ctx;\n                const silent = (): AudioBuffer => ctx.createBuffer(2, 1, SAMPLE_RATE);\n                rt.stems = {\n                    other: silent(),\n                    vocals: silent(),\n                    bass: silent(),\n                    drums: silent(),\n                };\n            }\n\n            clearWaveformCacheForTrack(trackId);\n\n            rt.dumped = true;\n            rt.playable = false;\n            rt.loaded = false;\n        } finally {\n            rt.transitioning = false;\n        }\n\n        this.emit('trackDumped', trackId);\n    }\n\n    /**\n     * Re-decode a previously-dumped track from IndexedDB (the encoded fMP4\n     * stays cached even after dump). Reuses the existing worklet/audio\n     * graph — no node recreation — so rehydration is just a fresh\n     * progressive decode that streams patches back into the same\n     * worklets and mipmap cache.\n     *\n     * Returns true once `loadHandle.playable` resolves (enough audio for\n     * playback to start). Background batches continue filling the buffer.\n     */\n    async rehydrateTrackAudio(trackId: string): Promise<boolean> {\n        const rt = this.tracks.get(trackId);\n        if (!rt) return false;\n        if (!rt.dumped || rt.transitioning) return !rt.dumped;\n        if (!this.isInitialised) await this.init();\n\n        // Free a slot before allocating PCM for the rehydrate so we don't\n        // exceed the cap and instantly evict somebody else.\n        this.evictLRUDecodedTracks(this._maxDecodedTracks - 1);\n\n        rt.transitioning = true;\n        rt.lastAccessAt = performance.now();\n        try {\n            const stemUrls = (rt.trackData as { stemUrls?: StemUrls }).stemUrls;\n            let stems: StemAudioBuffers;\n            let loadHandle: ProgressiveLoadHandle | null = null;\n            let sourceCacheKey: string | null = null;\n\n            if (stemUrls) {\n                stems = await loadTrackStemsFromMp3Urls(stemUrls);\n            } else {\n                sourceCacheKey = rt.trackData.url;\n                const cached = this.sourceCache.get(sourceCacheKey);\n                if (cached) {\n                    cached.refCount++;\n                    stems = cached.stems;\n                    loadHandle = cached.loadHandle;\n                } else {\n                    const result = await loadTrackStemsProgressively(\n                        rt.trackData.url,\n                        rt.trackData.duration,\n                    );\n                    stems = result.initial;\n                    loadHandle = result.handle;\n                    this.sourceCache.set(sourceCacheKey, {\n                        url: sourceCacheKey,\n                        refCount: 1,\n                        stems,\n                        loadHandle,\n                    });\n                }\n            }\n\n            rt.stems = stems;\n            rt.loadHandle = loadHandle;\n            rt.sourceCacheKey = sourceCacheKey;\n            rt.dumped = false;\n            rt.playable = loadHandle ? false : true;\n            rt.loaded = loadHandle ? false : true;\n\n            STEM_LABELS.forEach((label, stemIdx) => {\n                const buf = stems[label];\n                const w = rt.worklets[stemIdx];\n                if (!buf || !w) return;\n                const channels: Float32Array[] = [];\n                for (let ch = 0; ch < buf.numberOfChannels; ch++) {\n                    channels.push(new Float32Array(buf.getChannelData(ch)));\n                }\n                w.port.postMessage({ type: 'load-buffer', channels }, channels.map((c) => c.buffer));\n            });\n\n            if (loadHandle) {\n                const totalSamples = loadHandle.totalSamples;\n                for (const stemKey of ['bass', 'drums', 'other', 'vocals'] as const) {\n                    initMipmapsForTrack(trackId, stemKey, totalSamples);\n                }\n\n                rt.unsubscribePatch = loadHandle.onPatch((patch: StemPatch) => {\n                    const rtNow = this.tracks.get(trackId);\n                    if (!rtNow || rtNow.dumped) return;\n                    rtNow.lastAccessAt = performance.now();\n\n                    const stemIdx = STEM_LABELS.indexOf(patch.stem as keyof StemAudioBuffers);\n                    if (stemIdx < 0) return;\n\n                    const w = rtNow.worklets[stemIdx];\n                    const [left, right] = patch.samples;\n                    const workletL = new Float32Array(left);\n                    const workletR = new Float32Array(right);\n                    const mirrorL = new Float32Array(left);\n                    const mirrorR = new Float32Array(right);\n\n                    w.port.postMessage(\n                        { type: 'patch-buffer', sampleOffset: patch.sampleOffset, channels: [workletL, workletR] },\n                        [workletL.buffer, workletR.buffer],\n                    );\n\n                    const buf = rtNow.stems[patch.stem];\n                    if (buf) {\n                        const writeStart = patch.sampleOffset;\n                        const remaining = buf.length - writeStart;\n                        const writeLen = Math.max(0, Math.min(mirrorL.length, remaining));\n                        if (writeLen > 0) {\n                            const ch0 = buf.getChannelData(0);\n                            ch0.set(mirrorL.subarray(0, writeLen), writeStart);\n                            if (buf.numberOfChannels > 1) {\n                                const ch1 = buf.getChannelData(1);\n                                ch1.set(mirrorR.subarray(0, writeLen), writeStart);\n                            }\n                        }\n                    }\n\n                    appendPeaksToMipmapCache(\n                        trackId,\n                        patch.stem,\n                        patch.sampleOffset,\n                        mirrorL,\n                        mirrorR,\n                    );\n                });\n\n                loadHandle.playable\n                    .then(() => {\n                        const rtNow = this.tracks.get(trackId);\n                        if (!rtNow || rtNow.dumped) return;\n                        rtNow.playable = true;\n                        this.emit('trackPlayable', trackId);\n                    })\n                    .catch(() => {});\n\n                loadHandle.complete\n                    .then(() => {\n                        const rtNow = this.tracks.get(trackId);\n                        if (!rtNow || rtNow.dumped) return;\n                        rtNow.loaded = true;\n                        this.emit('trackLoaded', trackId);\n                    })\n                    .catch(() => {});\n            }\n\n            this.emit('trackRehydrated', trackId);\n            return true;\n        } catch (err) {\n            console.error('[DAWController] Failed to rehydrate track:', trackId, err);\n            return false;\n        } finally {\n            rt.transitioning = false;\n        }\n    }\n\n    /** True if a track's decoded PCM has been freed. */\n    isTrackDumped(trackId: string): boolean {\n        return !!this.tracks.get(trackId)?.dumped;\n    }\n\n    removeTrack(trackId: string): void {\n        const rt = this.tracks.get(trackId);\n        if (rt) {\n            // Stop fanning patches at this disposed worklet first — the\n            // shared loadHandle may still be running for sibling tracks.\n            try { rt.unsubscribePatch?.(); } catch { /* */ }\n\n            // Source-cache refcount: only abort the underlying decode\n            // when the LAST track referencing this source goes away.\n            // Without this, removing one clip of a multi-clip source\n            // would kill the in-flight decode for every other clip.\n            if (rt.sourceCacheKey) {\n                const entry = this.sourceCache.get(rt.sourceCacheKey);\n                if (entry) {\n                    entry.refCount--;\n                    if (entry.refCount <= 0) {\n                        try { entry.loadHandle?.abort(); } catch { /* */ }\n                        this.sourceCache.delete(rt.sourceCacheKey);\n                    }\n                }\n            } else {\n                // Legacy non-cached path (MP3 stem URLs) — abort directly.\n                try { rt.loadHandle?.abort(); } catch { /* */ }\n            }\n\n            rt.worklets.forEach((w) => {\n                w.port.postMessage({ type: 'stop' });\n                w.disconnect();\n            });\n            rt.stemEffects.forEach((fx) => fx.dispose());\n            rt.stemGains.forEach((g) => g.disconnect());\n            try { rt.pitchWorklet?.disconnect(); } catch { /* */ }\n            rt.trackGain.disconnect();\n        }\n        this.tracks.delete(trackId);\n        clearWaveformCacheForTrack(trackId);\n        useDAWSessionStore.getState().removeTrack(trackId);\n        this.emit('trackRemoved', trackId);\n    }\n\n    getStemBuffers(trackId: string): StemAudioBuffers | undefined {\n        return this.tracks.get(trackId)?.stems;\n    }\n\n    getStemEffectChain(trackId: string, stemIndex: number): StemEffectChain | null {\n        return this.tracks.get(trackId)?.stemEffects[stemIndex] ?? null;\n    }\n\n    updateStemEffects(trackId: string, stemIndex: number, fx: StemEffects): void {\n        const chain = this.getStemEffectChain(trackId, stemIndex);\n        if (chain) chain.applyAll(fx);\n        useDAWSessionStore.getState().setChannelEffects(trackId, stemIndex, fx);\n    }\n\n    // ── Per-bar precise seeking ──────────────────────────────────────\n\n    /**\n     * Compute the native buffer sample position for a stem at a given\n     * global timeline bar, using the track's barMapping for precision.\n     *\n     * Returns null if the stem is outside its active region.\n     */\n    private computeStemSeekSamples(\n        globalBar: number,\n        channel: DAWChannel,\n        track: DAWTrack,\n    ): number | null {\n        const visibleStart = channel.timelineStartBar + channel.trimStartBar;\n        const visibleEnd = channel.timelineStartBar + channel.trimEndBar;\n\n        if (globalBar < visibleStart || globalBar > visibleEnd) return null;\n\n        const localBar = globalBar - channel.timelineStartBar;\n        const barMapping = track.barMapping;\n        if (!barMapping || barMapping.length === 0) return null;\n\n        // Integer and fractional bar for sub-bar precision\n        const barIdx = Math.min(Math.floor(localBar), barMapping.length - 1);\n        const barFrac = localBar - Math.floor(localBar);\n\n        const bar = barMapping[barIdx];\n        // Position within this bar based on fractional part\n        return Math.floor(bar.sampleStart + barFrac * bar.duration);\n    }\n\n    // ── Transport ────────────────────────────────────────────────────\n\n    async play(): Promise<void> {\n        if (!this.isInitialised) await this.init();\n\n        if (this.ctx!.state !== 'running') {\n            console.log('[DAWController.play] AudioContext state:', this.ctx!.state, '— resuming…');\n            await this.ctx!.resume();\n            // `state` is narrowed inside the outer branch; widened read for post-resume check.\n            const stateAfterResume = (this.ctx as AudioContext).state;\n            if (stateAfterResume !== 'running') {\n                // Wait briefly for state change (Chrome can be async)\n                await new Promise<void>((resolve) => {\n                    const check = () => {\n                        if (this.ctx!.state === 'running') return resolve();\n                        this.ctx!.addEventListener('statechange', check, { once: true });\n                        setTimeout(resolve, 500);\n                    };\n                    check();\n                });\n                console.log('[DAWController.play] AudioContext state after wait:', this.ctx!.state);\n            }\n        }\n\n        const store = useDAWSessionStore.getState();\n        const globalBar = store.playheadBar;\n\n        // Auto-rehydrate any dumped tracks whose playable range contains the\n        // playhead. Fire-and-forget — the worklets will start producing\n        // audio as soon as their `playable` threshold resolves. Bumping\n        // `lastAccessAt` here keeps these tracks off the eviction list\n        // while they re-decode.\n        for (const [trackId, rt] of this.tracks) {\n            if (!rt.dumped) continue;\n            if (!this.isTrackAudible(trackId)) continue;\n            rt.lastAccessAt = performance.now();\n            void this.rehydrateTrackAudio(trackId);\n        }\n\n        let stemsActivated = 0;\n        let stemsPaused = 0;\n        for (const [trackId, rt] of this.tracks) {\n            const track = store.getTrackById(trackId);\n            if (!track) continue;\n\n            track.channels.forEach((ch, i) => {\n                const key = `${trackId}:${i}`;\n                const seekPos = this.computeStemSeekSamples(globalBar, ch, track);\n                const w = rt.worklets[i];\n                if (seekPos !== null) {\n                    w.port.postMessage({ type: 'seek', position: seekPos });\n                    w.port.postMessage({ type: 'play' });\n                    this._stemActive.set(key, true);\n                    stemsActivated++;\n                    rt.lastAccessAt = performance.now();\n                } else {\n                    w.port.postMessage({ type: 'pause' });\n                    this._stemActive.set(key, false);\n                    stemsPaused++;\n                }\n            });\n        }\n\n        if (stemsActivated === 0 && this.tracks.size > 0) {\n            const debugInfo: Record<string, unknown>[] = [];\n            for (const [trackId] of this.tracks) {\n                const t = store.getTrackById(trackId);\n                debugInfo.push({\n                    trackId,\n                    found: !!t,\n                    barMappingLen: t?.barMapping?.length ?? -1,\n                    channels: t?.channels.map((ch) => ({\n                        timelineStart: ch.timelineStartBar,\n                        trimStart: ch.trimStartBar,\n                        trimEnd: ch.trimEndBar,\n                        visibleStart: ch.timelineStartBar + ch.trimStartBar,\n                        visibleEnd: ch.timelineStartBar + ch.trimEndBar,\n                    })),\n                });\n            }\n            console.warn('[DAWController.play] No stems activated!', {\n                globalBar,\n                trackCount: this.tracks.size,\n                stemsPaused,\n                audioCtxState: this.ctx!.state,\n                tracks: debugInfo,\n            });\n        }\n\n        this._playStartCtxTime = this.ctx!.currentTime;\n        this._playStartOffset = this.barToSeconds(globalBar);\n        this._playing = true;\n        store.setPlaying(true);\n\n        // In native mode, record the native sample position we're starting from\n        if (this._nativeModeTrackId) {\n            const nTrack = store.getTrackById(this._nativeModeTrackId);\n            if (nTrack) {\n                const ch0 = nTrack.channels[0];\n                const seekPos = ch0 ? this.computeStemSeekSamples(globalBar, ch0, nTrack) : null;\n                this._nativeStartSample = seekPos ?? 0;\n            }\n        }\n\n        // Apply per-bar pitch for the starting bar (one-shot, before worklet starts firing)\n        for (const [trackId, rt] of this.tracks) {\n            rt.currentPitchBar = -1;\n            rt.lastPitchStr = '';\n            this.sendInitialBarPitch(trackId, globalBar);\n        }\n\n        this.startPlayheadAnimation();\n        if (this._metronomeOn) this.startMetronomeScheduler();\n        this.emit('playing');\n    }\n\n    pause(): void {\n        if (this._playing) {\n            if (this._nativeModeTrackId) {\n                this._nativeStartSample = this.getNativePositionSec() * SAMPLE_RATE;\n            }\n            const posSec = this.getCurrentPositionSec();\n            useDAWSessionStore.getState().setPlayheadBar(this.secondsToBar(posSec));\n        }\n\n        for (const [, rt] of this.tracks) {\n            rt.currentPitchBar = -1;\n            rt.lastPitchStr = '';\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'pause' });\n            }\n        }\n\n        this._playing = false;\n        useDAWSessionStore.getState().setPlaying(false);\n        this.stopPlayheadAnimation();\n        this.stopMetronomeScheduler();\n\n        // Reset pitch to static correction when not playing\n        for (const trackId of this.tracks.keys()) {\n            this.syncPitchForTrack(trackId);\n        }\n\n        this.emit('paused');\n    }\n\n    stop(): void {\n        for (const [, rt] of this.tracks) {\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'stop' });\n            }\n        }\n\n        this._playing = false;\n        const store = useDAWSessionStore.getState();\n        store.setPlaying(false);\n        store.setPlayheadBar(0);\n        this.stopPlayheadAnimation();\n        this.stopMetronomeScheduler();\n        this.emit('stopped');\n    }\n\n    async seekToBar(bar: number): Promise<void> {\n        const gen = ++this._seekGen;\n        const wasPlaying = this._playing;\n        const store = useDAWSessionStore.getState();\n\n        store.setPlayheadBar(bar);\n\n        // Same auto-rehydrate sweep as `play()` — without this a dumped\n        // track seeked into mid-playback would stay silent until the next\n        // explicit play() call.\n        if (wasPlaying) {\n            for (const [trackId, rt] of this.tracks) {\n                if (!rt.dumped) continue;\n                if (!this.isTrackAudible(trackId)) continue;\n                rt.lastAccessAt = performance.now();\n                void this.rehydrateTrackAudio(trackId);\n            }\n        }\n\n        for (const [trackId, rt] of this.tracks) {\n            const track = store.getTrackById(trackId);\n            if (!track) continue;\n\n            track.channels.forEach((ch, i) => {\n                const key = `${trackId}:${i}`;\n                const seekPos = this.computeStemSeekSamples(bar, ch, track);\n                const w = rt.worklets[i];\n                if (seekPos !== null) {\n                    w.port.postMessage({ type: 'seek', position: seekPos });\n                    if (wasPlaying) w.port.postMessage({ type: 'play' });\n                    this._stemActive.set(key, wasPlaying);\n                    if (wasPlaying) rt.lastAccessAt = performance.now();\n                } else if (wasPlaying) {\n                    w.port.postMessage({ type: 'pause' });\n                    this._stemActive.set(key, false);\n                }\n            });\n        }\n\n        if (wasPlaying && gen === this._seekGen) {\n            this._playStartCtxTime = this.ctx!.currentTime;\n            this._playStartOffset = this.barToSeconds(bar);\n            const seekStep = this._metronomeDownbeatOnly ? 1 : 0.25;\n            this._metronomeNextBeat = Math.ceil(bar / seekStep) * seekStep;\n        }\n\n        // Track native position for beat lab\n        if (this._nativeModeTrackId) {\n            const nTrack = store.getTrackById(this._nativeModeTrackId);\n            if (nTrack) {\n                const ch0 = nTrack.channels[0];\n                const seekPos = ch0 ? this.computeStemSeekSamples(bar, ch0, nTrack) : null;\n                this._nativeStartSample = seekPos ?? 0;\n            }\n        }\n\n        // Immediately apply the correct per-bar pitch for the seek target\n        for (const [trackId, rt] of this.tracks) {\n            rt.currentPitchBar = -1;\n            rt.lastPitchStr = '';\n            this.sendInitialBarPitch(trackId, bar);\n        }\n    }\n\n    // ── Master tempo / playback rate ────────────────────────────────\n\n    setMasterTempo(bpm: number): void {\n        const wasPlaying = this._playing;\n        const store = useDAWSessionStore.getState();\n        const currentBar = wasPlaying\n            ? this.secondsToBar(this.getCurrentPositionSec())\n            : store.playheadBar;\n\n        store.setMasterTempo(bpm);\n\n        const newStore = useDAWSessionStore.getState();\n\n        // Recompute and send new bar maps + seek to correct positions\n        for (const [trackId, rt] of this.tracks) {\n            const track = newStore.getTrackById(trackId);\n            if (!track) continue;\n\n            const barMap = buildBarMap(\n                track.barMapping,\n                newStore.masterBarLength48000,\n                track.nativeBarLength,\n                newStore.barSubdivisions,\n            );\n\n            rt.barPitchMap = buildBarPitchMap(barMap);\n            rt.currentPitchBar = -1;\n            this.syncWorkletPitchMap(trackId);\n\n            track.channels.forEach((ch, i) => {\n                const w = rt.worklets[i];\n                w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n\n                const seekPos = this.computeStemSeekSamples(currentBar, ch, track);\n                if (seekPos !== null) {\n                    w.port.postMessage({ type: 'seek', position: seekPos });\n                }\n            });\n        }\n\n        if (wasPlaying) {\n            this._playStartOffset = this.barToSeconds(currentBar);\n            this._playStartCtxTime = this.ctx!.currentTime;\n        } else {\n            newStore.setPlayheadBar(currentBar);\n        }\n        this.emit('tempoChanged', bpm);\n    }\n\n    /**\n     * Reorder a track and, if it becomes the new first track, promote it as master.\n     * Handles worklet bar map / pitch resync the same way setMasterTempo does.\n     */\n    reorderTrack(trackId: string, newIndex: number): void {\n        const wasPlaying = this._playing;\n        const store = useDAWSessionStore.getState();\n        const currentBar = wasPlaying\n            ? this.secondsToBar(this.getCurrentPositionSec())\n            : store.playheadBar;\n\n        const oldMasterId = store.tracks[0]?.id;\n        store.reorderTrack(trackId, newIndex);\n        const newStore = useDAWSessionStore.getState();\n        const newMasterId = newStore.tracks[0]?.id;\n\n        if (newMasterId !== oldMasterId) {\n            for (const [tid, rt] of this.tracks) {\n                const track = newStore.getTrackById(tid);\n                if (!track) continue;\n\n                const barMap = buildBarMap(\n                    track.barMapping,\n                    newStore.masterBarLength48000,\n                    track.nativeBarLength,\n                    newStore.barSubdivisions,\n                );\n\n                rt.barPitchMap = buildBarPitchMap(barMap);\n                rt.currentPitchBar = -1;\n                this.syncWorkletPitchMap(tid);\n                this.syncPitchForTrack(tid);\n\n                track.channels.forEach((ch, i) => {\n                    const w = rt.worklets[i];\n                    w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n\n                    const seekPos = this.computeStemSeekSamples(currentBar, ch, track);\n                    if (seekPos !== null) {\n                        w.port.postMessage({ type: 'seek', position: seekPos });\n                    }\n                });\n            }\n\n            if (wasPlaying) {\n                this._playStartOffset = this.barToSeconds(currentBar);\n                this._playStartCtxTime = this.ctx!.currentTime;\n            } else {\n                newStore.setPlayheadBar(currentBar);\n            }\n            this.emit('tempoChanged', newStore.masterTempo);\n        }\n    }\n\n    setGlobalPlaybackRate(rate: number): void {\n        if (this._playing) {\n            const posSec = this.getCurrentPositionSec();\n            useDAWSessionStore.getState().setGlobalPlaybackRate(rate);\n            this._playStartOffset = posSec;\n            this._playStartCtxTime = this.ctx!.currentTime;\n        } else {\n            useDAWSessionStore.getState().setGlobalPlaybackRate(rate);\n        }\n        this.broadcastGlobalRate();\n        this.emit('playbackRateChanged', rate);\n    }\n\n    private broadcastGlobalRate(): void {\n        const rate = useDAWSessionStore.getState().globalPlaybackRate;\n        for (const [, rt] of this.tracks) {\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'set-global-rate', rate });\n            }\n        }\n    }\n\n    // ── Gain / mute / solo sync ──────────────────────────────────────\n\n    syncGainsForTrack(trackId: string): void {\n        const state = useDAWSessionStore.getState();\n        const track = state.getTrackById(trackId);\n        const rt = this.tracks.get(trackId);\n        if (!track || !rt) return;\n\n        const hasSoloedTrack = state.tracks.some((t) => t.soloed);\n        const trackAudible = hasSoloedTrack ? track.soloed : !track.muted;\n        rt.trackGain.gain.setValueAtTime(trackAudible ? track.volume : 0, this.ctx!.currentTime);\n\n        const globalSoloedStems = new Set<number>();\n        for (const t of state.tracks) {\n            for (const ch of t.channels) {\n                if (ch.soloed && ch.visible) globalSoloedStems.add(ch.stemIndex);\n            }\n        }\n        const hasAnySoloedStem = globalSoloedStems.size > 0;\n\n        track.channels.forEach((ch, i) => {\n            const audible = ch.visible && (hasAnySoloedStem ? ch.soloed : !ch.muted);\n            rt.stemGains[i]?.gain.setValueAtTime(audible ? ch.volume : 0, this.ctx!.currentTime);\n        });\n    }\n\n    syncAllGains(): void {\n        for (const trackId of this.tracks.keys()) this.syncGainsForTrack(trackId);\n    }\n\n    /** Send mute regions for a specific stem to its worklet (bar-based → sample-based) */\n    syncMuteRegions(trackId: string, stemIndex: number): void {\n        const rt = this.tracks.get(trackId);\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!rt || !track) return;\n\n        const ch = track.channels.find((c) => c.stemIndex === stemIndex);\n        if (!ch) return;\n\n        const barMapping = track.barMapping;\n        const sampleRegions = ch.muteRegions.map((r) => {\n            const sStart = barToSample(r.startBar, barMapping);\n            const sEnd = barToSample(r.endBar, barMapping);\n            return { sampleStart: sStart, sampleEnd: sEnd };\n        });\n\n        const w = rt.worklets[stemIndex];\n        if (w) w.port.postMessage({ type: 'load-mute-regions', regions: sampleRegions });\n    }\n\n    /** Sync mute regions for all stems of a track */\n    syncAllMuteRegionsForTrack(trackId: string): void {\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track) return;\n        for (const ch of track.channels) {\n            this.syncMuteRegions(trackId, ch.stemIndex);\n        }\n    }\n\n    /**\n     * Rebuild and re-send the bar map for a track to all its stem worklets.\n     * Call after the track's barMapping has been modified (e.g. beat grid nudge).\n     */\n    reloadTrackBarMap(trackId: string): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt) return;\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!track) return;\n\n        // In native mode, keep speed=1.0 for the active track (beat lab)\n        const isNativeTrack = this._nativeModeTrackId === trackId;\n\n        const barMap = isNativeTrack\n            ? track.barMapping.map((bar) => ({ sampleStart: bar.sampleStart, speed: 1.0 }))\n            : buildBarMap(\n                track.barMapping,\n                store.masterBarLength48000,\n                track.nativeBarLength,\n                store.barSubdivisions,\n            );\n\n        if (isNativeTrack) {\n            rt.barPitchMap = barMap.map(() => ({ pitchCorrectionSemitones: 0, speed: 1.0 }));\n        } else {\n            rt.barPitchMap = buildBarPitchMap(barMap);\n        }\n        rt.currentPitchBar = -1;\n\n        const wasPlaying = this._playing;\n        const currentBar = store.playheadBar;\n        track.channels.forEach((ch, i) => {\n            const w = rt.worklets[i];\n            if (!w) return;\n            w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n\n            const seekPos = this.computeStemSeekSamples(currentBar, ch, track);\n            if (seekPos !== null) {\n                w.port.postMessage({ type: 'seek', position: seekPos });\n                if (wasPlaying) w.port.postMessage({ type: 'play' });\n            }\n        });\n\n        if (!isNativeTrack) {\n            this.syncWorkletPitchMap(trackId);\n            this.syncPitchForTrack(trackId);\n        }\n        this.syncAllMuteRegionsForTrack(trackId);\n\n        if (wasPlaying) {\n            this._playStartOffset = this.barToSeconds(currentBar);\n            this._playStartCtxTime = this.ctx!.currentTime;\n        }\n    }\n\n    // ── Native playback mode (Beat Lab) ────────────────────────────────\n\n    /**\n     * Enable or disable native playback for a track. When enabled, the track\n     * plays at its original recorded tempo (speed = 1.0 for every bar) with no\n     * pitch compensation. This allows the Beat Lab to display beat markers at\n     * their true sample positions and have them match the audible audio exactly.\n     */\n    setNativePlaybackMode(trackId: string, enabled: boolean): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt) return;\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!track) return;\n\n        const wasPlaying = this._playing;\n        if (wasPlaying) this.pause();\n\n        if (enabled) {\n            // Save the current quantized bar map so we can restore it later\n            const quantizedBarMap = buildBarMap(\n                track.barMapping,\n                store.masterBarLength48000,\n                track.nativeBarLength,\n                store.barSubdivisions,\n            );\n            this._nativeModeOriginalBarMap = quantizedBarMap;\n            this._nativeModeTrackId = trackId;\n\n            // Build a flat speed=1.0 bar map (native playback)\n            const nativeBarMap: BarMapEntry[] = track.barMapping.map((bar) => ({\n                sampleStart: bar.sampleStart,\n                speed: 1.0,\n            }));\n\n            // Send flat map to all stem worklets\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'load-bar-map', entries: nativeBarMap });\n            }\n\n            // Clear pitch compensation — we want the original pitch too\n            rt.barPitchMap = nativeBarMap.map(() => ({ pitchCorrectionSemitones: 0, speed: 1.0 }));\n            rt.currentPitchBar = -1;\n\n            // CRITICAL: Bypass the pitch (Rubber Band) worklet entirely.\n            // Even with pitch factor = 1.0 it buffers audio and adds ~40-60ms\n            // latency, which makes the audible audio lag the visual playhead.\n            // In native mode we want sample-exact timing for beat analysis.\n            if (rt.pitchWorklet) {\n                try {\n                    for (const g of rt.stemGains) {\n                        g.disconnect();\n                        g.connect(rt.trackGain);\n                    }\n                } catch (err) {\n                    console.warn('[DAWController] Failed to bypass pitch worklet for native mode:', err);\n                }\n            }\n        } else {\n            this._nativeModeTrackId = null;\n            this._nativeModeOriginalBarMap = null;\n\n            // Restore pitch worklet routing\n            if (rt.pitchWorklet) {\n                try {\n                    for (const g of rt.stemGains) {\n                        g.disconnect();\n                        g.connect(rt.pitchWorklet);\n                    }\n                } catch (err) {\n                    console.warn('[DAWController] Failed to restore pitch worklet routing:', err);\n                }\n            }\n\n            // Restore the quantized bar map\n            const barMap = buildBarMap(\n                track.barMapping,\n                store.masterBarLength48000,\n                track.nativeBarLength,\n                store.barSubdivisions,\n            );\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n            }\n\n            rt.barPitchMap = buildBarPitchMap(barMap);\n            rt.currentPitchBar = -1;\n            this.syncWorkletPitchMap(trackId);\n            this.syncPitchForTrack(trackId);\n        }\n\n        // Reset playhead to start\n        store.setPlayheadBar(0);\n    }\n\n    isNativePlaybackMode(): boolean {\n        return this._nativeModeTrackId !== null;\n    }\n\n    get nativeModeTrackId(): string | null {\n        return this._nativeModeTrackId;\n    }\n\n    /**\n     * Get the current playback position in native (unquantized) seconds.\n     * Only meaningful when native playback mode is active.\n     */\n    getNativePositionSec(): number {\n        if (!this.ctx || !this._nativeModeTrackId) return 0;\n        if (!this._playing) return this._nativeStartSample / SAMPLE_RATE;\n        const globalRate = useDAWSessionStore.getState().globalPlaybackRate;\n        const elapsedSec = (this.ctx.currentTime - this._playStartCtxTime) * globalRate;\n        return (this._nativeStartSample / SAMPLE_RATE) + elapsedSec;\n    }\n\n    /**\n     * Seek to a native sample position (used by beat lab scrubbing).\n     * Only works when native playback mode is active.\n     */\n    seekToNativeSample(sample: number): void {\n        if (!this._nativeModeTrackId) return;\n        const rt = this.tracks.get(this._nativeModeTrackId);\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(this._nativeModeTrackId);\n        if (!rt || !track) return;\n\n        this._nativeStartSample = sample;\n\n        for (let i = 0; i < rt.worklets.length; i++) {\n            const w = rt.worklets[i];\n            w.port.postMessage({ type: 'seek', position: Math.floor(sample) });\n            if (this._playing) w.port.postMessage({ type: 'play' });\n        }\n\n        if (this.ctx) {\n            this._playStartCtxTime = this.ctx.currentTime;\n            this._playStartOffset = sample / SAMPLE_RATE;\n        }\n\n        // Update the store playheadBar so play() can resume from the right spot.\n        // Convert native sample → approximate bar position using barMapping.\n        const bm = track.barMapping;\n        if (bm.length > 0) {\n            let barIdx = 0;\n            for (let i = bm.length - 1; i >= 0; i--) {\n                if (bm[i].sampleStart <= sample) { barIdx = i; break; }\n            }\n            const bar = bm[barIdx];\n            const barFrac = bar.duration > 0\n                ? Math.max(0, Math.min(1, (sample - bar.sampleStart) / bar.duration))\n                : 0;\n            const tlStart = track.channels[0]?.timelineStartBar ?? 0;\n            store.setPlayheadBar(barIdx + barFrac + tlStart);\n        }\n    }\n\n    // ── Pitch / transpose ─────────────────────────────────────────────\n\n    /**\n     * Send the combined pitch factor (correction + transpose) to a track's pitch worklet.\n     * When perBarPitchCompensation is enabled, the per-bar correction is added on top\n     * via updatePerBarPitch() during playback — this method sends the base correction.\n     */\n    syncPitchForTrack(trackId: string): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt?.pitchWorklet) return;\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track) return;\n\n        if (track.perBarPitchCompensation && this._playing) {\n            // During playback with per-bar compensation, updatePerBarPitch handles it\n            return;\n        }\n\n        const totalSemitones = track.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset;\n        const factor = semitonesToPitchFactor(totalSemitones);\n        this.sendPitchToWorklet(rt.pitchWorklet, factor);\n    }\n\n    /**\n     * Check if we've crossed a bar boundary and send an updated pitch factor\n     * that includes the per-bar pitch compensation for that specific bar.\n     */\n    /**\n     * rAF-driven per-bar pitch update (fallback for seeks / playback start).\n     * handleWorkletBarChange is the primary path — this catches missed bars.\n     */\n    private updatePerBarPitch(_trackId: string, _globalBar: number): void {\n        // No-op during playback. All pitch updates are driven by the stem\n        // worklet's bar-changed messages via handleWorkletBarChange, which is\n        // sample-accurate. The rAF loop calls this but it intentionally does\n        // nothing — initial pitch is set by sendInitialBarPitch on play/seek.\n    }\n\n    /** One-shot pitch send for play/seek — sends the bar's pitch and sets\n     *  lastPitchStr so the rAF path won't duplicate it. */\n    private sendInitialBarPitch(trackId: string, globalBar: number): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt?.pitchWorklet || !this.ctx) return;\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!track?.perBarPitchCompensation) return;\n\n        const ch0 = track.channels[0];\n        if (!ch0) return;\n\n        const subs = store.barSubdivisions ?? 1;\n        const localBarF = (globalBar - ch0.timelineStartBar) * subs;\n        const localBar = Math.floor(localBarF);\n        if (localBar < 0 || localBar >= rt.barPitchMap.length) return;\n\n        rt.currentPitchBar = localBar;\n        const barPitch = rt.barPitchMap[localBar];\n        const totalSemitones = barPitch.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset;\n        const factor = semitonesToPitchFactor(totalSemitones);\n        const pitchStr = factor.toFixed(2);\n        rt.lastPitchStr = pitchStr;\n        this.sendPitchToWorklet(rt.pitchWorklet, factor);\n    }\n\n    /** Set user transpose in semitones and update the worklet. */\n    setTrackTranspose(trackId: string, semitones: number): void {\n        useDAWSessionStore.getState().setTrackTranspose(trackId, semitones);\n        this.syncWorkletPitchMap(trackId);\n        this.syncPitchForTrack(trackId);\n        this.emit('transposeChanged', trackId, semitones);\n    }\n\n    /** Toggle per-bar pitch compensation for a track. */\n    setPerBarPitchCompensation(trackId: string, enabled: boolean): void {\n        useDAWSessionStore.getState().setPerBarPitchCompensation(trackId, enabled);\n        const rt = this.tracks.get(trackId);\n        if (rt) {\n            rt.currentPitchBar = -1;\n            rt.lastPitchStr = '';\n        }\n        this.syncWorkletPitchMap(trackId);\n        if (!enabled) {\n            this.syncPitchForTrack(trackId);\n        } else if (this._playing) {\n            const bar = useDAWSessionStore.getState().playheadBar;\n            this.sendInitialBarPitch(trackId, bar);\n        }\n        this.emit('perBarPitchChanged', trackId, enabled);\n    }\n\n    /**\n     * Send the per-bar pitch map to the first stem worklet so it can apply a\n     * short crossfade at bar boundaries to mask the ~3ms gap between the\n     * speed change (instant, audio-thread) and the pitch change (next render quantum).\n     * Also sends the pitch map so the worklet can forward pitch commands to\n     * Rubberband via the main thread at the exact bar crossing moment.\n     */\n    private syncWorkletPitchMap(trackId: string): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt || rt.worklets.length === 0) return;\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track) return;\n\n        const stemWorklet = rt.worklets[0];\n\n        if (!track.perBarPitchCompensation || !rt.pitchWorklet) {\n            stemWorklet.port.postMessage({ type: 'load-pitch-map', entries: [] });\n            return;\n        }\n\n        const entries = rt.barPitchMap.map((bp) => ({\n            pitchFactor: semitonesToPitchFactor(\n                bp.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset\n            ),\n        }));\n\n        stemWorklet.port.postMessage({ type: 'load-pitch-map', entries });\n    }\n\n    /** Update rubberband settings for a track. Requires worklet recreation. */\n    setTrackPitchSettings(trackId: string, settings: PitchEngineSettings): void {\n        useDAWSessionStore.getState().setTrackPitchSettings(trackId, settings);\n        const rt = this.tracks.get(trackId);\n        if (!rt?.pitchWorklet || !this.ctx || USE_PHAZE) return;\n\n        // Recreate the worklet with new settings — configure via messages since\n        // the worklet ignores processorOptions.\n        const oldWorklet = rt.pitchWorklet;\n\n        try {\n            const newWorklet = new AudioWorkletNode(this.ctx, PITCH_WORKLET_NAME, {\n                numberOfInputs: 1,\n                numberOfOutputs: 1,\n                channelCount: 2,\n                channelCountMode: 'explicit',\n                outputChannelCount: [2],\n            });\n\n            this.configurePitchWorklet(newWorklet, settings);\n\n            // Reconnect: stemGains -> newWorklet -> trackGain\n            newWorklet.connect(rt.trackGain);\n            for (const sg of rt.stemGains) {\n                sg.disconnect();\n                sg.connect(newWorklet);\n            }\n            oldWorklet.disconnect();\n            rt.pitchWorklet = newWorklet;\n\n            // Re-apply pitch\n            this.syncPitchForTrack(trackId);\n        } catch (err) {\n            console.warn('[DAWController] Failed to recreate pitch worklet:', err);\n        }\n    }\n\n    /**\n     * Apply an autotune pitch offset on top of the current pitch chain.\n     * Called from the autotune panel at ~60fps during playback.\n     * Set autoSemitones=0 to clear the override.\n     */\n    applyAutotuneCorrection(trackId: string, autoSemitones: number): void {\n        const rt = this.tracks.get(trackId);\n        if (!rt?.pitchWorklet) return;\n\n        // Store the autotune offset so other pitch paths (per-bar, bar-change) include it\n        rt.autotuneOffset = autoSemitones;\n\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track) return;\n\n        let baseSemitones = track.transposeSemitones;\n\n        if (track.perBarPitchCompensation && this._playing && rt.currentPitchBar >= 0 && rt.currentPitchBar < rt.barPitchMap.length) {\n            baseSemitones += rt.barPitchMap[rt.currentPitchBar].pitchCorrectionSemitones;\n        } else {\n            baseSemitones += track.pitchCorrectionSemitones;\n        }\n\n        const factor = semitonesToPitchFactor(baseSemitones + autoSemitones);\n        this.sendPitchToWorklet(rt.pitchWorklet, factor);\n    }\n\n    /** Send a pitch factor to the Rubberband pitch worklet. */\n    private sendPitchToWorklet(worklet: AudioWorkletNode, pitchFactor: number): void {\n        worklet.port.postMessage(JSON.stringify(['pitch', pitchFactor.toFixed(2)]));\n    }\n\n    /**\n     * Configure a Rubberband pitch worklet via port messages.\n     * The worklet ignores processorOptions — all config must be sent as messages.\n     *\n     * pitchDelayMs controls how many ms the worklet waits after receiving a pitch\n     * command before applying it to Rubberband. This is the single control for\n     * aligning speed changes (instant) with pitch changes (delayed by FFT pipeline).\n     */\n    private configurePitchWorklet(worklet: AudioWorkletNode, settings: PitchEngineSettings): void {\n        const delaySamples = Math.round((settings.pitchDelayMs / 1000) * SAMPLE_RATE);\n        worklet.port.postMessage(JSON.stringify(['set-pitch-delay', { delaySamples }]));\n        worklet.port.postMessage(JSON.stringify(['pitchTransitionStep', 100.0]));\n        worklet.port.postMessage(JSON.stringify(['pitchOption', settings.pitchOption]));\n        worklet.port.postMessage(JSON.stringify(['formantOption', settings.formantOption]));\n        worklet.port.postMessage(JSON.stringify(['engineOption', settings.engineOption]));\n\n        if (!worklet.port.onmessage) {\n            worklet.port.onmessage = (e) => {\n                try {\n                    const msg = typeof e.data === 'string' ? JSON.parse(e.data) : e.data;\n                    if (!msg?.type) return;\n                    if (msg.type === 'pitch-delayed') {\n                        const d = msg.data ?? msg;\n                        console.log(\n                            `%c[RB Worklet] pitch-delayed target=${d.targetPitch} delay=${d.delaySamples}smp (${((d.delaySamples / SAMPLE_RATE) * 1000).toFixed(1)}ms)`,\n                            'color: #4fc3f7'\n                        );\n                    } else if (msg.type === 'pitch-applied') {\n                        console.log(\n                            `%c[RB Worklet] pitch-applied value=${msg.data?.value ?? '?'} @audioTime=${msg.timestamp?.toFixed(3) ?? '?'}`,\n                            'color: #4ade80; font-weight: bold'\n                        );\n                    } else if (msg.type === 'pitch-delay-updated' || msg.type === 'api-created') {\n                        console.log(`%c[RB Worklet] ${msg.type}`, 'color: #4fc3f7', msg.data ?? msg);\n                    }\n                } catch { /* non-JSON message */ }\n            };\n        }\n\n        console.log(\n            `%c[DAW] Rubberband worklet configured`,\n            'color: #81c784; font-weight: bold',\n            `pitchDelay=${delaySamples} samples (${settings.pitchDelayMs}ms)` +\n            ` | ${settings.pitchOption} | ${settings.formantOption} | ${settings.engineOption}` +\n            ` | step=100 (instant)`\n        );\n    }\n\n    get pitchWorkletAvailable(): boolean { return this._pitchWorkletAvailable; }\n    get audioContext(): AudioContext | null { return this.ctx; }\n\n    /** Get the total semitones currently being applied to a track's pitch worklet. */\n    getCurrentPitchSemitones(trackId: string): number {\n        const rt = this.tracks.get(trackId);\n        if (!rt) return 0;\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track) return 0;\n\n        if (track.perBarPitchCompensation && rt.currentPitchBar >= 0 && rt.currentPitchBar < rt.barPitchMap.length) {\n            const barPitch = rt.barPitchMap[rt.currentPitchBar];\n            return barPitch.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset;\n        }\n        return track.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset;\n    }\n\n    /**\n     * Get the pitch artifact (semitones) that the current bar's speed change introduces.\n     * This is the inverse of the compensation: if speed > 1, pitch goes up, etc.\n     * Returns 0 if per-bar compensation isn't active or no bar is current.\n     */\n    getCurrentSpeedPitchArtifact(trackId: string): number {\n        const rt = this.tracks.get(trackId);\n        if (!rt || rt.currentPitchBar < 0 || rt.currentPitchBar >= rt.barPitchMap.length) return 0;\n        // The artifact is -compensation: speed > 1 → compensation is negative → artifact is positive\n        return -rt.barPitchMap[rt.currentPitchBar].pitchCorrectionSemitones;\n    }\n\n    /** Update pitchDelayMs for a track and push it to the Rubberband worklet immediately. */\n    setTrackPitchDelay(trackId: string, ms: number): void {\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!track) return;\n        const settings = { ...(track.pitchSettings ?? DEFAULT_PITCH_SETTINGS), pitchDelayMs: ms };\n        store.setTrackPitchSettings(trackId, settings);\n\n        const rt = this.tracks.get(trackId);\n        if (rt?.pitchWorklet) {\n            const delaySamples = Math.round((ms / 1000) * SAMPLE_RATE);\n            rt.pitchWorklet.port.postMessage(JSON.stringify(['set-pitch-delay', { delaySamples }]));\n            console.log(`%c[DAW] Pitch delay updated`, 'color: #81c784; font-weight: bold',\n                `${ms}ms → ${delaySamples} samples`);\n        }\n    }\n\n    /** Change bar subdivision and reload all bar maps + pitch maps. */\n    setBarSubdivisions(n: number): void {\n        const store = useDAWSessionStore.getState();\n        store.setBarSubdivisions(n);\n        const subs = store.barSubdivisions; // re-read after clamping\n\n        for (const [trackId, rt] of this.tracks) {\n            const track = store.getTrackById(trackId);\n            if (!track) continue;\n\n            const barMap = buildBarMap(\n                track.barMapping,\n                store.masterBarLength48000,\n                track.nativeBarLength,\n                n,\n            );\n\n            rt.barPitchMap = buildBarPitchMap(barMap);\n            rt.currentPitchBar = -1;\n\n            for (const w of rt.worklets) {\n                w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n            }\n        }\n\n        console.log(\n            `%c[DAW] Bar subdivisions set to ${n}`,\n            'color: #81c784; font-weight: bold',\n            `(${n === 1 ? 'bar-level' : n + ' sub-bars per bar'})`\n        );\n    }\n\n    // ── CPU monitoring ──────────────────────────────────────────────\n\n    /** Start polling AudioContext render capacity (if supported). */\n    startCpuMonitor(intervalMs = 500): void {\n        this.stopCpuMonitor();\n        const ac = this.ctx as any;\n        if (!ac?.renderCapacity) {\n            // renderCapacity is behind a flag in some browsers — fall back to a rough estimate\n            this._cpuPollId = setInterval(() => {\n                // Rough heuristic: count active worklets\n                const trackCount = this.tracks.size;\n                const workletCount = trackCount * 4 + (this._pitchWorkletAvailable ? trackCount : 0);\n                this._cpuUsage = Math.min(1, workletCount * 0.03);\n                this.emit('cpuUsage', this._cpuUsage);\n            }, intervalMs);\n            return;\n        }\n\n        // Use AudioContext.renderCapacity if available (Chrome 130+)\n        ac.renderCapacity.addEventListener('update', (e: any) => {\n            this._cpuUsage = e.load ?? e.averageLoad ?? 0;\n            this.emit('cpuUsage', this._cpuUsage);\n        });\n        ac.renderCapacity.start({ updateInterval: intervalMs / 1000 });\n    }\n\n    stopCpuMonitor(): void {\n        if (this._cpuPollId) {\n            clearInterval(this._cpuPollId);\n            this._cpuPollId = null;\n        }\n        const ac = this.ctx as any;\n        if (ac?.renderCapacity) {\n            try { ac.renderCapacity.stop(); } catch { /* */ }\n        }\n    }\n\n    get cpuUsage(): number { return this._cpuUsage; }\n\n    // ── Playhead animation ───────────────────────────────────────────\n\n    private startPlayheadAnimation(): void {\n        this.stopPlayheadAnimation();\n        const animate = () => {\n            if (!this._playing) return;\n            const posSec = this.getCurrentPositionSec();\n            const store = useDAWSessionStore.getState();\n            const currentBar = this.secondsToBar(posSec);\n            store.setPlayheadBar(currentBar);\n\n            this.updateStemPlayState(currentBar);\n            this.updateAllPerBarPitch(currentBar);\n\n            if (store.loopRegion && currentBar >= store.loopRegion.endBar) {\n                this.seekToBar(store.loopRegion.startBar);\n            }\n            this.playheadAnimationId = requestAnimationFrame(animate);\n        };\n        this.playheadAnimationId = requestAnimationFrame(animate);\n    }\n\n    /**\n     * Authoritative pitch update from the worklet's bar-changed message.\n     * Fires at audio-thread timing. The worklet's internal pitchDelayMs\n     * handles the alignment between speed (instant) and pitch (delayed).\n     */\n    private handleWorkletBarChange(trackId: string, localBarIndex: number): void {\n        if (!this._playing) return;\n        const rt = this.tracks.get(trackId);\n        if (!rt?.pitchWorklet) return;\n        const track = useDAWSessionStore.getState().getTrackById(trackId);\n        if (!track?.perBarPitchCompensation) return;\n        if (localBarIndex < 0 || localBarIndex >= rt.barPitchMap.length) return;\n        if (localBarIndex === rt.currentPitchBar) return;\n\n        rt.currentPitchBar = localBarIndex;\n        const barPitch = rt.barPitchMap[localBarIndex];\n        const totalSemitones = barPitch.pitchCorrectionSemitones + track.transposeSemitones + rt.autotuneOffset;\n        const factor = semitonesToPitchFactor(totalSemitones);\n        const pitchStr = factor.toFixed(2);\n\n        // Skip if pitch value hasn't changed (e.g. subdivided sub-bars with same speed)\n        if (pitchStr === rt.lastPitchStr) return;\n        rt.lastPitchStr = pitchStr;\n\n        this.sendPitchToWorklet(rt.pitchWorklet, factor);\n        const subs = useDAWSessionStore.getState().barSubdivisions ?? 1;\n        const parentBar = subs > 1 ? Math.floor(localBarIndex / subs) : localBarIndex;\n        const subIdx = subs > 1 ? localBarIndex % subs : 0;\n        const prevSt = rt.lastCorrectionSt ?? barPitch.pitchCorrectionSemitones;\n        const deltaSt = barPitch.pitchCorrectionSemitones - prevSt;\n        rt.lastCorrectionSt = barPitch.pitchCorrectionSemitones;\n        const deltaColor = Math.abs(deltaSt) > 0.5 ? 'color: #ef4444; font-weight: bold' : 'color: #81c784';\n        console.log(\n            `%c[Bar Pitch] parentBar=${parentBar} sub=${subIdx}/${subs}` +\n            ` | idx=${localBarIndex} pitch=${pitchStr} speed=${barPitch.speed.toFixed(4)}` +\n            ` | correction=${barPitch.pitchCorrectionSemitones.toFixed(2)}st` +\n            ` | Δ=${deltaSt >= 0 ? '+' : ''}${deltaSt.toFixed(2)}st`,\n            deltaColor\n        );\n    }\n\n    /** Update per-bar pitch compensation for all tracks that have it enabled. */\n    private updateAllPerBarPitch(globalBar: number): void {\n        for (const trackId of this.tracks.keys()) {\n            this.updatePerBarPitch(trackId, globalBar);\n        }\n    }\n\n    /** Start/stop individual stem worklets as the playhead crosses their timeline boundaries */\n    private updateStemPlayState(globalBar: number): void {\n        const store = useDAWSessionStore.getState();\n        for (const [trackId, rt] of this.tracks) {\n            const track = store.getTrackById(trackId);\n            if (!track) continue;\n\n            let anyShouldBeActive = false;\n            track.channels.forEach((ch, i) => {\n                const key = `${trackId}:${i}`;\n                const seekPos = this.computeStemSeekSamples(globalBar, ch, track);\n                const wasActive = this._stemActive.get(key) ?? false;\n                const shouldBeActive = seekPos !== null;\n                if (shouldBeActive) anyShouldBeActive = true;\n\n                if (shouldBeActive && !wasActive) {\n                    const w = rt.worklets[i];\n                    w.port.postMessage({ type: 'seek', position: seekPos });\n                    w.port.postMessage({ type: 'play' });\n                    this._stemActive.set(key, true);\n                } else if (!shouldBeActive && wasActive) {\n                    const w = rt.worklets[i];\n                    w.port.postMessage({ type: 'pause' });\n                    this._stemActive.set(key, false);\n                }\n            });\n\n            if (anyShouldBeActive) {\n                rt.lastAccessAt = performance.now();\n                if (rt.dumped && !rt.transitioning) {\n                    // Playhead just entered this dumped track's range —\n                    // bring its audio back from IndexedDB. Fire-and-forget;\n                    // worklet will produce silence until patches arrive.\n                    void this.rehydrateTrackAudio(trackId);\n                }\n            }\n        }\n    }\n\n    private stopPlayheadAnimation(): void {\n        if (this.playheadAnimationId !== null) {\n            cancelAnimationFrame(this.playheadAnimationId);\n            this.playheadAnimationId = null;\n        }\n    }\n\n    private getCurrentPositionSec(): number {\n        if (!this.ctx) return 0;\n        const globalRate = useDAWSessionStore.getState().globalPlaybackRate;\n        return this._playStartOffset + (this.ctx.currentTime - this._playStartCtxTime) * globalRate;\n    }\n\n    // ── Time / bar conversion ────────────────────────────────────────\n\n    barToSeconds(bar: number): number {\n        const masterBar = useDAWSessionStore.getState().masterBarLength48000;\n        if (!masterBar) return 0;\n        return (bar * masterBar) / SAMPLE_RATE;\n    }\n\n    secondsToBar(sec: number): number {\n        const masterBar = useDAWSessionStore.getState().masterBarLength48000;\n        if (!masterBar) return 0;\n        return (sec * SAMPLE_RATE) / masterBar;\n    }\n\n    // ── Metronome ─────────────────────────────────────────────────────\n\n    get metronomeEnabled(): boolean { return this._metronomeOn; }\n    get metronomeVolume(): number { return this._metronomeGain?.gain.value ?? 0.35; }\n    get metronomeDownbeatOnly(): boolean { return this._metronomeDownbeatOnly; }\n\n    setMetronomeDownbeatOnly(downbeatOnly: boolean): void {\n        this._metronomeDownbeatOnly = downbeatOnly;\n        if (this._metronomeOn && this._playing) {\n            this.startMetronomeScheduler();\n        }\n    }\n\n    toggleMetronome(on?: boolean): void {\n        this._metronomeOn = on ?? !this._metronomeOn;\n        if (this._metronomeOn && !this._metronomeGain && this.ctx) {\n            this._metronomeGain = this.ctx.createGain();\n            this._metronomeGain.gain.value = 0.35;\n            this._metronomeGain.connect(this.ctx.destination);\n        }\n        if (this._metronomeOn && this._playing) {\n            this.startMetronomeScheduler();\n        } else {\n            this.stopMetronomeScheduler();\n        }\n        this.emit('metronomeChanged', this._metronomeOn);\n    }\n\n    setMetronomeVolume(volume: number): void {\n        if (this._metronomeGain) {\n            this._metronomeGain.gain.value = Math.max(0, Math.min(1, volume));\n        }\n        this.emit('metronomeChanged', this._metronomeOn);\n    }\n\n    private startMetronomeScheduler(): void {\n        this.stopMetronomeScheduler();\n        if (!this.ctx || !this._metronomeGain) return;\n\n        const store = useDAWSessionStore.getState();\n        const currentBar = store.playheadBar;\n        const step = this._metronomeDownbeatOnly ? 1 : 0.25;\n        this._metronomeNextBeat = Math.ceil(currentBar / step) * step;\n\n        this._metronomeScheduleId = setInterval(() => this.scheduleMetronomeClicks(), METRO_SCHEDULE_INTERVAL_MS);\n    }\n\n    private stopMetronomeScheduler(): void {\n        if (this._metronomeScheduleId !== null) {\n            clearInterval(this._metronomeScheduleId);\n            this._metronomeScheduleId = null;\n        }\n    }\n\n    private scheduleMetronomeClicks(): void {\n        if (!this.ctx || !this._metronomeGain || !this._playing) return;\n\n        const store = useDAWSessionStore.getState();\n        const masterBar = store.masterBarLength48000;\n        if (!masterBar) return;\n\n        const globalRate = store.globalPlaybackRate;\n        const lookAheadBar = METRO_LOOK_AHEAD_SEC / (masterBar / SAMPLE_RATE);\n        const currentBar = this.secondsToBar(this.getCurrentPositionSec());\n        const horizonBar = currentBar + lookAheadBar;\n\n        const step = this._metronomeDownbeatOnly ? 1 : 0.25;\n        while (this._metronomeNextBeat < horizonBar) {\n            const beatBar = this._metronomeNextBeat;\n            if (beatBar >= currentBar) {\n                const deltaSec = (beatBar - currentBar) * masterBar / SAMPLE_RATE / globalRate;\n                const when = this.ctx.currentTime + deltaSec;\n                this.playClick(when, Math.abs(beatBar % 1) < 0.01);\n            }\n            this._metronomeNextBeat += step;\n        }\n    }\n\n    private playClick(when: number, downbeat: boolean): void {\n        if (!this.ctx || !this._metronomeGain) return;\n        const osc = this.ctx.createOscillator();\n        const env = this.ctx.createGain();\n        osc.type = 'sine';\n        osc.frequency.value = downbeat ? 1000 : 800;\n        env.gain.setValueAtTime(downbeat ? 1.0 : 0.5, when);\n        env.gain.exponentialRampToValueAtTime(0.001, when + 0.04);\n        osc.connect(env);\n        env.connect(this._metronomeGain);\n        osc.start(when);\n        osc.stop(when + 0.05);\n    }\n\n    // ── Copy/paste audio between bar ranges ─────────────────────────\n\n    /**\n     * Copy audio samples from one bar range and overwrite another bar range\n     * inside a track's stem buffers. Source and destination stems may be the\n     * same (default) or different stems on the same track. After writing, the\n     * modified destination buffer is sent to the worklet so playback reflects\n     * the change immediately.\n     *\n     * Both `sourceStart/End` and `targetStartBar` are in local bars (relative\n     * to the track's barMapping, not the global timeline).\n     *\n     * `targetStemIndex` defaults to `stemIndex` (source) for backwards-compatible\n     * same-stem behaviour.\n     */\n    pasteAudioRegion(\n        trackId: string,\n        stemIndex: number,\n        sourceStartBar: number,\n        sourceEndBar: number,\n        targetStartBar: number,\n        targetStemIndex: number = stemIndex,\n    ): void {\n        const rt = this.tracks.get(trackId);\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!rt || !track || !this.ctx) {\n            console.warn('[DAW pasteAudio] Missing runtime/track/ctx', { hasRt: !!rt, hasTrack: !!track, hasCtx: !!this.ctx });\n            return;\n        }\n\n        const stemKeys: (keyof StemAudioBuffers)[] = ['other', 'vocals', 'bass', 'drums'];\n        const srcKey = stemKeys[stemIndex];\n        const dstKey = stemKeys[targetStemIndex];\n        const srcBuf = rt.stems[srcKey];\n        const dstBuf = rt.stems[dstKey];\n        if (!srcBuf || !dstBuf) {\n            console.warn('[DAW pasteAudio] No buffer for stem', { srcKey, dstKey });\n            return;\n        }\n\n        const barMapping = track.barMapping;\n        const srcSampleStart = barToSample(sourceStartBar, barMapping);\n        const srcSampleEnd = barToSample(sourceEndBar, barMapping);\n        const srcLen = srcSampleEnd - srcSampleStart;\n\n        const dstSampleStart = barToSample(targetStartBar, barMapping);\n        const clipLen = sourceEndBar - sourceStartBar;\n        const targetEndBar = Math.min(targetStartBar + clipLen, barMapping.length);\n        const dstSampleEnd = barToSample(targetEndBar, barMapping);\n        const dstLen = dstSampleEnd - dstSampleStart;\n\n        if (srcLen <= 0) { console.warn('[DAW pasteAudio] srcLen <= 0, aborting'); return; }\n        if (dstLen <= 0) { console.warn('[DAW pasteAudio] dstLen <= 0, aborting'); return; }\n\n        const copyLen = Math.min(srcLen, dstLen);\n        const channelCount = Math.min(srcBuf.numberOfChannels, dstBuf.numberOfChannels);\n\n        for (let ch = 0; ch < channelCount; ch++) {\n            const srcData = srcBuf.getChannelData(ch);\n            const dstData = dstBuf.getChannelData(ch);\n            // Snapshot the source samples first so cross-stem (or overlapping\n            // same-stem) writes never clobber data they still need to read.\n            const srcCopy = srcData.slice(srcSampleStart, srcSampleStart + copyLen);\n            dstData.set(srcCopy, dstSampleStart);\n\n            if (dstLen > copyLen) {\n                for (let s = dstSampleStart + copyLen; s < dstSampleEnd; s++) {\n                    dstData[s] = 0;\n                }\n            }\n        }\n\n        // Re-send the modified DESTINATION buffer to its worklet\n        const w = rt.worklets[targetStemIndex];\n        const channels: Float32Array[] = [];\n        for (let ch = 0; ch < dstBuf.numberOfChannels; ch++) {\n            channels.push(new Float32Array(dstBuf.getChannelData(ch)));\n        }\n        w.port.postMessage({ type: 'load-buffer', channels }, channels.map(c => c.buffer));\n\n        const freshStore = useDAWSessionStore.getState();\n        const freshTrack = freshStore.getTrackById(trackId);\n        if (freshTrack) {\n            const barMap = buildBarMap(freshTrack.barMapping, freshStore.masterBarLength48000, freshTrack.nativeBarLength, freshStore.barSubdivisions);\n            w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n            rt.barPitchMap = buildBarPitchMap(barMap);\n            rt.currentPitchBar = -1;\n        }\n\n        this.syncMuteRegions(trackId, targetStemIndex);\n    }\n\n    // ── Selection-based batch operations ────────────────────────────\n    //\n    // These wrap pasteAudioRegion / mute regions / clipboard with the right\n    // undo/redo semantics so the keyboard handlers and the SelectionActionBar\n    // popover share one implementation.\n\n    /**\n     * Copy each selection's bar range into the clipboard. No mutation of\n     * the audio buffers — just snapshots the source bar ranges. No undo step\n     * (copy is non-destructive).\n     */\n    copySelections(selections: import('../interfaces').StemSelectionRange[]): void {\n        const store = useDAWSessionStore.getState();\n        if (selections.length === 0) return;\n        const entries: import('../interfaces').ClipboardEntry[] = selections\n            .filter((s) => s.endBar > s.startBar)\n            .map((s) => ({\n                trackId: s.trackId,\n                stemIndex: s.stemIndex,\n                sourceStartBar: Math.round(s.startBar),\n                sourceEndBar: Math.round(s.endBar),\n            }));\n        store.setClipboardEntries(entries);\n    }\n\n    /**\n     * Mute (silence) each selection's bar range. Single undo step covers all\n     * ranges. After muting, clears the selection set.\n     */\n    muteSelections(selections: import('../interfaces').StemSelectionRange[]): void {\n        if (selections.length === 0) return;\n        const store = useDAWSessionStore.getState();\n        store.pushUndo();\n        const touched = new Set<string>();\n        for (const sel of selections) {\n            store.addMuteRegion(sel.trackId, sel.stemIndex, {\n                startBar: Math.round(sel.startBar),\n                endBar: Math.round(sel.endBar),\n            });\n            touched.add(`${sel.trackId}:${sel.stemIndex}`);\n        }\n        for (const key of touched) {\n            const [trackId, idx] = key.split(':');\n            this.syncMuteRegions(trackId, parseInt(idx, 10));\n        }\n        store.clearSelections();\n    }\n\n    /**\n     * Cut: copy each selection's bar range to clipboard AND mute it. Single\n     * undo step. Clears the selection set after.\n     */\n    cutSelections(selections: import('../interfaces').StemSelectionRange[]): void {\n        if (selections.length === 0) return;\n        this.copySelections(selections);\n        this.muteSelections(selections);\n    }\n\n    /**\n     * Paste the current clipboard at a global timeline bar. Each clipboard\n     * entry pastes into its own stemIndex on the focused track. Cross-track\n     * paste is supported as same-stemIndex-different-track when the source\n     * track no longer exists.\n     *\n     * For a SINGLE-entry clipboard, the optional `targetStemIndex` allows\n     * cross-stem paste (e.g. paste bass clip into drums lane).\n     */\n    pasteClipboardAtBar(\n        targetGlobalBar: number,\n        targetTrackId: string,\n        targetStemIndex?: number,\n    ): void {\n        const store = useDAWSessionStore.getState();\n        const entries = store.clipboardEntries;\n        if (entries.length === 0) return;\n\n        const track = store.getTrackById(targetTrackId);\n        if (!track) return;\n\n        store.pushUndo();\n\n        const isSingleEntryCrossStem =\n            entries.length === 1 && targetStemIndex !== undefined && targetStemIndex !== entries[0].stemIndex;\n\n        for (const cb of entries) {\n            const dstStemIndex = isSingleEntryCrossStem ? (targetStemIndex as number) : cb.stemIndex;\n            const ch = store.getChannel(targetTrackId, dstStemIndex);\n            if (!ch) continue;\n\n            const clipLen = cb.sourceEndBar - cb.sourceStartBar;\n            const maxBar = track.barMapping.length;\n            let localPasteBar = Math.round(targetGlobalBar) - ch.timelineStartBar;\n\n            // Pasting before the buffer start: shift the stem earlier so it\n            // falls inside the buffer (Logic-style \"make room\").\n            if (localPasteBar < 0) {\n                const shift = -localPasteBar;\n                const newTimelineStart = Math.max(0, ch.timelineStartBar - shift);\n                const actualShift = ch.timelineStartBar - newTimelineStart;\n                store.setStemTimelineStart(targetTrackId, dstStemIndex, newTimelineStart);\n                store.setStemTrim(\n                    targetTrackId,\n                    dstStemIndex,\n                    ch.trimStartBar + actualShift,\n                    ch.trimEndBar + actualShift,\n                );\n                localPasteBar = Math.round(targetGlobalBar) - newTimelineStart;\n            }\n\n            const pasteEnd = Math.min(localPasteBar + clipLen, maxBar);\n            if (pasteEnd <= localPasteBar || localPasteBar < 0 || localPasteBar >= maxBar) {\n                continue;\n            }\n\n            // Copy buffer audio. If the clipboard entry is from a different\n            // track we can't cross-track paste yet (different barMappings /\n            // sample lengths), so fall through to same-track only.\n            const srcTrackId = cb.trackId === targetTrackId ? targetTrackId : targetTrackId;\n            this.pasteAudioRegion(\n                srcTrackId,\n                cb.stemIndex,\n                cb.sourceStartBar,\n                cb.sourceEndBar,\n                localPasteBar,\n                dstStemIndex,\n            );\n\n            const freshCh = store.getChannel(targetTrackId, dstStemIndex);\n            if (!freshCh) continue;\n\n            const newTrimStart = Math.min(freshCh.trimStartBar, localPasteBar);\n            const newTrimEnd = Math.max(freshCh.trimEndBar, pasteEnd);\n            if (newTrimStart !== freshCh.trimStartBar || newTrimEnd !== freshCh.trimEndBar) {\n                store.setStemTrim(targetTrackId, dstStemIndex, newTrimStart, newTrimEnd);\n            }\n\n            // Clear mute regions covering the paste target so the audio plays.\n            const updatedCh = store.getChannel(targetTrackId, dstStemIndex);\n            if (updatedCh) {\n                const newMuteRegions = updatedCh.muteRegions.flatMap((r) => {\n                    if (r.endBar <= localPasteBar || r.startBar >= pasteEnd) return [r];\n                    const parts: { startBar: number; endBar: number }[] = [];\n                    if (r.startBar < localPasteBar) parts.push({ startBar: r.startBar, endBar: localPasteBar });\n                    if (r.endBar > pasteEnd) parts.push({ startBar: pasteEnd, endBar: r.endBar });\n                    return parts;\n                });\n                store.setMuteRegionsBatch(targetTrackId, dstStemIndex, newMuteRegions);\n                this.syncMuteRegions(targetTrackId, dstStemIndex);\n            }\n        }\n\n        // Advance playhead to immediately after the last pasted bar.\n        const longestClipLen = entries.reduce((m, e) => Math.max(m, e.sourceEndBar - e.sourceStartBar), 0);\n        store.setPlayheadBar(Math.round(targetGlobalBar) + longestClipLen);\n        store.clearSelections();\n    }\n\n    /**\n     * Duplicate each selection: copy its audio in-buffer to immediately after\n     * the selection end on the same stem. Single undo step.\n     */\n    duplicateSelections(selections: import('../interfaces').StemSelectionRange[]): void {\n        if (selections.length === 0) return;\n        const store = useDAWSessionStore.getState();\n        store.pushUndo();\n\n        const newSelections: import('../interfaces').StemSelectionRange[] = [];\n        for (const sel of selections) {\n            const track = store.getTrackById(sel.trackId);\n            const ch = store.getChannel(sel.trackId, sel.stemIndex);\n            if (!track || !ch) continue;\n\n            const start = Math.round(sel.startBar);\n            const end = Math.round(sel.endBar);\n            const len = end - start;\n            if (len <= 0) continue;\n\n            const maxBar = track.barMapping.length;\n            const dstStart = end;\n            const dstEnd = Math.min(dstStart + len, maxBar);\n            if (dstEnd <= dstStart) continue;\n\n            this.pasteAudioRegion(sel.trackId, sel.stemIndex, start, end, dstStart, sel.stemIndex);\n\n            // Expand trim if the duplicate spills past the current visible end.\n            const freshCh = store.getChannel(sel.trackId, sel.stemIndex);\n            if (freshCh && dstEnd > freshCh.trimEndBar) {\n                store.setStemTrim(sel.trackId, sel.stemIndex, freshCh.trimStartBar, dstEnd);\n            }\n\n            // Carve out any mute region covering the duplicate destination so\n            // the new copy actually plays.\n            const updatedCh = store.getChannel(sel.trackId, sel.stemIndex);\n            if (updatedCh) {\n                const newMuteRegions = updatedCh.muteRegions.flatMap((r) => {\n                    if (r.endBar <= dstStart || r.startBar >= dstEnd) return [r];\n                    const parts: { startBar: number; endBar: number }[] = [];\n                    if (r.startBar < dstStart) parts.push({ startBar: r.startBar, endBar: dstStart });\n                    if (r.endBar > dstEnd) parts.push({ startBar: dstEnd, endBar: r.endBar });\n                    return parts;\n                });\n                store.setMuteRegionsBatch(sel.trackId, sel.stemIndex, newMuteRegions);\n                this.syncMuteRegions(sel.trackId, sel.stemIndex);\n            }\n\n            newSelections.push({\n                trackId: sel.trackId,\n                stemIndex: sel.stemIndex,\n                startBar: dstStart,\n                endBar: dstEnd,\n            });\n        }\n\n        if (newSelections.length > 0) {\n            store.setSelections(newSelections);\n        }\n    }\n\n    // ── Resampled region insertion ──────────────────────────────────\n\n    /**\n     * Insert a resampled AudioBuffer into a stem, replacing the specified bar range.\n     * Rebuilds the worklet buffer and bar map, and records the resampled region in the store.\n     */\n    insertResampledRegion(\n        trackId: string,\n        stemIndex: number,\n        startBar: number,\n        endBar: number,\n        resampledBuffer: AudioBuffer,\n    ): void {\n        const rt = this.tracks.get(trackId);\n        const store = useDAWSessionStore.getState();\n        const track = store.getTrackById(trackId);\n        if (!rt || !track || !this.ctx) return;\n\n        const stemKeys: (keyof import('./daw-track-loader').StemAudioBuffers)[] = ['other', 'vocals', 'bass', 'drums'];\n        const stemKey = stemKeys[stemIndex];\n        const originalBuf = rt.stems[stemKey];\n        if (!originalBuf) return;\n\n        const barMapping = track.barMapping;\n        const insertSample = barMapping[startBar]?.sampleStart ?? 0;\n        const endSample = endBar < barMapping.length\n            ? barMapping[endBar].sampleStart\n            : (barMapping[barMapping.length - 1]?.sampleStart ?? 0) + (barMapping[barMapping.length - 1]?.duration ?? 0);\n        const replaceLen = endSample - insertSample;\n\n        // Build a new buffer with the resampled content spliced in\n        const newLen = originalBuf.length - replaceLen + resampledBuffer.length;\n        const newBuf = this.ctx.createBuffer(originalBuf.numberOfChannels, Math.max(newLen, 1), originalBuf.sampleRate);\n        for (let ch = 0; ch < originalBuf.numberOfChannels; ch++) {\n            const src = originalBuf.getChannelData(ch);\n            const dst = newBuf.getChannelData(ch);\n            const rep = resampledBuffer.numberOfChannels > ch\n                ? resampledBuffer.getChannelData(ch)\n                : resampledBuffer.getChannelData(0);\n            dst.set(src.subarray(0, insertSample));\n            dst.set(rep, insertSample);\n            if (endSample < src.length) {\n                dst.set(src.subarray(endSample), insertSample + resampledBuffer.length);\n            }\n        }\n\n        // Update the runtime stems\n        (rt.stems as unknown as Record<string, AudioBuffer>)[stemKey] = newBuf;\n\n        // Send the new buffer to the worklet\n        const w = rt.worklets[stemIndex];\n        const channels: Float32Array[] = [];\n        for (let ch = 0; ch < newBuf.numberOfChannels; ch++) {\n            channels.push(new Float32Array(newBuf.getChannelData(ch)));\n        }\n        w.port.postMessage({ type: 'load-buffer', channels }, channels.map((c) => c.buffer));\n\n        // Rebuild and send bar map\n        const freshStore = useDAWSessionStore.getState();\n        const freshTrack = freshStore.getTrackById(trackId);\n        if (freshTrack) {\n            const barMap = buildBarMap(freshTrack.barMapping, freshStore.masterBarLength48000, freshTrack.nativeBarLength, freshStore.barSubdivisions);\n            w.port.postMessage({ type: 'load-bar-map', entries: barMap });\n            rt.barPitchMap = buildBarPitchMap(barMap);\n            rt.currentPitchBar = -1;\n        }\n\n        // Re-sync mute regions\n        this.syncMuteRegions(trackId, stemIndex);\n\n        // Record the resampled region in the store for visual indication\n        store.addResampledRegion(trackId, stemIndex, {\n            startBar,\n            endBar,\n            sliceCount: 0,\n        });\n\n        this.emit('resampledRegionInserted', trackId, stemIndex);\n    }\n\n    // ── Cleanup ──────────────────────────────────────────────────────\n\n    destroy(): void {\n        this.stopPlayheadAnimation();\n        this.stopMetronomeScheduler();\n        this.stopCpuMonitor();\n        this._metronomeGain?.disconnect();\n        this._metronomeGain = null;\n        for (const [trackId, rt] of this.tracks) {\n            try { rt.unsubscribePatch?.(); } catch { /* */ }\n            rt.worklets.forEach((w) => {\n                w.port.postMessage({ type: 'stop' });\n                w.disconnect();\n            });\n            rt.stemGains.forEach((g) => g.disconnect());\n            try { rt.pitchWorklet?.disconnect(); } catch { /* */ }\n            rt.trackGain.disconnect();\n            clearWaveformCacheForTrack(trackId);\n        }\n        // Abort every shared decode in one go now that no track refers\n        // to them. Refcounts are no longer relevant — the controller is\n        // being torn down.\n        for (const entry of this.sourceCache.values()) {\n            try { entry.loadHandle?.abort(); } catch { /* */ }\n        }\n        this.sourceCache.clear();\n        this.tracks.clear();\n        this.masterGain?.disconnect();\n        this.ctx?.close();\n        this.ctx = null;\n        this.masterGain = null;\n        this.isInitialised = false;\n        this._pitchWorkletAvailable = false;\n        this.removeAllListeners();\n        DAWController.instance = null;\n    }\n}\n"]}