@waveform-playlist/recording 12.2.0 → 13.0.0

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Files changed (8) hide show
  1. package/README.md +153 -139
  2. package/dist/index.d.mts +35 -10
  3. package/dist/index.d.ts +35 -10
  4. package/dist/index.js +104 -52
  5. package/dist/index.js.map +1 -1
  6. package/dist/index.mjs +110 -57
  7. package/dist/index.mjs.map +1 -1
  8. package/package.json +5 -5
package/dist/index.mjs.map CHANGED
@@ -1 +1 @@
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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 { addRecordingWorkletModule } 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 // Resolved when the worklet posts the final done message after a stop command.\n const stopAckResolveRef = useRef<(() => void) | null>(null);\n // True from the start of stopRecording until it resolves. Distinct from\n // isPausedRef (user explicitly paused) — both freeze the duration tick,\n // but conflating them risks pauseRecording becoming a no-op while we're\n // stopping, or vice versa.\n const isStoppingRef = useRef<boolean>(false);\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 && !isStoppingRef.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 // Use addRecordingWorkletModule with rawContext.audioWorklet.addModule callback.\n // Can't use Tone.js addAudioWorkletModule — it caches a single _workletPromise\n // per context, silently skipping subsequent URLs.\n const rawCtx = context.rawContext as AudioContext;\n await addRecordingWorkletModule((url) => rawCtx.audioWorklet.addModule(url));\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, done } = event.data as {\n channels: Float32Array[];\n done?: boolean;\n };\n\n // Empty messages can arrive from the final stop ack when nothing was buffered.\n // Handle channels first if present, then resolve the stop barrier.\n if (channels && channels.length > 0 && channels[0] && channels[0].length > 0) {\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\n // Skip live-preview peak updates while stop is in flight. The\n // queue may have many flushes pending; updating peaks for each\n // would let the live-preview waveform grow visibly past where\n // the user stopped, only to be replaced moments later by the\n // final clip (which is shorter due to latency compensation).\n // Chunks are already pushed above so no audio is lost.\n if (stopAckResolveRef.current === null) {\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 }\n\n if (done) {\n stopAckResolveRef.current?.();\n stopAckResolveRef.current = null;\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 // Freeze the duration tick immediately so the live-preview width\n // (durationSamples = duration * sampleRate) doesn't keep growing\n // during the worklet stop handshake / queue drain. isStoppingRef\n // stops the rAF tick from rescheduling; cancelling animationFrameRef\n // kills any frame already queued.\n isStoppingRef.current = true;\n if (animationFrameRef.current !== null) {\n cancelAnimationFrame(animationFrameRef.current);\n animationFrameRef.current = null;\n }\n\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 and wait for its final flush message before reading chunks.\n // Without the await, the partial buffer at the end of recording arrives after\n // the AudioBuffer is built and is silently dropped.\n // Snapshot the worklet node once — the drain loop yields the event loop\n // for up to 250ms, during which an unmount-cleanup could null the ref.\n const node = workletNodeRef.current;\n if (node) {\n const stopAck = new Promise<void>((resolve) => {\n stopAckResolveRef.current = resolve;\n });\n let timeoutId: ReturnType<typeof setTimeout> | undefined;\n // Safety timeout (1s) — purely a circuit breaker for a truly failed\n // worklet (crashed, context closed). The drain loop below catches\n // any straggler flush messages that were queued before the timeout\n // fired, so chunks are complete even when this fires.\n const timeout = new Promise<void>((resolve) => {\n timeoutId = setTimeout(resolve, 1000);\n });\n\n node.port.postMessage({ command: 'stop' });\n await Promise.race([stopAck, timeout]);\n clearTimeout(timeoutId);\n stopAckResolveRef.current = null;\n\n // Drain the event-loop queue. During recording the worklet posts a\n // flush every ~16ms; if main was slower than that, messages back up.\n // Yield repeatedly until totalSamplesRef stabilizes for several\n // consecutive ticks — at that point the queue is empty.\n let lastSamples = -1;\n let stable = 0;\n for (let i = 0; i < 50; i++) {\n if (totalSamplesRef.current === lastSamples) {\n if (++stable >= 3) break;\n } else {\n stable = 0;\n lastSamples = totalSamplesRef.current;\n }\n await new Promise<void>((r) => setTimeout(r, 5));\n }\n\n // Null the handler so any late delivery from this worklet doesn't\n // contaminate a subsequent recording session's chunks.\n node.port.onmessage = null;\n\n // Disconnect worklet from source\n if (mediaStreamSourceRef.current) {\n try {\n mediaStreamSourceRef.current.disconnect(node);\n } catch (err) {\n console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n }\n }\n node.disconnect();\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 return null;\n }\n\n const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n setAudioBuffer(buffer);\n setDuration(buffer.duration);\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 } finally {\n // Always clear recording flags, even on failure — otherwise the UI gets\n // stuck in \"recording\" state if AudioBuffer creation throws.\n isRecordingRef.current = false;\n isPausedRef.current = false;\n isStoppingRef.current = false;\n setIsRecording(false);\n setIsPaused(false);\n setLevel(0);\n }\n }, [channelCount]);\n stopRecordingRef.current = stopRecording;\n\n // Pause recording\n const pauseRecording = useCallback(() => {\n if (isRecording && !isPaused) {\n workletNodeRef.current?.port.postMessage({ command: 'pause' });\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 workletNodeRef.current?.port.postMessage({ command: 'resume' });\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 { addMeterWorkletModule, 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 // Use addMeterWorkletModule with rawContext.audioWorklet.addModule callback.\n // Can't use Tone.js addAudioWorkletModule — it caches a single _workletPromise\n // per context, silently skipping subsequent URLs.\n const rawCtx = context.rawContext as AudioContext;\n await addMeterWorkletModule((url) => rawCtx.audioWorklet.addModule(url));\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 ? 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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, latencyOffset, ...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. 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The recorded audio could not be saved.`\n );\n console.warn(`[waveform-playlist] ${err.message}`);\n setHookError(err);\n return;\n }\n\n const selectedTrack = tracks[selectedTrackIndex];\n\n // Use the captured start time (not live currentTime which advances during overdub)\n const recordStartTimeSamples = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n let lastClipEndSample = 0;\n if (selectedTrack.clips.length > 0) {\n const endSamples = selectedTrack.clips.map(\n (clip) => clip.startSample + clip.durationSamples\n );\n lastClipEndSample = Math.max(...endSamples);\n }\n\n const startSample = Math.max(recordStartTimeSamples, lastClipEndSample);\n\n // Latency compensation: an explicit latencyOffset (seconds) replaces the\n // auto-computed outputLatency + lookAhead window. Shared resolver with the\n // live preview in PlaylistVisualization — keep both using the same value so\n // trim widths match.\n const audioContext = getGlobalAudioContext();\n const outputLatency = audioContext.outputLatency ?? 0;\n const toneContext = getGlobalContext();\n const lookAhead = toneContext.lookAhead ?? 0;\n const latencyOffsetSamples = resolveRecordingOffsetSamples({\n overrideSeconds: latencyOffset,\n outputLatency,\n lookAhead,\n sampleRate: buffer.sampleRate,\n });\n\n // Guard: very short recordings (< latency compensation) would produce negative duration\n const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n if (effectiveDuration === 0) {\n console.warn(\n '[waveform-playlist] Recording too short for latency compensation — discarding'\n );\n setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n return;\n }\n\n // Create new clip from recording\n const newClip: AudioClip = {\n id: `clip-${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, latencyOffset]);\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 { addRecordingWorkletModule } 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 // Resolved when the worklet posts the final done message after a stop command.\n const stopAckResolveRef = useRef<(() => void) | null>(null);\n // True from the start of stopRecording until it resolves. Distinct from\n // isPausedRef (user explicitly paused) — both freeze the duration tick,\n // but conflating them risks pauseRecording becoming a no-op while we're\n // stopping, or vice versa.\n const isStoppingRef = useRef<boolean>(false);\n // In-flight stop result — re-entrant stopRecording calls await this\n // instead of running a second handshake.\n const stopPromiseRef = useRef<Promise<AudioBuffer | null> | null>(null);\n // Assigned each render below (same pattern as stopRecordingRef) so the\n // stable stopRecording callback can invoke the handshake implementation.\n const finishStopRecordingRef = useRef<(() => Promise<AudioBuffer | null>) | null>(null);\n // Set synchronously at startRecording entry. isRecordingRef only flips\n // after several awaits — without this, two rapid start calls both pass\n // the guard and build two capture pipelines feeding one chunk store.\n const isStartingRef = useRef<boolean>(false);\n // Set by the unmount cleanup so a startRecording continuation that was\n // parked on an await doesn't build an ownerless capture graph.\n const isUnmountedRef = useRef<boolean>(false);\n // Mirrors the duration state for callbacks that must not close over it.\n const durationRef = useRef<number>(0);\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 && !isStoppingRef.current) {\n const elapsed = (performance.now() - startTimeRef.current) / 1000;\n durationRef.current = elapsed;\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 // Use addRecordingWorkletModule with rawContext.audioWorklet.addModule callback.\n // Can't use Tone.js addAudioWorkletModule — it caches a single _workletPromise\n // per context, silently skipping subsequent URLs.\n const rawCtx = context.rawContext as AudioContext;\n await addRecordingWorkletModule((url) => rawCtx.audioWorklet.addModule(url));\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. Resolves true when the capture pipeline started.\n const startRecording = useCallback(async (): Promise<boolean> => {\n if (isRecordingRef.current || isStartingRef.current) return false;\n if (!stream) {\n setError(new Error('No microphone stream available'));\n return false;\n }\n isStartingRef.current = true;\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 // Unmounted while awaiting resume/module load — abort before any graph\n // nodes exist. A continuation past this point would start an ownerless\n // session: the rAF loop reschedules forever and chunks accumulate with\n // nothing left to stop them.\n if (isUnmountedRef.current) return false;\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, done } = event.data as {\n channels: Float32Array[];\n done?: boolean;\n };\n\n // Empty messages can arrive from the final stop ack when nothing was buffered.\n // Handle channels first if present, then resolve the stop barrier.\n if (channels && channels.length > 0 && channels[0] && channels[0].length > 0) {\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\n // Skip live-preview peak updates while stop is in flight. The\n // queue may have many flushes pending; updating peaks for each\n // would let the live-preview waveform grow visibly past where\n // the user stopped, only to be replaced moments later by the\n // final clip (which is shorter due to latency compensation).\n // Chunks are already pushed above so no audio is lost.\n if (stopAckResolveRef.current === null) {\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 }\n\n if (done) {\n stopAckResolveRef.current?.();\n stopAckResolveRef.current = null;\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 return true;\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 return false;\n } finally {\n isStartingRef.current = false;\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 // Re-entrant stop (double-click, unmount cleanup while a stop is\n // pending): a second handshake would overwrite the ack resolver —\n // stranding the first call on its safety timeout — and post a redundant\n // stop command. Both callers observe the same in-flight stop.\n if (isStoppingRef.current) {\n return stopPromiseRef.current ?? null;\n }\n // Freeze the duration tick immediately so the live-preview width\n // (durationSamples = duration * sampleRate) doesn't keep growing\n // during the worklet stop handshake / queue drain. isStoppingRef\n // stops the rAF tick from rescheduling; cancelling animationFrameRef\n // kills any frame already queued.\n isStoppingRef.current = true;\n const stopPromise = finishStopRecordingRef.current!();\n stopPromiseRef.current = stopPromise;\n return stopPromise;\n }, []);\n\n // The actual stop handshake — only ever invoked via stopRecording's\n // re-entrancy gate above.\n const finishStopRecording = useCallback(async (): Promise<AudioBuffer | null> => {\n try {\n if (animationFrameRef.current !== null) {\n cancelAnimationFrame(animationFrameRef.current);\n animationFrameRef.current = null;\n }\n\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 and wait for its final flush message before reading chunks.\n // Without the await, the partial buffer at the end of recording arrives after\n // the AudioBuffer is built and is silently dropped.\n // Snapshot the worklet node once — the drain loop yields the event loop\n // for up to 250ms, during which an unmount-cleanup could null the ref.\n const node = workletNodeRef.current;\n if (node) {\n const stopAck = new Promise<void>((resolve) => {\n stopAckResolveRef.current = resolve;\n });\n let timeoutId: ReturnType<typeof setTimeout> | undefined;\n // Safety timeout (1s) — purely a circuit breaker for a truly failed\n // worklet (crashed, context closed). The drain loop below catches\n // any straggler flush messages that were queued before the timeout\n // fired, so chunks are complete even when this fires.\n const timeout = new Promise<void>((resolve) => {\n timeoutId = setTimeout(resolve, 1000);\n });\n\n node.port.postMessage({ command: 'stop' });\n await Promise.race([stopAck, timeout]);\n clearTimeout(timeoutId);\n stopAckResolveRef.current = null;\n\n // Drain the event-loop queue. During recording the worklet posts a\n // flush every ~16ms; if main was slower than that, messages back up.\n // Yield repeatedly until totalSamplesRef stabilizes for several\n // consecutive ticks — at that point the queue is empty.\n let lastSamples = -1;\n let stable = 0;\n for (let i = 0; i < 50; i++) {\n if (totalSamplesRef.current === lastSamples) {\n if (++stable >= 3) break;\n } else {\n stable = 0;\n lastSamples = totalSamplesRef.current;\n }\n await new Promise<void>((r) => setTimeout(r, 5));\n }\n\n // Null the handler so any late delivery from this worklet doesn't\n // contaminate a subsequent recording session's chunks.\n node.port.onmessage = null;\n\n // Disconnect worklet from source\n if (mediaStreamSourceRef.current) {\n try {\n mediaStreamSourceRef.current.disconnect(node);\n } catch (err) {\n console.warn('[waveform-playlist] Source disconnect during stop:', String(err));\n }\n }\n node.disconnect();\n\n // Session fully torn down — null the refs so the unmount cleanup\n // doesn't post a redundant stop to the already-stopped worklet.\n workletNodeRef.current = null;\n mediaStreamSourceRef.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 return null;\n }\n\n const buffer = createAudioBuffer(rawContext, channelData, rawContext.sampleRate, numChannels);\n\n setAudioBuffer(buffer);\n durationRef.current = buffer.duration;\n setDuration(buffer.duration);\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 } finally {\n // Always clear recording flags, even on failure — otherwise the UI gets\n // stuck in \"recording\" state if AudioBuffer creation throws.\n isRecordingRef.current = false;\n isPausedRef.current = false;\n isStoppingRef.current = false;\n setIsRecording(false);\n setIsPaused(false);\n setLevel(0);\n }\n }, [channelCount]);\n stopRecordingRef.current = stopRecording;\n finishStopRecordingRef.current = finishStopRecording;\n\n // Pause recording — guards read refs, not state: a pause in the same tick\n // as start/resume closes over stale state and silently no-ops while the\n // worklet keeps capturing (only visible in the resulting buffer length).\n const pauseRecording = useCallback(() => {\n if (isRecordingRef.current && !isPausedRef.current && !isStoppingRef.current) {\n workletNodeRef.current?.port.postMessage({ command: 'pause' });\n if (animationFrameRef.current !== null) {\n cancelAnimationFrame(animationFrameRef.current);\n animationFrameRef.current = null;\n }\n isPausedRef.current = true;\n setIsPaused(true);\n }\n }, []);\n\n // Resume recording — same ref-based guards as pauseRecording.\n const resumeRecording = useCallback(() => {\n if (isRecordingRef.current && isPausedRef.current && !isStoppingRef.current) {\n workletNodeRef.current?.port.postMessage({ command: 'resume' });\n isPausedRef.current = false;\n setIsPaused(false);\n startTimeRef.current = performance.now() - durationRef.current * 1000;\n startDurationLoop();\n }\n }, [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 // Reset on (re)mount — StrictMode dev remounts reuse the same hook\n // instance, so a flag left true by the probe-unmount's cleanup would\n // make startRecording abort forever.\n isUnmountedRef.current = false;\n return () => {\n // Abort any startRecording continuation parked on an await — it must\n // not build the capture graph after unmount.\n isUnmountedRef.current = true;\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, useRef } from 'react';\nimport { enumerateMicrophones } from '@waveform-playlist/core';\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 // Mirrors the stream state so unmount cleanup and overlapping requests\n // operate on the CURRENT stream, not a stale closure capture.\n const streamRef = useRef<MediaStream | null>(null);\n // Generation token: bumped by every requestAccess (and by stopStream /\n // unmount). A getUserMedia grant belonging to an older generation is a\n // stale request — its stream must be stopped, not installed, or the mic\n // stays hot with no owner (permission prompt resolving after unmount,\n // or the slower of two overlapping requests).\n const requestGenerationRef = useRef(0);\n // Set by unmount cleanup (plain assignment — the exhaustive-deps rule\n // rejects read-modify-writes like a generation bump in cleanup functions).\n const isUnmountedRef = useRef(false);\n\n // Enumerate audio input devices — delegates to the framework-agnostic core\n // helper (redaction-aware fallback labels, SSR/insecure-context safe).\n const enumerateDevices = useCallback(async () => {\n try {\n const { devices: audioInputs } = await enumerateMicrophones();\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 const generation = ++requestGenerationRef.current;\n setIsLoading(true);\n setError(null);\n\n try {\n // Stop and clear the existing stream. Clearing state (not just\n // stopping tracks) matters on failure: consumers left holding a\n // stopped stream silently record silence with no error.\n if (streamRef.current) {\n streamRef.current.getTracks().forEach((track) => track.stop());\n streamRef.current = null;\n setStream(null);\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\n // Superseded by a newer request, stopStream, or unmount while the\n // permission prompt was open — stop the just-granted stream instead\n // of installing it (a hot mic nobody owns).\n if (generation !== requestGenerationRef.current || isUnmountedRef.current) {\n newStream.getTracks().forEach((track) => track.stop());\n return;\n }\n\n streamRef.current = newStream;\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 if (generation === requestGenerationRef.current) {\n setError(err instanceof Error ? err : new Error('Failed to access microphone'));\n setHasPermission(false);\n }\n } finally {\n if (generation === requestGenerationRef.current) {\n setIsLoading(false);\n }\n }\n },\n [enumerateDevices]\n );\n\n // Stop the stream and revoke access\n const stopStream = useCallback(() => {\n // Invalidate any in-flight request so a pending grant can't re-arm the mic\n requestGenerationRef.current++;\n if (streamRef.current) {\n streamRef.current.getTracks().forEach((track) => track.stop());\n streamRef.current = null;\n setStream(null);\n setHasPermission(false);\n }\n }, []);\n\n // Check initial permission state, enumerate devices, and listen for hot-plug changes\n useEffect(() => {\n // Reset on (re)mount — StrictMode dev remounts reuse the same hook\n // instance, so a flag left true by the probe-unmount's cleanup would\n // stop every future getUserMedia grant as \"stale\".\n isUnmountedRef.current = false;\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 // Invalidate any in-flight request (a grant arriving after unmount is\n // stopped by the isUnmounted check) and stop the current stream.\n isUnmountedRef.current = true;\n if (streamRef.current) {\n streamRef.current.getTracks().forEach((track) => track.stop());\n streamRef.current = null;\n }\n };\n }, [enumerateDevices]);\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 { addMeterWorkletModule, 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 the metered channel\n * count: the mic's auto-detected channels, with mono mirrored up to the\n * requested `channelCount`. A mic with MORE channels than `channelCount`\n * yields one entry per actual channel.\n * True peak: max absolute sample value per analysis frame.\n */\n levels: number[];\n\n /**\n * Per-channel held peak levels (0-1). Same length contract as `levels`.\n */\n peakLevels: number[];\n\n /**\n * Per-channel RMS levels (0-1). Same length contract as `levels`.\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 // Use addMeterWorkletModule with rawContext.audioWorklet.addModule callback.\n // Can't use Tone.js addAudioWorkletModule — it caches a single _workletPromise\n // per context, silently skipping subsequent URLs.\n const rawCtx = context.rawContext as AudioContext;\n await addMeterWorkletModule((url) => rawCtx.audioWorklet.addModule(url));\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 // MeterMessage layout: [0..N-1] peak, [N..2N-1] rms\n const data = event.data as MeterMessage;\n const workletChannels = data.length / 2;\n const smoothed = smoothedPeakRef.current;\n\n const peakValues: number[] = [];\n const rmsValues: number[] = [];\n\n for (let ch = 0; ch < workletChannels; ch++) {\n smoothed[ch] = Math.max(data[ch], (smoothed[ch] ?? 0) * PEAK_DECAY);\n peakValues.push(gainToNormalized(smoothed[ch]));\n rmsValues.push(gainToNormalized(data[workletChannels + ch]));\n }\n\n // Mirror mono to fill requested channelCount\n const mirroredPeaks =\n workletChannels < channelCount ? new Array(channelCount).fill(peakValues[0]) : peakValues;\n const mirroredRms =\n workletChannels < 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 // Setup succeeded — clear any error from a previous failed attempt so\n // the UI doesn't show a permanent failure while metering works.\n setMeterError(null);\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 {\n type ClipTrack,\n type AudioClip,\n carveClipRange,\n resolveRecordingOffsetSamples,\n} 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 * Punch-in semantics (#579): the recorded clip lands exactly at this\n * position and REPLACES any existing clip content it overlaps — partial\n * overlaps are trimmed, fully-covered clips removed, spanning clips split.\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 /**\n * Latency offset to skip at the start of the recording, in **seconds**.\n * Absolute replacement for the auto-computed `outputLatency + lookAhead` value\n * — use this to apply an externally-measured round-trip latency. `0` disables\n * compensation; when omitted, the auto-computed value is used. Pass the same\n * value into the provider's `recordingState.latencyOffset` so the live preview\n * matches the finalized clip. Negative or non-finite values are treated as 0.\n */\n latencyOffset?: 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 /**\n * Per-channel peak levels (0-1). Array length matches the metered channel\n * count (auto-detected mic channels; mono mirrored up to channelCount —\n * see UseMicrophoneLevelReturn.levels).\n */\n levels: number[];\n /** Per-channel held peak levels (0-1). Same length contract as `levels`. */\n peakLevels: number[];\n /** Per-channel RMS levels (0-1). Same length contract as `levels`. */\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 /** Start capturing on the selected track. Resolves `true` when the capture\n * pipeline actually started — callers that synchronize playback with\n * recording (overdub) should check this before starting playback. */\n startRecording: () => Promise<boolean>;\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, latencyOffset, ...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 // Latest tracks for reads that happen AFTER an await — the stop handshake\n // can take >1s, and finalizing against a closure capture would discard any\n // track edits that landed in that window.\n const tracksRef = useRef(tracks);\n tracksRef.current = tracks;\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 (): Promise<boolean> => {\n if (!selectedTrackIdRef.current) {\n setHookError(\n new Error('Cannot start recording: no track selected. Select or create a track first.')\n );\n return false;\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 return await startRec();\n } catch (err) {\n setHookError(err instanceof Error ? err : new Error(String(err)));\n return false;\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. Read tracks through the\n // ref — the awaited stop handshake above can span >1s, and the closure's\n // tracks capture would silently revert concurrent edits.\n const trackId = selectedTrackIdRef.current;\n const currentTracks = tracksRef.current;\n if (buffer && trackId) {\n const selectedTrackIndex = currentTracks.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] ${err.message}`);\n setHookError(err);\n return;\n }\n\n // Use the captured start time (not live currentTime which advances during overdub).\n // Punch-in replace (#579): the take lands exactly here; overlapped\n // content is carved out below when the clip list is assembled.\n const startSample = Math.floor(recordingStartTimeRef.current * buffer.sampleRate);\n\n // Latency compensation: an explicit latencyOffset (seconds) replaces the\n // auto-computed outputLatency + lookAhead window. Shared resolver with the\n // live preview in PlaylistVisualization — keep both using the same value so\n // trim widths match.\n const audioContext = getGlobalAudioContext();\n const outputLatency = audioContext.outputLatency ?? 0;\n const toneContext = getGlobalContext();\n const lookAhead = toneContext.lookAhead ?? 0;\n const latencyOffsetSamples = resolveRecordingOffsetSamples({\n overrideSeconds: latencyOffset,\n outputLatency,\n lookAhead,\n sampleRate: buffer.sampleRate,\n });\n\n // Guard: very short recordings (< latency compensation) would produce negative duration\n const effectiveDuration = Math.max(0, buffer.length - latencyOffsetSamples);\n if (effectiveDuration === 0) {\n console.warn(\n '[waveform-playlist] Recording too short for latency compensation — discarding'\n );\n setHookError(new Error('Recording was too short to save. Try recording for longer.'));\n return;\n }\n\n // Create new clip from recording\n const newClip: AudioClip = {\n id: `clip-${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 — the recorded clip owns [startSample,\n // startSample + effectiveDuration): trim/remove/split whatever it overlaps.\n const newTracks = currentTracks.map((track, index) => {\n if (index === selectedTrackIndex) {\n return {\n ...track,\n clips: [\n ...carveClipRange(track.clips, startSample, startSample + effectiveDuration),\n newClip,\n ],\n };\n }\n return track;\n });\n\n setTracks(newTracks);\n }\n }, [setTracks, stopRec, latencyOffset]);\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 // Mirror isRecording for the device-switch guards below — reading the\n // state directly would churn the callbacks every recording toggle.\n const isRecordingRef = useRef(isRecording);\n isRecordingRef.current = isRecording;\n\n // Request microphone access\n const requestMicAccess = useCallback(async () => {\n // requestAccess stops the current stream's tracks, and track.stop() fires\n // no 'ended' event — the recording's source is never rewired, so the rest\n // of the take silently records silence. Refuse while recording.\n if (isRecordingRef.current) {\n setHookError(new Error('Cannot request microphone access while recording. Stop first.'));\n return;\n }\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 // Same silent-silence hazard as requestMicAccess above.\n if (isRecordingRef.current) {\n setHookError(new Error('Cannot switch microphone while recording. Stop first.'));\n return;\n }\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/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "@waveform-playlist/recording",
3
- "version": "12.2.0",
3
+ "version": "13.0.0",
4
4
  "description": "Audio recording support for waveform-playlist using AudioWorklet",
5
5
  "main": "./dist/index.js",
6
6
  "module": "./dist/index.mjs",
@@ -47,10 +47,10 @@
47
47
  "vitest": "^3.2.6"
48
48
  },
49
49
  "dependencies": {
50
- "@waveform-playlist/core": "12.3.0",
51
- "@waveform-playlist/playout": "12.4.0",
52
- "@waveform-playlist/worklets": "12.2.0",
53
- "@waveform-playlist/ui-components": "13.0.1"
50
+ "@waveform-playlist/core": "12.6.0",
51
+ "@waveform-playlist/ui-components": "13.1.2",
52
+ "@waveform-playlist/playout": "12.5.1",
53
+ "@waveform-playlist/worklets": "13.0.0"
54
54
  },
55
55
  "peerDependencies": {
56
56
  "react": "^18.0.0",