@siteed/expo-audio-studio 2.18.5 → 3.0.0

package/build/esm/workers/InlineFeaturesExtractor.web.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"InlineFeaturesExtractor.web.js","sourceRoot":"","sources":["../../../src/workers/InlineFeaturesExtractor.web.tsx"],"names":[],"mappings":"AAAA,yEAAyE;AACzE,MAAM,CAAC,MAAM,uBAAuB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CAq1BtC,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        }\n    }\n\n    try {\n        // Measure actual processing time using performance.now() for higher precision\n        const processingStartTime = performance.now()\n        \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        const processingEndTime = performance.now()\n        const actualExtractionTimeMs = processingEndTime - processingStartTime\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                extractionTimeMs: actualExtractionTimeMs,\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/esm/workers/inlineAudioWebWorker.web.js

@@ -1,181 +0,0 @@
-// packages/expo-audio-stream/src/workers/inlineAudioWebWorker.web.tsx
-export const 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/esm/workers/inlineAudioWebWorker.web.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"inlineAudioWebWorker.web.js","sourceRoot":"","sources":["../../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":"AAAA,sEAAsE;AACtE,MAAM,CAAC,MAAM,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/types/AudioAnalysis/AudioAnalysis.types.d.ts

@@ -1,196 +0,0 @@
-import { BitDepth, ConsoleLike } from '../ExpoAudioStream.types';
-/**
- * Represents the configuration for decoding audio data.
- */
-export interface DecodingConfig {
-    /** Target sample rate for decoded audio (Android and Web) */
-    targetSampleRate?: number;
-    /** Target number of channels (Android and Web) */
-    targetChannels?: number;
-    /** Target bit depth (Android and Web) */
-    targetBitDepth?: BitDepth;
-    /** Whether to normalize audio levels (Android and Web) */
-    normalizeAudio?: boolean;
-}
-/**
- * Represents speech-related features extracted from audio.
- */
-export interface SpeechFeatures {
-    isActive: boolean;
-    speakerId?: number;
-}
-/**
- * Represents various audio features extracted from an audio signal.
- */
-export interface AudioFeatures {
-    energy?: number;
-    mfcc?: number[];
-    rms?: number;
-    minAmplitude?: number;
-    maxAmplitude?: number;
-    zcr?: number;
-    spectralCentroid?: number;
-    spectralFlatness?: number;
-    spectralRolloff?: number;
-    spectralBandwidth?: number;
-    chromagram?: number[];
-    tempo?: number;
-    hnr?: number;
-    melSpectrogram?: number[];
-    spectralContrast?: number[];
-    tonnetz?: number[];
-    pitch?: number;
-    crc32?: number;
-}
-/**
- * Options to specify which audio features to extract.
- * Note: Advanced features (spectral features, chromagram, pitch, etc.) are experimental,
- * especially during live recording, due to high processing requirements.
- */
-export interface AudioFeaturesOptions {
-    energy?: boolean;
-    rms?: boolean;
-    zcr?: boolean;
-    mfcc?: boolean;
-    spectralCentroid?: boolean;
-    spectralFlatness?: boolean;
-    spectralRolloff?: boolean;
-    spectralBandwidth?: boolean;
-    chromagram?: boolean;
-    tempo?: boolean;
-    hnr?: boolean;
-    melSpectrogram?: boolean;
-    spectralContrast?: boolean;
-    tonnetz?: boolean;
-    pitch?: boolean;
-    crc32?: boolean;
-}
-/**
- * Represents a single data point in the audio analysis.
- */
-export interface DataPoint {
-    id: number;
-    amplitude: number;
-    rms: number;
-    dB: number;
-    silent: boolean;
-    features?: AudioFeatures;
-    speech?: SpeechFeatures;
-    startTime?: number;
-    endTime?: number;
-    startPosition?: number;
-    endPosition?: number;
-    samples?: number;
-}
-/**
- * Represents the complete data from the audio analysis.
- */
-export interface AudioAnalysis {
-    segmentDurationMs: number;
-    durationMs: number;
-    /**
-     * Bit depth used for audio analysis processing.
-     *
-     * **Important**: This represents the internal processing bit depth, which may differ
-     * from the recording bit depth. Audio is typically converted to 32-bit float for
-     * analysis to ensure precision in calculations, regardless of the original recording format.
-     *
-     * Platform behavior:
-     * - iOS: Always 32 (float processing)
-     * - Android: Always 32 (float processing)
-     * - Web: Always 32 (Web Audio API standard)
-     *
-     * The actual recorded file will maintain the requested bit depth (8, 16, or 32).
-     */
-    bitDepth: number;
-    samples: number;
-    numberOfChannels: number;
-    sampleRate: number;
-    dataPoints: DataPoint[];
-    amplitudeRange: {
-        min: number;
-        max: number;
-    };
-    rmsRange: {
-        min: number;
-        max: number;
-    };
-    extractionTimeMs: number;
-    speechAnalysis?: {
-        speakerChanges: {
-            timestamp: number;
-            speakerId: number;
-        }[];
-    };
-}
-/**
- * Options for specifying a time range within an audio file.
- */
-export interface AudioRangeOptions {
-    /** Start time in milliseconds */
-    startTimeMs?: number;
-    /** End time in milliseconds */
-    endTimeMs?: number;
-}
-/**
- * Options for generating a quick preview of audio waveform.
- * This is optimized for UI rendering with a specified number of points.
- */
-export interface PreviewOptions extends AudioRangeOptions {
-    /** URI of the audio file to analyze */
-    fileUri: string;
-    /**
-     * Total number of points to generate for the preview.
-     * @default 100
-     */
-    numberOfPoints?: number;
-    /**
-     * Optional logger for debugging.
-     */
-    logger?: ConsoleLike;
-    /**
-     * Optional configuration for decoding the audio file.
-     * Defaults to:
-     * - targetSampleRate: undefined (keep original)
-     * - targetChannels: undefined (keep original)
-     * - targetBitDepth: 16
-     * - normalizeAudio: false
-     */
-    decodingOptions?: DecodingConfig;
-}
-/**
- * Options for mel-spectrogram extraction
- *
- * @experimental This feature is experimental and currently only available on Android.
- * The API may change in future versions.
- */
-export interface ExtractMelSpectrogramOptions {
-    fileUri?: string;
-    arrayBuffer?: ArrayBuffer;
-    windowSizeMs: number;
-    hopLengthMs: number;
-    nMels: number;
-    fMin?: number;
-    fMax?: number;
-    windowType?: 'hann' | 'hamming';
-    normalize?: boolean;
-    logScale?: boolean;
-    decodingOptions?: DecodingConfig;
-    startTimeMs?: number;
-    endTimeMs?: number;
-    logger?: ConsoleLike;
-}
-/**
- * Return type for mel spectrogram extraction
- *
- * @experimental This feature is experimental and currently only available on Android.
- * The API may change in future versions.
- */
-export interface MelSpectrogram {
-    spectrogram: number[][];
-    sampleRate: number;
-    nMels: number;
-    timeSteps: number;
-    durationMs: number;
-}
-//# sourceMappingURL=AudioAnalysis.types.d.ts.map

package/build/types/AudioAnalysis/AudioAnalysis.types.d.ts.map

@@ -1 +0,0 @@
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package/build/types/AudioAnalysis/extractAudioAnalysis.d.ts

@@ -1,74 +0,0 @@
-/**
- * This module provides functions for extracting and analyzing audio data.
- * - `extractAudioAnalysis`: For detailed analysis with customizable ranges and decoding options.
- * - `extractWavAudioAnalysis`: For analyzing WAV files without decoding, preserving original PCM values.
- * - `extractPreview`: For generating quick previews of audio waveforms, optimized for UI rendering.
- */
-import { ConsoleLike } from '../ExpoAudioStream.types';
-import { AudioAnalysis, AudioFeaturesOptions, DecodingConfig } from './AudioAnalysis.types';
-import { WavFileInfo } from '../utils/getWavFileInfo';
-export interface ExtractWavAudioAnalysisProps {
-    fileUri?: string;
-    wavMetadata?: WavFileInfo;
-    arrayBuffer?: ArrayBuffer;
-    bitDepth?: number;
-    durationMs?: number;
-    sampleRate?: number;
-    numberOfChannels?: number;
-    position?: number;
-    length?: number;
-    segmentDurationMs?: number;
-    features?: AudioFeaturesOptions;
-    featuresExtratorUrl?: string;
-    logger?: ConsoleLike;
-    decodingOptions?: DecodingConfig;
-}
-interface BaseExtractOptions {
-    fileUri?: string;
-    arrayBuffer?: ArrayBuffer;
-    /**
-     * Duration of each analysis segment in milliseconds. Defaults to 100ms if not specified.
-     */
-    segmentDurationMs?: number;
-    features?: AudioFeaturesOptions;
-    decodingOptions?: DecodingConfig;
-    logger?: ConsoleLike;
-}
-interface TimeRangeOptions extends BaseExtractOptions {
-    startTimeMs?: number;
-    endTimeMs?: number;
-    position?: never;
-    length?: never;
-}
-interface ByteRangeOptions extends BaseExtractOptions {
-    position?: number;
-    length?: number;
-    startTimeMs?: never;
-    endTimeMs?: never;
-}
-/**
- * Options for extracting audio analysis.
- * - For time-based analysis, provide `startTimeMs` and `endTimeMs`.
- * - For byte-based analysis, provide `position` and `length`.
- * - Do not mix time and byte ranges.
- */
-export type ExtractAudioAnalysisProps = TimeRangeOptions | ByteRangeOptions;
-/**
- * Extracts detailed audio analysis from the specified audio file or buffer.
- * Supports either time-based or byte-based ranges for flexibility in analysis.
- *
- * @param props - The options for extraction, including file URI, ranges, and decoding settings.
- * @returns A promise that resolves to the audio analysis data.
- * @throws {Error} If both time and byte ranges are provided or if required parameters are missing.
- */
-export declare function extractAudioAnalysis(props: ExtractAudioAnalysisProps): Promise<AudioAnalysis>;
-/**
- * Analyzes WAV files without decoding, preserving original PCM values.
- * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.
- *
- * @param props - The options for WAV analysis, including file URI and range.
- * @returns A promise that resolves to the audio analysis data.
- */
-export declare const extractRawWavAnalysis: ({ fileUri, segmentDurationMs, arrayBuffer, bitDepth, durationMs, sampleRate, numberOfChannels, features, logger, position, length, }: ExtractWavAudioAnalysisProps) => Promise<AudioAnalysis>;
-export {};
-//# sourceMappingURL=extractAudioAnalysis.d.ts.map

package/build/types/AudioAnalysis/extractAudioAnalysis.d.ts.map

@@ -1 +0,0 @@
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package/build/types/AudioAnalysis/extractAudioData.d.ts

@@ -1,3 +0,0 @@
-import { ExtractAudioDataOptions } from '../ExpoAudioStream.types';
-export declare const extractAudioData: (props: ExtractAudioDataOptions) => Promise<any>;
-//# sourceMappingURL=extractAudioData.d.ts.map

package/build/types/AudioAnalysis/extractAudioData.d.ts.map

@@ -1 +0,0 @@
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package/build/types/AudioAnalysis/extractMelSpectrogram.d.ts

@@ -1,14 +0,0 @@
-/**
- * @experimental This feature is experimental and currently only available on Android.
- * The API may change in future versions. The web implementation is a placeholder.
- */
-import { ExtractMelSpectrogramOptions, MelSpectrogram } from './AudioAnalysis.types';
-/**
- * Extracts a mel spectrogram from audio data
- *
- * @experimental This feature is experimental and currently only available on Android.
- * The iOS implementation will throw an "UNSUPPORTED_PLATFORM" error.
- * The web implementation is a placeholder that returns dummy data.
- */
-export declare function extractMelSpectrogram(options: ExtractMelSpectrogramOptions): Promise<MelSpectrogram>;
-//# sourceMappingURL=extractMelSpectrogram.d.ts.map

package/build/types/AudioAnalysis/extractMelSpectrogram.d.ts.map

@@ -1 +0,0 @@
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package/build/types/AudioAnalysis/extractPreview.d.ts

@@ -1,11 +0,0 @@
-import { PreviewOptions, AudioAnalysis } from './AudioAnalysis.types';
-/**
- * Generates a simplified preview of the audio waveform for quick visualization.
- * Ideal for UI rendering with a specified number of points.
- *
- * @param options - The options for the preview, including file URI and time range.
- * @returns A promise that resolves to the audio preview data.
- */
-export declare function extractPreview({ fileUri, numberOfPoints, startTimeMs, endTimeMs, // First 30 seconds
-decodingOptions, logger, }: PreviewOptions): Promise<AudioAnalysis>;
-//# sourceMappingURL=extractPreview.d.ts.map

package/build/types/AudioAnalysis/extractPreview.d.ts.map

@@ -1 +0,0 @@
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package/build/types/AudioAnalysis/extractWaveform.d.ts

@@ -1,8 +0,0 @@
-export interface ExtractWaveformProps {
-    fileUri: string;
-    numberOfSamples: number;
-    offset?: number;
-    length?: number;
-}
-export declare const extractWaveform: ({ fileUri, numberOfSamples, offset, length, }: ExtractWaveformProps) => Promise<unknown>;
-//# sourceMappingURL=extractWaveform.d.ts.map

package/build/types/AudioAnalysis/extractWaveform.d.ts.map

@@ -1 +0,0 @@
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