@waveform-playlist/recording 11.3.1 → 12.0.0

package/dist/index.d.mts

@@ -211,36 +211,4 @@ interface UseIntegratedRecordingReturn {
 }
 declare function useIntegratedRecording(tracks: ClipTrack[], setTracks: (tracks: ClipTrack[]) => void, selectedTrackId: string | null, options?: IntegratedRecordingOptions): UseIntegratedRecordingReturn;
 
-/**
- * Peak generation for real-time waveform visualization during recording
- * Matches the format used by webaudio-peaks: min/max pairs with bit depth
- */
-/**
- * Generate peaks from audio samples in standard min/max pair format
- *
- * @param samples - Audio samples to process
- * @param samplesPerPixel - Number of samples to represent in each peak
- * @param bits - Bit depth for peak values (8 or 16)
- * @returns Int8Array or Int16Array of peak values (min/max pairs)
- */
-declare function generatePeaks(samples: Float32Array, samplesPerPixel: number, bits?: 8 | 16): Int8Array | Int16Array;
-/**
- * Append new peaks to existing peaks array
- * This is used for incremental peak updates during recording
- */
-declare function appendPeaks(existingPeaks: Int8Array | Int16Array, newSamples: Float32Array, samplesPerPixel: number, totalSamplesProcessed: number, bits?: 8 | 16): Int8Array | Int16Array;
-
-/**
- * Utility functions for working with AudioBuffers during recording
- */
-/**
- * Concatenate multiple Float32Arrays into a single array
- */
-declare function concatenateAudioData(chunks: Float32Array[]): Float32Array;
-/**
- * Convert channel data to AudioBuffer.
- * Accepts either per-channel Float32Array[] or a single Float32Array (mono, backwards compatible).
- */
-declare function createAudioBuffer(audioContext: AudioContext, channelData: Float32Array[] | Float32Array, sampleRate: number, channelCount?: number): AudioBuffer;
-
-export { type IntegratedRecordingOptions, type MicrophoneDevice, type RecordingData, type RecordingOptions, type RecordingState, type UseIntegratedRecordingReturn, type UseMicrophoneAccessReturn, type UseMicrophoneLevelOptions, type UseMicrophoneLevelReturn, type UseRecordingReturn, appendPeaks, concatenateAudioData, createAudioBuffer, generatePeaks, useIntegratedRecording, useMicrophoneAccess, useMicrophoneLevel, useRecording };
+export { type IntegratedRecordingOptions, type MicrophoneDevice, type RecordingData, type RecordingOptions, type RecordingState, type UseIntegratedRecordingReturn, type UseMicrophoneAccessReturn, type UseMicrophoneLevelOptions, type UseMicrophoneLevelReturn, type UseRecordingReturn, useIntegratedRecording, useMicrophoneAccess, useMicrophoneLevel, useRecording };

package/dist/index.d.ts

@@ -211,36 +211,4 @@ interface UseIntegratedRecordingReturn {
 }
 declare function useIntegratedRecording(tracks: ClipTrack[], setTracks: (tracks: ClipTrack[]) => void, selectedTrackId: string | null, options?: IntegratedRecordingOptions): UseIntegratedRecordingReturn;
 
-/**
- * Peak generation for real-time waveform visualization during recording
- * Matches the format used by webaudio-peaks: min/max pairs with bit depth
- */
-/**
- * Generate peaks from audio samples in standard min/max pair format
- *
- * @param samples - Audio samples to process
- * @param samplesPerPixel - Number of samples to represent in each peak
- * @param bits - Bit depth for peak values (8 or 16)
- * @returns Int8Array or Int16Array of peak values (min/max pairs)
- */
-declare function generatePeaks(samples: Float32Array, samplesPerPixel: number, bits?: 8 | 16): Int8Array | Int16Array;
-/**
- * Append new peaks to existing peaks array
- * This is used for incremental peak updates during recording
- */
-declare function appendPeaks(existingPeaks: Int8Array | Int16Array, newSamples: Float32Array, samplesPerPixel: number, totalSamplesProcessed: number, bits?: 8 | 16): Int8Array | Int16Array;
-
-/**
- * Utility functions for working with AudioBuffers during recording
- */
-/**
- * Concatenate multiple Float32Arrays into a single array
- */
-declare function concatenateAudioData(chunks: Float32Array[]): Float32Array;
-/**
- * Convert channel data to AudioBuffer.
- * Accepts either per-channel Float32Array[] or a single Float32Array (mono, backwards compatible).
- */
-declare function createAudioBuffer(audioContext: AudioContext, channelData: Float32Array[] | Float32Array, sampleRate: number, channelCount?: number): AudioBuffer;
-
-export { type IntegratedRecordingOptions, type MicrophoneDevice, type RecordingData, type RecordingOptions, type RecordingState, type UseIntegratedRecordingReturn, type UseMicrophoneAccessReturn, type UseMicrophoneLevelOptions, type UseMicrophoneLevelReturn, type UseRecordingReturn, appendPeaks, concatenateAudioData, createAudioBuffer, generatePeaks, useIntegratedRecording, useMicrophoneAccess, useMicrophoneLevel, useRecording };
+export { type IntegratedRecordingOptions, type MicrophoneDevice, type RecordingData, type RecordingOptions, type RecordingState, type UseIntegratedRecordingReturn, type UseMicrophoneAccessReturn, type UseMicrophoneLevelOptions, type UseMicrophoneLevelReturn, type UseRecordingReturn, useIntegratedRecording, useMicrophoneAccess, useMicrophoneLevel, useRecording };

package/dist/index.js

@@ -20,10 +20,6 @@ var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: tru
 // src/index.ts
 var src_exports = {};
 __export(src_exports, {
-  appendPeaks: () => appendPeaks,
-  concatenateAudioData: () => concatenateAudioData,
-  createAudioBuffer: () => createAudioBuffer,
-  generatePeaks: () => generatePeaks,
   useIntegratedRecording: () => useIntegratedRecording,
   useMicrophoneAccess: () => useMicrophoneAccess,
   useMicrophoneLevel: () => useMicrophoneLevel,
@@ -33,81 +29,7 @@ module.exports = __toCommonJS(src_exports);
 
 // src/hooks/useRecording.ts
 var import_react = require("react");
-
-// src/utils/audioBufferUtils.ts
-function concatenateAudioData(chunks) {
-  const totalLength = chunks.reduce((sum, chunk) => sum + chunk.length, 0);
-  const result = new Float32Array(totalLength);
-  let offset = 0;
-  for (const chunk of chunks) {
-    result.set(chunk, offset);
-    offset += chunk.length;
-  }
-  return result;
-}
-function createAudioBuffer(audioContext, channelData, sampleRate, channelCount = 1) {
-  const channels = channelData instanceof Float32Array ? [channelData] : channelData;
-  const length = channels[0]?.length ?? 0;
-  const buffer = audioContext.createBuffer(channelCount, length, sampleRate);
-  for (let ch = 0; ch < Math.min(channelCount, channels.length); ch++) {
-    buffer.copyToChannel(new Float32Array(channels[ch]), ch);
-  }
-  return buffer;
-}
-
-// src/utils/peaksGenerator.ts
-function generatePeaks(samples, samplesPerPixel, bits = 16) {
-  const numPeaks = Math.ceil(samples.length / samplesPerPixel);
-  const peakArray = bits === 8 ? new Int8Array(numPeaks * 2) : new Int16Array(numPeaks * 2);
-  const maxValue = 2 ** (bits - 1);
-  for (let i = 0; i < numPeaks; i++) {
-    const start = i * samplesPerPixel;
-    const end = Math.min(start + samplesPerPixel, samples.length);
-    let min = 0;
-    let max = 0;
-    for (let j = start; j < end; j++) {
-      const value = samples[j];
-      if (value < min) min = value;
-      if (value > max) max = value;
-    }
-    peakArray[i * 2] = Math.max(-maxValue, Math.floor(min * maxValue));
-    peakArray[i * 2 + 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));
-  }
-  return peakArray;
-}
-function appendPeaks(existingPeaks, newSamples, samplesPerPixel, totalSamplesProcessed, bits = 16) {
-  const maxValue = 2 ** (bits - 1);
-  const remainder = totalSamplesProcessed % samplesPerPixel;
-  let offset = 0;
-  if (remainder > 0 && existingPeaks.length > 0) {
-    const samplesToComplete = samplesPerPixel - remainder;
-    const endIndex = Math.min(samplesToComplete, newSamples.length);
-    let min = existingPeaks[existingPeaks.length - 2] / maxValue;
-    let max = existingPeaks[existingPeaks.length - 1] / maxValue;
-    for (let i = 0; i < endIndex; i++) {
-      const value = newSamples[i];
-      if (value < min) min = value;
-      if (value > max) max = value;
-    }
-    const updated = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length);
-    updated.set(existingPeaks);
-    updated[existingPeaks.length - 2] = Math.max(-maxValue, Math.floor(min * maxValue));
-    updated[existingPeaks.length - 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));
-    offset = endIndex;
-    const newPeaks2 = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);
-    const result2 = new (bits === 8 ? Int8Array : Int16Array)(updated.length + newPeaks2.length);
-    result2.set(updated);
-    result2.set(newPeaks2, updated.length);
-    return result2;
-  }
-  const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);
-  const result = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length + newPeaks.length);
-  result.set(existingPeaks);
-  result.set(newPeaks, existingPeaks.length);
-  return result;
-}
-
-// src/hooks/useRecording.ts
+var import_core = require("@waveform-playlist/core");
 var import_playout = require("@waveform-playlist/playout");
 var import_worklets = require("@waveform-playlist/worklets");
 function emptyPeaks(bits) {
@@ -219,7 +141,7 @@ function useRecording(stream, options = {}) {
           for (let ch = 0; ch < channels.length; ch++) {
             const prev = prevPeaks[ch] ?? emptyPeaks(bits);
             updated.push(
-              appendPeaks(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)
+              (0, import_core.appendPeaks)(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)
             );
           }
           return updated;
@@ -277,7 +199,7 @@ function useRecording(stream, options = {}) {
       const context = (0, import_playout.getGlobalContext)();
       const rawContext = context.rawContext;
       const numChannels = recordedChunksRef.current.length || channelCount;
-      const channelData = recordedChunksRef.current.map((chunks) => concatenateAudioData(chunks));
+      const channelData = recordedChunksRef.current.map((chunks) => (0, import_core.concatenateAudioData)(chunks));
       const totalSamples = channelData[0]?.length ?? 0;
       if (totalSamples === 0) {
         console.warn("[waveform-playlist] Recording stopped with 0 samples captured \u2014 discarding");
@@ -288,7 +210,7 @@ function useRecording(stream, options = {}) {
         setLevel(0);
         return null;
       }
-      const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);
+      const buffer = (0, import_core.createAudioBuffer)(rawContext, channelData, rawContext.sampleRate, numChannels);
       setAudioBuffer(buffer);
       setDuration(buffer.duration);
       isRecordingRef.current = false;
@@ -454,7 +376,7 @@ function useMicrophoneAccess() {
 // src/hooks/useMicrophoneLevel.ts
 var import_react3 = require("react");
 var import_playout2 = require("@waveform-playlist/playout");
-var import_core = require("@waveform-playlist/core");
+var import_core2 = require("@waveform-playlist/core");
 var import_worklets2 = require("@waveform-playlist/worklets");
 var PEAK_DECAY = 0.98;
 function useMicrophoneLevel(stream, options = {}) {
@@ -515,8 +437,8 @@ function useMicrophoneLevel(stream, options = {}) {
         const rmsValues = [];
         for (let ch = 0; ch < peak.length; ch++) {
           smoothed[ch] = Math.max(peak[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);
-          peakValues.push((0, import_core.gainToNormalized)(smoothed[ch]));
-          rmsValues.push((0, import_core.gainToNormalized)(rms[ch]));
+          peakValues.push((0, import_core2.gainToNormalized)(smoothed[ch]));
+          rmsValues.push((0, import_core2.gainToNormalized)(rms[ch]));
         }
         const mirroredPeaks = peak.length < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;
         const mirroredRms = peak.length < channelCount ? new Array(channelCount).fill(rmsValues[0]) : rmsValues;
@@ -752,10 +674,6 @@ function useIntegratedRecording(tracks, setTracks, selectedTrackId, options = {}
 }
 // Annotate the CommonJS export names for ESM import in node:
 0 && (module.exports = {
-  appendPeaks,
-  concatenateAudioData,
-  createAudioBuffer,
-  generatePeaks,
   useIntegratedRecording,
   useMicrophoneAccess,
   useMicrophoneLevel,

package/dist/index.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/index.ts","../src/hooks/useRecording.ts","../src/utils/audioBufferUtils.ts","../src/utils/peaksGenerator.ts","../src/hooks/useMicrophoneAccess.ts","../src/hooks/useMicrophoneLevel.ts","../src/hooks/useIntegratedRecording.ts"],"sourcesContent":["/**\n * @waveform-playlist/recording\n *\n * Audio recording support using AudioWorklet for waveform-playlist\n */\n\n// Hooks\nexport {\n  useRecording,\n  useMicrophoneAccess,\n  useMicrophoneLevel,\n  useIntegratedRecording,\n} from './hooks';\nexport type {\n  UseMicrophoneLevelOptions,\n  UseMicrophoneLevelReturn,\n  UseIntegratedRecordingReturn,\n  IntegratedRecordingOptions,\n} from './hooks';\n\n// Types\nexport type {\n  RecordingState,\n  RecordingData,\n  MicrophoneDevice,\n  RecordingOptions,\n  UseRecordingReturn,\n  UseMicrophoneAccessReturn,\n} from './types';\n\n// Utilities\nexport { generatePeaks, appendPeaks } from './utils/peaksGenerator';\nexport { createAudioBuffer, concatenateAudioData } from './utils/audioBufferUtils';\n","/**\n * Main recording hook using AudioWorklet\n */\n\nimport { useState, useRef, useCallback, useEffect } from 'react';\nimport { UseRecordingReturn, RecordingOptions } from '../types';\nimport { concatenateAudioData, createAudioBuffer } from '../utils/audioBufferUtils';\nimport { appendPeaks } from '../utils/peaksGenerator';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { recordingProcessorUrl } from '@waveform-playlist/worklets';\n\nfunction emptyPeaks(bits: 8 | 16): Int8Array | Int16Array {\n  return bits === 8 ? new Int8Array(0) : new Int16Array(0);\n}\n\nexport function useRecording(\n  stream: MediaStream | null,\n  options: RecordingOptions = {}\n): UseRecordingReturn {\n  const { channelCount = 1, samplesPerPixel = 1024, bits = 16 } = options;\n\n  // State\n  const [isRecording, setIsRecording] = useState(false);\n  const [isPaused, setIsPaused] = useState(false);\n  const [duration, setDuration] = useState(0);\n  // Per-channel peaks for multi-channel live preview\n  const [peaks, setPeaks] = useState<(Int8Array | Int16Array)[]>([emptyPeaks(bits)]);\n  const [audioBuffer, setAudioBuffer] = useState<AudioBuffer | null>(null);\n  const [error, setError] = useState<Error | null>(null);\n  const [level, setLevel] = useState(0); // Current RMS level (0-1)\n  const [peakLevel, setPeakLevel] = useState(0); // Peak level since recording started (0-1)\n\n  // Per-instance flag to prevent loading worklet multiple times within the same hook instance.\n  // Note: Multiple hook instances each have their own ref — see \"Multi-Instance Worklet\n  // Registration Gap\" in recording/CLAUDE.md for the known limitation and planned fix.\n  const workletLoadedRef = useRef<boolean>(false);\n\n  // Refs for AudioWorklet and data accumulation\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const mediaStreamSourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  // Per-channel sample accumulation: recordedChunksRef[channelIndex] = Float32Array[]\n  const recordedChunksRef = useRef<Float32Array[][]>([]);\n  const totalSamplesRef = useRef(0);\n  const animationFrameRef = useRef<number | null>(null);\n  const startTimeRef = useRef<number>(0);\n  const isRecordingRef = useRef<boolean>(false);\n  const isPausedRef = useRef<boolean>(false);\n  const audioTrackRef = useRef<MediaStreamTrack | null>(null);\n  const onTrackEndedRef = useRef<(() => void) | null>(null);\n  const stopRecordingRef = useRef<(() => Promise<AudioBuffer | null>) | null>(null);\n\n  // Shared duration update loop — starts a rAF loop that updates duration\n  // from performance.now(). Used by both startRecording and resumeRecording.\n  const startDurationLoop = useCallback(() => {\n    const tick = () => {\n      if (isRecordingRef.current && !isPausedRef.current) {\n        const elapsed = (performance.now() - startTimeRef.current) / 1000;\n        setDuration(elapsed);\n        animationFrameRef.current = requestAnimationFrame(tick);\n      }\n    };\n    tick();\n  }, []);\n\n  // Load AudioWorklet module\n  const loadWorklet = useCallback(async () => {\n    // Skip if already loaded to prevent \"already registered\" error\n    if (workletLoadedRef.current) {\n      return;\n    }\n\n    try {\n      const context = getGlobalContext();\n      // Load the worklet module directly on the raw AudioContext.\n      // Tone.js's addAudioWorkletModule only loads ONE module per context\n      // (caches _workletPromise). If meter-processor was loaded first by\n      // useMicrophoneLevel, recording-processor is silently skipped.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(recordingProcessorUrl);\n      workletLoadedRef.current = true;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to load AudioWorklet module:', String(err));\n      const error = new Error('Failed to load recording processor: ' + String(err));\n      throw error;\n    }\n  }, []);\n\n  // Start recording\n  const startRecording = useCallback(async () => {\n    if (isRecordingRef.current) return;\n    if (!stream) {\n      setError(new Error('No microphone stream available'));\n      return;\n    }\n\n    try {\n      setError(null);\n\n      // Use Tone.js Context for cross-browser compatibility\n      const context = getGlobalContext();\n\n      // Resume AudioContext if suspended\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n\n      // Load worklet module\n      await loadWorklet();\n\n      // Create MediaStreamSource from Tone's context\n      // Each hook creates its own source to avoid cross-context issues in Firefox\n      const source = context.createMediaStreamSource(stream);\n      mediaStreamSourceRef.current = source;\n\n      // Use the stream track's actual channel count from getSettings().\n      // source.channelCount defaults to 2 per Web Audio spec (not the mic's\n      // real count). Fall back to user-provided channelCount if unavailable.\n      const detectedChannelCount = stream.getAudioTracks()[0]?.getSettings().channelCount;\n      if (detectedChannelCount === undefined) {\n        console.warn(\n          `[waveform-playlist] Could not detect stream channel count, using fallback: ${channelCount}`\n        );\n      }\n      const streamChannelCount = detectedChannelCount ?? channelCount;\n\n      const workletNode = context.createAudioWorkletNode('recording-processor', {\n        channelCount: streamChannelCount,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Recording worklet processor error:', String(event));\n        setError(new Error('Recording processor encountered an error'));\n      };\n\n      // Reset state before connecting — prevents race where a worklet message\n      // arrives before refs are cleared, corrupting samplesProcessedBefore calculations\n      recordedChunksRef.current = Array.from({ length: streamChannelCount }, () => []);\n      totalSamplesRef.current = 0;\n      setPeaks(Array.from({ length: streamChannelCount }, () => emptyPeaks(bits)));\n      setAudioBuffer(null);\n      setLevel(0);\n      setPeakLevel(0);\n\n      // Listen for audio data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        const { channels } = event.data as { channels: Float32Array[] };\n\n        if (!channels || channels.length === 0) {\n          console.warn('[waveform-playlist] Recording worklet sent empty or missing channels data');\n          return;\n        }\n\n        // Accumulate per-channel samples\n        for (let ch = 0; ch < channels.length; ch++) {\n          if (!recordedChunksRef.current[ch]) {\n            console.warn(\n              `[waveform-playlist] Unexpected channel ${ch} from worklet (expected ${recordedChunksRef.current.length})`\n            );\n            recordedChunksRef.current[ch] = [];\n          }\n          recordedChunksRef.current[ch].push(channels[ch]);\n        }\n        // Capture sample offset before incrementing — used by peak alignment\n        const samplesProcessedBefore = totalSamplesRef.current;\n        totalSamplesRef.current += channels[0].length;\n        setPeaks((prevPeaks) => {\n          // Ensure we have an entry per channel\n          const updated: (Int8Array | Int16Array)[] = [];\n          for (let ch = 0; ch < channels.length; ch++) {\n            const prev = prevPeaks[ch] ?? emptyPeaks(bits);\n            updated.push(\n              appendPeaks(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)\n            );\n          }\n          return updated;\n        });\n\n        // Note: VU meter levels come from useMicrophoneLevel (meter-processor worklet)\n        // We don't update level/peakLevel here to avoid conflicting state updates\n      };\n\n      // Connect and start — after state reset and handler setup\n      source.connect(workletNode);\n      // Do NOT send sampleRate — worklet uses its global sampleRate (always correct)\n      workletNode.port.postMessage({ command: 'start', channelCount: streamChannelCount });\n\n      // Listen on MediaStreamTrack for mic unplug (MediaStream has no 'ended' event)\n      const audioTrack = stream.getAudioTracks()[0] ?? null;\n      audioTrackRef.current = audioTrack;\n      if (audioTrack) {\n        const onEnded = () => {\n          console.warn('[waveform-playlist] Audio track ended (mic unplugged or revoked)');\n          // Stop recording to clean up worklet, rAF loop, and UI state\n          stopRecordingRef.current?.();\n          setError(new Error('Microphone disconnected during recording'));\n        };\n        onTrackEndedRef.current = onEnded;\n        audioTrack.addEventListener('ended', onEnded);\n      }\n\n      isRecordingRef.current = true;\n      isPausedRef.current = false;\n      setIsRecording(true);\n      setIsPaused(false);\n      startTimeRef.current = performance.now();\n      startDurationLoop();\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to start recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to start recording'));\n    }\n  }, [stream, channelCount, samplesPerPixel, bits, loadWorklet, startDurationLoop]);\n\n  // Stop recording\n  const stopRecording = useCallback(async (): Promise<AudioBuffer | null> => {\n    if (!isRecordingRef.current) {\n      return null;\n    }\n\n    try {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n        audioTrackRef.current = null;\n        onTrackEndedRef.current = null;\n      }\n\n      // Stop the worklet\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n          }\n        }\n        workletNodeRef.current.disconnect();\n      }\n\n      // Cancel animation frame\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n\n      // Create final AudioBuffer from accumulated per-channel chunks\n      const context = getGlobalContext();\n      const rawContext = context.rawContext as AudioContext;\n      const numChannels = recordedChunksRef.current.length || channelCount;\n      const channelData = recordedChunksRef.current.map((chunks) => concatenateAudioData(chunks));\n      const totalSamples = channelData[0]?.length ?? 0;\n\n      // Guard: if no samples were captured (e.g., stop called immediately after start),\n      // return null instead of creating a 0-length AudioBuffer which throws\n      if (totalSamples === 0) {\n        console.warn('[waveform-playlist] Recording stopped with 0 samples captured — discarding');\n        isRecordingRef.current = false;\n        isPausedRef.current = false;\n        setIsRecording(false);\n        setIsPaused(false);\n        setLevel(0);\n        return null;\n      }\n\n      const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n      setAudioBuffer(buffer);\n      setDuration(buffer.duration);\n      isRecordingRef.current = false;\n      isPausedRef.current = false;\n      setIsRecording(false);\n      setIsPaused(false);\n      setLevel(0);\n      // Keep peakLevel to show the peak reached during recording\n\n      return buffer;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to stop recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to stop recording'));\n      return null;\n    }\n  }, [channelCount]);\n  stopRecordingRef.current = stopRecording;\n\n  // Pause recording\n  const pauseRecording = useCallback(() => {\n    if (isRecording && !isPaused) {\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n      isPausedRef.current = true;\n      setIsPaused(true);\n    }\n  }, [isRecording, isPaused]);\n\n  // Resume recording\n  const resumeRecording = useCallback(() => {\n    if (isRecording && isPaused) {\n      isPausedRef.current = false;\n      setIsPaused(false);\n      startTimeRef.current = performance.now() - duration * 1000;\n      startDurationLoop();\n    }\n  }, [isRecording, isPaused, duration, startDurationLoop]);\n\n  // Cleanup on unmount — read refs in cleanup, not effect body.\n  // Pattern #16 (copy refs in effect body) doesn't apply to [] deps — refs are\n  // null at mount and only set later during startRecording.\n  useEffect(() => {\n    return () => {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n      }\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during cleanup:', String(err));\n          }\n        }\n        try {\n          workletNodeRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Worklet disconnect during cleanup:', String(err));\n        }\n      }\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n      }\n      // Don't close the global AudioContext - it's shared across the app\n    };\n  }, []);\n\n  return {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer,\n    level,\n    peakLevel,\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    error,\n  };\n}\n","/**\n * Utility functions for working with AudioBuffers during recording\n */\n\n/**\n * Concatenate multiple Float32Arrays into a single array\n */\nexport function concatenateAudioData(chunks: Float32Array[]): Float32Array {\n  const totalLength = chunks.reduce((sum, chunk) => sum + chunk.length, 0);\n  const result = new Float32Array(totalLength);\n\n  let offset = 0;\n  for (const chunk of chunks) {\n    result.set(chunk, offset);\n    offset += chunk.length;\n  }\n\n  return result;\n}\n\n/**\n * Convert channel data to AudioBuffer.\n * Accepts either per-channel Float32Array[] or a single Float32Array (mono, backwards compatible).\n */\nexport function createAudioBuffer(\n  audioContext: AudioContext,\n  channelData: Float32Array[] | Float32Array,\n  sampleRate: number,\n  channelCount: number = 1\n): AudioBuffer {\n  // Backwards compatibility: single Float32Array → wrap as mono\n  const channels: Float32Array[] =\n    channelData instanceof Float32Array ? [channelData] : channelData;\n\n  const length = channels[0]?.length ?? 0;\n  const buffer = audioContext.createBuffer(channelCount, length, sampleRate);\n\n  for (let ch = 0; ch < Math.min(channelCount, channels.length); ch++) {\n    buffer.copyToChannel(new Float32Array(channels[ch]), ch);\n  }\n\n  return buffer;\n}\n\n/**\n * Append new samples to an existing AudioBuffer (mono convenience)\n */\nexport function appendToAudioBuffer(\n  audioContext: AudioContext,\n  existingBuffer: AudioBuffer | null,\n  newSamples: Float32Array,\n  sampleRate: number\n): AudioBuffer {\n  if (!existingBuffer) {\n    return createAudioBuffer(audioContext, [newSamples], sampleRate);\n  }\n\n  // Get existing samples\n  const existingData = existingBuffer.getChannelData(0);\n\n  // Concatenate using concatenateAudioData helper\n  const combined = concatenateAudioData([existingData, newSamples]);\n\n  // Create new buffer\n  return createAudioBuffer(audioContext, [combined], sampleRate);\n}\n\n/**\n * Calculate duration in seconds from sample count and sample rate\n */\nexport function calculateDuration(sampleCount: number, sampleRate: number): number {\n  return sampleCount / sampleRate;\n}\n","/**\n * Peak generation for real-time waveform visualization during recording\n * Matches the format used by webaudio-peaks: min/max pairs with bit depth\n */\n\n/**\n * Generate peaks from audio samples in standard min/max pair format\n *\n * @param samples - Audio samples to process\n * @param samplesPerPixel - Number of samples to represent in each peak\n * @param bits - Bit depth for peak values (8 or 16)\n * @returns Int8Array or Int16Array of peak values (min/max pairs)\n */\nexport function generatePeaks(\n  samples: Float32Array,\n  samplesPerPixel: number,\n  bits: 8 | 16 = 16\n): Int8Array | Int16Array {\n  const numPeaks = Math.ceil(samples.length / samplesPerPixel);\n  const peakArray = bits === 8 ? new Int8Array(numPeaks * 2) : new Int16Array(numPeaks * 2);\n  const maxValue = 2 ** (bits - 1);\n\n  for (let i = 0; i < numPeaks; i++) {\n    const start = i * samplesPerPixel;\n    const end = Math.min(start + samplesPerPixel, samples.length);\n\n    let min = 0;\n    let max = 0;\n\n    for (let j = start; j < end; j++) {\n      const value = samples[j];\n      if (value < min) min = value;\n      if (value > max) max = value;\n    }\n\n    // Store as min/max pairs scaled to bit depth\n    // Clamp to valid range: Int16 is [-32768, 32767], Int8 is [-128, 127]\n    peakArray[i * 2] = Math.max(-maxValue, Math.floor(min * maxValue));\n    peakArray[i * 2 + 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));\n  }\n\n  return peakArray;\n}\n\n/**\n * Append new peaks to existing peaks array\n * This is used for incremental peak updates during recording\n */\nexport function appendPeaks(\n  existingPeaks: Int8Array | Int16Array,\n  newSamples: Float32Array,\n  samplesPerPixel: number,\n  totalSamplesProcessed: number,\n  bits: 8 | 16 = 16\n): Int8Array | Int16Array {\n  const maxValue = 2 ** (bits - 1);\n\n  // Check if we have a partial peak from the last update\n  const remainder = totalSamplesProcessed % samplesPerPixel;\n  let offset = 0;\n\n  // If there's a partial peak, we need to update the last peak\n  if (remainder > 0 && existingPeaks.length > 0) {\n    const samplesToComplete = samplesPerPixel - remainder;\n    const endIndex = Math.min(samplesToComplete, newSamples.length);\n\n    // Get current min/max from last peak\n    let min = existingPeaks[existingPeaks.length - 2] / maxValue;\n    let max = existingPeaks[existingPeaks.length - 1] / maxValue;\n\n    // Update with new samples\n    for (let i = 0; i < endIndex; i++) {\n      const value = newSamples[i];\n      if (value < min) min = value;\n      if (value > max) max = value;\n    }\n\n    // Update last peak\n    const updated = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length);\n    updated.set(existingPeaks);\n    updated[existingPeaks.length - 2] = Math.max(-maxValue, Math.floor(min * maxValue));\n    updated[existingPeaks.length - 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));\n\n    offset = endIndex;\n\n    // Generate peaks for remaining samples and concatenate\n    const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);\n    const result = new (bits === 8 ? Int8Array : Int16Array)(updated.length + newPeaks.length);\n    result.set(updated);\n    result.set(newPeaks, updated.length);\n    return result;\n  }\n\n  // No partial peak, just concatenate\n  const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);\n  const result = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length + newPeaks.length);\n  result.set(existingPeaks);\n  result.set(newPeaks, existingPeaks.length);\n  return result;\n}\n","/**\n * Hook for managing microphone access and device enumeration\n */\n\nimport { useState, useEffect, useCallback } from 'react';\nimport { UseMicrophoneAccessReturn, MicrophoneDevice } from '../types';\n\nexport function useMicrophoneAccess(): UseMicrophoneAccessReturn {\n  const [stream, setStream] = useState<MediaStream | null>(null);\n  const [devices, setDevices] = useState<MicrophoneDevice[]>([]);\n  const [hasPermission, setHasPermission] = useState(false);\n  const [isLoading, setIsLoading] = useState(false);\n  const [error, setError] = useState<Error | null>(null);\n\n  // Enumerate audio input devices\n  const enumerateDevices = useCallback(async () => {\n    try {\n      const allDevices = await navigator.mediaDevices.enumerateDevices();\n      const audioInputs = allDevices\n        .filter((device) => device.kind === 'audioinput')\n        .map((device) => ({\n          deviceId: device.deviceId,\n          label: device.label || `Microphone ${device.deviceId.slice(0, 8)}`,\n          groupId: device.groupId,\n        }));\n\n      setDevices(audioInputs);\n    } catch (err) {\n      console.error('Failed to enumerate devices:', err);\n      setError(err instanceof Error ? err : new Error('Failed to enumerate devices'));\n    }\n  }, []);\n\n  // Request microphone access\n  const requestAccess = useCallback(\n    async (deviceId?: string, audioConstraints?: MediaTrackConstraints) => {\n      setIsLoading(true);\n      setError(null);\n\n      try {\n        // Stop existing stream if any\n        if (stream) {\n          stream.getTracks().forEach((track) => track.stop());\n        }\n\n        // Build audio constraints\n        const audio: MediaTrackConstraints & { latency?: number } = {\n          // Recording-optimized defaults: prioritize raw audio quality and low latency\n          echoCancellation: false,\n          noiseSuppression: false,\n          autoGainControl: false,\n          latency: 0, // Low latency mode (not in TS types yet, but supported in modern browsers)\n          // User-provided constraints override defaults\n          ...audioConstraints,\n          // Device ID override (if specified)\n          ...(deviceId && { deviceId: { exact: deviceId } }),\n        };\n\n        const constraints: MediaStreamConstraints = {\n          audio,\n          video: false,\n        };\n\n        const newStream = await navigator.mediaDevices.getUserMedia(constraints);\n        setStream(newStream);\n        setHasPermission(true);\n\n        // Enumerate devices after getting permission (labels will be available)\n        await enumerateDevices();\n      } catch (err) {\n        console.error('Failed to access microphone:', err);\n        setError(err instanceof Error ? err : new Error('Failed to access microphone'));\n        setHasPermission(false);\n      } finally {\n        setIsLoading(false);\n      }\n    },\n    [stream, enumerateDevices]\n  );\n\n  // Stop the stream and revoke access\n  const stopStream = useCallback(() => {\n    if (stream) {\n      stream.getTracks().forEach((track) => track.stop());\n      setStream(null);\n      setHasPermission(false);\n    }\n  }, [stream]);\n\n  // Check initial permission state, enumerate devices, and listen for hot-plug changes\n  useEffect(() => {\n    // Try to enumerate devices (labels won't be available without permission)\n    enumerateDevices();\n\n    // Re-enumerate when devices are plugged in or removed\n    navigator.mediaDevices.addEventListener('devicechange', enumerateDevices);\n\n    // Cleanup on unmount\n    return () => {\n      navigator.mediaDevices.removeEventListener('devicechange', enumerateDevices);\n      if (stream) {\n        stream.getTracks().forEach((track) => track.stop());\n      }\n    };\n  }, [enumerateDevices, stream]);\n\n  return {\n    stream,\n    devices,\n    hasPermission,\n    isLoading,\n    requestAccess,\n    stopStream,\n    error,\n  };\n}\n","/**\n * Hook for monitoring microphone input levels\n *\n * Uses an AudioWorklet-based meter processor for sample-accurate\n * peak and RMS metering without requestAnimationFrame overhead.\n */\n\nimport { useEffect, useState, useRef, useCallback } from 'react';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { gainToNormalized } from '@waveform-playlist/core';\nimport { meterProcessorUrl, type MeterMessage } from '@waveform-playlist/worklets';\n\n/** Peak decay constant — exponential decay for smooth peak hold (~800ms to 1/e at 60fps) */\nconst PEAK_DECAY = 0.98;\n\nexport interface UseMicrophoneLevelOptions {\n  /**\n   * How often to update the level (in Hz)\n   * Default: 60 (60fps)\n   */\n  updateRate?: number;\n\n  /**\n   * Number of channels to meter (1 = mono, 2 = stereo)\n   * Default: 1\n   */\n  channelCount?: number;\n}\n\nexport interface UseMicrophoneLevelReturn {\n  /**\n   * Current peak audio level (0-1)\n   * For single channel: channel 0 level\n   * For multi-channel: max across all channels\n   */\n  level: number;\n\n  /**\n   * Held peak level since last reset (0-1)\n   * For single channel: channel 0 peak\n   * For multi-channel: max across all channels\n   */\n  peakLevel: number;\n\n  /**\n   * Reset the held peak level\n   */\n  resetPeak: () => void;\n\n  /**\n   * Per-channel peak levels (0-1). Array length matches channelCount.\n   * True peak: max absolute sample value per analysis frame.\n   */\n  levels: number[];\n\n  /**\n   * Per-channel held peak levels (0-1). Array length matches channelCount.\n   */\n  peakLevels: number[];\n\n  /**\n   * Per-channel RMS levels (0-1). Array length matches channelCount.\n   * RMS: root mean square of samples per analysis frame.\n   */\n  rmsLevels: number[];\n\n  /**\n   * Error from meter setup (worklet load failure, context issues, etc.)\n   * Null when metering is working normally.\n   */\n  error: Error | null;\n}\n\n/**\n * Monitor microphone input levels in real-time\n *\n * @param stream - MediaStream from getUserMedia\n * @param options - Configuration options\n * @returns Object with current peak level, RMS level, and held peak level\n *\n * @example\n * ```typescript\n * const { stream } = useMicrophoneAccess();\n * const { levels, rmsLevels, peakLevels } = useMicrophoneLevel(stream, { channelCount: 2 });\n *\n * return <SegmentedVUMeter levels={levels} peakLevels={peakLevels} />;\n * ```\n */\nexport function useMicrophoneLevel(\n  stream: MediaStream | null,\n  options: UseMicrophoneLevelOptions = {}\n): UseMicrophoneLevelReturn {\n  const { updateRate = 60, channelCount = 1 } = options;\n\n  const [levels, setLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [peakLevels, setPeakLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [rmsLevels, setRmsLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [meterError, setMeterError] = useState<Error | null>(null);\n\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  const smoothedPeakRef = useRef<number[]>(new Array(channelCount).fill(0));\n\n  const resetPeak = useCallback(\n    () => setPeakLevels(new Array(channelCount).fill(0)),\n    [channelCount]\n  );\n\n  useEffect(() => {\n    if (!stream) {\n      setLevels(new Array(channelCount).fill(0));\n      setPeakLevels(new Array(channelCount).fill(0));\n      setRmsLevels(new Array(channelCount).fill(0));\n      smoothedPeakRef.current = new Array(channelCount).fill(0);\n      return;\n    }\n\n    let isMounted = true;\n\n    const setupMonitoring = async () => {\n      if (!isMounted) return;\n\n      const context = getGlobalContext();\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n      if (!isMounted) return;\n\n      // Auto-detect actual mic channel count from stream\n      const trackSettings = stream.getAudioTracks()[0]?.getSettings();\n      const actualChannels = trackSettings?.channelCount ?? channelCount;\n\n      // Load worklet directly on rawContext — Tone.js's addAudioWorkletModule\n      // only loads ONE module per context (caches _workletPromise), silently\n      // skipping subsequent calls with different URLs.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(meterProcessorUrl);\n      if (!isMounted) return;\n\n      // Use Tone.js's createAudioWorkletNode — avoids rawContext identity issues\n      // in webpack-aliased environments (Docusaurus)\n      const workletNode = context.createAudioWorkletNode('meter-processor', {\n        channelCount: actualChannels,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n        processorOptions: {\n          numberOfChannels: actualChannels,\n          updateRate,\n        },\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Mic meter worklet processor error:', String(event));\n      };\n\n      // Create source and connect: source → meter\n      // Don't connect output to destination — mic monitoring would cause feedback\n      const source = context.createMediaStreamSource(stream);\n      sourceRef.current = source;\n      source.connect(workletNode);\n\n      smoothedPeakRef.current = new Array(actualChannels).fill(0);\n\n      // Listen for meter data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        if (!isMounted) return;\n\n        const { peak, rms } = event.data as MeterMessage;\n        const smoothed = smoothedPeakRef.current;\n\n        const peakValues: number[] = [];\n        const rmsValues: number[] = [];\n\n        for (let ch = 0; ch < peak.length; ch++) {\n          smoothed[ch] = Math.max(peak[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);\n          peakValues.push(gainToNormalized(smoothed[ch]));\n          rmsValues.push(gainToNormalized(rms[ch]));\n        }\n\n        // Mirror mono to fill requested channelCount\n        const mirroredPeaks =\n          peak.length < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;\n        const mirroredRms =\n          peak.length < channelCount ? new Array(channelCount).fill(rmsValues[0]) : rmsValues;\n\n        setLevels(mirroredPeaks);\n        setRmsLevels(mirroredRms);\n        setPeakLevels((prev) => mirroredPeaks.map((val, i) => Math.max(prev[i] ?? 0, val)));\n      };\n    };\n\n    setupMonitoring().catch((err) => {\n      console.warn('[waveform-playlist] Failed to set up mic level monitoring:', String(err));\n      if (isMounted) {\n        setMeterError(err instanceof Error ? err : new Error(String(err)));\n      }\n    });\n\n    return () => {\n      isMounted = false;\n\n      if (sourceRef.current) {\n        try {\n          sourceRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic source disconnect during cleanup:', String(err));\n        }\n        sourceRef.current = null;\n      }\n\n      if (workletNodeRef.current) {\n        try {\n          workletNodeRef.current.disconnect();\n          workletNodeRef.current.port.close();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic meter disconnect during cleanup:', String(err));\n        }\n        workletNodeRef.current = null;\n      }\n    };\n  }, [stream, updateRate, channelCount]);\n\n  // Backwards-compatible scalar values\n  const level = channelCount === 1 ? (levels[0] ?? 0) : Math.max(...levels);\n  const peakLevel = channelCount === 1 ? (peakLevels[0] ?? 0) : Math.max(...peakLevels);\n\n  return {\n    level,\n    peakLevel,\n    resetPeak,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: meterError,\n  };\n}\n","/**\n * Hook for integrated multi-track recording\n * Combines recording functionality with track management\n */\n\nimport { useState, useCallback, useEffect, useRef } from 'react';\nimport { useRecording } from './useRecording';\nimport { useMicrophoneAccess } from './useMicrophoneAccess';\nimport { useMicrophoneLevel } from './useMicrophoneLevel';\nimport type { MicrophoneDevice } from '../types';\nimport { type ClipTrack, type AudioClip } from '@waveform-playlist/core';\nimport {\n  resumeGlobalAudioContext,\n  getGlobalAudioContext,\n  getGlobalContext,\n} from '@waveform-playlist/playout';\n\nexport interface IntegratedRecordingOptions {\n  /**\n   * Current playback/cursor position in seconds\n   * Recording will start from max(currentTime, lastClipEndTime)\n   */\n  currentTime?: number;\n\n  /**\n   * MediaTrackConstraints for audio recording\n   * These will override the recording-optimized defaults (echo cancellation off, low latency)\n   */\n  audioConstraints?: MediaTrackConstraints;\n\n  /**\n   * Number of channels to record (1 = mono, 2 = stereo)\n   * Default: 1 (mono)\n   */\n  channelCount?: number;\n\n  /**\n   * Samples per pixel for peak generation\n   * Default: 1024\n   */\n  samplesPerPixel?: number;\n}\n\nexport interface UseIntegratedRecordingReturn {\n  // Recording state\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number;\n  level: number;\n  peakLevel: number;\n  /** Per-channel peak levels (0-1). Array length matches channelCount. */\n  levels: number[];\n  /** Per-channel held peak levels (0-1). Array length matches channelCount. */\n  peakLevels: number[];\n  /** Per-channel RMS levels (0-1). Array length matches channelCount. */\n  rmsLevels: number[];\n  error: Error | null;\n\n  // Microphone state\n  stream: MediaStream | null;\n  devices: MicrophoneDevice[];\n  hasPermission: boolean;\n  selectedDevice: string | null;\n\n  // Recording controls\n  startRecording: () => void;\n  stopRecording: () => void;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  requestMicAccess: () => Promise<void>;\n  changeDevice: (deviceId: string) => Promise<void>;\n\n  // Track state (for live waveform during recording)\n  recordingPeaks: (Int8Array | Int16Array)[];\n}\n\nexport function useIntegratedRecording(\n  tracks: ClipTrack[],\n  setTracks: (tracks: ClipTrack[]) => void,\n  selectedTrackId: string | null,\n  options: IntegratedRecordingOptions = {}\n): UseIntegratedRecordingReturn {\n  const { currentTime = 0, audioConstraints, ...recordingOptions } = options;\n\n  // Track if we're currently monitoring (for auto-resume audio context)\n  const [isMonitoring, setIsMonitoring] = useState(false);\n  const [selectedDevice, setSelectedDevice] = useState<string | null>(null);\n  const [hookError, setHookError] = useState<Error | null>(null);\n\n  // Capture timeline position when recording starts (not at stop time)\n  const recordingStartTimeRef = useRef(0);\n\n  // Keep selectedTrackId and currentTime in refs for use in callbacks.\n  // Avoids stale closures and prevents 60fps useCallback recreation\n  // (currentTime updates at animation-frame rate during playback).\n  const selectedTrackIdRef = useRef(selectedTrackId);\n  selectedTrackIdRef.current = selectedTrackId;\n  const currentTimeRef = useRef(currentTime);\n  currentTimeRef.current = currentTime;\n\n  // Microphone access\n  const { stream, devices, hasPermission, requestAccess, error: micError } = useMicrophoneAccess();\n\n  // Microphone level (for VU meter)\n  const {\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    resetPeak,\n    error: meterError,\n  } = useMicrophoneLevel(stream, {\n    channelCount: recordingOptions.channelCount,\n  });\n\n  // Recording\n  const {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer: _recordedAudioBuffer,\n    startRecording: startRec,\n    stopRecording: stopRec,\n    pauseRecording,\n    resumeRecording,\n    error: recError,\n  } = useRecording(stream, recordingOptions);\n\n  // Start recording handler\n  // Reads selectedTrackId from ref to avoid stale closures when\n  // auto-create track + start recording happen in the same render cycle\n  const startRecording = useCallback(async () => {\n    if (!selectedTrackIdRef.current) {\n      setHookError(\n        new Error('Cannot start recording: no track selected. Select or create a track first.')\n      );\n      return;\n    }\n\n    try {\n      setHookError(null);\n      // Resume audio context if needed\n      if (!isMonitoring) {\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      }\n\n      // Capture timeline position NOW — before recording starts.\n      // Using currentTime at stop time would be wrong during overdub\n      // (playback advances currentTime while recording).\n      recordingStartTimeRef.current = currentTimeRef.current;\n\n      await startRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [isMonitoring, startRec]);\n\n  // Stop recording and add clip to selected track\n  const stopRecording = useCallback(async () => {\n    let buffer: AudioBuffer | null;\n    try {\n      buffer = await stopRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n      return;\n    }\n\n    // Add clip to track after recording completes\n    const trackId = selectedTrackIdRef.current;\n    if (buffer && trackId) {\n      const selectedTrackIndex = tracks.findIndex((t) => t.id === trackId);\n      if (selectedTrackIndex === -1) {\n        const err = new Error(\n          `Recording completed but track \"${trackId}\" no longer exists. The recorded audio could not be saved.`\n        );\n        console.warn(`[waveform-playlist] ${err.message}`);\n        setHookError(err);\n        return;\n      }\n\n      const selectedTrack = tracks[selectedTrackIndex];\n\n      // Use the captured start time (not live currentTime which advances during overdub)\n      const recordStartTimeSamples = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n      let lastClipEndSample = 0;\n      if (selectedTrack.clips.length > 0) {\n        const endSamples = selectedTrack.clips.map(\n          (clip) => clip.startSample + clip.durationSamples\n        );\n        lastClipEndSample = Math.max(...endSamples);\n      }\n\n      const startSample = Math.max(recordStartTimeSamples, lastClipEndSample);\n\n      // Latency compensation:\n      // Two sources of delay between recording start and audible playback:\n      // 1. Tone.js lookAhead (~100ms) — Transport schedules audio ahead of real time\n      // 2. Output latency — hardware DAC delay before audio reaches speakers\n      // The user hears playback delayed by both, so they perform late relative\n      // to the timeline. Skip that duration at the start of the recorded audio.\n      const audioContext = getGlobalAudioContext();\n      const outputLatency = audioContext.outputLatency ?? 0;\n      const toneContext = getGlobalContext();\n      const lookAhead = toneContext.lookAhead ?? 0;\n      const totalLatency = outputLatency + lookAhead;\n      const latencyOffsetSamples = Math.floor(totalLatency * buffer.sampleRate);\n\n      // Guard: very short recordings (< latency compensation) would produce negative duration\n      const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n      if (effectiveDuration === 0) {\n        console.warn(\n          '[waveform-playlist] Recording too short for latency compensation — discarding'\n        );\n        setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n        return;\n      }\n\n      // Create new clip from recording\n      const newClip: AudioClip = {\n        id: `clip-${Date.now()}`,\n        audioBuffer: buffer,\n        startSample,\n        durationSamples: effectiveDuration,\n        offsetSamples: latencyOffsetSamples,\n        sampleRate: buffer.sampleRate,\n        sourceDurationSamples: buffer.length,\n        gain: 1.0,\n        name: `Recording ${new Date().toLocaleTimeString()}`,\n      };\n\n      // Add clip to track\n      const newTracks = tracks.map((track, index) => {\n        if (index === selectedTrackIndex) {\n          return {\n            ...track,\n            clips: [...track.clips, newClip],\n          };\n        }\n        return track;\n      });\n\n      setTracks(newTracks);\n    }\n  }, [tracks, setTracks, stopRec]);\n\n  // Auto-select first device when available, or fallback if selected device was unplugged\n  useEffect(() => {\n    if (!hasPermission || devices.length === 0) return;\n\n    if (selectedDevice === null) {\n      // First-time selection\n      setSelectedDevice(devices[0].deviceId);\n    } else if (!devices.some((d) => d.deviceId === selectedDevice)) {\n      // Selected device was removed — fall back to first available\n      const fallbackId = devices[0].deviceId;\n      setSelectedDevice(fallbackId);\n      resetPeak();\n      requestAccess(fallbackId, audioConstraints);\n    }\n  }, [hasPermission, devices, selectedDevice, resetPeak, requestAccess, audioConstraints]);\n\n  // Request microphone access\n  const requestMicAccess = useCallback(async () => {\n    try {\n      setHookError(null);\n      await requestAccess(undefined, audioConstraints);\n      await resumeGlobalAudioContext();\n      setIsMonitoring(true);\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [requestAccess, audioConstraints]);\n\n  // Change device\n  const changeDevice = useCallback(\n    async (deviceId: string) => {\n      try {\n        setHookError(null);\n        setSelectedDevice(deviceId);\n        resetPeak();\n        await requestAccess(deviceId, audioConstraints);\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      } catch (err) {\n        setHookError(err instanceof Error ? err : new Error(String(err)));\n      }\n    },\n    [requestAccess, audioConstraints, resetPeak]\n  );\n\n  return {\n    // Recording state\n    isRecording,\n    isPaused,\n    duration,\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: hookError || micError || meterError || recError,\n\n    // Microphone state\n    stream,\n    devices,\n    hasPermission,\n    selectedDevice,\n\n    // Recording controls\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    requestMicAccess,\n    changeDevice,\n\n    // Track state\n    recordingPeaks: peaks,\n  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+{"version":3,"sources":["../src/index.ts","../src/hooks/useRecording.ts","../src/hooks/useMicrophoneAccess.ts","../src/hooks/useMicrophoneLevel.ts","../src/hooks/useIntegratedRecording.ts"],"sourcesContent":["/**\n * @waveform-playlist/recording\n *\n * Audio recording support using AudioWorklet for waveform-playlist\n */\n\n// Hooks\nexport {\n  useRecording,\n  useMicrophoneAccess,\n  useMicrophoneLevel,\n  useIntegratedRecording,\n} from './hooks';\nexport type {\n  UseMicrophoneLevelOptions,\n  UseMicrophoneLevelReturn,\n  UseIntegratedRecordingReturn,\n  IntegratedRecordingOptions,\n} from './hooks';\n\n// Types\nexport type {\n  RecordingState,\n  RecordingData,\n  MicrophoneDevice,\n  RecordingOptions,\n  UseRecordingReturn,\n  UseMicrophoneAccessReturn,\n} from './types';\n","/**\n * Main recording hook using AudioWorklet\n */\n\nimport { useState, useRef, useCallback, useEffect } from 'react';\nimport { UseRecordingReturn, RecordingOptions } from '../types';\nimport { concatenateAudioData, createAudioBuffer, appendPeaks } from '@waveform-playlist/core';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { recordingProcessorUrl } from '@waveform-playlist/worklets';\n\nfunction emptyPeaks(bits: 8 | 16): Int8Array | Int16Array {\n  return bits === 8 ? new Int8Array(0) : new Int16Array(0);\n}\n\nexport function useRecording(\n  stream: MediaStream | null,\n  options: RecordingOptions = {}\n): UseRecordingReturn {\n  const { channelCount = 1, samplesPerPixel = 1024, bits = 16 } = options;\n\n  // State\n  const [isRecording, setIsRecording] = useState(false);\n  const [isPaused, setIsPaused] = useState(false);\n  const [duration, setDuration] = useState(0);\n  // Per-channel peaks for multi-channel live preview\n  const [peaks, setPeaks] = useState<(Int8Array | Int16Array)[]>([emptyPeaks(bits)]);\n  const [audioBuffer, setAudioBuffer] = useState<AudioBuffer | null>(null);\n  const [error, setError] = useState<Error | null>(null);\n  const [level, setLevel] = useState(0); // Current RMS level (0-1)\n  const [peakLevel, setPeakLevel] = useState(0); // Peak level since recording started (0-1)\n\n  // Per-instance flag to prevent loading worklet multiple times within the same hook instance.\n  // Note: Multiple hook instances each have their own ref — see \"Multi-Instance Worklet\n  // Registration Gap\" in recording/CLAUDE.md for the known limitation and planned fix.\n  const workletLoadedRef = useRef<boolean>(false);\n\n  // Refs for AudioWorklet and data accumulation\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const mediaStreamSourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  // Per-channel sample accumulation: recordedChunksRef[channelIndex] = Float32Array[]\n  const recordedChunksRef = useRef<Float32Array[][]>([]);\n  const totalSamplesRef = useRef(0);\n  const animationFrameRef = useRef<number | null>(null);\n  const startTimeRef = useRef<number>(0);\n  const isRecordingRef = useRef<boolean>(false);\n  const isPausedRef = useRef<boolean>(false);\n  const audioTrackRef = useRef<MediaStreamTrack | null>(null);\n  const onTrackEndedRef = useRef<(() => void) | null>(null);\n  const stopRecordingRef = useRef<(() => Promise<AudioBuffer | null>) | null>(null);\n\n  // Shared duration update loop — starts a rAF loop that updates duration\n  // from performance.now(). Used by both startRecording and resumeRecording.\n  const startDurationLoop = useCallback(() => {\n    const tick = () => {\n      if (isRecordingRef.current && !isPausedRef.current) {\n        const elapsed = (performance.now() - startTimeRef.current) / 1000;\n        setDuration(elapsed);\n        animationFrameRef.current = requestAnimationFrame(tick);\n      }\n    };\n    tick();\n  }, []);\n\n  // Load AudioWorklet module\n  const loadWorklet = useCallback(async () => {\n    // Skip if already loaded to prevent \"already registered\" error\n    if (workletLoadedRef.current) {\n      return;\n    }\n\n    try {\n      const context = getGlobalContext();\n      // Load the worklet module directly on the raw AudioContext.\n      // Tone.js's addAudioWorkletModule only loads ONE module per context\n      // (caches _workletPromise). If meter-processor was loaded first by\n      // useMicrophoneLevel, recording-processor is silently skipped.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(recordingProcessorUrl);\n      workletLoadedRef.current = true;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to load AudioWorklet module:', String(err));\n      const error = new Error('Failed to load recording processor: ' + String(err));\n      throw error;\n    }\n  }, []);\n\n  // Start recording\n  const startRecording = useCallback(async () => {\n    if (isRecordingRef.current) return;\n    if (!stream) {\n      setError(new Error('No microphone stream available'));\n      return;\n    }\n\n    try {\n      setError(null);\n\n      // Use Tone.js Context for cross-browser compatibility\n      const context = getGlobalContext();\n\n      // Resume AudioContext if suspended\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n\n      // Load worklet module\n      await loadWorklet();\n\n      // Create MediaStreamSource from Tone's context\n      // Each hook creates its own source to avoid cross-context issues in Firefox\n      const source = context.createMediaStreamSource(stream);\n      mediaStreamSourceRef.current = source;\n\n      // Use the stream track's actual channel count from getSettings().\n      // source.channelCount defaults to 2 per Web Audio spec (not the mic's\n      // real count). Fall back to user-provided channelCount if unavailable.\n      const detectedChannelCount = stream.getAudioTracks()[0]?.getSettings().channelCount;\n      if (detectedChannelCount === undefined) {\n        console.warn(\n          `[waveform-playlist] Could not detect stream channel count, using fallback: ${channelCount}`\n        );\n      }\n      const streamChannelCount = detectedChannelCount ?? channelCount;\n\n      const workletNode = context.createAudioWorkletNode('recording-processor', {\n        channelCount: streamChannelCount,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Recording worklet processor error:', String(event));\n        setError(new Error('Recording processor encountered an error'));\n      };\n\n      // Reset state before connecting — prevents race where a worklet message\n      // arrives before refs are cleared, corrupting samplesProcessedBefore calculations\n      recordedChunksRef.current = Array.from({ length: streamChannelCount }, () => []);\n      totalSamplesRef.current = 0;\n      setPeaks(Array.from({ length: streamChannelCount }, () => emptyPeaks(bits)));\n      setAudioBuffer(null);\n      setLevel(0);\n      setPeakLevel(0);\n\n      // Listen for audio data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        const { channels } = event.data as { channels: Float32Array[] };\n\n        if (!channels || channels.length === 0) {\n          console.warn('[waveform-playlist] Recording worklet sent empty or missing channels data');\n          return;\n        }\n\n        // Accumulate per-channel samples\n        for (let ch = 0; ch < channels.length; ch++) {\n          if (!recordedChunksRef.current[ch]) {\n            console.warn(\n              `[waveform-playlist] Unexpected channel ${ch} from worklet (expected ${recordedChunksRef.current.length})`\n            );\n            recordedChunksRef.current[ch] = [];\n          }\n          recordedChunksRef.current[ch].push(channels[ch]);\n        }\n        // Capture sample offset before incrementing — used by peak alignment\n        const samplesProcessedBefore = totalSamplesRef.current;\n        totalSamplesRef.current += channels[0].length;\n        setPeaks((prevPeaks) => {\n          // Ensure we have an entry per channel\n          const updated: (Int8Array | Int16Array)[] = [];\n          for (let ch = 0; ch < channels.length; ch++) {\n            const prev = prevPeaks[ch] ?? emptyPeaks(bits);\n            updated.push(\n              appendPeaks(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)\n            );\n          }\n          return updated;\n        });\n\n        // Note: VU meter levels come from useMicrophoneLevel (meter-processor worklet)\n        // We don't update level/peakLevel here to avoid conflicting state updates\n      };\n\n      // Connect and start — after state reset and handler setup\n      source.connect(workletNode);\n      // Do NOT send sampleRate — worklet uses its global sampleRate (always correct)\n      workletNode.port.postMessage({ command: 'start', channelCount: streamChannelCount });\n\n      // Listen on MediaStreamTrack for mic unplug (MediaStream has no 'ended' event)\n      const audioTrack = stream.getAudioTracks()[0] ?? null;\n      audioTrackRef.current = audioTrack;\n      if (audioTrack) {\n        const onEnded = () => {\n          console.warn('[waveform-playlist] Audio track ended (mic unplugged or revoked)');\n          // Stop recording to clean up worklet, rAF loop, and UI state\n          stopRecordingRef.current?.();\n          setError(new Error('Microphone disconnected during recording'));\n        };\n        onTrackEndedRef.current = onEnded;\n        audioTrack.addEventListener('ended', onEnded);\n      }\n\n      isRecordingRef.current = true;\n      isPausedRef.current = false;\n      setIsRecording(true);\n      setIsPaused(false);\n      startTimeRef.current = performance.now();\n      startDurationLoop();\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to start recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to start recording'));\n    }\n  }, [stream, channelCount, samplesPerPixel, bits, loadWorklet, startDurationLoop]);\n\n  // Stop recording\n  const stopRecording = useCallback(async (): Promise<AudioBuffer | null> => {\n    if (!isRecordingRef.current) {\n      return null;\n    }\n\n    try {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n        audioTrackRef.current = null;\n        onTrackEndedRef.current = null;\n      }\n\n      // Stop the worklet\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n          }\n        }\n        workletNodeRef.current.disconnect();\n      }\n\n      // Cancel animation frame\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n\n      // Create final AudioBuffer from accumulated per-channel chunks\n      const context = getGlobalContext();\n      const rawContext = context.rawContext as AudioContext;\n      const numChannels = recordedChunksRef.current.length || channelCount;\n      const channelData = recordedChunksRef.current.map((chunks) => concatenateAudioData(chunks));\n      const totalSamples = channelData[0]?.length ?? 0;\n\n      // Guard: if no samples were captured (e.g., stop called immediately after start),\n      // return null instead of creating a 0-length AudioBuffer which throws\n      if (totalSamples === 0) {\n        console.warn('[waveform-playlist] Recording stopped with 0 samples captured — discarding');\n        isRecordingRef.current = false;\n        isPausedRef.current = false;\n        setIsRecording(false);\n        setIsPaused(false);\n        setLevel(0);\n        return null;\n      }\n\n      const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n      setAudioBuffer(buffer);\n      setDuration(buffer.duration);\n      isRecordingRef.current = false;\n      isPausedRef.current = false;\n      setIsRecording(false);\n      setIsPaused(false);\n      setLevel(0);\n      // Keep peakLevel to show the peak reached during recording\n\n      return buffer;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to stop recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to stop recording'));\n      return null;\n    }\n  }, [channelCount]);\n  stopRecordingRef.current = stopRecording;\n\n  // Pause recording\n  const pauseRecording = useCallback(() => {\n    if (isRecording && !isPaused) {\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n      isPausedRef.current = true;\n      setIsPaused(true);\n    }\n  }, [isRecording, isPaused]);\n\n  // Resume recording\n  const resumeRecording = useCallback(() => {\n    if (isRecording && isPaused) {\n      isPausedRef.current = false;\n      setIsPaused(false);\n      startTimeRef.current = performance.now() - duration * 1000;\n      startDurationLoop();\n    }\n  }, [isRecording, isPaused, duration, startDurationLoop]);\n\n  // Cleanup on unmount — read refs in cleanup, not effect body.\n  // Pattern #16 (copy refs in effect body) doesn't apply to [] deps — refs are\n  // null at mount and only set later during startRecording.\n  useEffect(() => {\n    return () => {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n      }\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during cleanup:', String(err));\n          }\n        }\n        try {\n          workletNodeRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Worklet disconnect during cleanup:', String(err));\n        }\n      }\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n      }\n      // Don't close the global AudioContext - it's shared across the app\n    };\n  }, []);\n\n  return {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer,\n    level,\n    peakLevel,\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    error,\n  };\n}\n","/**\n * Hook for managing microphone access and device enumeration\n */\n\nimport { useState, useEffect, useCallback } from 'react';\nimport { UseMicrophoneAccessReturn, MicrophoneDevice } from '../types';\n\nexport function useMicrophoneAccess(): UseMicrophoneAccessReturn {\n  const [stream, setStream] = useState<MediaStream | null>(null);\n  const [devices, setDevices] = useState<MicrophoneDevice[]>([]);\n  const [hasPermission, setHasPermission] = useState(false);\n  const [isLoading, setIsLoading] = useState(false);\n  const [error, setError] = useState<Error | null>(null);\n\n  // Enumerate audio input devices\n  const enumerateDevices = useCallback(async () => {\n    try {\n      const allDevices = await navigator.mediaDevices.enumerateDevices();\n      const audioInputs = allDevices\n        .filter((device) => device.kind === 'audioinput')\n        .map((device) => ({\n          deviceId: device.deviceId,\n          label: device.label || `Microphone ${device.deviceId.slice(0, 8)}`,\n          groupId: device.groupId,\n        }));\n\n      setDevices(audioInputs);\n    } catch (err) {\n      console.error('Failed to enumerate devices:', err);\n      setError(err instanceof Error ? err : new Error('Failed to enumerate devices'));\n    }\n  }, []);\n\n  // Request microphone access\n  const requestAccess = useCallback(\n    async (deviceId?: string, audioConstraints?: MediaTrackConstraints) => {\n      setIsLoading(true);\n      setError(null);\n\n      try {\n        // Stop existing stream if any\n        if (stream) {\n          stream.getTracks().forEach((track) => track.stop());\n        }\n\n        // Build audio constraints\n        const audio: MediaTrackConstraints & { latency?: number } = {\n          // Recording-optimized defaults: prioritize raw audio quality and low latency\n          echoCancellation: false,\n          noiseSuppression: false,\n          autoGainControl: false,\n          latency: 0, // Low latency mode (not in TS types yet, but supported in modern browsers)\n          // User-provided constraints override defaults\n          ...audioConstraints,\n          // Device ID override (if specified)\n          ...(deviceId && { deviceId: { exact: deviceId } }),\n        };\n\n        const constraints: MediaStreamConstraints = {\n          audio,\n          video: false,\n        };\n\n        const newStream = await navigator.mediaDevices.getUserMedia(constraints);\n        setStream(newStream);\n        setHasPermission(true);\n\n        // Enumerate devices after getting permission (labels will be available)\n        await enumerateDevices();\n      } catch (err) {\n        console.error('Failed to access microphone:', err);\n        setError(err instanceof Error ? err : new Error('Failed to access microphone'));\n        setHasPermission(false);\n      } finally {\n        setIsLoading(false);\n      }\n    },\n    [stream, enumerateDevices]\n  );\n\n  // Stop the stream and revoke access\n  const stopStream = useCallback(() => {\n    if (stream) {\n      stream.getTracks().forEach((track) => track.stop());\n      setStream(null);\n      setHasPermission(false);\n    }\n  }, [stream]);\n\n  // Check initial permission state, enumerate devices, and listen for hot-plug changes\n  useEffect(() => {\n    // Try to enumerate devices (labels won't be available without permission)\n    enumerateDevices();\n\n    // Re-enumerate when devices are plugged in or removed\n    navigator.mediaDevices.addEventListener('devicechange', enumerateDevices);\n\n    // Cleanup on unmount\n    return () => {\n      navigator.mediaDevices.removeEventListener('devicechange', enumerateDevices);\n      if (stream) {\n        stream.getTracks().forEach((track) => track.stop());\n      }\n    };\n  }, [enumerateDevices, stream]);\n\n  return {\n    stream,\n    devices,\n    hasPermission,\n    isLoading,\n    requestAccess,\n    stopStream,\n    error,\n  };\n}\n","/**\n * Hook for monitoring microphone input levels\n *\n * Uses an AudioWorklet-based meter processor for sample-accurate\n * peak and RMS metering without requestAnimationFrame overhead.\n */\n\nimport { useEffect, useState, useRef, useCallback } from 'react';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { gainToNormalized } from '@waveform-playlist/core';\nimport { meterProcessorUrl, type MeterMessage } from '@waveform-playlist/worklets';\n\n/** Peak decay constant — exponential decay for smooth peak hold (~800ms to 1/e at 60fps) */\nconst PEAK_DECAY = 0.98;\n\nexport interface UseMicrophoneLevelOptions {\n  /**\n   * How often to update the level (in Hz)\n   * Default: 60 (60fps)\n   */\n  updateRate?: number;\n\n  /**\n   * Number of channels to meter (1 = mono, 2 = stereo)\n   * Default: 1\n   */\n  channelCount?: number;\n}\n\nexport interface UseMicrophoneLevelReturn {\n  /**\n   * Current peak audio level (0-1)\n   * For single channel: channel 0 level\n   * For multi-channel: max across all channels\n   */\n  level: number;\n\n  /**\n   * Held peak level since last reset (0-1)\n   * For single channel: channel 0 peak\n   * For multi-channel: max across all channels\n   */\n  peakLevel: number;\n\n  /**\n   * Reset the held peak level\n   */\n  resetPeak: () => void;\n\n  /**\n   * Per-channel peak levels (0-1). Array length matches channelCount.\n   * True peak: max absolute sample value per analysis frame.\n   */\n  levels: number[];\n\n  /**\n   * Per-channel held peak levels (0-1). Array length matches channelCount.\n   */\n  peakLevels: number[];\n\n  /**\n   * Per-channel RMS levels (0-1). Array length matches channelCount.\n   * RMS: root mean square of samples per analysis frame.\n   */\n  rmsLevels: number[];\n\n  /**\n   * Error from meter setup (worklet load failure, context issues, etc.)\n   * Null when metering is working normally.\n   */\n  error: Error | null;\n}\n\n/**\n * Monitor microphone input levels in real-time\n *\n * @param stream - MediaStream from getUserMedia\n * @param options - Configuration options\n * @returns Object with current peak level, RMS level, and held peak level\n *\n * @example\n * ```typescript\n * const { stream } = useMicrophoneAccess();\n * const { levels, rmsLevels, peakLevels } = useMicrophoneLevel(stream, { channelCount: 2 });\n *\n * return <SegmentedVUMeter levels={levels} peakLevels={peakLevels} />;\n * ```\n */\nexport function useMicrophoneLevel(\n  stream: MediaStream | null,\n  options: UseMicrophoneLevelOptions = {}\n): UseMicrophoneLevelReturn {\n  const { updateRate = 60, channelCount = 1 } = options;\n\n  const [levels, setLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [peakLevels, setPeakLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [rmsLevels, setRmsLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [meterError, setMeterError] = useState<Error | null>(null);\n\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  const smoothedPeakRef = useRef<number[]>(new Array(channelCount).fill(0));\n\n  const resetPeak = useCallback(\n    () => setPeakLevels(new Array(channelCount).fill(0)),\n    [channelCount]\n  );\n\n  useEffect(() => {\n    if (!stream) {\n      setLevels(new Array(channelCount).fill(0));\n      setPeakLevels(new Array(channelCount).fill(0));\n      setRmsLevels(new Array(channelCount).fill(0));\n      smoothedPeakRef.current = new Array(channelCount).fill(0);\n      return;\n    }\n\n    let isMounted = true;\n\n    const setupMonitoring = async () => {\n      if (!isMounted) return;\n\n      const context = getGlobalContext();\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n      if (!isMounted) return;\n\n      // Auto-detect actual mic channel count from stream\n      const trackSettings = stream.getAudioTracks()[0]?.getSettings();\n      const actualChannels = trackSettings?.channelCount ?? channelCount;\n\n      // Load worklet directly on rawContext — Tone.js's addAudioWorkletModule\n      // only loads ONE module per context (caches _workletPromise), silently\n      // skipping subsequent calls with different URLs.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(meterProcessorUrl);\n      if (!isMounted) return;\n\n      // Use Tone.js's createAudioWorkletNode — avoids rawContext identity issues\n      // in webpack-aliased environments (Docusaurus)\n      const workletNode = context.createAudioWorkletNode('meter-processor', {\n        channelCount: actualChannels,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n        processorOptions: {\n          numberOfChannels: actualChannels,\n          updateRate,\n        },\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Mic meter worklet processor error:', String(event));\n      };\n\n      // Create source and connect: source → meter\n      // Don't connect output to destination — mic monitoring would cause feedback\n      const source = context.createMediaStreamSource(stream);\n      sourceRef.current = source;\n      source.connect(workletNode);\n\n      smoothedPeakRef.current = new Array(actualChannels).fill(0);\n\n      // Listen for meter data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        if (!isMounted) return;\n\n        const { peak, rms } = event.data as MeterMessage;\n        const smoothed = smoothedPeakRef.current;\n\n        const peakValues: number[] = [];\n        const rmsValues: number[] = [];\n\n        for (let ch = 0; ch < peak.length; ch++) {\n          smoothed[ch] = Math.max(peak[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);\n          peakValues.push(gainToNormalized(smoothed[ch]));\n          rmsValues.push(gainToNormalized(rms[ch]));\n        }\n\n        // Mirror mono to fill requested channelCount\n        const mirroredPeaks =\n          peak.length < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;\n        const mirroredRms =\n          peak.length < channelCount ? new Array(channelCount).fill(rmsValues[0]) : rmsValues;\n\n        setLevels(mirroredPeaks);\n        setRmsLevels(mirroredRms);\n        setPeakLevels((prev) => mirroredPeaks.map((val, i) => Math.max(prev[i] ?? 0, val)));\n      };\n    };\n\n    setupMonitoring().catch((err) => {\n      console.warn('[waveform-playlist] Failed to set up mic level monitoring:', String(err));\n      if (isMounted) {\n        setMeterError(err instanceof Error ? err : new Error(String(err)));\n      }\n    });\n\n    return () => {\n      isMounted = false;\n\n      if (sourceRef.current) {\n        try {\n          sourceRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic source disconnect during cleanup:', String(err));\n        }\n        sourceRef.current = null;\n      }\n\n      if (workletNodeRef.current) {\n        try {\n          workletNodeRef.current.disconnect();\n          workletNodeRef.current.port.close();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic meter disconnect during cleanup:', String(err));\n        }\n        workletNodeRef.current = null;\n      }\n    };\n  }, [stream, updateRate, channelCount]);\n\n  // Backwards-compatible scalar values\n  const level = channelCount === 1 ? (levels[0] ?? 0) : Math.max(...levels);\n  const peakLevel = channelCount === 1 ? (peakLevels[0] ?? 0) : Math.max(...peakLevels);\n\n  return {\n    level,\n    peakLevel,\n    resetPeak,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: meterError,\n  };\n}\n","/**\n * Hook for integrated multi-track recording\n * Combines recording functionality with track management\n */\n\nimport { useState, useCallback, useEffect, useRef } from 'react';\nimport { useRecording } from './useRecording';\nimport { useMicrophoneAccess } from './useMicrophoneAccess';\nimport { useMicrophoneLevel } from './useMicrophoneLevel';\nimport type { MicrophoneDevice } from '../types';\nimport { type ClipTrack, type AudioClip } from '@waveform-playlist/core';\nimport {\n  resumeGlobalAudioContext,\n  getGlobalAudioContext,\n  getGlobalContext,\n} from '@waveform-playlist/playout';\n\nexport interface IntegratedRecordingOptions {\n  /**\n   * Current playback/cursor position in seconds\n   * Recording will start from max(currentTime, lastClipEndTime)\n   */\n  currentTime?: number;\n\n  /**\n   * MediaTrackConstraints for audio recording\n   * These will override the recording-optimized defaults (echo cancellation off, low latency)\n   */\n  audioConstraints?: MediaTrackConstraints;\n\n  /**\n   * Number of channels to record (1 = mono, 2 = stereo)\n   * Default: 1 (mono)\n   */\n  channelCount?: number;\n\n  /**\n   * Samples per pixel for peak generation\n   * Default: 1024\n   */\n  samplesPerPixel?: number;\n}\n\nexport interface UseIntegratedRecordingReturn {\n  // Recording state\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number;\n  level: number;\n  peakLevel: number;\n  /** Per-channel peak levels (0-1). Array length matches channelCount. */\n  levels: number[];\n  /** Per-channel held peak levels (0-1). Array length matches channelCount. */\n  peakLevels: number[];\n  /** Per-channel RMS levels (0-1). Array length matches channelCount. */\n  rmsLevels: number[];\n  error: Error | null;\n\n  // Microphone state\n  stream: MediaStream | null;\n  devices: MicrophoneDevice[];\n  hasPermission: boolean;\n  selectedDevice: string | null;\n\n  // Recording controls\n  startRecording: () => void;\n  stopRecording: () => void;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  requestMicAccess: () => Promise<void>;\n  changeDevice: (deviceId: string) => Promise<void>;\n\n  // Track state (for live waveform during recording)\n  recordingPeaks: (Int8Array | Int16Array)[];\n}\n\nexport function useIntegratedRecording(\n  tracks: ClipTrack[],\n  setTracks: (tracks: ClipTrack[]) => void,\n  selectedTrackId: string | null,\n  options: IntegratedRecordingOptions = {}\n): UseIntegratedRecordingReturn {\n  const { currentTime = 0, audioConstraints, ...recordingOptions } = options;\n\n  // Track if we're currently monitoring (for auto-resume audio context)\n  const [isMonitoring, setIsMonitoring] = useState(false);\n  const [selectedDevice, setSelectedDevice] = useState<string | null>(null);\n  const [hookError, setHookError] = useState<Error | null>(null);\n\n  // Capture timeline position when recording starts (not at stop time)\n  const recordingStartTimeRef = useRef(0);\n\n  // Keep selectedTrackId and currentTime in refs for use in callbacks.\n  // Avoids stale closures and prevents 60fps useCallback recreation\n  // (currentTime updates at animation-frame rate during playback).\n  const selectedTrackIdRef = useRef(selectedTrackId);\n  selectedTrackIdRef.current = selectedTrackId;\n  const currentTimeRef = useRef(currentTime);\n  currentTimeRef.current = currentTime;\n\n  // Microphone access\n  const { stream, devices, hasPermission, requestAccess, error: micError } = useMicrophoneAccess();\n\n  // Microphone level (for VU meter)\n  const {\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    resetPeak,\n    error: meterError,\n  } = useMicrophoneLevel(stream, {\n    channelCount: recordingOptions.channelCount,\n  });\n\n  // Recording\n  const {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer: _recordedAudioBuffer,\n    startRecording: startRec,\n    stopRecording: stopRec,\n    pauseRecording,\n    resumeRecording,\n    error: recError,\n  } = useRecording(stream, recordingOptions);\n\n  // Start recording handler\n  // Reads selectedTrackId from ref to avoid stale closures when\n  // auto-create track + start recording happen in the same render cycle\n  const startRecording = useCallback(async () => {\n    if (!selectedTrackIdRef.current) {\n      setHookError(\n        new Error('Cannot start recording: no track selected. Select or create a track first.')\n      );\n      return;\n    }\n\n    try {\n      setHookError(null);\n      // Resume audio context if needed\n      if (!isMonitoring) {\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      }\n\n      // Capture timeline position NOW — before recording starts.\n      // Using currentTime at stop time would be wrong during overdub\n      // (playback advances currentTime while recording).\n      recordingStartTimeRef.current = currentTimeRef.current;\n\n      await startRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [isMonitoring, startRec]);\n\n  // Stop recording and add clip to selected track\n  const stopRecording = useCallback(async () => {\n    let buffer: AudioBuffer | null;\n    try {\n      buffer = await stopRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n      return;\n    }\n\n    // Add clip to track after recording completes\n    const trackId = selectedTrackIdRef.current;\n    if (buffer && trackId) {\n      const selectedTrackIndex = tracks.findIndex((t) => t.id === trackId);\n      if (selectedTrackIndex === -1) {\n        const err = new Error(\n          `Recording completed but track \"${trackId}\" no longer exists. The recorded audio could not be saved.`\n        );\n        console.warn(`[waveform-playlist] ${err.message}`);\n        setHookError(err);\n        return;\n      }\n\n      const selectedTrack = tracks[selectedTrackIndex];\n\n      // Use the captured start time (not live currentTime which advances during overdub)\n      const recordStartTimeSamples = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n      let lastClipEndSample = 0;\n      if (selectedTrack.clips.length > 0) {\n        const endSamples = selectedTrack.clips.map(\n          (clip) => clip.startSample + clip.durationSamples\n        );\n        lastClipEndSample = Math.max(...endSamples);\n      }\n\n      const startSample = Math.max(recordStartTimeSamples, lastClipEndSample);\n\n      // Latency compensation:\n      // Two sources of delay between recording start and audible playback:\n      // 1. Tone.js lookAhead (~100ms) — Transport schedules audio ahead of real time\n      // 2. Output latency — hardware DAC delay before audio reaches speakers\n      // The user hears playback delayed by both, so they perform late relative\n      // to the timeline. Skip that duration at the start of the recorded audio.\n      const audioContext = getGlobalAudioContext();\n      const outputLatency = audioContext.outputLatency ?? 0;\n      const toneContext = getGlobalContext();\n      const lookAhead = toneContext.lookAhead ?? 0;\n      const totalLatency = outputLatency + lookAhead;\n      const latencyOffsetSamples = Math.floor(totalLatency * buffer.sampleRate);\n\n      // Guard: very short recordings (< latency compensation) would produce negative duration\n      const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n      if (effectiveDuration === 0) {\n        console.warn(\n          '[waveform-playlist] Recording too short for latency compensation — discarding'\n        );\n        setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n        return;\n      }\n\n      // Create new clip from recording\n      const newClip: AudioClip = {\n        id: `clip-${Date.now()}`,\n        audioBuffer: buffer,\n        startSample,\n        durationSamples: effectiveDuration,\n        offsetSamples: latencyOffsetSamples,\n        sampleRate: buffer.sampleRate,\n        sourceDurationSamples: buffer.length,\n        gain: 1.0,\n        name: `Recording ${new Date().toLocaleTimeString()}`,\n      };\n\n      // Add clip to track\n      const newTracks = tracks.map((track, index) => {\n        if (index === selectedTrackIndex) {\n          return {\n            ...track,\n            clips: [...track.clips, newClip],\n          };\n        }\n        return track;\n      });\n\n      setTracks(newTracks);\n    }\n  }, [tracks, setTracks, stopRec]);\n\n  // Auto-select first device when available, or fallback if selected device was unplugged\n  useEffect(() => {\n    if (!hasPermission || devices.length === 0) return;\n\n    if (selectedDevice === null) {\n      // First-time selection\n      setSelectedDevice(devices[0].deviceId);\n    } else if (!devices.some((d) => d.deviceId === selectedDevice)) {\n      // Selected device was removed — fall back to first available\n      const fallbackId = devices[0].deviceId;\n      setSelectedDevice(fallbackId);\n      resetPeak();\n      requestAccess(fallbackId, audioConstraints);\n    }\n  }, [hasPermission, devices, selectedDevice, resetPeak, requestAccess, audioConstraints]);\n\n  // Request microphone access\n  const requestMicAccess = useCallback(async () => {\n    try {\n      setHookError(null);\n      await requestAccess(undefined, audioConstraints);\n      await resumeGlobalAudioContext();\n      setIsMonitoring(true);\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [requestAccess, audioConstraints]);\n\n  // Change device\n  const changeDevice = useCallback(\n    async (deviceId: string) => {\n      try {\n        setHookError(null);\n        setSelectedDevice(deviceId);\n        resetPeak();\n        await requestAccess(deviceId, audioConstraints);\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      } catch (err) {\n        setHookError(err instanceof Error ? err : new Error(String(err)));\n      }\n    },\n    [requestAccess, audioConstraints, resetPeak]\n  );\n\n  return {\n    // Recording state\n    isRecording,\n    isPaused,\n    duration,\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: hookError || micError || meterError || recError,\n\n    // Microphone state\n    stream,\n    devices,\n    hasPermission,\n    selectedDevice,\n\n    // Recording controls\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    requestMicAccess,\n    changeDevice,\n\n    // Track state\n    recordingPeaks: peaks,\n  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package/dist/index.mjs

@@ -1,80 +1,6 @@
 // src/hooks/useRecording.ts
 import { useState, useRef, useCallback, useEffect } from "react";
-
-// src/utils/audioBufferUtils.ts
-function concatenateAudioData(chunks) {
-  const totalLength = chunks.reduce((sum, chunk) => sum + chunk.length, 0);
-  const result = new Float32Array(totalLength);
-  let offset = 0;
-  for (const chunk of chunks) {
-    result.set(chunk, offset);
-    offset += chunk.length;
-  }
-  return result;
-}
-function createAudioBuffer(audioContext, channelData, sampleRate, channelCount = 1) {
-  const channels = channelData instanceof Float32Array ? [channelData] : channelData;
-  const length = channels[0]?.length ?? 0;
-  const buffer = audioContext.createBuffer(channelCount, length, sampleRate);
-  for (let ch = 0; ch < Math.min(channelCount, channels.length); ch++) {
-    buffer.copyToChannel(new Float32Array(channels[ch]), ch);
-  }
-  return buffer;
-}
-
-// src/utils/peaksGenerator.ts
-function generatePeaks(samples, samplesPerPixel, bits = 16) {
-  const numPeaks = Math.ceil(samples.length / samplesPerPixel);
-  const peakArray = bits === 8 ? new Int8Array(numPeaks * 2) : new Int16Array(numPeaks * 2);
-  const maxValue = 2 ** (bits - 1);
-  for (let i = 0; i < numPeaks; i++) {
-    const start = i * samplesPerPixel;
-    const end = Math.min(start + samplesPerPixel, samples.length);
-    let min = 0;
-    let max = 0;
-    for (let j = start; j < end; j++) {
-      const value = samples[j];
-      if (value < min) min = value;
-      if (value > max) max = value;
-    }
-    peakArray[i * 2] = Math.max(-maxValue, Math.floor(min * maxValue));
-    peakArray[i * 2 + 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));
-  }
-  return peakArray;
-}
-function appendPeaks(existingPeaks, newSamples, samplesPerPixel, totalSamplesProcessed, bits = 16) {
-  const maxValue = 2 ** (bits - 1);
-  const remainder = totalSamplesProcessed % samplesPerPixel;
-  let offset = 0;
-  if (remainder > 0 && existingPeaks.length > 0) {
-    const samplesToComplete = samplesPerPixel - remainder;
-    const endIndex = Math.min(samplesToComplete, newSamples.length);
-    let min = existingPeaks[existingPeaks.length - 2] / maxValue;
-    let max = existingPeaks[existingPeaks.length - 1] / maxValue;
-    for (let i = 0; i < endIndex; i++) {
-      const value = newSamples[i];
-      if (value < min) min = value;
-      if (value > max) max = value;
-    }
-    const updated = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length);
-    updated.set(existingPeaks);
-    updated[existingPeaks.length - 2] = Math.max(-maxValue, Math.floor(min * maxValue));
-    updated[existingPeaks.length - 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));
-    offset = endIndex;
-    const newPeaks2 = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);
-    const result2 = new (bits === 8 ? Int8Array : Int16Array)(updated.length + newPeaks2.length);
-    result2.set(updated);
-    result2.set(newPeaks2, updated.length);
-    return result2;
-  }
-  const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);
-  const result = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length + newPeaks.length);
-  result.set(existingPeaks);
-  result.set(newPeaks, existingPeaks.length);
-  return result;
-}
-
-// src/hooks/useRecording.ts
+import { concatenateAudioData, createAudioBuffer, appendPeaks } from "@waveform-playlist/core";
 import { getGlobalContext } from "@waveform-playlist/playout";
 import { recordingProcessorUrl } from "@waveform-playlist/worklets";
 function emptyPeaks(bits) {
@@ -722,10 +648,6 @@ function useIntegratedRecording(tracks, setTracks, selectedTrackId, options = {}
   };
 }
 export {
-  appendPeaks,
-  concatenateAudioData,
-  createAudioBuffer,
-  generatePeaks,
   useIntegratedRecording,
   useMicrophoneAccess,
   useMicrophoneLevel,

package/dist/index.mjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/hooks/useRecording.ts","../src/utils/audioBufferUtils.ts","../src/utils/peaksGenerator.ts","../src/hooks/useMicrophoneAccess.ts","../src/hooks/useMicrophoneLevel.ts","../src/hooks/useIntegratedRecording.ts"],"sourcesContent":["/**\n * Main recording hook using AudioWorklet\n */\n\nimport { useState, useRef, useCallback, useEffect } from 'react';\nimport { UseRecordingReturn, RecordingOptions } from '../types';\nimport { concatenateAudioData, createAudioBuffer } from '../utils/audioBufferUtils';\nimport { appendPeaks } from '../utils/peaksGenerator';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { recordingProcessorUrl } from '@waveform-playlist/worklets';\n\nfunction emptyPeaks(bits: 8 | 16): Int8Array | Int16Array {\n  return bits === 8 ? new Int8Array(0) : new Int16Array(0);\n}\n\nexport function useRecording(\n  stream: MediaStream | null,\n  options: RecordingOptions = {}\n): UseRecordingReturn {\n  const { channelCount = 1, samplesPerPixel = 1024, bits = 16 } = options;\n\n  // State\n  const [isRecording, setIsRecording] = useState(false);\n  const [isPaused, setIsPaused] = useState(false);\n  const [duration, setDuration] = useState(0);\n  // Per-channel peaks for multi-channel live preview\n  const [peaks, setPeaks] = useState<(Int8Array | Int16Array)[]>([emptyPeaks(bits)]);\n  const [audioBuffer, setAudioBuffer] = useState<AudioBuffer | null>(null);\n  const [error, setError] = useState<Error | null>(null);\n  const [level, setLevel] = useState(0); // Current RMS level (0-1)\n  const [peakLevel, setPeakLevel] = useState(0); // Peak level since recording started (0-1)\n\n  // Per-instance flag to prevent loading worklet multiple times within the same hook instance.\n  // Note: Multiple hook instances each have their own ref — see \"Multi-Instance Worklet\n  // Registration Gap\" in recording/CLAUDE.md for the known limitation and planned fix.\n  const workletLoadedRef = useRef<boolean>(false);\n\n  // Refs for AudioWorklet and data accumulation\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const mediaStreamSourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  // Per-channel sample accumulation: recordedChunksRef[channelIndex] = Float32Array[]\n  const recordedChunksRef = useRef<Float32Array[][]>([]);\n  const totalSamplesRef = useRef(0);\n  const animationFrameRef = useRef<number | null>(null);\n  const startTimeRef = useRef<number>(0);\n  const isRecordingRef = useRef<boolean>(false);\n  const isPausedRef = useRef<boolean>(false);\n  const audioTrackRef = useRef<MediaStreamTrack | null>(null);\n  const onTrackEndedRef = useRef<(() => void) | null>(null);\n  const stopRecordingRef = useRef<(() => Promise<AudioBuffer | null>) | null>(null);\n\n  // Shared duration update loop — starts a rAF loop that updates duration\n  // from performance.now(). Used by both startRecording and resumeRecording.\n  const startDurationLoop = useCallback(() => {\n    const tick = () => {\n      if (isRecordingRef.current && !isPausedRef.current) {\n        const elapsed = (performance.now() - startTimeRef.current) / 1000;\n        setDuration(elapsed);\n        animationFrameRef.current = requestAnimationFrame(tick);\n      }\n    };\n    tick();\n  }, []);\n\n  // Load AudioWorklet module\n  const loadWorklet = useCallback(async () => {\n    // Skip if already loaded to prevent \"already registered\" error\n    if (workletLoadedRef.current) {\n      return;\n    }\n\n    try {\n      const context = getGlobalContext();\n      // Load the worklet module directly on the raw AudioContext.\n      // Tone.js's addAudioWorkletModule only loads ONE module per context\n      // (caches _workletPromise). If meter-processor was loaded first by\n      // useMicrophoneLevel, recording-processor is silently skipped.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(recordingProcessorUrl);\n      workletLoadedRef.current = true;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to load AudioWorklet module:', String(err));\n      const error = new Error('Failed to load recording processor: ' + String(err));\n      throw error;\n    }\n  }, []);\n\n  // Start recording\n  const startRecording = useCallback(async () => {\n    if (isRecordingRef.current) return;\n    if (!stream) {\n      setError(new Error('No microphone stream available'));\n      return;\n    }\n\n    try {\n      setError(null);\n\n      // Use Tone.js Context for cross-browser compatibility\n      const context = getGlobalContext();\n\n      // Resume AudioContext if suspended\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n\n      // Load worklet module\n      await loadWorklet();\n\n      // Create MediaStreamSource from Tone's context\n      // Each hook creates its own source to avoid cross-context issues in Firefox\n      const source = context.createMediaStreamSource(stream);\n      mediaStreamSourceRef.current = source;\n\n      // Use the stream track's actual channel count from getSettings().\n      // source.channelCount defaults to 2 per Web Audio spec (not the mic's\n      // real count). Fall back to user-provided channelCount if unavailable.\n      const detectedChannelCount = stream.getAudioTracks()[0]?.getSettings().channelCount;\n      if (detectedChannelCount === undefined) {\n        console.warn(\n          `[waveform-playlist] Could not detect stream channel count, using fallback: ${channelCount}`\n        );\n      }\n      const streamChannelCount = detectedChannelCount ?? channelCount;\n\n      const workletNode = context.createAudioWorkletNode('recording-processor', {\n        channelCount: streamChannelCount,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Recording worklet processor error:', String(event));\n        setError(new Error('Recording processor encountered an error'));\n      };\n\n      // Reset state before connecting — prevents race where a worklet message\n      // arrives before refs are cleared, corrupting samplesProcessedBefore calculations\n      recordedChunksRef.current = Array.from({ length: streamChannelCount }, () => []);\n      totalSamplesRef.current = 0;\n      setPeaks(Array.from({ length: streamChannelCount }, () => emptyPeaks(bits)));\n      setAudioBuffer(null);\n      setLevel(0);\n      setPeakLevel(0);\n\n      // Listen for audio data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        const { channels } = event.data as { channels: Float32Array[] };\n\n        if (!channels || channels.length === 0) {\n          console.warn('[waveform-playlist] Recording worklet sent empty or missing channels data');\n          return;\n        }\n\n        // Accumulate per-channel samples\n        for (let ch = 0; ch < channels.length; ch++) {\n          if (!recordedChunksRef.current[ch]) {\n            console.warn(\n              `[waveform-playlist] Unexpected channel ${ch} from worklet (expected ${recordedChunksRef.current.length})`\n            );\n            recordedChunksRef.current[ch] = [];\n          }\n          recordedChunksRef.current[ch].push(channels[ch]);\n        }\n        // Capture sample offset before incrementing — used by peak alignment\n        const samplesProcessedBefore = totalSamplesRef.current;\n        totalSamplesRef.current += channels[0].length;\n        setPeaks((prevPeaks) => {\n          // Ensure we have an entry per channel\n          const updated: (Int8Array | Int16Array)[] = [];\n          for (let ch = 0; ch < channels.length; ch++) {\n            const prev = prevPeaks[ch] ?? emptyPeaks(bits);\n            updated.push(\n              appendPeaks(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)\n            );\n          }\n          return updated;\n        });\n\n        // Note: VU meter levels come from useMicrophoneLevel (meter-processor worklet)\n        // We don't update level/peakLevel here to avoid conflicting state updates\n      };\n\n      // Connect and start — after state reset and handler setup\n      source.connect(workletNode);\n      // Do NOT send sampleRate — worklet uses its global sampleRate (always correct)\n      workletNode.port.postMessage({ command: 'start', channelCount: streamChannelCount });\n\n      // Listen on MediaStreamTrack for mic unplug (MediaStream has no 'ended' event)\n      const audioTrack = stream.getAudioTracks()[0] ?? null;\n      audioTrackRef.current = audioTrack;\n      if (audioTrack) {\n        const onEnded = () => {\n          console.warn('[waveform-playlist] Audio track ended (mic unplugged or revoked)');\n          // Stop recording to clean up worklet, rAF loop, and UI state\n          stopRecordingRef.current?.();\n          setError(new Error('Microphone disconnected during recording'));\n        };\n        onTrackEndedRef.current = onEnded;\n        audioTrack.addEventListener('ended', onEnded);\n      }\n\n      isRecordingRef.current = true;\n      isPausedRef.current = false;\n      setIsRecording(true);\n      setIsPaused(false);\n      startTimeRef.current = performance.now();\n      startDurationLoop();\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to start recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to start recording'));\n    }\n  }, [stream, channelCount, samplesPerPixel, bits, loadWorklet, startDurationLoop]);\n\n  // Stop recording\n  const stopRecording = useCallback(async (): Promise<AudioBuffer | null> => {\n    if (!isRecordingRef.current) {\n      return null;\n    }\n\n    try {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n        audioTrackRef.current = null;\n        onTrackEndedRef.current = null;\n      }\n\n      // Stop the worklet\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n          }\n        }\n        workletNodeRef.current.disconnect();\n      }\n\n      // Cancel animation frame\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n\n      // Create final AudioBuffer from accumulated per-channel chunks\n      const context = getGlobalContext();\n      const rawContext = context.rawContext as AudioContext;\n      const numChannels = recordedChunksRef.current.length || channelCount;\n      const channelData = recordedChunksRef.current.map((chunks) => concatenateAudioData(chunks));\n      const totalSamples = channelData[0]?.length ?? 0;\n\n      // Guard: if no samples were captured (e.g., stop called immediately after start),\n      // return null instead of creating a 0-length AudioBuffer which throws\n      if (totalSamples === 0) {\n        console.warn('[waveform-playlist] Recording stopped with 0 samples captured — discarding');\n        isRecordingRef.current = false;\n        isPausedRef.current = false;\n        setIsRecording(false);\n        setIsPaused(false);\n        setLevel(0);\n        return null;\n      }\n\n      const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n      setAudioBuffer(buffer);\n      setDuration(buffer.duration);\n      isRecordingRef.current = false;\n      isPausedRef.current = false;\n      setIsRecording(false);\n      setIsPaused(false);\n      setLevel(0);\n      // Keep peakLevel to show the peak reached during recording\n\n      return buffer;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to stop recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to stop recording'));\n      return null;\n    }\n  }, [channelCount]);\n  stopRecordingRef.current = stopRecording;\n\n  // Pause recording\n  const pauseRecording = useCallback(() => {\n    if (isRecording && !isPaused) {\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n      isPausedRef.current = true;\n      setIsPaused(true);\n    }\n  }, [isRecording, isPaused]);\n\n  // Resume recording\n  const resumeRecording = useCallback(() => {\n    if (isRecording && isPaused) {\n      isPausedRef.current = false;\n      setIsPaused(false);\n      startTimeRef.current = performance.now() - duration * 1000;\n      startDurationLoop();\n    }\n  }, [isRecording, isPaused, duration, startDurationLoop]);\n\n  // Cleanup on unmount — read refs in cleanup, not effect body.\n  // Pattern #16 (copy refs in effect body) doesn't apply to [] deps — refs are\n  // null at mount and only set later during startRecording.\n  useEffect(() => {\n    return () => {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n      }\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during cleanup:', String(err));\n          }\n        }\n        try {\n          workletNodeRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Worklet disconnect during cleanup:', String(err));\n        }\n      }\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n      }\n      // Don't close the global AudioContext - it's shared across the app\n    };\n  }, []);\n\n  return {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer,\n    level,\n    peakLevel,\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    error,\n  };\n}\n","/**\n * Utility functions for working with AudioBuffers during recording\n */\n\n/**\n * Concatenate multiple Float32Arrays into a single array\n */\nexport function concatenateAudioData(chunks: Float32Array[]): Float32Array {\n  const totalLength = chunks.reduce((sum, chunk) => sum + chunk.length, 0);\n  const result = new Float32Array(totalLength);\n\n  let offset = 0;\n  for (const chunk of chunks) {\n    result.set(chunk, offset);\n    offset += chunk.length;\n  }\n\n  return result;\n}\n\n/**\n * Convert channel data to AudioBuffer.\n * Accepts either per-channel Float32Array[] or a single Float32Array (mono, backwards compatible).\n */\nexport function createAudioBuffer(\n  audioContext: AudioContext,\n  channelData: Float32Array[] | Float32Array,\n  sampleRate: number,\n  channelCount: number = 1\n): AudioBuffer {\n  // Backwards compatibility: single Float32Array → wrap as mono\n  const channels: Float32Array[] =\n    channelData instanceof Float32Array ? [channelData] : channelData;\n\n  const length = channels[0]?.length ?? 0;\n  const buffer = audioContext.createBuffer(channelCount, length, sampleRate);\n\n  for (let ch = 0; ch < Math.min(channelCount, channels.length); ch++) {\n    buffer.copyToChannel(new Float32Array(channels[ch]), ch);\n  }\n\n  return buffer;\n}\n\n/**\n * Append new samples to an existing AudioBuffer (mono convenience)\n */\nexport function appendToAudioBuffer(\n  audioContext: AudioContext,\n  existingBuffer: AudioBuffer | null,\n  newSamples: Float32Array,\n  sampleRate: number\n): AudioBuffer {\n  if (!existingBuffer) {\n    return createAudioBuffer(audioContext, [newSamples], sampleRate);\n  }\n\n  // Get existing samples\n  const existingData = existingBuffer.getChannelData(0);\n\n  // Concatenate using concatenateAudioData helper\n  const combined = concatenateAudioData([existingData, newSamples]);\n\n  // Create new buffer\n  return createAudioBuffer(audioContext, [combined], sampleRate);\n}\n\n/**\n * Calculate duration in seconds from sample count and sample rate\n */\nexport function calculateDuration(sampleCount: number, sampleRate: number): number {\n  return sampleCount / sampleRate;\n}\n","/**\n * Peak generation for real-time waveform visualization during recording\n * Matches the format used by webaudio-peaks: min/max pairs with bit depth\n */\n\n/**\n * Generate peaks from audio samples in standard min/max pair format\n *\n * @param samples - Audio samples to process\n * @param samplesPerPixel - Number of samples to represent in each peak\n * @param bits - Bit depth for peak values (8 or 16)\n * @returns Int8Array or Int16Array of peak values (min/max pairs)\n */\nexport function generatePeaks(\n  samples: Float32Array,\n  samplesPerPixel: number,\n  bits: 8 | 16 = 16\n): Int8Array | Int16Array {\n  const numPeaks = Math.ceil(samples.length / samplesPerPixel);\n  const peakArray = bits === 8 ? new Int8Array(numPeaks * 2) : new Int16Array(numPeaks * 2);\n  const maxValue = 2 ** (bits - 1);\n\n  for (let i = 0; i < numPeaks; i++) {\n    const start = i * samplesPerPixel;\n    const end = Math.min(start + samplesPerPixel, samples.length);\n\n    let min = 0;\n    let max = 0;\n\n    for (let j = start; j < end; j++) {\n      const value = samples[j];\n      if (value < min) min = value;\n      if (value > max) max = value;\n    }\n\n    // Store as min/max pairs scaled to bit depth\n    // Clamp to valid range: Int16 is [-32768, 32767], Int8 is [-128, 127]\n    peakArray[i * 2] = Math.max(-maxValue, Math.floor(min * maxValue));\n    peakArray[i * 2 + 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));\n  }\n\n  return peakArray;\n}\n\n/**\n * Append new peaks to existing peaks array\n * This is used for incremental peak updates during recording\n */\nexport function appendPeaks(\n  existingPeaks: Int8Array | Int16Array,\n  newSamples: Float32Array,\n  samplesPerPixel: number,\n  totalSamplesProcessed: number,\n  bits: 8 | 16 = 16\n): Int8Array | Int16Array {\n  const maxValue = 2 ** (bits - 1);\n\n  // Check if we have a partial peak from the last update\n  const remainder = totalSamplesProcessed % samplesPerPixel;\n  let offset = 0;\n\n  // If there's a partial peak, we need to update the last peak\n  if (remainder > 0 && existingPeaks.length > 0) {\n    const samplesToComplete = samplesPerPixel - remainder;\n    const endIndex = Math.min(samplesToComplete, newSamples.length);\n\n    // Get current min/max from last peak\n    let min = existingPeaks[existingPeaks.length - 2] / maxValue;\n    let max = existingPeaks[existingPeaks.length - 1] / maxValue;\n\n    // Update with new samples\n    for (let i = 0; i < endIndex; i++) {\n      const value = newSamples[i];\n      if (value < min) min = value;\n      if (value > max) max = value;\n    }\n\n    // Update last peak\n    const updated = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length);\n    updated.set(existingPeaks);\n    updated[existingPeaks.length - 2] = Math.max(-maxValue, Math.floor(min * maxValue));\n    updated[existingPeaks.length - 1] = Math.min(maxValue - 1, Math.floor(max * maxValue));\n\n    offset = endIndex;\n\n    // Generate peaks for remaining samples and concatenate\n    const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);\n    const result = new (bits === 8 ? Int8Array : Int16Array)(updated.length + newPeaks.length);\n    result.set(updated);\n    result.set(newPeaks, updated.length);\n    return result;\n  }\n\n  // No partial peak, just concatenate\n  const newPeaks = generatePeaks(newSamples.slice(offset), samplesPerPixel, bits);\n  const result = new (bits === 8 ? Int8Array : Int16Array)(existingPeaks.length + newPeaks.length);\n  result.set(existingPeaks);\n  result.set(newPeaks, existingPeaks.length);\n  return result;\n}\n","/**\n * Hook for managing microphone access and device enumeration\n */\n\nimport { useState, useEffect, useCallback } from 'react';\nimport { UseMicrophoneAccessReturn, MicrophoneDevice } from '../types';\n\nexport function useMicrophoneAccess(): UseMicrophoneAccessReturn {\n  const [stream, setStream] = useState<MediaStream | null>(null);\n  const [devices, setDevices] = useState<MicrophoneDevice[]>([]);\n  const [hasPermission, setHasPermission] = useState(false);\n  const [isLoading, setIsLoading] = useState(false);\n  const [error, setError] = useState<Error | null>(null);\n\n  // Enumerate audio input devices\n  const enumerateDevices = useCallback(async () => {\n    try {\n      const allDevices = await navigator.mediaDevices.enumerateDevices();\n      const audioInputs = allDevices\n        .filter((device) => device.kind === 'audioinput')\n        .map((device) => ({\n          deviceId: device.deviceId,\n          label: device.label || `Microphone ${device.deviceId.slice(0, 8)}`,\n          groupId: device.groupId,\n        }));\n\n      setDevices(audioInputs);\n    } catch (err) {\n      console.error('Failed to enumerate devices:', err);\n      setError(err instanceof Error ? err : new Error('Failed to enumerate devices'));\n    }\n  }, []);\n\n  // Request microphone access\n  const requestAccess = useCallback(\n    async (deviceId?: string, audioConstraints?: MediaTrackConstraints) => {\n      setIsLoading(true);\n      setError(null);\n\n      try {\n        // Stop existing stream if any\n        if (stream) {\n          stream.getTracks().forEach((track) => track.stop());\n        }\n\n        // Build audio constraints\n        const audio: MediaTrackConstraints & { latency?: number } = {\n          // Recording-optimized defaults: prioritize raw audio quality and low latency\n          echoCancellation: false,\n          noiseSuppression: false,\n          autoGainControl: false,\n          latency: 0, // Low latency mode (not in TS types yet, but supported in modern browsers)\n          // User-provided constraints override defaults\n          ...audioConstraints,\n          // Device ID override (if specified)\n          ...(deviceId && { deviceId: { exact: deviceId } }),\n        };\n\n        const constraints: MediaStreamConstraints = {\n          audio,\n          video: false,\n        };\n\n        const newStream = await navigator.mediaDevices.getUserMedia(constraints);\n        setStream(newStream);\n        setHasPermission(true);\n\n        // Enumerate devices after getting permission (labels will be available)\n        await enumerateDevices();\n      } catch (err) {\n        console.error('Failed to access microphone:', err);\n        setError(err instanceof Error ? err : new Error('Failed to access microphone'));\n        setHasPermission(false);\n      } finally {\n        setIsLoading(false);\n      }\n    },\n    [stream, enumerateDevices]\n  );\n\n  // Stop the stream and revoke access\n  const stopStream = useCallback(() => {\n    if (stream) {\n      stream.getTracks().forEach((track) => track.stop());\n      setStream(null);\n      setHasPermission(false);\n    }\n  }, [stream]);\n\n  // Check initial permission state, enumerate devices, and listen for hot-plug changes\n  useEffect(() => {\n    // Try to enumerate devices (labels won't be available without permission)\n    enumerateDevices();\n\n    // Re-enumerate when devices are plugged in or removed\n    navigator.mediaDevices.addEventListener('devicechange', enumerateDevices);\n\n    // Cleanup on unmount\n    return () => {\n      navigator.mediaDevices.removeEventListener('devicechange', enumerateDevices);\n      if (stream) {\n        stream.getTracks().forEach((track) => track.stop());\n      }\n    };\n  }, [enumerateDevices, stream]);\n\n  return {\n    stream,\n    devices,\n    hasPermission,\n    isLoading,\n    requestAccess,\n    stopStream,\n    error,\n  };\n}\n","/**\n * Hook for monitoring microphone input levels\n *\n * Uses an AudioWorklet-based meter processor for sample-accurate\n * peak and RMS metering without requestAnimationFrame overhead.\n */\n\nimport { useEffect, useState, useRef, useCallback } from 'react';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { gainToNormalized } from '@waveform-playlist/core';\nimport { meterProcessorUrl, type MeterMessage } from '@waveform-playlist/worklets';\n\n/** Peak decay constant — exponential decay for smooth peak hold (~800ms to 1/e at 60fps) */\nconst PEAK_DECAY = 0.98;\n\nexport interface UseMicrophoneLevelOptions {\n  /**\n   * How often to update the level (in Hz)\n   * Default: 60 (60fps)\n   */\n  updateRate?: number;\n\n  /**\n   * Number of channels to meter (1 = mono, 2 = stereo)\n   * Default: 1\n   */\n  channelCount?: number;\n}\n\nexport interface UseMicrophoneLevelReturn {\n  /**\n   * Current peak audio level (0-1)\n   * For single channel: channel 0 level\n   * For multi-channel: max across all channels\n   */\n  level: number;\n\n  /**\n   * Held peak level since last reset (0-1)\n   * For single channel: channel 0 peak\n   * For multi-channel: max across all channels\n   */\n  peakLevel: number;\n\n  /**\n   * Reset the held peak level\n   */\n  resetPeak: () => void;\n\n  /**\n   * Per-channel peak levels (0-1). Array length matches channelCount.\n   * True peak: max absolute sample value per analysis frame.\n   */\n  levels: number[];\n\n  /**\n   * Per-channel held peak levels (0-1). Array length matches channelCount.\n   */\n  peakLevels: number[];\n\n  /**\n   * Per-channel RMS levels (0-1). Array length matches channelCount.\n   * RMS: root mean square of samples per analysis frame.\n   */\n  rmsLevels: number[];\n\n  /**\n   * Error from meter setup (worklet load failure, context issues, etc.)\n   * Null when metering is working normally.\n   */\n  error: Error | null;\n}\n\n/**\n * Monitor microphone input levels in real-time\n *\n * @param stream - MediaStream from getUserMedia\n * @param options - Configuration options\n * @returns Object with current peak level, RMS level, and held peak level\n *\n * @example\n * ```typescript\n * const { stream } = useMicrophoneAccess();\n * const { levels, rmsLevels, peakLevels } = useMicrophoneLevel(stream, { channelCount: 2 });\n *\n * return <SegmentedVUMeter levels={levels} peakLevels={peakLevels} />;\n * ```\n */\nexport function useMicrophoneLevel(\n  stream: MediaStream | null,\n  options: UseMicrophoneLevelOptions = {}\n): UseMicrophoneLevelReturn {\n  const { updateRate = 60, channelCount = 1 } = options;\n\n  const [levels, setLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [peakLevels, setPeakLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [rmsLevels, setRmsLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [meterError, setMeterError] = useState<Error | null>(null);\n\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  const smoothedPeakRef = useRef<number[]>(new Array(channelCount).fill(0));\n\n  const resetPeak = useCallback(\n    () => setPeakLevels(new Array(channelCount).fill(0)),\n    [channelCount]\n  );\n\n  useEffect(() => {\n    if (!stream) {\n      setLevels(new Array(channelCount).fill(0));\n      setPeakLevels(new Array(channelCount).fill(0));\n      setRmsLevels(new Array(channelCount).fill(0));\n      smoothedPeakRef.current = new Array(channelCount).fill(0);\n      return;\n    }\n\n    let isMounted = true;\n\n    const setupMonitoring = async () => {\n      if (!isMounted) return;\n\n      const context = getGlobalContext();\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n      if (!isMounted) return;\n\n      // Auto-detect actual mic channel count from stream\n      const trackSettings = stream.getAudioTracks()[0]?.getSettings();\n      const actualChannels = trackSettings?.channelCount ?? channelCount;\n\n      // Load worklet directly on rawContext — Tone.js's addAudioWorkletModule\n      // only loads ONE module per context (caches _workletPromise), silently\n      // skipping subsequent calls with different URLs.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(meterProcessorUrl);\n      if (!isMounted) return;\n\n      // Use Tone.js's createAudioWorkletNode — avoids rawContext identity issues\n      // in webpack-aliased environments (Docusaurus)\n      const workletNode = context.createAudioWorkletNode('meter-processor', {\n        channelCount: actualChannels,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n        processorOptions: {\n          numberOfChannels: actualChannels,\n          updateRate,\n        },\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Mic meter worklet processor error:', String(event));\n      };\n\n      // Create source and connect: source → meter\n      // Don't connect output to destination — mic monitoring would cause feedback\n      const source = context.createMediaStreamSource(stream);\n      sourceRef.current = source;\n      source.connect(workletNode);\n\n      smoothedPeakRef.current = new Array(actualChannels).fill(0);\n\n      // Listen for meter data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        if (!isMounted) return;\n\n        const { peak, rms } = event.data as MeterMessage;\n        const smoothed = smoothedPeakRef.current;\n\n        const peakValues: number[] = [];\n        const rmsValues: number[] = [];\n\n        for (let ch = 0; ch < peak.length; ch++) {\n          smoothed[ch] = Math.max(peak[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);\n          peakValues.push(gainToNormalized(smoothed[ch]));\n          rmsValues.push(gainToNormalized(rms[ch]));\n        }\n\n        // Mirror mono to fill requested channelCount\n        const mirroredPeaks =\n          peak.length < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;\n        const mirroredRms =\n          peak.length < channelCount ? new Array(channelCount).fill(rmsValues[0]) : rmsValues;\n\n        setLevels(mirroredPeaks);\n        setRmsLevels(mirroredRms);\n        setPeakLevels((prev) => mirroredPeaks.map((val, i) => Math.max(prev[i] ?? 0, val)));\n      };\n    };\n\n    setupMonitoring().catch((err) => {\n      console.warn('[waveform-playlist] Failed to set up mic level monitoring:', String(err));\n      if (isMounted) {\n        setMeterError(err instanceof Error ? err : new Error(String(err)));\n      }\n    });\n\n    return () => {\n      isMounted = false;\n\n      if (sourceRef.current) {\n        try {\n          sourceRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic source disconnect during cleanup:', String(err));\n        }\n        sourceRef.current = null;\n      }\n\n      if (workletNodeRef.current) {\n        try {\n          workletNodeRef.current.disconnect();\n          workletNodeRef.current.port.close();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic meter disconnect during cleanup:', String(err));\n        }\n        workletNodeRef.current = null;\n      }\n    };\n  }, [stream, updateRate, channelCount]);\n\n  // Backwards-compatible scalar values\n  const level = channelCount === 1 ? (levels[0] ?? 0) : Math.max(...levels);\n  const peakLevel = channelCount === 1 ? (peakLevels[0] ?? 0) : Math.max(...peakLevels);\n\n  return {\n    level,\n    peakLevel,\n    resetPeak,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: meterError,\n  };\n}\n","/**\n * Hook for integrated multi-track recording\n * Combines recording functionality with track management\n */\n\nimport { useState, useCallback, useEffect, useRef } from 'react';\nimport { useRecording } from './useRecording';\nimport { useMicrophoneAccess } from './useMicrophoneAccess';\nimport { useMicrophoneLevel } from './useMicrophoneLevel';\nimport type { MicrophoneDevice } from '../types';\nimport { type ClipTrack, type AudioClip } from '@waveform-playlist/core';\nimport {\n  resumeGlobalAudioContext,\n  getGlobalAudioContext,\n  getGlobalContext,\n} from '@waveform-playlist/playout';\n\nexport interface IntegratedRecordingOptions {\n  /**\n   * Current playback/cursor position in seconds\n   * Recording will start from max(currentTime, lastClipEndTime)\n   */\n  currentTime?: number;\n\n  /**\n   * MediaTrackConstraints for audio recording\n   * These will override the recording-optimized defaults (echo cancellation off, low latency)\n   */\n  audioConstraints?: MediaTrackConstraints;\n\n  /**\n   * Number of channels to record (1 = mono, 2 = stereo)\n   * Default: 1 (mono)\n   */\n  channelCount?: number;\n\n  /**\n   * Samples per pixel for peak generation\n   * Default: 1024\n   */\n  samplesPerPixel?: number;\n}\n\nexport interface UseIntegratedRecordingReturn {\n  // Recording state\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number;\n  level: number;\n  peakLevel: number;\n  /** Per-channel peak levels (0-1). Array length matches channelCount. */\n  levels: number[];\n  /** Per-channel held peak levels (0-1). Array length matches channelCount. */\n  peakLevels: number[];\n  /** Per-channel RMS levels (0-1). Array length matches channelCount. */\n  rmsLevels: number[];\n  error: Error | null;\n\n  // Microphone state\n  stream: MediaStream | null;\n  devices: MicrophoneDevice[];\n  hasPermission: boolean;\n  selectedDevice: string | null;\n\n  // Recording controls\n  startRecording: () => void;\n  stopRecording: () => void;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  requestMicAccess: () => Promise<void>;\n  changeDevice: (deviceId: string) => Promise<void>;\n\n  // Track state (for live waveform during recording)\n  recordingPeaks: (Int8Array | Int16Array)[];\n}\n\nexport function useIntegratedRecording(\n  tracks: ClipTrack[],\n  setTracks: (tracks: ClipTrack[]) => void,\n  selectedTrackId: string | null,\n  options: IntegratedRecordingOptions = {}\n): UseIntegratedRecordingReturn {\n  const { currentTime = 0, audioConstraints, ...recordingOptions } = options;\n\n  // Track if we're currently monitoring (for auto-resume audio context)\n  const [isMonitoring, setIsMonitoring] = useState(false);\n  const [selectedDevice, setSelectedDevice] = useState<string | null>(null);\n  const [hookError, setHookError] = useState<Error | null>(null);\n\n  // Capture timeline position when recording starts (not at stop time)\n  const recordingStartTimeRef = useRef(0);\n\n  // Keep selectedTrackId and currentTime in refs for use in callbacks.\n  // Avoids stale closures and prevents 60fps useCallback recreation\n  // (currentTime updates at animation-frame rate during playback).\n  const selectedTrackIdRef = useRef(selectedTrackId);\n  selectedTrackIdRef.current = selectedTrackId;\n  const currentTimeRef = useRef(currentTime);\n  currentTimeRef.current = currentTime;\n\n  // Microphone access\n  const { stream, devices, hasPermission, requestAccess, error: micError } = useMicrophoneAccess();\n\n  // Microphone level (for VU meter)\n  const {\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    resetPeak,\n    error: meterError,\n  } = useMicrophoneLevel(stream, {\n    channelCount: recordingOptions.channelCount,\n  });\n\n  // Recording\n  const {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer: _recordedAudioBuffer,\n    startRecording: startRec,\n    stopRecording: stopRec,\n    pauseRecording,\n    resumeRecording,\n    error: recError,\n  } = useRecording(stream, recordingOptions);\n\n  // Start recording handler\n  // Reads selectedTrackId from ref to avoid stale closures when\n  // auto-create track + start recording happen in the same render cycle\n  const startRecording = useCallback(async () => {\n    if (!selectedTrackIdRef.current) {\n      setHookError(\n        new Error('Cannot start recording: no track selected. Select or create a track first.')\n      );\n      return;\n    }\n\n    try {\n      setHookError(null);\n      // Resume audio context if needed\n      if (!isMonitoring) {\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      }\n\n      // Capture timeline position NOW — before recording starts.\n      // Using currentTime at stop time would be wrong during overdub\n      // (playback advances currentTime while recording).\n      recordingStartTimeRef.current = currentTimeRef.current;\n\n      await startRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [isMonitoring, startRec]);\n\n  // Stop recording and add clip to selected track\n  const stopRecording = useCallback(async () => {\n    let buffer: AudioBuffer | null;\n    try {\n      buffer = await stopRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n      return;\n    }\n\n    // Add clip to track after recording completes\n    const trackId = selectedTrackIdRef.current;\n    if (buffer && trackId) {\n      const selectedTrackIndex = tracks.findIndex((t) => t.id === trackId);\n      if (selectedTrackIndex === -1) {\n        const err = new Error(\n          `Recording completed but track \"${trackId}\" no longer exists. The recorded audio could not be saved.`\n        );\n        console.warn(`[waveform-playlist] ${err.message}`);\n        setHookError(err);\n        return;\n      }\n\n      const selectedTrack = tracks[selectedTrackIndex];\n\n      // Use the captured start time (not live currentTime which advances during overdub)\n      const recordStartTimeSamples = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n      let lastClipEndSample = 0;\n      if (selectedTrack.clips.length > 0) {\n        const endSamples = selectedTrack.clips.map(\n          (clip) => clip.startSample + clip.durationSamples\n        );\n        lastClipEndSample = Math.max(...endSamples);\n      }\n\n      const startSample = Math.max(recordStartTimeSamples, lastClipEndSample);\n\n      // Latency compensation:\n      // Two sources of delay between recording start and audible playback:\n      // 1. Tone.js lookAhead (~100ms) — Transport schedules audio ahead of real time\n      // 2. Output latency — hardware DAC delay before audio reaches speakers\n      // The user hears playback delayed by both, so they perform late relative\n      // to the timeline. Skip that duration at the start of the recorded audio.\n      const audioContext = getGlobalAudioContext();\n      const outputLatency = audioContext.outputLatency ?? 0;\n      const toneContext = getGlobalContext();\n      const lookAhead = toneContext.lookAhead ?? 0;\n      const totalLatency = outputLatency + lookAhead;\n      const latencyOffsetSamples = Math.floor(totalLatency * buffer.sampleRate);\n\n      // Guard: very short recordings (< latency compensation) would produce negative duration\n      const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n      if (effectiveDuration === 0) {\n        console.warn(\n          '[waveform-playlist] Recording too short for latency compensation — discarding'\n        );\n        setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n        return;\n      }\n\n      // Create new clip from recording\n      const newClip: AudioClip = {\n        id: `clip-${Date.now()}`,\n        audioBuffer: buffer,\n        startSample,\n        durationSamples: effectiveDuration,\n        offsetSamples: latencyOffsetSamples,\n        sampleRate: buffer.sampleRate,\n        sourceDurationSamples: buffer.length,\n        gain: 1.0,\n        name: `Recording ${new Date().toLocaleTimeString()}`,\n      };\n\n      // Add clip to track\n      const newTracks = tracks.map((track, index) => {\n        if (index === selectedTrackIndex) {\n          return {\n            ...track,\n            clips: [...track.clips, newClip],\n          };\n        }\n        return track;\n      });\n\n      setTracks(newTracks);\n    }\n  }, [tracks, setTracks, stopRec]);\n\n  // Auto-select first device when available, or fallback if selected device was unplugged\n  useEffect(() => {\n    if (!hasPermission || devices.length === 0) return;\n\n    if (selectedDevice === null) {\n      // First-time selection\n      setSelectedDevice(devices[0].deviceId);\n    } else if (!devices.some((d) => d.deviceId === selectedDevice)) {\n      // Selected device was removed — fall back to first available\n      const fallbackId = devices[0].deviceId;\n      setSelectedDevice(fallbackId);\n      resetPeak();\n      requestAccess(fallbackId, audioConstraints);\n    }\n  }, [hasPermission, devices, selectedDevice, resetPeak, requestAccess, audioConstraints]);\n\n  // Request microphone access\n  const requestMicAccess = useCallback(async () => {\n    try {\n      setHookError(null);\n      await requestAccess(undefined, audioConstraints);\n      await resumeGlobalAudioContext();\n      setIsMonitoring(true);\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [requestAccess, audioConstraints]);\n\n  // Change device\n  const changeDevice = useCallback(\n    async (deviceId: string) => {\n      try {\n        setHookError(null);\n        setSelectedDevice(deviceId);\n        resetPeak();\n        await requestAccess(deviceId, audioConstraints);\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      } catch (err) {\n        setHookError(err instanceof Error ? err : new Error(String(err)));\n      }\n    },\n    [requestAccess, audioConstraints, resetPeak]\n  );\n\n  return {\n    // Recording state\n    isRecording,\n    isPaused,\n    duration,\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: hookError || micError || meterError || recError,\n\n    // Microphone state\n    stream,\n    devices,\n    hasPermission,\n    selectedDevice,\n\n    // Recording controls\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    requestMicAccess,\n    changeDevice,\n\n    // Track state\n    recordingPeaks: peaks,\n  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+{"version":3,"sources":["../src/hooks/useRecording.ts","../src/hooks/useMicrophoneAccess.ts","../src/hooks/useMicrophoneLevel.ts","../src/hooks/useIntegratedRecording.ts"],"sourcesContent":["/**\n * Main recording hook using AudioWorklet\n */\n\nimport { useState, useRef, useCallback, useEffect } from 'react';\nimport { UseRecordingReturn, RecordingOptions } from '../types';\nimport { concatenateAudioData, createAudioBuffer, appendPeaks } from '@waveform-playlist/core';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { recordingProcessorUrl } from '@waveform-playlist/worklets';\n\nfunction emptyPeaks(bits: 8 | 16): Int8Array | Int16Array {\n  return bits === 8 ? new Int8Array(0) : new Int16Array(0);\n}\n\nexport function useRecording(\n  stream: MediaStream | null,\n  options: RecordingOptions = {}\n): UseRecordingReturn {\n  const { channelCount = 1, samplesPerPixel = 1024, bits = 16 } = options;\n\n  // State\n  const [isRecording, setIsRecording] = useState(false);\n  const [isPaused, setIsPaused] = useState(false);\n  const [duration, setDuration] = useState(0);\n  // Per-channel peaks for multi-channel live preview\n  const [peaks, setPeaks] = useState<(Int8Array | Int16Array)[]>([emptyPeaks(bits)]);\n  const [audioBuffer, setAudioBuffer] = useState<AudioBuffer | null>(null);\n  const [error, setError] = useState<Error | null>(null);\n  const [level, setLevel] = useState(0); // Current RMS level (0-1)\n  const [peakLevel, setPeakLevel] = useState(0); // Peak level since recording started (0-1)\n\n  // Per-instance flag to prevent loading worklet multiple times within the same hook instance.\n  // Note: Multiple hook instances each have their own ref — see \"Multi-Instance Worklet\n  // Registration Gap\" in recording/CLAUDE.md for the known limitation and planned fix.\n  const workletLoadedRef = useRef<boolean>(false);\n\n  // Refs for AudioWorklet and data accumulation\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const mediaStreamSourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  // Per-channel sample accumulation: recordedChunksRef[channelIndex] = Float32Array[]\n  const recordedChunksRef = useRef<Float32Array[][]>([]);\n  const totalSamplesRef = useRef(0);\n  const animationFrameRef = useRef<number | null>(null);\n  const startTimeRef = useRef<number>(0);\n  const isRecordingRef = useRef<boolean>(false);\n  const isPausedRef = useRef<boolean>(false);\n  const audioTrackRef = useRef<MediaStreamTrack | null>(null);\n  const onTrackEndedRef = useRef<(() => void) | null>(null);\n  const stopRecordingRef = useRef<(() => Promise<AudioBuffer | null>) | null>(null);\n\n  // Shared duration update loop — starts a rAF loop that updates duration\n  // from performance.now(). Used by both startRecording and resumeRecording.\n  const startDurationLoop = useCallback(() => {\n    const tick = () => {\n      if (isRecordingRef.current && !isPausedRef.current) {\n        const elapsed = (performance.now() - startTimeRef.current) / 1000;\n        setDuration(elapsed);\n        animationFrameRef.current = requestAnimationFrame(tick);\n      }\n    };\n    tick();\n  }, []);\n\n  // Load AudioWorklet module\n  const loadWorklet = useCallback(async () => {\n    // Skip if already loaded to prevent \"already registered\" error\n    if (workletLoadedRef.current) {\n      return;\n    }\n\n    try {\n      const context = getGlobalContext();\n      // Load the worklet module directly on the raw AudioContext.\n      // Tone.js's addAudioWorkletModule only loads ONE module per context\n      // (caches _workletPromise). If meter-processor was loaded first by\n      // useMicrophoneLevel, recording-processor is silently skipped.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(recordingProcessorUrl);\n      workletLoadedRef.current = true;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to load AudioWorklet module:', String(err));\n      const error = new Error('Failed to load recording processor: ' + String(err));\n      throw error;\n    }\n  }, []);\n\n  // Start recording\n  const startRecording = useCallback(async () => {\n    if (isRecordingRef.current) return;\n    if (!stream) {\n      setError(new Error('No microphone stream available'));\n      return;\n    }\n\n    try {\n      setError(null);\n\n      // Use Tone.js Context for cross-browser compatibility\n      const context = getGlobalContext();\n\n      // Resume AudioContext if suspended\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n\n      // Load worklet module\n      await loadWorklet();\n\n      // Create MediaStreamSource from Tone's context\n      // Each hook creates its own source to avoid cross-context issues in Firefox\n      const source = context.createMediaStreamSource(stream);\n      mediaStreamSourceRef.current = source;\n\n      // Use the stream track's actual channel count from getSettings().\n      // source.channelCount defaults to 2 per Web Audio spec (not the mic's\n      // real count). Fall back to user-provided channelCount if unavailable.\n      const detectedChannelCount = stream.getAudioTracks()[0]?.getSettings().channelCount;\n      if (detectedChannelCount === undefined) {\n        console.warn(\n          `[waveform-playlist] Could not detect stream channel count, using fallback: ${channelCount}`\n        );\n      }\n      const streamChannelCount = detectedChannelCount ?? channelCount;\n\n      const workletNode = context.createAudioWorkletNode('recording-processor', {\n        channelCount: streamChannelCount,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Recording worklet processor error:', String(event));\n        setError(new Error('Recording processor encountered an error'));\n      };\n\n      // Reset state before connecting — prevents race where a worklet message\n      // arrives before refs are cleared, corrupting samplesProcessedBefore calculations\n      recordedChunksRef.current = Array.from({ length: streamChannelCount }, () => []);\n      totalSamplesRef.current = 0;\n      setPeaks(Array.from({ length: streamChannelCount }, () => emptyPeaks(bits)));\n      setAudioBuffer(null);\n      setLevel(0);\n      setPeakLevel(0);\n\n      // Listen for audio data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        const { channels } = event.data as { channels: Float32Array[] };\n\n        if (!channels || channels.length === 0) {\n          console.warn('[waveform-playlist] Recording worklet sent empty or missing channels data');\n          return;\n        }\n\n        // Accumulate per-channel samples\n        for (let ch = 0; ch < channels.length; ch++) {\n          if (!recordedChunksRef.current[ch]) {\n            console.warn(\n              `[waveform-playlist] Unexpected channel ${ch} from worklet (expected ${recordedChunksRef.current.length})`\n            );\n            recordedChunksRef.current[ch] = [];\n          }\n          recordedChunksRef.current[ch].push(channels[ch]);\n        }\n        // Capture sample offset before incrementing — used by peak alignment\n        const samplesProcessedBefore = totalSamplesRef.current;\n        totalSamplesRef.current += channels[0].length;\n        setPeaks((prevPeaks) => {\n          // Ensure we have an entry per channel\n          const updated: (Int8Array | Int16Array)[] = [];\n          for (let ch = 0; ch < channels.length; ch++) {\n            const prev = prevPeaks[ch] ?? emptyPeaks(bits);\n            updated.push(\n              appendPeaks(prev, channels[ch], samplesPerPixel, samplesProcessedBefore, bits)\n            );\n          }\n          return updated;\n        });\n\n        // Note: VU meter levels come from useMicrophoneLevel (meter-processor worklet)\n        // We don't update level/peakLevel here to avoid conflicting state updates\n      };\n\n      // Connect and start — after state reset and handler setup\n      source.connect(workletNode);\n      // Do NOT send sampleRate — worklet uses its global sampleRate (always correct)\n      workletNode.port.postMessage({ command: 'start', channelCount: streamChannelCount });\n\n      // Listen on MediaStreamTrack for mic unplug (MediaStream has no 'ended' event)\n      const audioTrack = stream.getAudioTracks()[0] ?? null;\n      audioTrackRef.current = audioTrack;\n      if (audioTrack) {\n        const onEnded = () => {\n          console.warn('[waveform-playlist] Audio track ended (mic unplugged or revoked)');\n          // Stop recording to clean up worklet, rAF loop, and UI state\n          stopRecordingRef.current?.();\n          setError(new Error('Microphone disconnected during recording'));\n        };\n        onTrackEndedRef.current = onEnded;\n        audioTrack.addEventListener('ended', onEnded);\n      }\n\n      isRecordingRef.current = true;\n      isPausedRef.current = false;\n      setIsRecording(true);\n      setIsPaused(false);\n      startTimeRef.current = performance.now();\n      startDurationLoop();\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to start recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to start recording'));\n    }\n  }, [stream, channelCount, samplesPerPixel, bits, loadWorklet, startDurationLoop]);\n\n  // Stop recording\n  const stopRecording = useCallback(async (): Promise<AudioBuffer | null> => {\n    if (!isRecordingRef.current) {\n      return null;\n    }\n\n    try {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n        audioTrackRef.current = null;\n        onTrackEndedRef.current = null;\n      }\n\n      // Stop the worklet\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n          }\n        }\n        workletNodeRef.current.disconnect();\n      }\n\n      // Cancel animation frame\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n\n      // Create final AudioBuffer from accumulated per-channel chunks\n      const context = getGlobalContext();\n      const rawContext = context.rawContext as AudioContext;\n      const numChannels = recordedChunksRef.current.length || channelCount;\n      const channelData = recordedChunksRef.current.map((chunks) => concatenateAudioData(chunks));\n      const totalSamples = channelData[0]?.length ?? 0;\n\n      // Guard: if no samples were captured (e.g., stop called immediately after start),\n      // return null instead of creating a 0-length AudioBuffer which throws\n      if (totalSamples === 0) {\n        console.warn('[waveform-playlist] Recording stopped with 0 samples captured — discarding');\n        isRecordingRef.current = false;\n        isPausedRef.current = false;\n        setIsRecording(false);\n        setIsPaused(false);\n        setLevel(0);\n        return null;\n      }\n\n      const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n      setAudioBuffer(buffer);\n      setDuration(buffer.duration);\n      isRecordingRef.current = false;\n      isPausedRef.current = false;\n      setIsRecording(false);\n      setIsPaused(false);\n      setLevel(0);\n      // Keep peakLevel to show the peak reached during recording\n\n      return buffer;\n    } catch (err) {\n      console.warn('[waveform-playlist] Failed to stop recording:', String(err));\n      setError(err instanceof Error ? err : new Error('Failed to stop recording'));\n      return null;\n    }\n  }, [channelCount]);\n  stopRecordingRef.current = stopRecording;\n\n  // Pause recording\n  const pauseRecording = useCallback(() => {\n    if (isRecording && !isPaused) {\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n        animationFrameRef.current = null;\n      }\n      isPausedRef.current = true;\n      setIsPaused(true);\n    }\n  }, [isRecording, isPaused]);\n\n  // Resume recording\n  const resumeRecording = useCallback(() => {\n    if (isRecording && isPaused) {\n      isPausedRef.current = false;\n      setIsPaused(false);\n      startTimeRef.current = performance.now() - duration * 1000;\n      startDurationLoop();\n    }\n  }, [isRecording, isPaused, duration, startDurationLoop]);\n\n  // Cleanup on unmount — read refs in cleanup, not effect body.\n  // Pattern #16 (copy refs in effect body) doesn't apply to [] deps — refs are\n  // null at mount and only set later during startRecording.\n  useEffect(() => {\n    return () => {\n      // Remove mic-unplug listener\n      if (audioTrackRef.current && onTrackEndedRef.current) {\n        audioTrackRef.current.removeEventListener('ended', onTrackEndedRef.current);\n      }\n      if (workletNodeRef.current) {\n        workletNodeRef.current.port.postMessage({ command: 'stop' });\n\n        // Disconnect worklet from source\n        if (mediaStreamSourceRef.current) {\n          try {\n            mediaStreamSourceRef.current.disconnect(workletNodeRef.current);\n          } catch (err) {\n            console.warn('[waveform-playlist] Source disconnect during cleanup:', String(err));\n          }\n        }\n        try {\n          workletNodeRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Worklet disconnect during cleanup:', String(err));\n        }\n      }\n      if (animationFrameRef.current !== null) {\n        cancelAnimationFrame(animationFrameRef.current);\n      }\n      // Don't close the global AudioContext - it's shared across the app\n    };\n  }, []);\n\n  return {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer,\n    level,\n    peakLevel,\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    error,\n  };\n}\n","/**\n * Hook for managing microphone access and device enumeration\n */\n\nimport { useState, useEffect, useCallback } from 'react';\nimport { UseMicrophoneAccessReturn, MicrophoneDevice } from '../types';\n\nexport function useMicrophoneAccess(): UseMicrophoneAccessReturn {\n  const [stream, setStream] = useState<MediaStream | null>(null);\n  const [devices, setDevices] = useState<MicrophoneDevice[]>([]);\n  const [hasPermission, setHasPermission] = useState(false);\n  const [isLoading, setIsLoading] = useState(false);\n  const [error, setError] = useState<Error | null>(null);\n\n  // Enumerate audio input devices\n  const enumerateDevices = useCallback(async () => {\n    try {\n      const allDevices = await navigator.mediaDevices.enumerateDevices();\n      const audioInputs = allDevices\n        .filter((device) => device.kind === 'audioinput')\n        .map((device) => ({\n          deviceId: device.deviceId,\n          label: device.label || `Microphone ${device.deviceId.slice(0, 8)}`,\n          groupId: device.groupId,\n        }));\n\n      setDevices(audioInputs);\n    } catch (err) {\n      console.error('Failed to enumerate devices:', err);\n      setError(err instanceof Error ? err : new Error('Failed to enumerate devices'));\n    }\n  }, []);\n\n  // Request microphone access\n  const requestAccess = useCallback(\n    async (deviceId?: string, audioConstraints?: MediaTrackConstraints) => {\n      setIsLoading(true);\n      setError(null);\n\n      try {\n        // Stop existing stream if any\n        if (stream) {\n          stream.getTracks().forEach((track) => track.stop());\n        }\n\n        // Build audio constraints\n        const audio: MediaTrackConstraints & { latency?: number } = {\n          // Recording-optimized defaults: prioritize raw audio quality and low latency\n          echoCancellation: false,\n          noiseSuppression: false,\n          autoGainControl: false,\n          latency: 0, // Low latency mode (not in TS types yet, but supported in modern browsers)\n          // User-provided constraints override defaults\n          ...audioConstraints,\n          // Device ID override (if specified)\n          ...(deviceId && { deviceId: { exact: deviceId } }),\n        };\n\n        const constraints: MediaStreamConstraints = {\n          audio,\n          video: false,\n        };\n\n        const newStream = await navigator.mediaDevices.getUserMedia(constraints);\n        setStream(newStream);\n        setHasPermission(true);\n\n        // Enumerate devices after getting permission (labels will be available)\n        await enumerateDevices();\n      } catch (err) {\n        console.error('Failed to access microphone:', err);\n        setError(err instanceof Error ? err : new Error('Failed to access microphone'));\n        setHasPermission(false);\n      } finally {\n        setIsLoading(false);\n      }\n    },\n    [stream, enumerateDevices]\n  );\n\n  // Stop the stream and revoke access\n  const stopStream = useCallback(() => {\n    if (stream) {\n      stream.getTracks().forEach((track) => track.stop());\n      setStream(null);\n      setHasPermission(false);\n    }\n  }, [stream]);\n\n  // Check initial permission state, enumerate devices, and listen for hot-plug changes\n  useEffect(() => {\n    // Try to enumerate devices (labels won't be available without permission)\n    enumerateDevices();\n\n    // Re-enumerate when devices are plugged in or removed\n    navigator.mediaDevices.addEventListener('devicechange', enumerateDevices);\n\n    // Cleanup on unmount\n    return () => {\n      navigator.mediaDevices.removeEventListener('devicechange', enumerateDevices);\n      if (stream) {\n        stream.getTracks().forEach((track) => track.stop());\n      }\n    };\n  }, [enumerateDevices, stream]);\n\n  return {\n    stream,\n    devices,\n    hasPermission,\n    isLoading,\n    requestAccess,\n    stopStream,\n    error,\n  };\n}\n","/**\n * Hook for monitoring microphone input levels\n *\n * Uses an AudioWorklet-based meter processor for sample-accurate\n * peak and RMS metering without requestAnimationFrame overhead.\n */\n\nimport { useEffect, useState, useRef, useCallback } from 'react';\nimport { getGlobalContext } from '@waveform-playlist/playout';\nimport { gainToNormalized } from '@waveform-playlist/core';\nimport { meterProcessorUrl, type MeterMessage } from '@waveform-playlist/worklets';\n\n/** Peak decay constant — exponential decay for smooth peak hold (~800ms to 1/e at 60fps) */\nconst PEAK_DECAY = 0.98;\n\nexport interface UseMicrophoneLevelOptions {\n  /**\n   * How often to update the level (in Hz)\n   * Default: 60 (60fps)\n   */\n  updateRate?: number;\n\n  /**\n   * Number of channels to meter (1 = mono, 2 = stereo)\n   * Default: 1\n   */\n  channelCount?: number;\n}\n\nexport interface UseMicrophoneLevelReturn {\n  /**\n   * Current peak audio level (0-1)\n   * For single channel: channel 0 level\n   * For multi-channel: max across all channels\n   */\n  level: number;\n\n  /**\n   * Held peak level since last reset (0-1)\n   * For single channel: channel 0 peak\n   * For multi-channel: max across all channels\n   */\n  peakLevel: number;\n\n  /**\n   * Reset the held peak level\n   */\n  resetPeak: () => void;\n\n  /**\n   * Per-channel peak levels (0-1). Array length matches channelCount.\n   * True peak: max absolute sample value per analysis frame.\n   */\n  levels: number[];\n\n  /**\n   * Per-channel held peak levels (0-1). Array length matches channelCount.\n   */\n  peakLevels: number[];\n\n  /**\n   * Per-channel RMS levels (0-1). Array length matches channelCount.\n   * RMS: root mean square of samples per analysis frame.\n   */\n  rmsLevels: number[];\n\n  /**\n   * Error from meter setup (worklet load failure, context issues, etc.)\n   * Null when metering is working normally.\n   */\n  error: Error | null;\n}\n\n/**\n * Monitor microphone input levels in real-time\n *\n * @param stream - MediaStream from getUserMedia\n * @param options - Configuration options\n * @returns Object with current peak level, RMS level, and held peak level\n *\n * @example\n * ```typescript\n * const { stream } = useMicrophoneAccess();\n * const { levels, rmsLevels, peakLevels } = useMicrophoneLevel(stream, { channelCount: 2 });\n *\n * return <SegmentedVUMeter levels={levels} peakLevels={peakLevels} />;\n * ```\n */\nexport function useMicrophoneLevel(\n  stream: MediaStream | null,\n  options: UseMicrophoneLevelOptions = {}\n): UseMicrophoneLevelReturn {\n  const { updateRate = 60, channelCount = 1 } = options;\n\n  const [levels, setLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [peakLevels, setPeakLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [rmsLevels, setRmsLevels] = useState<number[]>(() => new Array(channelCount).fill(0));\n  const [meterError, setMeterError] = useState<Error | null>(null);\n\n  const workletNodeRef = useRef<AudioWorkletNode | null>(null);\n  const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);\n  const smoothedPeakRef = useRef<number[]>(new Array(channelCount).fill(0));\n\n  const resetPeak = useCallback(\n    () => setPeakLevels(new Array(channelCount).fill(0)),\n    [channelCount]\n  );\n\n  useEffect(() => {\n    if (!stream) {\n      setLevels(new Array(channelCount).fill(0));\n      setPeakLevels(new Array(channelCount).fill(0));\n      setRmsLevels(new Array(channelCount).fill(0));\n      smoothedPeakRef.current = new Array(channelCount).fill(0);\n      return;\n    }\n\n    let isMounted = true;\n\n    const setupMonitoring = async () => {\n      if (!isMounted) return;\n\n      const context = getGlobalContext();\n      if (context.state === 'suspended') {\n        await context.resume();\n      }\n      if (!isMounted) return;\n\n      // Auto-detect actual mic channel count from stream\n      const trackSettings = stream.getAudioTracks()[0]?.getSettings();\n      const actualChannels = trackSettings?.channelCount ?? channelCount;\n\n      // Load worklet directly on rawContext — Tone.js's addAudioWorkletModule\n      // only loads ONE module per context (caches _workletPromise), silently\n      // skipping subsequent calls with different URLs.\n      const rawCtx = context.rawContext as AudioContext;\n      await rawCtx.audioWorklet.addModule(meterProcessorUrl);\n      if (!isMounted) return;\n\n      // Use Tone.js's createAudioWorkletNode — avoids rawContext identity issues\n      // in webpack-aliased environments (Docusaurus)\n      const workletNode = context.createAudioWorkletNode('meter-processor', {\n        channelCount: actualChannels,\n        channelCountMode: 'explicit' as globalThis.ChannelCountMode,\n        processorOptions: {\n          numberOfChannels: actualChannels,\n          updateRate,\n        },\n      });\n      workletNodeRef.current = workletNode;\n\n      workletNode.onprocessorerror = (event) => {\n        console.warn('[waveform-playlist] Mic meter worklet processor error:', String(event));\n      };\n\n      // Create source and connect: source → meter\n      // Don't connect output to destination — mic monitoring would cause feedback\n      const source = context.createMediaStreamSource(stream);\n      sourceRef.current = source;\n      source.connect(workletNode);\n\n      smoothedPeakRef.current = new Array(actualChannels).fill(0);\n\n      // Listen for meter data from worklet\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        if (!isMounted) return;\n\n        const { peak, rms } = event.data as MeterMessage;\n        const smoothed = smoothedPeakRef.current;\n\n        const peakValues: number[] = [];\n        const rmsValues: number[] = [];\n\n        for (let ch = 0; ch < peak.length; ch++) {\n          smoothed[ch] = Math.max(peak[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);\n          peakValues.push(gainToNormalized(smoothed[ch]));\n          rmsValues.push(gainToNormalized(rms[ch]));\n        }\n\n        // Mirror mono to fill requested channelCount\n        const mirroredPeaks =\n          peak.length < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;\n        const mirroredRms =\n          peak.length < channelCount ? new Array(channelCount).fill(rmsValues[0]) : rmsValues;\n\n        setLevels(mirroredPeaks);\n        setRmsLevels(mirroredRms);\n        setPeakLevels((prev) => mirroredPeaks.map((val, i) => Math.max(prev[i] ?? 0, val)));\n      };\n    };\n\n    setupMonitoring().catch((err) => {\n      console.warn('[waveform-playlist] Failed to set up mic level monitoring:', String(err));\n      if (isMounted) {\n        setMeterError(err instanceof Error ? err : new Error(String(err)));\n      }\n    });\n\n    return () => {\n      isMounted = false;\n\n      if (sourceRef.current) {\n        try {\n          sourceRef.current.disconnect();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic source disconnect during cleanup:', String(err));\n        }\n        sourceRef.current = null;\n      }\n\n      if (workletNodeRef.current) {\n        try {\n          workletNodeRef.current.disconnect();\n          workletNodeRef.current.port.close();\n        } catch (err) {\n          console.warn('[waveform-playlist] Mic meter disconnect during cleanup:', String(err));\n        }\n        workletNodeRef.current = null;\n      }\n    };\n  }, [stream, updateRate, channelCount]);\n\n  // Backwards-compatible scalar values\n  const level = channelCount === 1 ? (levels[0] ?? 0) : Math.max(...levels);\n  const peakLevel = channelCount === 1 ? (peakLevels[0] ?? 0) : Math.max(...peakLevels);\n\n  return {\n    level,\n    peakLevel,\n    resetPeak,\n    levels,\n    peakLevels,\n    rmsLevels,\n    error: meterError,\n  };\n}\n","/**\n * Hook for integrated multi-track recording\n * Combines recording functionality with track management\n */\n\nimport { useState, useCallback, useEffect, useRef } from 'react';\nimport { useRecording } from './useRecording';\nimport { useMicrophoneAccess } from './useMicrophoneAccess';\nimport { useMicrophoneLevel } from './useMicrophoneLevel';\nimport type { MicrophoneDevice } from '../types';\nimport { type ClipTrack, type AudioClip } from '@waveform-playlist/core';\nimport {\n  resumeGlobalAudioContext,\n  getGlobalAudioContext,\n  getGlobalContext,\n} from '@waveform-playlist/playout';\n\nexport interface IntegratedRecordingOptions {\n  /**\n   * Current playback/cursor position in seconds\n   * Recording will start from max(currentTime, lastClipEndTime)\n   */\n  currentTime?: number;\n\n  /**\n   * MediaTrackConstraints for audio recording\n   * These will override the recording-optimized defaults (echo cancellation off, low latency)\n   */\n  audioConstraints?: MediaTrackConstraints;\n\n  /**\n   * Number of channels to record (1 = mono, 2 = stereo)\n   * Default: 1 (mono)\n   */\n  channelCount?: number;\n\n  /**\n   * Samples per pixel for peak generation\n   * Default: 1024\n   */\n  samplesPerPixel?: number;\n}\n\nexport interface UseIntegratedRecordingReturn {\n  // Recording state\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number;\n  level: number;\n  peakLevel: number;\n  /** Per-channel peak levels (0-1). Array length matches channelCount. */\n  levels: number[];\n  /** Per-channel held peak levels (0-1). Array length matches channelCount. */\n  peakLevels: number[];\n  /** Per-channel RMS levels (0-1). Array length matches channelCount. */\n  rmsLevels: number[];\n  error: Error | null;\n\n  // Microphone state\n  stream: MediaStream | null;\n  devices: MicrophoneDevice[];\n  hasPermission: boolean;\n  selectedDevice: string | null;\n\n  // Recording controls\n  startRecording: () => void;\n  stopRecording: () => void;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  requestMicAccess: () => Promise<void>;\n  changeDevice: (deviceId: string) => Promise<void>;\n\n  // Track state (for live waveform during recording)\n  recordingPeaks: (Int8Array | Int16Array)[];\n}\n\nexport function useIntegratedRecording(\n  tracks: ClipTrack[],\n  setTracks: (tracks: ClipTrack[]) => void,\n  selectedTrackId: string | null,\n  options: IntegratedRecordingOptions = {}\n): UseIntegratedRecordingReturn {\n  const { currentTime = 0, audioConstraints, ...recordingOptions } = options;\n\n  // Track if we're currently monitoring (for auto-resume audio context)\n  const [isMonitoring, setIsMonitoring] = useState(false);\n  const [selectedDevice, setSelectedDevice] = useState<string | null>(null);\n  const [hookError, setHookError] = useState<Error | null>(null);\n\n  // Capture timeline position when recording starts (not at stop time)\n  const recordingStartTimeRef = useRef(0);\n\n  // Keep selectedTrackId and currentTime in refs for use in callbacks.\n  // Avoids stale closures and prevents 60fps useCallback recreation\n  // (currentTime updates at animation-frame rate during playback).\n  const selectedTrackIdRef = useRef(selectedTrackId);\n  selectedTrackIdRef.current = selectedTrackId;\n  const currentTimeRef = useRef(currentTime);\n  currentTimeRef.current = currentTime;\n\n  // Microphone access\n  const { stream, devices, hasPermission, requestAccess, error: micError } = useMicrophoneAccess();\n\n  // Microphone level (for VU meter)\n  const {\n    level,\n    peakLevel,\n    levels,\n    peakLevels,\n    rmsLevels,\n    resetPeak,\n    error: meterError,\n  } = useMicrophoneLevel(stream, {\n    channelCount: recordingOptions.channelCount,\n  });\n\n  // Recording\n  const {\n    isRecording,\n    isPaused,\n    duration,\n    peaks,\n    audioBuffer: _recordedAudioBuffer,\n    startRecording: startRec,\n    stopRecording: stopRec,\n    pauseRecording,\n    resumeRecording,\n    error: recError,\n  } = useRecording(stream, recordingOptions);\n\n  // Start recording handler\n  // Reads selectedTrackId from ref to avoid stale closures when\n  // auto-create track + start recording happen in the same render cycle\n  const startRecording = useCallback(async () => {\n    if (!selectedTrackIdRef.current) {\n      setHookError(\n        new Error('Cannot start recording: no track selected. Select or create a track first.')\n      );\n      return;\n    }\n\n    try {\n      setHookError(null);\n      // Resume audio context if needed\n      if (!isMonitoring) {\n        await resumeGlobalAudioContext();\n        setIsMonitoring(true);\n      }\n\n      // Capture timeline position NOW — before recording starts.\n      // Using currentTime at stop time would be wrong during overdub\n      // (playback advances currentTime while recording).\n      recordingStartTimeRef.current = currentTimeRef.current;\n\n      await startRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [isMonitoring, startRec]);\n\n  // Stop recording and add clip to selected track\n  const stopRecording = useCallback(async () => {\n    let buffer: AudioBuffer | null;\n    try {\n      buffer = await stopRec();\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n      return;\n    }\n\n    // Add clip to track after recording completes\n    const trackId = selectedTrackIdRef.current;\n    if (buffer && trackId) {\n      const selectedTrackIndex = tracks.findIndex((t) => t.id === trackId);\n      if (selectedTrackIndex === -1) {\n        const err = new Error(\n          `Recording completed but track \"${trackId}\" no longer exists. The recorded audio could not be saved.`\n        );\n        console.warn(`[waveform-playlist] ${err.message}`);\n        setHookError(err);\n        return;\n      }\n\n      const selectedTrack = tracks[selectedTrackIndex];\n\n      // Use the captured start time (not live currentTime which advances during overdub)\n      const recordStartTimeSamples = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n      let lastClipEndSample = 0;\n      if (selectedTrack.clips.length > 0) {\n        const endSamples = selectedTrack.clips.map(\n          (clip) => clip.startSample + clip.durationSamples\n        );\n        lastClipEndSample = Math.max(...endSamples);\n      }\n\n      const startSample = Math.max(recordStartTimeSamples, lastClipEndSample);\n\n      // Latency compensation:\n      // Two sources of delay between recording start and audible playback:\n      // 1. Tone.js lookAhead (~100ms) — Transport schedules audio ahead of real time\n      // 2. Output latency — hardware DAC delay before audio reaches speakers\n      // The user hears playback delayed by both, so they perform late relative\n      // to the timeline. Skip that duration at the start of the recorded audio.\n      const audioContext = getGlobalAudioContext();\n      const outputLatency = audioContext.outputLatency ?? 0;\n      const toneContext = getGlobalContext();\n      const lookAhead = toneContext.lookAhead ?? 0;\n      const totalLatency = outputLatency + lookAhead;\n      const latencyOffsetSamples = Math.floor(totalLatency * buffer.sampleRate);\n\n      // Guard: very short recordings (< latency compensation) would produce negative duration\n      const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n      if (effectiveDuration === 0) {\n        console.warn(\n          '[waveform-playlist] Recording too short for latency compensation — discarding'\n        );\n        setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n        return;\n      }\n\n      // Create new clip from recording\n      const newClip: AudioClip = {\n        id: `clip-${Date.now()}`,\n        audioBuffer: buffer,\n        startSample,\n        durationSamples: effectiveDuration,\n        offsetSamples: latencyOffsetSamples,\n        sampleRate: buffer.sampleRate,\n        sourceDurationSamples: buffer.length,\n        gain: 1.0,\n        name: `Recording ${new Date().toLocaleTimeString()}`,\n      };\n\n      // Add clip to track\n      const newTracks = tracks.map((track, index) => {\n        if (index === selectedTrackIndex) {\n          return {\n            ...track,\n            clips: [...track.clips, newClip],\n          };\n        }\n        return track;\n      });\n\n      setTracks(newTracks);\n    }\n  }, [tracks, setTracks, stopRec]);\n\n  // Auto-select first device when available, or fallback if selected device was unplugged\n  useEffect(() => {\n    if (!hasPermission || devices.length === 0) return;\n\n    if (selectedDevice === null) {\n      // First-time selection\n      setSelectedDevice(devices[0].deviceId);\n    } else if (!devices.some((d) => d.deviceId === selectedDevice)) {\n      // Selected device was removed — fall back to first available\n      const fallbackId = devices[0].deviceId;\n      setSelectedDevice(fallbackId);\n      resetPeak();\n      requestAccess(fallbackId, audioConstraints);\n    }\n  }, [hasPermission, devices, selectedDevice, resetPeak, requestAccess, audioConstraints]);\n\n  // Request microphone access\n  const requestMicAccess = useCallback(async () => {\n    try {\n      setHookError(null);\n      await requestAccess(undefined, audioConstraints);\n      await resumeGlobalAudioContext();\n      setIsMonitoring(true);\n    } catch (err) {\n      setHookError(err instanceof Error ? err : new Error(String(err)));\n    }\n  }, [requestAccess, audioConstraints]);\n\n  // Change device\n  const changeDevice = 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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@waveform-playlist/recording",
-  "version": "11.3.1",
+  "version": "12.0.0",
   "description": "Audio recording support for waveform-playlist using AudioWorklet",
   "main": "./dist/index.js",
   "module": "./dist/index.mjs",
@@ -43,10 +43,10 @@
     "vitest": "^3.0.0"
   },
   "dependencies": {
-    "@waveform-playlist/core": "11.3.1",
-    "@waveform-playlist/playout": "11.3.1",
-    "@waveform-playlist/ui-components": "11.3.1",
-    "@waveform-playlist/worklets": "11.3.1"
+    "@waveform-playlist/core": "12.0.0",
+    "@waveform-playlist/playout": "12.0.0",
+    "@waveform-playlist/worklets": "12.0.0",
+    "@waveform-playlist/ui-components": "12.0.0"
   },
   "peerDependencies": {
     "react": "^18.0.0",