npm - @siteed/expo-audio-studio - Versions diffs - 2.8.0 → 2.8.1 - Mend

@siteed/expo-audio-studio 2.8.0 → 2.8.1

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package/build/cjs/workers/InlineFeaturesExtractor.web.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"InlineFeaturesExtractor.web.js","sourceRoot":"","sources":["../../../src/workers/InlineFeaturesExtractor.web.tsx"],"names":[],"mappings":";;;AAAA,yEAAyE;AAC5D,QAAA,uBAAuB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CA+0BtC,CAAA","sourcesContent":["// packages/expo-audio-studio/src/workers/InlineFeaturesExtractor.web.tsx\nexport const InlineFeaturesExtractor = `\n// Constants\nconst N_FFT = 1024; // Default FFT size\nconst MAX_FFT_SIZE = 8192; // Maximum FFT size to prevent memory issues\nconst N_CHROMA = 12;\n\n// FFT Implementation with normalized Hann window\nfunction FFT(n) {\n this.n = n;\n this.cosTable = new Float32Array(n / 2);\n this.sinTable = new Float32Array(n / 2);\n this.hannWindow = new Float32Array(n);\n \n // Match Android implementation with precomputed tables\n const normalizationFactor = Math.sqrt(2.0 / n);\n for (var i = 0; i < n / 2; i++) {\n this.cosTable[i] = Math.cos(2.0 * Math.PI * i / n);\n this.sinTable[i] = Math.sin(2.0 * Math.PI * i / n);\n }\n \n // Precompute normalized Hann window to match Android\n for (var i = 0; i < n; i++) {\n this.hannWindow[i] = normalizationFactor * 0.5 * (1 - Math.cos(2.0 * Math.PI * i / (n - 1)));\n }\n}\n\nFFT.prototype.transform = function(data) {\n const n = data.length;\n \n // Validate input length is power of 2\n if ((n & (n - 1)) !== 0) {\n throw new Error('FFT length must be power of 2');\n }\n\n // Use iterative bit reversal instead of recursive\n const bitReversedIndices = new Uint32Array(n);\n for (let i = 0; i < n; i++) {\n let reversed = 0;\n let j = i;\n let bits = Math.log2(n);\n while (bits--) {\n reversed = (reversed << 1) | (j & 1);\n j >>= 1;\n }\n bitReversedIndices[i] = reversed;\n }\n\n // Apply bit reversal\n for (let i = 0; i < n; i++) {\n const j = bitReversedIndices[i];\n if (i < j) {\n const temp = data[i];\n data[i] = data[j];\n data[j] = temp;\n }\n }\n\n // Iterative FFT computation with optimized memory usage\n for (let step = 1; step < n; step <<= 1) {\n const jump = step << 1;\n const angleStep = Math.PI / step;\n\n for (let group = 0; group < n; group += jump) {\n for (let pair = group; pair < group + step; pair++) {\n const match = pair + step;\n const angle = angleStep * (pair - group);\n \n const currentCos = Math.cos(angle);\n const currentSin = Math.sin(angle);\n\n const real = currentCos * data[match] - currentSin * data[match + 1];\n const imag = currentCos * data[match + 1] + currentSin * data[match];\n\n data[match] = data[pair] - real;\n data[match + 1] = data[pair + 1] - imag;\n data[pair] += real;\n data[pair + 1] += imag;\n }\n }\n }\n};\n\n// Add realInverse method\nFFT.prototype.realInverse = function(powerSpectrum, output) {\n const n = powerSpectrum.length;\n const complexData = new Float32Array(n * 2);\n \n // Copy power spectrum to complex format\n for (let i = 0; i < n/2 + 1; i++) {\n complexData[2 * i] = powerSpectrum[i];\n if (2 * i + 1 < complexData.length) {\n complexData[2 * i + 1] = 0;\n }\n }\n \n // Conjugate for inverse FFT\n for (let i = 0; i < n; i++) {\n if (2 * i + 1 < complexData.length) {\n complexData[2 * i + 1] = -complexData[2 * i + 1];\n }\n }\n \n this.transform(complexData);\n \n // Copy real part to output and scale\n for (let i = 0; i < n; i++) {\n output[i] = complexData[2 * i] / n;\n }\n};\n\n// Add helper functions to match Android\nfunction nextPowerOfTwo(n) {\n let value = 1;\n while (value < n) {\n value *= 2;\n }\n return value;\n}\n\nfunction applyHannWindow(samples) {\n const output = new Float32Array(samples.length);\n for (let i = 0; i < samples.length; i++) {\n const multiplier = 0.5 * (1 - Math.cos(2 * Math.PI * i / (samples.length - 1)));\n output[i] = samples[i] * multiplier;\n }\n return output;\n}\n\n// Update spectral feature computation to match Android\nfunction computeSpectralFeatures(segment, sampleRate, featureOptions = {}) {\n try {\n // Early return if no spectral features are requested\n if (!featureOptions.spectralCentroid && \n !featureOptions.spectralFlatness && \n !featureOptions.spectralRollOff && \n !featureOptions.spectralBandwidth &&\n !featureOptions.magnitudeSpectrum) {\n return {\n centroid: 0,\n flatness: 0,\n rollOff: 0,\n bandwidth: 0,\n magnitudeSpectrum: []\n };\n }\n\n // Ensure we have valid data\n if (!segment || segment.length === 0) {\n throw new Error('Invalid segment data');\n }\n\n // Process in fixed-size chunks\n const chunkSize = N_FFT;\n const numChunks = Math.ceil(segment.length / chunkSize);\n \n let results = {\n centroid: 0,\n flatness: 0,\n rollOff: 0,\n bandwidth: 0,\n magnitudeSpectrum: new Float32Array(N_FFT / 2 + 1).fill(0)\n };\n \n let validChunks = 0;\n \n // Iterate through chunks\n for (let i = 0; i < numChunks; i++) {\n const start = i * chunkSize;\n const end = Math.min(start + chunkSize, segment.length);\n const chunk = segment.slice(start, end);\n \n if (chunk.length < N_FFT / 4) continue; // Skip very small chunks\n\n // Process the chunk\n const paddedChunk = new Float32Array(N_FFT);\n paddedChunk.set(applyHannWindow(chunk));\n\n const fft = new FFT(N_FFT);\n fft.transform(paddedChunk);\n\n // Calculate magnitude spectrum\n const chunkMagnitudeSpectrum = new Float32Array(N_FFT / 2 + 1);\n let hasSignal = false;\n \n for (let j = 0; j < N_FFT / 2; j++) {\n const re = paddedChunk[2 * j];\n const im = paddedChunk[2 * j + 1];\n const magnitude = Math.sqrt(re * re + im * im);\n chunkMagnitudeSpectrum[j] = magnitude;\n if (magnitude > Number.EPSILON) hasSignal = true;\n }\n \n if (!hasSignal) continue;\n validChunks++;\n\n // Accumulate results\n if (featureOptions.spectralCentroid) {\n const centroid = computeSpectralCentroid(chunkMagnitudeSpectrum, sampleRate);\n if (!isNaN(centroid)) results.centroid += centroid;\n }\n \n if (featureOptions.spectralFlatness) {\n const flatness = computeSpectralFlatness(chunkMagnitudeSpectrum);\n if (!isNaN(flatness)) results.flatness += flatness;\n }\n \n if (featureOptions.spectralRollOff) {\n const rolloff = computeSpectralRollOff(chunkMagnitudeSpectrum, sampleRate);\n if (!isNaN(rolloff)) results.rollOff += rolloff;\n }\n \n if (featureOptions.spectralBandwidth && !isNaN(results.centroid)) {\n const bandwidth = computeSpectralBandwidth(chunkMagnitudeSpectrum, sampleRate, results.centroid);\n if (!isNaN(bandwidth)) results.bandwidth += bandwidth;\n }\n \n if (featureOptions.magnitudeSpectrum) {\n for (let j = 0; j < results.magnitudeSpectrum.length; j++) {\n results.magnitudeSpectrum[j] += chunkMagnitudeSpectrum[j];\n }\n }\n }\n\n // Average the accumulated results\n if (validChunks > 0) {\n results.centroid /= validChunks;\n results.flatness /= validChunks;\n results.rollOff /= validChunks;\n results.bandwidth /= validChunks;\n \n if (featureOptions.magnitudeSpectrum) {\n for (let i = 0; i < results.magnitudeSpectrum.length; i++) {\n results.magnitudeSpectrum[i] /= validChunks;\n }\n }\n }\n\n return results;\n } catch (error) {\n console.error('[Worker] Spectral feature computation error:', error);\n return {\n centroid: 0,\n flatness: 0,\n rollOff: 0,\n bandwidth: 0,\n magnitudeSpectrum: []\n };\n }\n}\n\nfunction computeSpectralCentroid(magnitudeSpectrum, sampleRate) {\n const sum = magnitudeSpectrum.reduce((a, b) => a + (b || 0), 0);\n if (sum <= Number.EPSILON) return 0;\n \n const weightedSum = magnitudeSpectrum.reduce((acc, value, index) => \n acc + (index * (sampleRate / N_FFT) * (value || 0)), 0);\n \n return weightedSum / sum;\n}\n\nfunction computeSpectralFlatness(powerSpectrum) {\n // Add small epsilon to avoid log(0)\n const epsilon = Number.EPSILON;\n const validSpectrum = powerSpectrum.map(v => Math.max(v, epsilon));\n \n const geometricMean = Math.exp(\n validSpectrum\n .map(v => Math.log(v))\n .reduce((a, b) => a + b) / validSpectrum.length\n );\n \n const arithmeticMean =\n validSpectrum.reduce((a, b) => a + b) / validSpectrum.length;\n \n return geometricMean / arithmeticMean;\n}\n\nfunction computeSpectralRollOff(magnitudeSpectrum, sampleRate) {\n const totalEnergy = magnitudeSpectrum.reduce((a, b) => a + b, 0);\n const rollOffThreshold = totalEnergy * 0.85;\n let cumulativeEnergy = 0;\n\n for (let i = 0; i < magnitudeSpectrum.length; i++) {\n cumulativeEnergy += magnitudeSpectrum[i];\n if (cumulativeEnergy >= rollOffThreshold) {\n return (i / magnitudeSpectrum.length) * (sampleRate / 2);\n }\n }\n\n return 0;\n}\n\nfunction computeSpectralBandwidth(magnitudeSpectrum, sampleRate, centroid) {\n const sum = magnitudeSpectrum.reduce((a, b) => a + (b || 0), 0);\n if (sum <= Number.EPSILON) return 0;\n\n const weightedSum = magnitudeSpectrum.reduce(\n (acc, value, index) => {\n const freq = index * sampleRate / (2 * magnitudeSpectrum.length);\n return acc + (value || 0) * Math.pow(freq - centroid, 2);\n }, 0\n );\n\n return Math.sqrt(weightedSum / sum);\n}\n\nfunction computeChroma(segmentData, sampleRate) {\n // Ensure we have valid input data\n if (!segmentData || segmentData.length === 0) {\n return new Array(N_CHROMA).fill(0);\n }\n\n const fftLength = nextPowerOfTwo(Math.max(segmentData.length, N_FFT));\n const windowed = applyHannWindow(segmentData);\n const padded = new Float32Array(fftLength);\n padded.set(windowed.slice(0, Math.min(windowed.length, fftLength)));\n\n const fft = new FFT(fftLength);\n try {\n fft.transform(padded);\n } catch (e) {\n console.error('[Worker] FFT transform failed in chromagram:', e);\n return new Array(N_CHROMA).fill(0);\n }\n\n const chroma = new Float32Array(N_CHROMA).fill(0);\n const freqsPerBin = sampleRate / fftLength;\n let totalEnergy = 0;\n\n // First pass: compute magnitudes and total energy\n for (let i = 0; i < fftLength / 2; i++) {\n const freq = i * freqsPerBin;\n if (freq > 20) { // Only consider frequencies above 20 Hz\n const re = padded[2 * i];\n const im = padded[2 * i + 1] || 0;\n const magnitude = Math.sqrt(re * re + im * im);\n \n if (magnitude > Number.EPSILON) {\n // Use a more stable pitch class calculation\n const midiNote = 69 + 12 * Math.log2(freq / 440.0);\n const pitchClass = Math.round(midiNote) % 12;\n \n if (pitchClass >= 0 && pitchClass < 12) {\n chroma[pitchClass] += magnitude;\n totalEnergy += magnitude;\n }\n }\n }\n }\n\n // Normalize chroma values only if we have energy\n if (totalEnergy > Number.EPSILON) {\n for (let i = 0; i < N_CHROMA; i++) {\n chroma[i] = chroma[i] / totalEnergy;\n }\n }\n\n // Convert to regular array and ensure no NaN values\n return Array.from(chroma, v => isNaN(v) ? 0 : v);\n}\n\nfunction extractHNR(segmentData) {\n const frameSize = segmentData.length;\n const autocorrelation = new Float32Array(frameSize);\n\n // Compute the autocorrelation iteratively\n for (let i = 0; i < frameSize; i++) {\n let sum = 0;\n for (let j = 0; j < frameSize - i; j++) {\n sum += segmentData[j] * segmentData[j + i];\n }\n autocorrelation[i] = sum;\n }\n\n // Find the maximum autocorrelation value iteratively\n let maxAutocorrelation = -Infinity;\n for (let i = 1; i < autocorrelation.length; i++) {\n if (autocorrelation[i] > maxAutocorrelation) {\n maxAutocorrelation = autocorrelation[i];\n }\n }\n\n // Compute the HNR\n return autocorrelation[0] !== 0\n ? 10 * Math.log10(maxAutocorrelation / (autocorrelation[0] - maxAutocorrelation))\n : 0;\n}\n\nfunction estimatePitch(segment, sampleRate) {\n // Early validation\n if (!segment || segment.length < 2 || !sampleRate) return 0;\n\n try {\n // Apply Hann window\n const windowed = applyHannWindow(segment);\n\n // Pad for FFT\n const fftLength = nextPowerOfTwo(segment.length * 2);\n const padded = new Float32Array(fftLength);\n padded.set(windowed);\n\n // Perform FFT\n const fft = new FFT(fftLength);\n fft.transform(padded);\n\n // Compute power spectrum\n const powerSpectrum = new Float32Array(fftLength / 2 + 1);\n for (let i = 0; i <= fftLength / 2; i++) {\n const re = padded[2 * i];\n const im = padded[2 * i + 1] || 0;\n powerSpectrum[i] = re * re + im * im;\n }\n\n // Find peak frequency\n let maxPower = 0;\n let peakIndex = 0;\n const minFreq = 50; // Minimum frequency to consider (Hz)\n const maxFreq = 1000; // Maximum frequency to consider (Hz)\n const minBin = Math.floor(minFreq * fftLength / sampleRate);\n const maxBin = Math.ceil(maxFreq * fftLength / sampleRate);\n\n for (let i = minBin; i <= maxBin; i++) {\n if (powerSpectrum[i] > maxPower) {\n maxPower = powerSpectrum[i];\n peakIndex = i;\n }\n }\n\n // Convert peak index to frequency\n const fundamentalFreq = peakIndex * sampleRate / fftLength;\n\n // Return 0 if the detected frequency is outside reasonable bounds\n return (fundamentalFreq >= minFreq && fundamentalFreq <= maxFreq) ? \n fundamentalFreq : 0;\n\n } catch (error) {\n console.error('[Worker] Pitch estimation error:', error);\n return 0;\n }\n}\n\n// Unique ID counter - the only state we need to maintain\nlet uniqueIdCounter = 0\nlet lastEmitTime = Date.now()\n\nself.onmessage = function (event) {\n // Extract enableLogging early so we can use it consistently\n const enableLogging = event.data.enableLogging || false;\n \n // Create consistent logger that only logs when enabled\n const logger = enableLogging ? {\n debug: (...args) => console.debug('[Worker]', ...args),\n log: (...args) => console.log('[Worker]', ...args),\n warn: (...args) => console.warn('[Worker]', ...args),\n error: (...args) => console.error('[Worker]', ...args)\n } : {\n debug: () => {},\n log: () => {},\n warn: () => {},\n error: () => {}\n };\n \n // Check if this is a reset command\n if (event.data.command === 'resetCounter') {\n const newValue = event.data.value;\n logger.log('Reset counter request received with value:', newValue);\n \n // Always respect explicit resets through the resetCounter command\n uniqueIdCounter = typeof newValue === 'number' ? newValue : 0;\n logger.log('Counter explicitly set to:', uniqueIdCounter);\n \n return; // Exit early, don't process audio\n }\n\n // Regular audio processing\n const {\n channelData,\n sampleRate,\n segmentDurationMs,\n algorithm,\n bitDepth,\n fullAudioDurationMs,\n numberOfChannels,\n features: _features,\n intervalAnalysis = 500,\n } = event.data\n\n // Calculate subChunkStartTime safely, defaulting to 0 if fullAudioDurationMs is not a valid number\n const subChunkStartTime = (typeof fullAudioDurationMs === 'number' && !isNaN(fullAudioDurationMs) && fullAudioDurationMs >= 0)\n ? fullAudioDurationMs / 1000\n : 0;\n\n const features = _features || {}\n const bytesPerSample = bitDepth / 8; // Calculate bytes per sample\n\n const SILENCE_THRESHOLD = 0.01\n const MIN_SILENCE_DURATION = 1.5 * sampleRate // 1.5 seconds of silence\n const SPEECH_INERTIA_DURATION = 0.1 * sampleRate // Speech inertia duration in samples\n const RMS_THRESHOLD = 0.01\n const ZCR_THRESHOLD = 0.1\n\n // Placeholder functions for feature extraction\n const extractMFCC = (segmentData, sampleRate) => {\n // Implement MFCC extraction logic here\n return []\n }\n\n const extractSpectralCentroid = (segmentData, sampleRate) => {\n const magnitudeSpectrum = segmentData.map((v) => v * v)\n const sum = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n if (sum === 0) return 0\n\n const weightedSum = magnitudeSpectrum.reduce(\n (acc, value, index) => acc + index * value,\n 0\n )\n return (\n ((weightedSum / sum) * (sampleRate / 2)) / magnitudeSpectrum.length\n )\n }\n\n const extractSpectralFlatness = (segmentData) => {\n const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n const geometricMean = Math.exp(\n magnitudeSpectrum\n .map((v) => Math.log(v + Number.MIN_VALUE))\n .reduce((a, b) => a + b) / magnitudeSpectrum.length\n )\n const arithmeticMean =\n magnitudeSpectrum.reduce((a, b) => a + b) / magnitudeSpectrum.length\n return arithmeticMean === 0 ? 0 : geometricMean / arithmeticMean\n }\n\n const extractSpectralRollOff = (segmentData, sampleRate) => {\n const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n const totalEnergy = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n const rollOffThreshold = totalEnergy * 0.85\n let cumulativeEnergy = 0\n\n for (let i = 0; i < magnitudeSpectrum.length; i++) {\n cumulativeEnergy += magnitudeSpectrum[i]\n if (cumulativeEnergy >= rollOffThreshold) {\n return (i / magnitudeSpectrum.length) * (sampleRate / 2)\n }\n }\n\n return 0\n }\n\n const extractSpectralBandwidth = (segmentData, sampleRate) => {\n const centroid = extractSpectralCentroid(segmentData, sampleRate)\n const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n const sum = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n if (sum === 0) return 0\n\n const weightedSum = magnitudeSpectrum.reduce(\n (acc, value, index) => acc + value * Math.pow(index - centroid, 2),\n 0\n )\n return Math.sqrt(weightedSum / sum)\n }\n\n const extractChromagram = (segmentData, sampleRate) => {\n return [] // TODO implement\n }\n\n /**\n * Creates a features object based on requested features\n */\n function createFeaturesObject(\n features,\n maxAmp,\n rms,\n sumSquares,\n zeroCrossings,\n remainingSamples,\n spectralFeatures,\n channelData,\n startIdx,\n endIdx,\n sampleRate,\n numberOfChannels,\n bytesPerSample\n ) {\n // If no features are requested, return undefined\n if (!Object.values(features).some(function(v) { return v; })) {\n return undefined;\n }\n\n const result = {};\n \n if (features.energy) {\n result.energy = sumSquares;\n }\n if (features.rms) {\n result.rms = rms;\n }\n // Always include min/max amplitude if any features are requested\n result.minAmplitude = -maxAmp;\n result.maxAmplitude = maxAmp;\n \n if (features.zcr) {\n result.zcr = zeroCrossings / remainingSamples;\n }\n if (features.spectralCentroid) {\n result.spectralCentroid = spectralFeatures.centroid;\n }\n if (features.spectralFlatness) {\n result.spectralFlatness = spectralFeatures.flatness;\n }\n if (features.spectralRolloff) {\n result.spectralRolloff = spectralFeatures.rollOff;\n }\n if (features.spectralBandwidth) {\n result.spectralBandwidth = spectralFeatures.bandwidth;\n }\n if (features.chromagram) {\n result.chromagram = computeChroma(channelData.slice(startIdx, endIdx), sampleRate);\n }\n if (features.hnr) {\n result.hnr = extractHNR(channelData.slice(startIdx, endIdx));\n }\n if (features.pitch) {\n result.pitch = estimatePitch(channelData.slice(startIdx, endIdx), sampleRate);\n }\n \n return result;\n }\n\n function extractWaveform(\n channelData,\n sampleRate,\n segmentDurationMs,\n numberOfChannels,\n bytesPerSample\n ) {\n const logger = enableLogging ? {\n debug: (...args) => console.debug('[Worker]', ...args),\n log: (...args) => console.log('[Worker]', ...args),\n error: (...args) => console.error('[Worker]', ...args)\n } : {\n debug: () => {},\n log: () => {},\n error: () => {}\n }\n\n // Calculate amplitude range\n let min = Infinity\n let max = -Infinity\n for (let i = 0; i < channelData.length; i++) {\n min = Math.min(min, channelData[i])\n max = Math.max(max, channelData[i])\n }\n\n const totalSamples = channelData.length\n const durationMs = (totalSamples / sampleRate) * 1000\n \n // Calculate fixed segment sizes\n const samplesPerSegment = Math.floor(sampleRate * (segmentDurationMs / 1000));\n const numPoints = Math.floor(totalSamples / samplesPerSegment);\n const remainingSamples = totalSamples % samplesPerSegment;\n\n const dataPoints = []\n\n // Process full segments\n for (let i = 0; i < numPoints; i++) {\n const startIdx = i * samplesPerSegment\n const endIdx = startIdx + samplesPerSegment\n \n let sumSquares = 0\n let maxAmp = 0\n let zeroCrossings = 0\n\n // Calculate segment features\n for (let j = startIdx; j < endIdx; j++) {\n const value = channelData[j]\n sumSquares += value * value\n maxAmp = Math.max(maxAmp, Math.abs(value))\n if (j > 0 && value * channelData[j - 1] < 0) {\n zeroCrossings++\n }\n }\n\n const rms = Math.sqrt(sumSquares / samplesPerSegment)\n const startTime = subChunkStartTime + (startIdx / sampleRate)\n const endTime = subChunkStartTime + (endIdx / sampleRate)\n // Calculate byte positions correctly based on numberOfChannels and bytesPerSample\n const startPosition = startIdx * numberOfChannels * bytesPerSample\n const endPosition = endIdx * numberOfChannels * bytesPerSample\n\n var spectralFeatures = computeSpectralFeatures(channelData.slice(startIdx, endIdx), sampleRate, features);\n\n // Simply use the counter, increment after assigning\n const dataPoint = {\n id: uniqueIdCounter++,\n amplitude: maxAmp,\n rms,\n startTime,\n endTime,\n dB: 20 * Math.log10(rms + 1e-6),\n silent: rms < 0.01,\n startPosition,\n endPosition,\n samples: samplesPerSegment,\n }\n\n // Extract features if any are requested\n const extractedFeatures = createFeaturesObject(\n features,\n maxAmp,\n rms,\n sumSquares,\n zeroCrossings,\n samplesPerSegment,\n spectralFeatures,\n channelData,\n startIdx,\n endIdx,\n sampleRate,\n numberOfChannels,\n bytesPerSample\n );\n \n if (extractedFeatures) {\n dataPoint.features = extractedFeatures;\n }\n\n dataPoints.push(dataPoint)\n }\n\n // Handle remaining samples if they exist and are enough to process\n if (remainingSamples > samplesPerSegment / 4) { // Only process if we have at least 1/4 of a segment\n const startIdx = numPoints * samplesPerSegment\n const endIdx = totalSamples\n \n let sumSquares = 0\n let maxAmp = 0\n let zeroCrossings = 0\n\n for (let j = startIdx; j < endIdx; j++) {\n const value = channelData[j]\n sumSquares += value * value\n maxAmp = Math.max(maxAmp, Math.abs(value))\n if (j > 0 && value * channelData[j - 1] < 0) {\n zeroCrossings++\n }\n }\n\n const rms = Math.sqrt(sumSquares / remainingSamples)\n const startTime = subChunkStartTime + (startIdx / sampleRate);\n const endTime = subChunkStartTime + (endIdx / sampleRate);\n // Calculate byte positions correctly based on numberOfChannels and bytesPerSample\n const startPosition = startIdx * numberOfChannels * bytesPerSample\n const endPosition = endIdx * numberOfChannels * bytesPerSample\n\n var spectralFeatures = computeSpectralFeatures(channelData.slice(startIdx, endIdx), sampleRate, features);\n\n // Simply use the counter, increment after assigning\n const dataPoint = {\n id: uniqueIdCounter++,\n amplitude: maxAmp,\n rms,\n startTime,\n endTime,\n dB: 20 * Math.log10(rms + 1e-6),\n silent: rms < 0.01,\n startPosition,\n endPosition,\n samples: remainingSamples,\n }\n\n logger.debug('extractWaveform - dataPoint', dataPoint);\n // Extract features if any are requested\n const extractedFeatures = createFeaturesObject(\n features,\n maxAmp,\n rms,\n sumSquares,\n zeroCrossings,\n remainingSamples,\n spectralFeatures,\n channelData,\n startIdx,\n endIdx,\n sampleRate,\n numberOfChannels,\n bytesPerSample\n );\n \n if (extractedFeatures) {\n dataPoint.features = extractedFeatures;\n }\n\n dataPoints.push(dataPoint)\n }\n\n return {\n durationMs,\n dataPoints,\n amplitudeRange: { min, max },\n rmsRange: {\n min: 0,\n max: Math.max(Math.abs(min), Math.abs(max))\n },\n extractionTimeMs: Date.now() - lastEmitTime\n }\n }\n\n try {\n const result = extractWaveform(\n channelData,\n sampleRate,\n segmentDurationMs,\n numberOfChannels || 1, // Default to 1 channel if not provided\n bytesPerSample\n )\n\n // Send complete result immediately\n self.postMessage({\n command: 'features',\n result: {\n bitDepth,\n samples: channelData.length,\n numberOfChannels,\n sampleRate,\n segmentDurationMs,\n durationMs: result.durationMs,\n dataPoints: result.dataPoints,\n amplitudeRange: result.amplitudeRange,\n rmsRange: result.rmsRange,\n }\n })\n } catch (error) {\n console.error('[Worker] Error', {\n message: error.message,\n stack: error.stack\n });\n \n self.postMessage({ \n error: {\n message: error.message,\n stack: error.stack,\n name: error.name\n }\n });\n }\n}\n`\n"]}

package/build/cjs/workers/inlineAudioWebWorker.web.js ADDED Viewed

@@ -0,0 +1,184 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.InlineAudioWebWorker = void 0;
+// packages/expo-audio-stream/src/workers/inlineAudioWebWorker.web.tsx
+exports.InlineAudioWebWorker = `
+const DEFAULT_BIT_DEPTH = 32
+const DEFAULT_SAMPLE_RATE = 44100
+class RecorderProcessor extends AudioWorkletProcessor {
+    constructor() {
+        super()
+        this.currentChunk = [] // Float32Array
+        this.samplesSinceLastExport = 0
+        this.recordSampleRate = DEFAULT_SAMPLE_RATE
+        this.exportSampleRate = DEFAULT_SAMPLE_RATE
+        this.recordBitDepth = DEFAULT_BIT_DEPTH
+        this.exportBitDepth = DEFAULT_BIT_DEPTH
+        this.numberOfChannels = 1
+        this.isRecording = true
+        this.port.onmessage = this.handleMessage.bind(this)
+        this.enableLogging = false
+        this.exportIntervalSamples = 0
+        this.currentPosition = 0 // Track current position in seconds
+    }
+    handleMessage(event) {
+        switch (event.data.command) {
+            case 'init':
+                this.enableLogging = event.data.enableLogging || false
+                this.recordSampleRate = event.data.recordSampleRate
+                this.exportSampleRate =
+                    event.data.exportSampleRate || event.data.recordSampleRate
+                this.exportIntervalSamples =
+                    this.recordSampleRate * (event.data.interval / 1000)
+                if (event.data.numberOfChannels) {
+                    this.numberOfChannels = event.data.numberOfChannels
+                }
+                if (event.data.recordBitDepth) {
+                    this.recordBitDepth = event.data.recordBitDepth
+                }
+                this.exportBitDepth =
+                    event.data.exportBitDepth || this.recordBitDepth
+                // Handle position parameter for device switching
+                if (typeof event.data.position === 'number' && event.data.position > 0) {
+                    this.currentPosition = event.data.position
+                    if (this.enableLogging) {
+                        console.log('AudioWorklet initialized with position:', this.currentPosition)
+                    }
+                }
+                break
+            case 'stop':
+                this.isRecording = false
+                if (this.currentChunk.length > 0) {
+                    this.processChunk()
+                }
+                break
+            case 'pause':
+                // Just a placeholder for pause handling
+                break
+            case 'resume':
+                // Just a placeholder for resume handling
+                break
+        }
+    }
+    process(inputs, _outputs, _parameters) {
+        if (!this.isRecording) return true
+        const input = inputs[0]
+        if (input.length > 0) {
+            const newBuffer = new Float32Array(input[0])
+            this.currentChunk.push(newBuffer)
+            this.samplesSinceLastExport += newBuffer.length
+            if (this.samplesSinceLastExport >= this.exportIntervalSamples) {
+                this.processChunk()
+                this.samplesSinceLastExport = 0
+            }
+        }
+        return true
+    }
+    mergeBuffers(bufferArray, recLength) {
+        const result = new Float32Array(recLength)
+        let offset = 0
+        for (let i = 0; i < bufferArray.length; i++) {
+            result.set(bufferArray[i], offset)
+            offset += bufferArray[i].length
+        }
+        return result
+    }
+    // Keep basic resampling for sample rate conversion
+    resample(samples, targetSampleRate) {
+        if (this.recordSampleRate === targetSampleRate) {
+            return samples
+        }
+        const resampledBuffer = new Float32Array(
+            Math.ceil(
+                (samples.length * targetSampleRate) / this.recordSampleRate
+            )
+        )
+        const ratio = this.recordSampleRate / targetSampleRate
+        let offset = 0
+        for (let i = 0; i < resampledBuffer.length; i++) {
+            const nextOffset = Math.floor((i + 1) * ratio)
+            let accum = 0
+            let count = 0
+            for (let j = offset; j < nextOffset && j < samples.length; j++) {
+                accum += samples[j]
+                count++
+            }
+            resampledBuffer[i] = count > 0 ? accum / count : 0
+            offset = nextOffset
+        }
+        return resampledBuffer
+    }
+    // Keep bit depth conversion if needed
+    convertBitDepth(input, targetBitDepth) {
+        if (targetBitDepth === 32) {
+            const output = new Int32Array(input.length)
+            for (let i = 0; i < input.length; i++) {
+                const s = Math.max(-1, Math.min(1, input[i]))
+                output[i] = s < 0 ? s * 0x80000000 : s * 0x7fffffff
+            }
+            return output
+        } else if (targetBitDepth === 16) {
+            const output = new Int16Array(input.length)
+            for (let i = 0; i < input.length; i++) {
+                const s = Math.max(-1, Math.min(1, input[i]))
+                output[i] = s < 0 ? s * 0x8000 : s * 0x7fff
+            }
+            return output
+        }
+        return input
+    }
+    processChunk() {
+        if (this.currentChunk.length === 0) return
+        // Merge buffers
+        const chunkLength = this.currentChunk.reduce(
+            (acc, buf) => acc + buf.length,
+            0
+        )
+        const mergedChunk = this.mergeBuffers(this.currentChunk, chunkLength)
+        // Resample if needed
+        const resampledChunk = this.resample(mergedChunk, this.exportSampleRate)
+        // Convert bit depth if needed
+        const finalBuffer =
+            this.recordBitDepth !== this.exportBitDepth
+                ? this.convertBitDepth(resampledChunk, this.exportBitDepth)
+                : resampledChunk
+        // Calculate the duration in seconds
+        const chunkDuration = finalBuffer.length / this.exportSampleRate
+        // Send processed chunk with the current position
+        this.port.postMessage({
+            command: 'newData',
+            recordedData: finalBuffer,
+            sampleRate: this.exportSampleRate,
+            bitDepth: this.exportBitDepth,
+            numberOfChannels: this.numberOfChannels,
+            position: this.currentPosition,
+        })
+        // Update the position
+        this.currentPosition += chunkDuration
+        // Clear the current chunk
+        this.currentChunk = []
+    }
+}
+registerProcessor('recorder-processor', RecorderProcessor)
+`;
+//# sourceMappingURL=inlineAudioWebWorker.web.js.map

package/build/cjs/workers/inlineAudioWebWorker.web.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"inlineAudioWebWorker.web.js","sourceRoot":"","sources":["../../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":";;;AAAA,sEAAsE;AACzD,QAAA,oBAAoB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CAkLnC,CAAA","sourcesContent":["// packages/expo-audio-stream/src/workers/inlineAudioWebWorker.web.tsx\nexport const InlineAudioWebWorker = `\nconst DEFAULT_BIT_DEPTH = 32\nconst DEFAULT_SAMPLE_RATE = 44100\n\nclass RecorderProcessor extends AudioWorkletProcessor {\n constructor() {\n super()\n this.currentChunk = [] // Float32Array\n this.samplesSinceLastExport = 0\n this.recordSampleRate = DEFAULT_SAMPLE_RATE\n this.exportSampleRate = DEFAULT_SAMPLE_RATE\n this.recordBitDepth = DEFAULT_BIT_DEPTH\n this.exportBitDepth = DEFAULT_BIT_DEPTH\n this.numberOfChannels = 1\n this.isRecording = true\n this.port.onmessage = this.handleMessage.bind(this)\n this.enableLogging = false\n this.exportIntervalSamples = 0\n this.currentPosition = 0 // Track current position in seconds\n }\n\n handleMessage(event) {\n switch (event.data.command) {\n case 'init':\n this.enableLogging = event.data.enableLogging || false\n this.recordSampleRate = event.data.recordSampleRate\n this.exportSampleRate =\n event.data.exportSampleRate || event.data.recordSampleRate\n this.exportIntervalSamples =\n this.recordSampleRate * (event.data.interval / 1000)\n if (event.data.numberOfChannels) {\n this.numberOfChannels = event.data.numberOfChannels\n }\n if (event.data.recordBitDepth) {\n this.recordBitDepth = event.data.recordBitDepth\n }\n this.exportBitDepth =\n event.data.exportBitDepth || this.recordBitDepth\n \n // Handle position parameter for device switching\n if (typeof event.data.position === 'number' && event.data.position > 0) {\n this.currentPosition = event.data.position\n if (this.enableLogging) {\n console.log('AudioWorklet initialized with position:', this.currentPosition)\n }\n }\n break\n\n case 'stop':\n this.isRecording = false\n if (this.currentChunk.length > 0) {\n this.processChunk()\n }\n break\n \n case 'pause':\n // Just a placeholder for pause handling\n break\n \n case 'resume':\n // Just a placeholder for resume handling\n break\n }\n }\n\n process(inputs, _outputs, _parameters) {\n if (!this.isRecording) return true\n const input = inputs[0]\n if (input.length > 0) {\n const newBuffer = new Float32Array(input[0])\n this.currentChunk.push(newBuffer)\n this.samplesSinceLastExport += newBuffer.length\n\n if (this.samplesSinceLastExport >= this.exportIntervalSamples) {\n this.processChunk()\n this.samplesSinceLastExport = 0\n }\n }\n return true\n }\n\n mergeBuffers(bufferArray, recLength) {\n const result = new Float32Array(recLength)\n let offset = 0\n for (let i = 0; i < bufferArray.length; i++) {\n result.set(bufferArray[i], offset)\n offset += bufferArray[i].length\n }\n return result\n }\n\n // Keep basic resampling for sample rate conversion\n resample(samples, targetSampleRate) {\n if (this.recordSampleRate === targetSampleRate) {\n return samples\n }\n const resampledBuffer = new Float32Array(\n Math.ceil(\n (samples.length * targetSampleRate) / this.recordSampleRate\n )\n )\n const ratio = this.recordSampleRate / targetSampleRate\n let offset = 0\n for (let i = 0; i < resampledBuffer.length; i++) {\n const nextOffset = Math.floor((i + 1) * ratio)\n let accum = 0\n let count = 0\n for (let j = offset; j < nextOffset && j < samples.length; j++) {\n accum += samples[j]\n count++\n }\n resampledBuffer[i] = count > 0 ? accum / count : 0\n offset = nextOffset\n }\n return resampledBuffer\n }\n\n // Keep bit depth conversion if needed\n convertBitDepth(input, targetBitDepth) {\n if (targetBitDepth === 32) {\n const output = new Int32Array(input.length)\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i]))\n output[i] = s < 0 ? s * 0x80000000 : s * 0x7fffffff\n }\n return output\n } else if (targetBitDepth === 16) {\n const output = new Int16Array(input.length)\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i]))\n output[i] = s < 0 ? s * 0x8000 : s * 0x7fff\n }\n return output\n }\n return input\n }\n\n processChunk() {\n if (this.currentChunk.length === 0) return\n\n // Merge buffers\n const chunkLength = this.currentChunk.reduce(\n (acc, buf) => acc + buf.length,\n 0\n )\n const mergedChunk = this.mergeBuffers(this.currentChunk, chunkLength)\n\n // Resample if needed\n const resampledChunk = this.resample(mergedChunk, this.exportSampleRate)\n\n // Convert bit depth if needed\n const finalBuffer =\n this.recordBitDepth !== this.exportBitDepth\n ? this.convertBitDepth(resampledChunk, this.exportBitDepth)\n : resampledChunk\n\n // Calculate the duration in seconds\n const chunkDuration = finalBuffer.length / this.exportSampleRate\n \n // Send processed chunk with the current position\n this.port.postMessage({\n command: 'newData',\n recordedData: finalBuffer,\n sampleRate: this.exportSampleRate,\n bitDepth: this.exportBitDepth,\n numberOfChannels: this.numberOfChannels,\n position: this.currentPosition,\n })\n \n // Update the position\n this.currentPosition += chunkDuration\n\n // Clear the current chunk\n this.currentChunk = []\n }\n}\n\nregisterProcessor('recorder-processor', RecorderProcessor)\n`\n"]}

package/build/esm/AudioAnalysis/AudioAnalysis.types.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,sEAAsE","sourcesContent":["// packages/expo-audio-stream/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../ExpoAudioStream.types'\n\n/**\n * Represents the configuration for decoding audio data.\n */\nexport interface DecodingConfig {\n /** Target sample rate for decoded audio (Android and Web) */\n targetSampleRate?: number\n /** Target number of channels (Android and Web) */\n targetChannels?: number\n /** Target bit depth (Android and Web) */\n targetBitDepth?: BitDepth\n /** Whether to normalize audio levels (Android and Web) */\n normalizeAudio?: boolean\n}\n\n/**\n * Represents speech-related features extracted from audio.\n */\nexport interface SpeechFeatures {\n isActive: boolean // Whether speech is detected in this segment\n speakerId?: number // Optional speaker identification\n // Could add more speech-related features here like:\n // confidence: number\n // language?: string\n // sentiment?: number\n // etc.\n}\n\n/**\n * Represents various audio features extracted from an audio signal.\n */\nexport interface AudioFeatures {\n energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n minAmplitude?: number // Minimum amplitude value in the audio signal.\n maxAmplitude?: number // Maximum amplitude value in the audio signal.\n zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n tonnetz?: number[] // Tonal network features representing harmonic relationships.\n pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/**\n * Options to specify which audio features to extract.\n */\nexport interface AudioFeaturesOptions {\n energy?: boolean\n mfcc?: boolean\n rms?: boolean\n zcr?: boolean\n spectralCentroid?: boolean\n spectralFlatness?: boolean\n spectralRolloff?: boolean\n spectralBandwidth?: boolean\n chromagram?: boolean\n tempo?: boolean\n hnr?: boolean\n melSpectrogram?: boolean\n spectralContrast?: boolean\n tonnetz?: boolean\n pitch?: boolean\n crc32?: boolean\n}\n\n/**\n * Represents a single data point in the audio analysis.\n */\nexport interface DataPoint {\n id: number\n amplitude: number // Peak amplitude for the segment\n rms: number // Root mean square value\n dB: number // dBFS (decibels relative to full scale) computed from RMS value\n silent: boolean // Always computed\n features?: AudioFeatures\n speech?: SpeechFeatures\n startTime?: number\n endTime?: number\n // start / end position in bytes\n startPosition?: number\n endPosition?: number\n // number of audio samples for this point (samples size depends on bit depth)\n samples?: number\n}\n\n/**\n * Represents the complete data from the audio analysis.\n */\nexport interface AudioAnalysis {\n segmentDurationMs: number // Duration of each segment in milliseconds\n durationMs: number // Duration of the audio in milliseconds\n bitDepth: number // Bit depth of the audio\n samples: number // Size of the audio in bytes\n numberOfChannels: number // Number of audio channels\n sampleRate: number // Sample rate of the audio\n dataPoints: DataPoint[] // Array of data points from the analysis.\n amplitudeRange: {\n min: number\n max: number\n }\n rmsRange: {\n min: number\n max: number\n }\n // TODO: speaker changes into a broader speech analysis section\n speechAnalysis?: {\n speakerChanges: {\n timestamp: number\n speakerId: number\n }[]\n // Could add more speech analysis data here like:\n // dominantSpeaker?: number\n // totalSpeechDuration?: number\n // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n }\n}\n\n/**\n * Options for specifying a time range within an audio file.\n */\nexport interface AudioRangeOptions {\n /** Start time in milliseconds */\n startTimeMs?: number\n /** End time in milliseconds */\n endTimeMs?: number\n}\n\n/**\n * Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n */\nexport interface PreviewOptions extends AudioRangeOptions {\n /** URI of the audio file to analyze */\n fileUri: string\n /**\n * Total number of points to generate for the preview.\n * @default 100\n */\n numberOfPoints?: number\n /**\n * Optional logger for debugging.\n */\n logger?: ConsoleLike\n /**\n * Optional configuration for decoding the audio file.\n * Defaults to:\n * - targetSampleRate: undefined (keep original)\n * - targetChannels: undefined (keep original)\n * - targetBitDepth: 16\n * - normalizeAudio: false\n */\n decodingOptions?: DecodingConfig\n}\n\n/**\n * Options for mel-spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface ExtractMelSpectrogramOptions {\n fileUri?: string // Path to audio file\n arrayBuffer?: ArrayBuffer // Raw audio buffer\n windowSizeMs: number // Window size in ms (e.g., 25)\n hopLengthMs: number // Hop length in ms (e.g., 10)\n nMels: number // Number of mel filters (e.g., 60)\n fMin?: number // Min frequency (default: 0)\n fMax?: number // Max frequency (default: sampleRate / 2)\n windowType?: 'hann' | 'hamming' // Window function (default: 'hann')\n normalize?: boolean // Mean normalization (default: false)\n logScale?: boolean // Log scaling of mel energies (default: true)\n decodingOptions?: DecodingConfig // Audio decoding settings\n startTimeMs?: number // Optional start time\n endTimeMs?: number // Optional end time\n logger?: ConsoleLike\n}\n\n/**\n * Return type for mel spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n spectrogram: number[][] // 2D array [time][mel]\n sampleRate: number // Audio sample rate\n nMels: number // Number of mel filters\n timeSteps: number // Number of time frames\n durationMs: number // Audio duration in ms\n}\n"]}

package/build/esm/AudioAnalysis/extractAudioAnalysis.js.map ADDED Viewed

@@ -0,0 +1 @@

+ 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32,\n numberOfChannels: processedBuffer.channels,\n fullAudioDurationMs: processedBuffer.durationMs,\n // enableLogging: !!logger,\n features,\n })\n })\n } finally {\n await audioContext.close()\n }\n } catch (error) {\n logger?.error('Failed to process audio:', error)\n throw error\n }\n } else {\n return await ExpoAudioStreamModule.extractAudioAnalysis(props)\n }\n}\n\n/**\n * Analyzes WAV files without decoding, preserving original PCM values.\n * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.\n *\n * @param props - The options for WAV analysis, including file URI and range.\n * @returns A promise that resolves to the audio analysis data.\n */\nexport const extractRawWavAnalysis = async ({\n fileUri,\n segmentDurationMs = 100, // Default to 100ms\n arrayBuffer,\n bitDepth,\n durationMs,\n sampleRate,\n numberOfChannels,\n features,\n logger,\n position = 0,\n length,\n}: ExtractWavAudioAnalysisProps): Promise<AudioAnalysis> => {\n if (isWeb) {\n if (!arrayBuffer && !fileUri) {\n throw new Error('Either arrayBuffer or fileUri must be provided')\n }\n\n if (!arrayBuffer) {\n logger?.log(`fetching fileUri`, fileUri)\n const response = await fetch(fileUri!)\n\n if (!response.ok) {\n throw new Error(\n `Failed to fetch fileUri: ${response.statusText}`\n )\n }\n\n arrayBuffer = await response.arrayBuffer()\n logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer)\n }\n\n // Create a new copy of the ArrayBuffer to avoid detachment issues\n const bufferCopy = arrayBuffer.slice(0)\n logger?.log(\n `extractAudioAnalysis bitDepth=${bitDepth} len=${bufferCopy.byteLength}`,\n bufferCopy.slice(0, 100)\n )\n\n let actualBitDepth = bitDepth\n if (!actualBitDepth) {\n logger?.log(\n `extractAudioAnalysis bitDepth not provided -- getting wav file info`\n )\n const fileInfo = await getWavFileInfo(bufferCopy)\n actualBitDepth = fileInfo.bitDepth\n }\n logger?.log(`extractAudioAnalysis actualBitDepth=${actualBitDepth}`)\n\n const {\n pcmValues: channelData,\n min,\n max,\n } = await convertPCMToFloat32({\n buffer: arrayBuffer,\n bitDepth: actualBitDepth,\n })\n logger?.log(\n `extractAudioAnalysis convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`\n )\n\n // Apply position and length constraints to channelData if specified\n const startIndex = position\n const endIndex = length ? startIndex + length : channelData.length\n const constrainedChannelData = channelData.slice(startIndex, endIndex)\n\n return new Promise((resolve, reject) => {\n const blob = new Blob([InlineFeaturesExtractor], {\n type: 'application/javascript',\n })\n const url = URL.createObjectURL(blob)\n const worker = new Worker(url)\n\n worker.onmessage = (event) => {\n resolve(event.data.result)\n }\n\n worker.onerror = (error) => {\n reject(error)\n }\n\n worker.postMessage({\n command: 'process',\n channelData: constrainedChannelData,\n sampleRate,\n segmentDurationMs,\n logger,\n bitDepth,\n fullAudioDurationMs: durationMs,\n numberOfChannels,\n })\n })\n } else {\n if (!fileUri) {\n throw new Error('fileUri is required')\n }\n logger?.log(`extractAudioAnalysis`, {\n fileUri,\n segmentDurationMs,\n })\n const res = await ExpoAudioStreamModule.extractAudioAnalysis({\n fileUri,\n segmentDurationMs,\n features,\n position,\n length,\n })\n logger?.log(`extractAudioAnalysis`, res)\n return res\n }\n}\n"]}

package/build/esm/AudioAnalysis/extractAudioData.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"extractAudioData.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractAudioData.ts"],"names":[],"mappings":"AACA,OAAO,qBAAqB,MAAM,0BAA0B,CAAA;AAE5D,MAAM,CAAC,MAAM,gBAAgB,GAAG,KAAK,EAAE,KAA8B,EAAE,EAAE;IACrE,OAAO,MAAM,qBAAqB,CAAC,gBAAgB,CAAC,KAAK,CAAC,CAAA;AAC9D,CAAC,CAAA","sourcesContent":["import { ExtractAudioDataOptions } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\n\nexport const extractAudioData = async (props: ExtractAudioDataOptions) => {\n return await ExpoAudioStreamModule.extractAudioData(props)\n}\n"]}

package/build/esm/AudioAnalysis/extractMelSpectrogram.js.map ADDED Viewed

@@ -0,0 +1 @@

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The web implementation is a placeholder.\n */\n\nimport { ExpoAudioStreamModule } from '..'\nimport { isWeb } from '../constants'\nimport {\n ExtractMelSpectrogramOptions,\n MelSpectrogram,\n} from './AudioAnalysis.types'\nimport {\n processAudioBuffer,\n ProcessedAudioData,\n} from '../utils/audioProcessing'\n\n/**\n * Extracts a mel spectrogram from audio data\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The iOS implementation will throw an \"UNSUPPORTED_PLATFORM\" error.\n * The web implementation is a placeholder that returns dummy data.\n */\nexport async function extractMelSpectrogram(\n options: ExtractMelSpectrogramOptions\n): Promise<MelSpectrogram> {\n const {\n fileUri,\n arrayBuffer,\n windowSizeMs,\n hopLengthMs,\n nMels,\n fMin = 0,\n fMax,\n windowType = 'hann',\n normalize = false,\n logScale = true,\n decodingOptions,\n startTimeMs,\n endTimeMs,\n logger,\n } = options\n\n if (isWeb) {\n // Create audio context\n const audioContext = new (window.AudioContext ||\n (window as any).webkitAudioContext)()\n\n try {\n // Process audio data using the existing utility\n const processedAudio: ProcessedAudioData = await processAudioBuffer(\n {\n arrayBuffer,\n fileUri,\n targetSampleRate:\n decodingOptions?.targetSampleRate || 16000,\n targetChannels: decodingOptions?.targetChannels || 1,\n normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n startTimeMs,\n endTimeMs,\n audioContext,\n logger: options.logger,\n }\n )\n\n // Calculate window and hop size in samples\n const sampleRate = processedAudio.sampleRate\n const windowSize = Math.floor((windowSizeMs * sampleRate) / 1000)\n const hopLength = Math.floor((hopLengthMs * sampleRate) / 1000)\n const maxFreq = fMax || sampleRate / 2\n\n // Extract the mel spectrogram from the processed audio\n const spectrogram = computeMelSpectrogram(\n processedAudio.channelData,\n sampleRate,\n nMels,\n windowSize,\n hopLength,\n fMin,\n maxFreq,\n windowType,\n normalize,\n logScale\n )\n\n const timeSteps = spectrogram.length\n\n return {\n spectrogram,\n sampleRate,\n nMels,\n timeSteps,\n durationMs: processedAudio.durationMs,\n }\n } catch (error) {\n logger?.error('Error extracting mel spectrogram:', error)\n throw error\n } finally {\n // Close the audio context\n await audioContext.close()\n }\n }\n return ExpoAudioStreamModule.extractMelSpectrogram(options)\n}\n\n/**\n * Computes a mel spectrogram from audio data\n *\n * @experimental This is a placeholder implementation that returns dummy data.\n * The actual implementation will be added in a future version.\n */\nfunction computeMelSpectrogram(\n audioData: Float32Array,\n sampleRate: number,\n nMels: number,\n windowSize: number,\n hopLength: number,\n fMin: number,\n fMax: number,\n windowType: 'hann' | 'hamming',\n normalize: boolean,\n logScale: boolean\n): number[][] {\n // Placeholder for the actual implementation\n // This would include:\n // 1. Windowing the audio data using the specified window type\n // 2. Computing the STFT (Short-Time Fourier Transform)\n // 3. Converting to power spectrogram\n // 4. Applying mel filterbanks\n // 5. Taking the logarithm if logScale is true\n // 6. Normalizing if normalize is true\n\n // For now, return a dummy implementation\n const numFrames =\n Math.floor((audioData.length - windowSize) / hopLength) + 1\n const spectrogram: number[][] = []\n\n // Create dummy mel spectrogram data\n for (let i = 0; i < numFrames; i++) {\n spectrogram.push(Array(nMels).fill(0))\n }\n\n return spectrogram\n}\n"]}

package/build/esm/AudioAnalysis/extractPreview.js.map ADDED Viewed

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+ {"version":3,"file":"extractPreview.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractPreview.ts"],"names":[],"mappings":"AACA,OAAO,EAAE,oBAAoB,EAAE,MAAM,wBAAwB,CAAA;AAE7D;;;;;;GAMG;AACH,MAAM,CAAC,KAAK,UAAU,cAAc,CAAC,EACjC,OAAO,EACP,cAAc,GAAG,GAAG,EACpB,WAAW,GAAG,CAAC,EACf,SAAS,GAAG,KAAK,EAAE,mBAAmB;AACtC,eAAe,EACf,MAAM,GACO;IACb,MAAM,UAAU,GAAG,SAAS,GAAG,WAAW,CAAA;IAC1C,MAAM,iBAAiB,GAAG,IAAI,CAAC,KAAK,CAAC,UAAU,GAAG,cAAc,CAAC,CAAA;IAEjE,uDAAuD;IACvD,MAAM,QAAQ,GAAG,MAAM,oBAAoB,CAAC;QACxC,OAAO;QACP,WAAW;QACX,SAAS;QACT,MAAM;QACN,iBAAiB;QACjB,eAAe;KAClB,CAAC,CAAA;IAEF,gDAAgD;IAChD,OAAO,QAAQ,CAAA;AACnB,CAAC","sourcesContent":["import { PreviewOptions, AudioAnalysis } from './AudioAnalysis.types'\nimport { extractAudioAnalysis } from './extractAudioAnalysis'\n\n/**\n * Generates a simplified preview of the audio waveform for quick visualization.\n * Ideal for UI rendering with a specified number of points.\n *\n * @param options - The options for the preview, including file URI and time range.\n * @returns A promise that resolves to the audio preview data.\n */\nexport async function extractPreview({\n fileUri,\n numberOfPoints = 100,\n startTimeMs = 0,\n endTimeMs = 30000, // First 30 seconds\n decodingOptions,\n logger,\n}: PreviewOptions): Promise<AudioAnalysis> {\n const durationMs = endTimeMs - startTimeMs\n const segmentDurationMs = Math.floor(durationMs / numberOfPoints)\n\n // Call extractAudioAnalysis with calculated parameters\n const analysis = await extractAudioAnalysis({\n fileUri,\n startTimeMs,\n endTimeMs,\n logger,\n segmentDurationMs,\n decodingOptions,\n })\n\n // Transform the result into AudioPreview format\n return analysis\n}\n"]}

package/build/esm/AudioAnalysis/extractWaveform.js.map ADDED Viewed

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+ {"version":3,"file":"extractWaveform.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractWaveform.ts"],"names":[],"mappings":"AAAA,OAAO,qBAAqB,MAAM,0BAA0B,CAAA;AAQ5D,MAAM,CAAC,MAAM,eAAe,GAAG,KAAK,EAAE,EAClC,OAAO,EACP,eAAe,EACf,MAAM,GAAG,CAAC,EACV,MAAM,GACa,EAAoB,EAAE;IACzC,MAAM,GAAG,GAAG,MAAM,qBAAqB,CAAC,oBAAoB,CAAC;QACzD,OAAO;QACP,eAAe;QACf,MAAM;QACN,MAAM;KACT,CAAC,CAAA;IACF,OAAO,GAAG,CAAA;AACd,CAAC,CAAA","sourcesContent":["import ExpoAudioStreamModule from '../ExpoAudioStreamModule'\n\nexport interface ExtractWaveformProps {\n fileUri: string\n numberOfSamples: number\n offset?: number\n length?: number\n}\nexport const extractWaveform = async ({\n fileUri,\n numberOfSamples,\n offset = 0,\n length,\n}: ExtractWaveformProps): Promise<unknown> => {\n const res = await ExpoAudioStreamModule.extractAudioAnalysis({\n fileUri,\n numberOfSamples,\n offset,\n length,\n })\n return res\n}\n"]}

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{ EventEmitter } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n AudioDevice,\n AudioDeviceCapabilities,\n DeviceDisconnectionBehavior,\n ConsoleLike,\n} from './ExpoAudioStream.types'\nimport ExpoAudioStreamModule from './ExpoAudioStreamModule'\n\n// Default device fallback for web and unsupported platforms\nconst DEFAULT_DEVICE: AudioDevice = {\n id: 'default',\n name: 'Default Microphone',\n type: 'builtin_mic',\n isDefault: true,\n isAvailable: true,\n capabilities: {\n sampleRates: [16000, 44100, 48000],\n channelCounts: [1, 2],\n bitDepths: [16, 24, 32],\n hasEchoCancellation: true,\n hasNoiseSuppression: true,\n hasAutomaticGainControl: true,\n },\n}\n\n// Helper function to map raw object to AudioDevice interface\n// This handles potential inconsistencies from the native module\nfunction mapRawDeviceToAudioDevice(rawDevice: any): AudioDevice {\n const capabilities = rawDevice.capabilities || {}\n return {\n id: rawDevice.id || 'unknown',\n name: rawDevice.name || 'Unknown Device',\n type: rawDevice.type || 'unknown',\n isDefault: rawDevice.isDefault || false,\n isAvailable:\n rawDevice.isAvailable !== undefined ? rawDevice.isAvailable : true, // Default to true if undefined\n capabilities: {\n sampleRates: capabilities.sampleRates || [16000, 44100, 48000], // Provide defaults\n channelCounts: capabilities.channelCounts || [1, 2],\n bitDepths: capabilities.bitDepths || [16, 24, 32],\n hasEchoCancellation: capabilities.hasEchoCancellation,\n hasNoiseSuppression: capabilities.hasNoiseSuppression,\n hasAutomaticGainControl: capabilities.hasAutomaticGainControl,\n },\n }\n}\n\n/**\n * Class that provides a cross-platform API for managing audio input devices\n */\nexport class AudioDeviceManager {\n private eventEmitter: InstanceType<typeof EventEmitter>\n private currentDeviceId: string | null = null\n private availableDevices: AudioDevice[] = []\n private deviceChangeListeners: Set<(devices: AudioDevice[]) => void> =\n new Set()\n private deviceListeners: Set<() => void> = new Set()\n private lastRefreshTime: number = 0\n private refreshInProgress: boolean = false\n private refreshDebounceMs: number = 500 // Minimum 500ms between refreshes\n private logger?: ConsoleLike\n\n constructor(options?: { logger?: ConsoleLike }) {\n this.eventEmitter = new EventEmitter(ExpoAudioStreamModule)\n this.logger = options?.logger\n\n // Listen for device change events from native modules if not on web\n if (Platform.OS !== 'web') {\n // Store the last event type to avoid duplicates\n let lastEventType: string | null = null\n let lastEventTime = 0\n\n this.eventEmitter.addListener(\n 'deviceChangedEvent',\n (event: any) => {\n // Skip processing duplicate events that occur too close together\n const now = Date.now()\n const isSimilarEvent =\n lastEventType === event.type &&\n now - lastEventTime < this.refreshDebounceMs\n\n if (isSimilarEvent) {\n this.logger?.debug(\n `Skipping similar device event (${event.type}) received too soon`\n )\n return\n }\n\n // Update the last event tracking\n lastEventType = event.type\n lastEventTime = now\n\n // Only refresh on meaningful events\n if (\n event.type === 'deviceConnected' ||\n event.type === 'deviceDisconnected' ||\n event.type === 'routeChanged'\n ) {\n this.logger?.debug(\n `Processing device event: ${event.type}`\n )\n // Refresh devices and notify listeners regardless of the direct return value\n this.refreshDevices()\n }\n }\n )\n }\n }\n\n /**\n * Initialize the device manager with a logger\n * @param logger A logger instance that implements the ConsoleLike interface\n * @returns The manager instance for chaining\n */\n initWithLogger(logger: ConsoleLike): AudioDeviceManager {\n this.setLogger(logger)\n return this\n }\n\n /**\n * Set the logger instance\n * @param logger A logger instance that implements the ConsoleLike interface\n */\n setLogger(logger: ConsoleLike) {\n this.logger = logger\n }\n\n /**\n * Get all available audio input devices\n * @param options Optional settings to force refresh the device list. Can include a refresh flag.\n * @returns Promise resolving to an array of audio devices conforming to AudioDevice interface\n */\n async getAvailableDevices(options?: {\n refresh?: boolean\n }): Promise<AudioDevice[]> {\n try {\n if (Platform.OS === 'web') {\n this.availableDevices = await this.getWebAudioDevices()\n } else if (ExpoAudioStreamModule.getAvailableInputDevices) {\n // Expecting an array of raw device objects from native\n const rawDevices: any[] =\n await ExpoAudioStreamModule.getAvailableInputDevices(\n options\n )\n // Map raw objects to the AudioDevice interface\n this.availableDevices = rawDevices.map(\n mapRawDeviceToAudioDevice\n )\n } else {\n // Fallback for unsupported platforms\n this.availableDevices = [DEFAULT_DEVICE]\n }\n return this.availableDevices\n } catch (error) {\n this.logger?.error('Failed to get available devices:', error)\n this.availableDevices = [DEFAULT_DEVICE] // Ensure state is updated on error\n return this.availableDevices\n }\n }\n\n /**\n * Get the currently selected audio input device\n * @returns Promise resolving to the current device (conforming to AudioDevice) or null\n */\n async getCurrentDevice(): Promise<AudioDevice | null> {\n try {\n if (Platform.OS === 'web') {\n if (!this.currentDeviceId) {\n // On web, return the typed default device if nothing is selected\n return DEFAULT_DEVICE\n }\n // Refresh web devices to ensure the current one is up-to-date\n const webDevices = await this.getWebAudioDevices()\n return (\n webDevices.find((d) => d.id === this.currentDeviceId) ||\n DEFAULT_DEVICE // Fallback to default if current ID not found\n )\n } else if (ExpoAudioStreamModule.getCurrentInputDevice) {\n // Expecting a single raw device object or null from native\n const rawDevice: any | null =\n await ExpoAudioStreamModule.getCurrentInputDevice()\n // Map to AudioDevice interface if not null\n return rawDevice ? mapRawDeviceToAudioDevice(rawDevice) : null\n } else {\n // Fallback for unsupported platforms\n return DEFAULT_DEVICE\n }\n } catch (error) {\n this.logger?.error('Failed to get current device:', error)\n return DEFAULT_DEVICE // Return default on error\n }\n }\n\n /**\n * Select a specific audio input device for recording\n * @param deviceId The ID of the device to select\n * @returns Promise resolving to a boolean indicating success\n */\n async selectDevice(deviceId: string): Promise<boolean> {\n try {\n let success = false\n if (Platform.OS === 'web') {\n // Check if the device exists before setting it\n const devices = await this.getWebAudioDevices()\n if (devices.some((d) => d.id === deviceId)) {\n this.currentDeviceId = deviceId\n success = true\n } else {\n this.logger?.warn(\n `Web: Device with ID ${deviceId} not found.`\n )\n success = false\n }\n } else if (ExpoAudioStreamModule.selectInputDevice) {\n success =\n await ExpoAudioStreamModule.selectInputDevice(deviceId)\n if (success) {\n this.currentDeviceId = deviceId\n }\n }\n // Refresh devices after selection attempt to update state\n await this.refreshDevices()\n return success\n } catch (error) {\n this.logger?.error('Failed to select device:', error)\n await this.refreshDevices() // Refresh even on error\n return false\n }\n }\n\n /**\n * Reset to the default audio input device\n * @returns Promise resolving to a boolean indicating success\n */\n async resetToDefaultDevice(): Promise<boolean> {\n try {\n let success = false\n if (Platform.OS === 'web') {\n this.currentDeviceId = 'default'\n success = true\n } else if (ExpoAudioStreamModule.resetToDefaultDevice) {\n success = await ExpoAudioStreamModule.resetToDefaultDevice()\n if (success) {\n this.currentDeviceId = null\n }\n }\n // Refresh devices after reset attempt\n await this.refreshDevices()\n return success\n } catch (error) {\n this.logger?.error('Failed to reset to default device:', error)\n await this.refreshDevices() // Refresh even on error\n return false\n }\n }\n\n /**\n * Register a listener for device changes\n * @param listener Function to call when devices change (receives AudioDevice[])\n * @returns Function to remove the listener\n */\n addDeviceChangeListener(\n listener: (devices: AudioDevice[]) => void\n ): () => void {\n this.deviceChangeListeners.add(listener)\n\n // Immediately call listener with current devices if available\n if (this.availableDevices.length > 0) {\n listener([...this.availableDevices])\n }\n\n // Return a function to remove the listener\n return () => {\n this.deviceChangeListeners.delete(listener)\n }\n }\n\n /**\n * Refresh the list of available devices with debouncing and notify listeners.\n * @returns Promise resolving to the updated device list (AudioDevice[])\n */\n async refreshDevices(): Promise<AudioDevice[]> {\n const now = Date.now()\n\n if (this.refreshInProgress) {\n this.logger?.debug('Refresh already in progress, skipping')\n return this.availableDevices\n }\n\n // Always allow refresh if forced by native event or longer than 2s debounce\n const timeSinceLastRefresh = now - this.lastRefreshTime\n const shouldDebounce =\n timeSinceLastRefresh < this.refreshDebounceMs &&\n timeSinceLastRefresh < 2000\n\n if (shouldDebounce) {\n this.logger?.debug(\n `Refresh debounced, skipping (last refresh was ${timeSinceLastRefresh}ms ago)`\n )\n return this.availableDevices\n }\n\n this.logger?.debug('Refreshing devices...')\n this.refreshInProgress = true\n try {\n // Fetch the latest devices; getAvailableDevices handles mapping now\n const devices = await this.getAvailableDevices({ refresh: true })\n // availableDevices state is updated within getAvailableDevices\n this.notifyListeners() // Notify listeners with the updated list\n this.lastRefreshTime = Date.now()\n return devices // Return the fetched & mapped list\n } catch (error) {\n this.logger?.error('Error during refreshDevices:', error)\n return this.availableDevices // Return potentially stale list on error\n } finally {\n this.refreshInProgress = false\n this.logger?.debug('Refresh finished.')\n }\n }\n\n /**\n * Get audio input devices using the Web Audio API\n * @returns Promise resolving to an array of AudioDevice objects\n */\n private async getWebAudioDevices(): Promise<AudioDevice[]> {\n if (\n typeof navigator === 'undefined' ||\n !navigator.mediaDevices ||\n !navigator.mediaDevices.enumerateDevices\n ) {\n return [DEFAULT_DEVICE]\n }\n\n try {\n const permissionStatus = await this.checkMicrophonePermission()\n\n if (permissionStatus === 'denied') {\n return [\n {\n ...DEFAULT_DEVICE,\n name: 'Microphone Access Denied',\n isAvailable: false,\n },\n ]\n }\n\n if (permissionStatus !== 'granted') {\n try {\n // Requesting stream often reveals device labels\n await navigator.mediaDevices.getUserMedia({ audio: true })\n } catch (error) {\n this.logger?.warn(\n 'Microphone permission request failed:',\n error\n )\n return [\n {\n ...DEFAULT_DEVICE,\n name: 'Microphone Access Required',\n isAvailable: false,\n },\n ]\n }\n }\n\n const devices = await navigator.mediaDevices.enumerateDevices()\n const audioInputDevices = devices\n .filter((device) => device.kind === 'audioinput')\n .map((device) => this.mapWebDeviceToAudioDevice(device))\n\n const hasUnlabeledDevices = audioInputDevices.some(\n (device) =>\n !device.name || device.name.startsWith('Microphone ')\n )\n\n let finalDevices = audioInputDevices\n if (hasUnlabeledDevices && this.isSafariOrIOS()) {\n finalDevices = this.enhanceDevicesForSafari(audioInputDevices)\n }\n\n if (finalDevices.length === 0) {\n finalDevices = [DEFAULT_DEVICE]\n }\n\n this.setupWebDeviceChangeListener()\n this.availableDevices = finalDevices // Update internal state\n return finalDevices\n } catch (error) {\n this.logger?.error('Failed to enumerate web audio devices:', error)\n this.availableDevices = [DEFAULT_DEVICE] // Update state on error\n return this.availableDevices\n }\n }\n\n /**\n * Check the current microphone permission status\n * @returns Permission state ('prompt', 'granted', or 'denied')\n */\n private async checkMicrophonePermission(): Promise<PermissionState> {\n if (!navigator.permissions || !navigator.permissions.query) {\n return 'prompt'\n }\n try {\n const permissionStatus = await navigator.permissions.query({\n name: 'microphone' as PermissionName,\n })\n permissionStatus.onchange = () => {\n // Refresh devices when permission changes\n this.refreshDevices()\n }\n return permissionStatus.state\n } catch (error) {\n this.logger?.warn('Permission query not supported:', error)\n return 'prompt'\n }\n }\n\n /**\n * Setup listener for device changes in web environment\n */\n private setupWebDeviceChangeListener() {\n if (\n typeof navigator === 'undefined' ||\n !navigator.mediaDevices ||\n this.deviceListeners.size > 0 // Avoid adding multiple listeners\n ) {\n return\n }\n\n const handleDeviceChange = () => {\n this.logger?.debug('Web device change detected.')\n // Refresh devices on change\n this.refreshDevices()\n }\n\n navigator.mediaDevices.addEventListener(\n 'devicechange',\n handleDeviceChange\n )\n this.deviceListeners.add(handleDeviceChange)\n this.logger?.debug('Web device change listener added.')\n }\n\n /**\n * Check if the current browser is Safari or iOS WebKit\n */\n private isSafariOrIOS(): boolean {\n if (typeof navigator === 'undefined') return false\n const ua = navigator.userAgent\n return (\n /^((?!chrome|android).)*safari/i.test(ua) ||\n /iPad|iPhone|iPod/.test(ua) ||\n (navigator.platform === 'MacIntel' && navigator.maxTouchPoints > 1)\n )\n }\n\n /**\n * Create enhanced device information for Safari and privacy-restricted browsers\n * @param devices Array of AudioDevice objects, potentially unlabeled\n * @returns Array of enhanced AudioDevice objects\n */\n private enhanceDevicesForSafari(devices: AudioDevice[]): AudioDevice[] {\n const defaultDevice = devices.find((d) => d.isDefault)\n\n if (devices.length <= 1) {\n // Return a typed default device\n return [\n {\n id: defaultDevice?.id || 'default',\n name: 'Microphone (Browser Managed)',\n type: 'builtin_mic',\n isDefault: true,\n isAvailable: true,\n capabilities:\n defaultDevice?.capabilities ||\n DEFAULT_DEVICE.capabilities,\n },\n ]\n }\n\n // Provide more descriptive names for unlabeled devices\n return devices.map((device, index) => {\n if (!device.name || device.name.startsWith('Microphone ')) {\n const deviceTypes = [\n 'Built-in Microphone',\n 'External Microphone',\n 'Headset Microphone',\n ]\n const typeName = deviceTypes[index % deviceTypes.length]\n return {\n ...device,\n name: device.isDefault ? `${typeName} (Default)` : typeName,\n }\n }\n return device\n })\n }\n\n /**\n * Map a Web MediaDeviceInfo to our AudioDevice format\n * @param device The MediaDeviceInfo object from the browser\n * @returns An object conforming to the AudioDevice interface\n */\n private mapWebDeviceToAudioDevice(device: MediaDeviceInfo): AudioDevice {\n const isDefault = device.deviceId === 'default'\n const deviceType = this.inferDeviceType(device.label || '')\n\n // Provide reasonable default capabilities for web devices\n const defaultWebCapabilities: AudioDeviceCapabilities = {\n sampleRates: [16000, 44100, 48000],\n channelCounts: [1, 2],\n bitDepths: [16, 32], // Web Audio uses float32, common PCM might be 16/32\n hasEchoCancellation: true, // Often handled by browser\n hasNoiseSuppression: true, // Often handled by browser\n hasAutomaticGainControl: true, // Often handled by browser\n }\n\n return {\n id: device.deviceId,\n name:\n device.label || `Microphone ${device.deviceId.substring(0, 8)}`,\n type: deviceType,\n isDefault,\n isAvailable: true, // Assume available if enumerated\n capabilities: defaultWebCapabilities,\n }\n }\n\n /**\n * Try to infer the device type from its name\n * @param deviceName The label of the device\n * @returns A string representing the inferred device type\n */\n private inferDeviceType(deviceName: string): string {\n const name = deviceName.toLowerCase()\n if (name.includes('bluetooth') || name.includes('airpods'))\n return 'bluetooth'\n if (name.includes('usb')) return 'usb'\n if (name.includes('headphone') || name.includes('headset')) {\n return name.includes('wired') ? 'wired_headset' : 'wired_headphones'\n }\n if (name.includes('speaker')) return 'speaker'\n return 'builtin_mic' // Default assumption\n }\n\n /**\n * Notify all registered listeners about device changes.\n */\n private notifyListeners(): void {\n // Pass a copy of the current devices array to listeners\n const devicesCopy = [...this.availableDevices]\n this.logger?.debug(\n `Notifying ${this.deviceChangeListeners.size} listeners with ${devicesCopy.length} devices.`\n )\n this.deviceChangeListeners.forEach((listener) => {\n try {\n listener(devicesCopy)\n } catch (error) {\n this.logger?.error('Error in device change listener:', error)\n }\n })\n }\n}\n\n// Create and export the singleton instance\nexport const audioDeviceManager = new AudioDeviceManager()\n\nexport { DeviceDisconnectionBehavior }\n"]}

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package/build/esm/ExpoAudioStream.types.js.map ADDED Viewed

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whether audio recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number\n /** Size of the recorded audio data in bytes */\n size: number\n /** Interval in milliseconds at which recording data is emitted */\n interval: number\n /** Interval in milliseconds at which analysis data is emitted */\n intervalAnalysis: number\n /** MIME type of the recorded audio (e.g., 'audio/wav') */\n mimeType: string\n /** Information about audio compression if enabled */\n compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n /** Audio data as base64 string (native) or Float32Array (web) */\n data: string | Float32Array\n /** Current position in the audio stream in bytes */\n position: number\n /** URI to the file being recorded */\n fileUri: string\n /** Size of the current data chunk in bytes */\n eventDataSize: number\n /** Total size of the recording so far in bytes */\n totalSize: number\n /** Information about compression if enabled, including the compressed data chunk */\n compression?: CompressionInfo & {\n /** Base64 (native) or Blob (web) encoded compressed data chunk */\n data?: string | Blob\n }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000\nexport type BitDepth = 8 | 16 | 32\nexport type PCMFormat = `pcm_${BitDepth}bit`\n\nexport type ConsoleLike = {\n /** Logs a message with optional arguments */\n log: (message: string, ...args: unknown[]) => void\n /** Logs a debug message with optional arguments */\n debug: (message: string, ...args: unknown[]) => void\n /** Logs an info message with optional arguments */\n info: (message: string, ...args: unknown[]) => void\n /** Logs a warning message with optional arguments */\n warn: (message: string, ...args: unknown[]) => void\n /** Logs an error message with optional arguments */\n error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n /** Transcribed text content */\n text: string\n /** Start and end timestamp in seconds [start, end] where end can be null if ongoing */\n timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n /** Unique identifier for the transcription */\n id: string\n /** Indicates if the transcriber is currently processing */\n isBusy: boolean\n /** Complete transcribed text */\n text: string\n /** Start time of the transcription in milliseconds */\n startTime: number\n /** End time of the transcription in milliseconds */\n endTime: number\n /** Array of transcribed text chunks with timestamps */\n chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n /** URI to the recorded audio file */\n fileUri: string\n /** Filename of the recorded audio */\n filename: string\n /** Duration of the recording in milliseconds */\n durationMs: number\n /** Size of the recording in bytes */\n size: number\n /** MIME type of the recorded audio */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate: SampleRate\n /** Timestamp when the recording was created */\n createdAt?: number\n /** Array of transcription data if available */\n transcripts?: TranscriberData[]\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface StartRecordingResult {\n /** URI to the file being recorded */\n fileUri: string\n /** MIME type of the recording */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth?: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate?: SampleRate\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface AudioSessionConfig {\n /**\n * Audio session category that defines the audio behavior\n * - 'Ambient': Audio continues with silent switch, mixes with other audio\n * - 'SoloAmbient': Audio continues with silent switch, interrupts other audio\n * - 'Playback': Audio continues in background, interrupts other audio\n * - 'Record': Optimized for recording, interrupts other audio\n * - 'PlayAndRecord': Allows simultaneous playback and recording\n * - 'MultiRoute': Routes audio to multiple outputs simultaneously\n */\n category?:\n | 'Ambient'\n | 'SoloAmbient'\n | 'Playback'\n | 'Record'\n | 'PlayAndRecord'\n | 'MultiRoute'\n /**\n * Audio session mode that defines the behavior for specific use cases\n * - 'Default': Standard audio behavior\n * - 'VoiceChat': Optimized for voice chat applications\n * - 'VideoChat': Optimized for video chat applications\n * - 'GameChat': Optimized for in-game chat\n * - 'VideoRecording': Optimized for video recording\n * - 'Measurement': Optimized for audio measurement\n * - 'MoviePlayback': Optimized for movie playback\n * - 'SpokenAudio': Optimized for spoken audio content\n */\n mode?:\n | 'Default'\n | 'VoiceChat'\n | 'VideoChat'\n | 'GameChat'\n | 'VideoRecording'\n | 'Measurement'\n | 'MoviePlayback'\n | 'SpokenAudio'\n /**\n * Options that modify the behavior of the audio session category\n * - 'MixWithOthers': Allows mixing with other active audio sessions\n * - 'DuckOthers': Reduces the volume of other audio sessions\n * - 'InterruptSpokenAudioAndMixWithOthers': Interrupts spoken audio and mixes with others\n * - 'AllowBluetooth': Allows audio routing to Bluetooth devices\n * - 'AllowBluetoothA2DP': Allows audio routing to Bluetooth A2DP devices\n * - 'AllowAirPlay': Allows audio routing to AirPlay devices\n * - 'DefaultToSpeaker': Routes audio to the speaker by default\n */\n categoryOptions?: (\n | 'MixWithOthers'\n | 'DuckOthers'\n | 'InterruptSpokenAudioAndMixWithOthers'\n | 'AllowBluetooth'\n | 'AllowBluetoothA2DP'\n | 'AllowAirPlay'\n | 'DefaultToSpeaker'\n )[]\n}\n\nexport interface IOSConfig {\n /** Configuration for the iOS audio session */\n audioSession?: AudioSessionConfig\n}\n\n/** Web platform specific configuration options */\nexport interface WebConfig {\n /**\n * Whether to store uncompressed audio data for WAV generation\n *\n * When true, all PCM chunks are stored in memory to create a WAV file when compression is disabled\n * When false, uncompressed audio won't be available, but memory usage will be lower\n *\n * Default: true (for backward compatibility)\n */\n storeUncompressedAudio?: boolean\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n /** Audio focus was lost to another app */\n | 'audioFocusLoss'\n /** Audio focus was regained */\n | 'audioFocusGain'\n /** Recording was interrupted by a phone call */\n | 'phoneCall'\n /** Phone call that interrupted recording has ended */\n | 'phoneCallEnded'\n /** Recording was stopped by the system or another app */\n | 'recordingStopped'\n /** Recording device was disconnected */\n | 'deviceDisconnected'\n /** Recording switched to default device after disconnection */\n | 'deviceFallback'\n /** A new audio device was connected */\n | 'deviceConnected'\n /** Device switching failed */\n | 'deviceSwitchFailed'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n /** The reason for the recording interruption */\n reason: RecordingInterruptionReason\n /** Indicates whether the recording is paused due to the interruption */\n isPaused: boolean\n}\n\nexport interface AudioDeviceCapabilities {\n /** Supported sample rates for the device */\n sampleRates: number[]\n /** Supported channel counts for the device */\n channelCounts: number[]\n /** Supported bit depths for the device */\n bitDepths: number[]\n /** Whether the device supports echo cancellation */\n hasEchoCancellation?: boolean\n /** Whether the device supports noise suppression */\n hasNoiseSuppression?: boolean\n /** Whether the device supports automatic gain control */\n hasAutomaticGainControl?: boolean\n}\n\nexport interface AudioDevice {\n /** Unique identifier for the device */\n id: string\n /** Human-readable name of the device */\n name: string\n /** Device type (builtin_mic, bluetooth, etc.) */\n type: string\n /** Whether this is the system default device */\n isDefault: boolean\n /** Audio capabilities for the device */\n capabilities: AudioDeviceCapabilities\n /** Whether the device is currently available */\n isAvailable: boolean\n}\n\n/** Defines how recording should behave when a device becomes unavailable */\nexport const DeviceDisconnectionBehavior = {\n /** Pause recording when device disconnects */\n PAUSE: 'pause',\n /** Switch to default device and continue recording */\n FALLBACK: 'fallback',\n} as const\n\n/** Type for DeviceDisconnectionBehavior values */\nexport type DeviceDisconnectionBehaviorType =\n (typeof DeviceDisconnectionBehavior)[keyof typeof DeviceDisconnectionBehavior]\n\nexport interface RecordingConfig {\n /** Sample rate for recording in Hz (16000, 44100, or 48000) */\n sampleRate?: SampleRate\n\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: 1 | 2\n\n /** Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit) */\n encoding?: EncodingType\n\n /** Interval in milliseconds at which to emit recording data */\n interval?: number\n\n /** Interval in milliseconds at which to emit analysis data */\n intervalAnalysis?: number\n\n /** Keep the device awake while recording (default is false) */\n keepAwake?: boolean\n\n /** Show a notification during recording (default is false) */\n showNotification?: boolean\n\n /** Show waveform in the notification (Android only, when showNotification is true) */\n showWaveformInNotification?: boolean\n\n /** Configuration for the notification */\n notification?: NotificationConfig\n\n /** Enable audio processing (default is false) */\n enableProcessing?: boolean\n\n /** iOS-specific configuration */\n ios?: IOSConfig\n\n /** Web-specific configuration options */\n web?: WebConfig\n\n /** Duration of each segment in milliseconds for analysis (default: 100) */\n segmentDurationMs?: number\n\n /** Feature options to extract during audio processing */\n features?: AudioFeaturesOptions\n\n /** Callback function to handle audio stream data */\n onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n /** Callback function to handle audio features extraction results */\n onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n /** Configuration for audio compression */\n compression?: {\n /** Enable audio compression */\n enabled: boolean\n /**\n * Format for compression\n * - 'aac': Advanced Audio Coding - supported on all platforms\n * - 'opus': Opus encoding - supported on Android and Web; on iOS will automatically fall back to AAC\n */\n format: 'aac' | 'opus'\n /** Bitrate for compression in bits per second */\n bitrate?: number\n }\n\n /** Whether to automatically resume recording after an interruption (default is false) */\n autoResumeAfterInterruption?: boolean\n\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n /** Optional directory path where output files will be saved */\n outputDirectory?: string // If not provided, uses default app directory\n /** Optional filename for the recording (uses UUID if not provided) */\n filename?: string // If not provided, uses UUID\n\n /** ID of the device to use for recording (if not specified, uses default) */\n deviceId?: string\n\n /** How to handle device disconnection during recording */\n deviceDisconnectionBehavior?: DeviceDisconnectionBehaviorType\n}\n\nexport interface NotificationConfig {\n /** Title of the notification */\n title?: string\n\n /** Main text content of the notification */\n text?: string\n\n /** Icon to be displayed in the notification (resource name or URI) */\n icon?: string\n\n /** Android-specific notification configuration */\n android?: {\n /** Unique identifier for the notification channel */\n channelId?: string\n\n /** User-visible name of the notification channel */\n channelName?: string\n\n /** User-visible description of the notification channel */\n channelDescription?: string\n\n /** Unique identifier for this notification */\n notificationId?: number\n\n /** List of actions that can be performed from the notification */\n actions?: NotificationAction[]\n\n /** Configuration for the waveform visualization in the notification */\n waveform?: WaveformConfig\n\n /** Color of the notification LED (if device supports it) */\n lightColor?: string\n\n /** Priority of the notification (affects how it's displayed) */\n priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n /** Accent color for the notification (used for the app icon and buttons) */\n accentColor?: string\n }\n\n /** iOS-specific notification configuration */\n ios?: {\n /** Identifier for the notification category (used for grouping similar notifications) */\n categoryIdentifier?: string\n }\n}\n\nexport interface NotificationAction {\n /** Display title for the action */\n title: string\n\n /** Unique identifier for the action */\n identifier: string\n\n /** Icon to be displayed for the action (Android only) */\n icon?: string\n}\n\nexport interface WaveformConfig {\n /** The color of the waveform (e.g., \"#FFFFFF\" for white) */\n color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n /** Opacity of the waveform (0.0 - 1.0) */\n opacity?: number // Opacity of the waveform (0.0 - 1.0)\n /** Width of the waveform line (default: 1.5) */\n strokeWidth?: number // Width of the waveform line (default: 1.5)\n /** Drawing style: \"stroke\" for outline, \"fill\" for solid */\n style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n /** Whether to mirror the waveform (symmetrical display) */\n mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n /** Height of the waveform view in dp (default: 64) */\n height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface ExtractAudioDataOptions {\n /** URI of the audio file to extract data from */\n fileUri: string\n /** Start time in milliseconds (for time-based range) */\n startTimeMs?: number\n /** End time in milliseconds (for time-based range) */\n endTimeMs?: number\n /** Start position in bytes (for byte-based range) */\n position?: number\n /** Length in bytes to extract (for byte-based range) */\n length?: number\n /** Include normalized audio data in [-1, 1] range */\n includeNormalizedData?: boolean\n /** Include base64 encoded string representation of the audio data */\n includeBase64Data?: boolean\n /** Include WAV header in the PCM data (makes it a valid WAV file) */\n includeWavHeader?: boolean\n /** Logger for debugging - can pass console directly. */\n logger?: ConsoleLike\n /** Compute the checksum of the PCM data */\n computeChecksum?: boolean\n /** Target config for the normalized audio (Android and Web) */\n decodingOptions?: DecodingConfig\n}\n\nexport interface ExtractedAudioData {\n /** Raw PCM audio data */\n pcmData: Uint8Array\n /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */\n normalizedData?: Float32Array\n /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */\n base64Data?: string\n /** Sample rate in Hz (e.g., 44100, 48000) */\n sampleRate: number\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bits per sample (8, 16, or 32) */\n bitDepth: BitDepth\n /** Duration of the audio in milliseconds */\n durationMs: number\n /** PCM format identifier (e.g., \"pcm_16bit\") */\n format: PCMFormat\n /** Total number of audio samples per channel */\n samples: number\n /** Whether the pcmData includes a WAV header */\n hasWavHeader?: boolean\n /** CRC32 Checksum of PCM data */\n checksum?: number\n}\n\nexport interface UseAudioRecorderState {\n /**\n * Prepares recording with the specified configuration without starting it.\n *\n * This method eliminates the latency between calling startRecording and the actual recording beginning.\n * It pre-initializes all audio resources, requests permissions, and sets up audio sessions in advance,\n * allowing for true zero-latency recording start when startRecording is called later.\n *\n * Technical benefits:\n * - Eliminates audio pipeline initialization delay (50-300ms depending on platform)\n * - Pre-allocates audio buffers to avoid memory allocation during recording start\n * - Initializes audio hardware in advance (particularly important on iOS)\n * - Requests and verifies permissions before the critical recording moment\n *\n * Use this method when:\n * - You need zero-latency recording start (e.g., voice commands, musical applications)\n * - You're building time-sensitive applications where missing initial audio would be problematic\n * - You want to prepare resources during app initialization, screen loading, or preceding user interaction\n * - You need to ensure recording starts reliably and instantly on all platforms\n *\n * @param config - The recording configuration, identical to what you would pass to startRecording\n * @returns A promise that resolves when preparation is complete\n *\n * @example\n * // Prepare during component mounting\n * useEffect(() => {\n * prepareRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n * }, []);\n *\n * // Later when user taps record button, it starts with zero latency\n * const handleRecordPress = () => startRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n */\n prepareRecording: (_: RecordingConfig) => Promise<void>\n /** Starts recording with the specified configuration */\n startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n /** Stops the current recording and returns the recording data */\n stopRecording: () => Promise<AudioRecording | null>\n /** Pauses the current recording */\n pauseRecording: () => Promise<void>\n /** Resumes a paused recording */\n resumeRecording: () => Promise<void>\n /** Indicates whether recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number // Duration of the recording\n /** Size of the recorded audio in bytes */\n size: number // Size in bytes of the recorded audio\n /** Information about compression if enabled */\n compression?: CompressionInfo\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n\n/**\n * Represents an event emitted during the trimming process to report progress.\n */\nexport interface TrimProgressEvent {\n /**\n * The percentage of the trimming process that has been completed, ranging from 0 to 100.\n */\n progress: number\n\n /**\n * The number of bytes that have been processed so far. This is optional and may not be provided in all implementations.\n */\n bytesProcessed?: number\n\n /**\n * The total number of bytes to process. This is optional and may not be provided in all implementations.\n */\n totalBytes?: number\n}\n\n/**\n * Defines a time range in milliseconds for trimming operations.\n */\nexport interface TimeRange {\n /**\n * The start time of the range in milliseconds.\n */\n startTimeMs: number\n\n /**\n * The end time of the range in milliseconds.\n */\n endTimeMs: number\n}\n\n/**\n * Options for configuring the audio trimming operation.\n */\nexport interface TrimAudioOptions {\n /**\n * The URI of the audio file to trim.\n */\n fileUri: string\n\n /**\n * The mode of trimming to apply.\n * - `'single'`: Trims the audio to a single range defined by `startTimeMs` and `endTimeMs`.\n * - `'keep'`: Keeps the specified `ranges` and removes all other portions of the audio.\n * - `'remove'`: Removes the specified `ranges` and keeps the remaining portions of the audio.\n * @default 'single'\n */\n mode?: 'single' | 'keep' | 'remove'\n\n /**\n * An array of time ranges to keep or remove, depending on the `mode`.\n * - Required for `'keep'` and `'remove'` modes.\n * - Ignored when `mode` is `'single'`.\n */\n ranges?: TimeRange[]\n\n /**\n * The start time in milliseconds for the `'single'` mode.\n * - If not provided, trimming starts from the beginning of the audio (0 ms).\n */\n startTimeMs?: number\n\n /**\n * The end time in milliseconds for the `'single'` mode.\n * - If not provided, trimming extends to the end of the audio.\n */\n endTimeMs?: number\n\n /**\n * The name of the output file. If not provided, a default name will be generated.\n */\n outputFileName?: string\n\n /**\n * Configuration for the output audio format.\n */\n outputFormat?: {\n /**\n * The format of the output audio file.\n * - `'wav'`: Waveform Audio File Format (uncompressed).\n * - `'aac'`: Advanced Audio Coding (compressed). Not supported on web platforms.\n * - `'opus'`: Opus Interactive Audio Codec (compressed).\n */\n format: 'wav' | 'aac' | 'opus'\n\n /**\n * The sample rate of the output audio in Hertz (Hz).\n * - If not provided, the input audio's sample rate is used.\n */\n sampleRate?: number\n\n /**\n * The number of channels in the output audio (e.g., 1 for mono, 2 for stereo).\n * - If not provided, the input audio's channel count is used.\n */\n channels?: number\n\n /**\n * The bit depth of the output audio, applicable to PCM formats like `'wav'`.\n * - If not provided, the input audio's bit depth is used.\n */\n bitDepth?: number\n\n /**\n * The bitrate of the output audio in bits per second, applicable to compressed formats like `'aac'`.\n * - If not provided, a default bitrate is used based on the format.\n */\n bitrate?: number\n }\n\n /**\n * Options for decoding the input audio file.\n * - See `DecodingConfig` for details.\n */\n decodingOptions?: DecodingConfig\n}\n\n/**\n * Result of the audio trimming operation.\n */\nexport interface TrimAudioResult {\n /**\n * The URI of the trimmed audio file.\n */\n uri: string\n\n /**\n * The filename of the trimmed audio file.\n */\n filename: string\n\n /**\n * The duration of the trimmed audio in milliseconds.\n */\n durationMs: number\n\n /**\n * The size of the trimmed audio file in bytes.\n */\n size: number\n\n /**\n * The sample rate of the trimmed audio in Hertz (Hz).\n */\n sampleRate: number\n\n /**\n * The number of channels in the trimmed audio (e.g., 1 for mono, 2 for stereo).\n */\n channels: number\n\n /**\n * The bit depth of the trimmed audio, applicable to PCM formats like `'wav'`.\n */\n bitDepth: number\n\n /**\n * The MIME type of the trimmed audio file (e.g., `'audio/wav'`, `'audio/mpeg'`).\n */\n mimeType: string\n\n /**\n * Information about compression if the output format is compressed.\n */\n compression?: {\n /**\n * The format of the compression (e.g., `'aac'`, `'mp3'`, `'opus'`).\n */\n format: string\n\n /**\n * The bitrate of the compressed audio in bits per second.\n */\n bitrate: number\n\n /**\n * The size of the compressed audio file in bytes.\n */\n size: number\n }\n\n /**\n * Information about the processing time.\n */\n processingInfo?: {\n /**\n * The time it took to process the audio in milliseconds.\n */\n durationMs: number\n }\n}\n"]}