@siteed/expo-audio-stream 2.1.0 → 2.2.0

package/build/utils/audioProcessing.js

@@ -1,133 +0,0 @@
-// packages/expo-audio-stream/src/utils/audioProcessing.ts
-import { Platform } from 'react-native';
-export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targetChannels, normalizeAudio, startTimeMs, endTimeMs, position, length, audioContext, logger, }) {
-    if (Platform.OS !== 'web') {
-        throw new Error('processAudioBuffer is only supported on web');
-    }
-    let ctx;
-    let buffer;
-    try {
-        // Log initial parameters
-        logger?.debug('Process audio buffer - Initial params:', {
-            hasArrayBuffer: !!arrayBuffer,
-            fileUri,
-            targetSampleRate,
-            targetChannels,
-            normalizeAudio,
-            startTimeMs,
-            endTimeMs,
-            position,
-            length,
-        });
-        // Get the audio data
-        let audioData;
-        if (arrayBuffer) {
-            audioData = arrayBuffer;
-        }
-        else if (fileUri) {
-            const response = await fetch(fileUri);
-            if (!response.ok) {
-                throw new Error(`Failed to fetch fileUri: ${response.statusText}`);
-            }
-            audioData = await response.arrayBuffer();
-        }
-        else {
-            throw new Error('Either arrayBuffer or fileUri must be provided');
-        }
-        logger?.debug('Audio data loaded:', {
-            byteLength: audioData.byteLength,
-            firstBytes: Array.from(new Uint8Array(audioData.slice(0, 16))),
-        });
-        // Create context at original sample rate first
-        ctx =
-            audioContext ||
-                new (window.AudioContext || window.webkitAudioContext)();
-        buffer = await ctx.decodeAudioData(audioData);
-        logger?.debug('Decoded audio buffer:', {
-            originalChannels: buffer.numberOfChannels,
-            originalSampleRate: buffer.sampleRate,
-            originalDuration: buffer.duration,
-            originalLength: buffer.length,
-        });
-        // Calculate time range
-        const startSample = startTimeMs !== undefined
-            ? Math.floor((startTimeMs / 1000) * buffer.sampleRate)
-            : position !== undefined
-                ? Math.floor(position / 2)
-                : 0;
-        // Fix: Adjust position calculation based on original sample rate
-        // When position is provided in bytes, we need to account for the original sample rate
-        const bytesPerSample = 2; // 16-bit audio = 2 bytes per sample
-        const adjustedStartSample = position !== undefined
-            ? Math.floor((position / bytesPerSample) *
-                (buffer.sampleRate / targetSampleRate))
-            : startSample;
-        const samplesNeeded = length !== undefined
-            ? Math.floor((length / bytesPerSample) *
-                (buffer.sampleRate / targetSampleRate))
-            : endTimeMs !== undefined && startTimeMs !== undefined
-                ? Math.floor(((endTimeMs - startTimeMs) / 1000) * buffer.sampleRate)
-                : buffer.length - adjustedStartSample;
-        logger?.debug('Sample calculations (adjusted):', {
-            originalStartSample: startSample,
-            adjustedStartSample,
-            samplesNeeded,
-            originalSampleRate: buffer.sampleRate,
-            targetSampleRate,
-            conversionRatio: buffer.sampleRate / targetSampleRate,
-            expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,
-        });
-        // Create temporary buffer for the segment
-        const segmentBuffer = ctx.createBuffer(buffer.numberOfChannels, samplesNeeded, buffer.sampleRate);
-        // Copy the segment
-        for (let channel = 0; channel < buffer.numberOfChannels; channel++) {
-            const channelData = buffer.getChannelData(channel);
-            const segmentData = segmentBuffer.getChannelData(channel);
-            for (let i = 0; i < samplesNeeded; i++) {
-                segmentData[i] = channelData[adjustedStartSample + i];
-            }
-        }
-        // Create offline context for resampling
-        const offlineCtx = new OfflineAudioContext(targetChannels, Math.ceil((samplesNeeded * targetSampleRate) / buffer.sampleRate), targetSampleRate);
-        // Create source and connect
-        const source = offlineCtx.createBufferSource();
-        source.buffer = segmentBuffer;
-        source.connect(offlineCtx.destination);
-        // Render at new sample rate
-        source.start();
-        const processedBuffer = await offlineCtx.startRendering();
-        // Get the final audio data
-        const channelData = processedBuffer.getChannelData(0);
-        const durationMs = Math.round((samplesNeeded / buffer.sampleRate) * 1000);
-        logger?.debug('Final processed audio:', {
-            outputSamples: channelData.length,
-            outputSampleRate: targetSampleRate,
-            durationMs,
-        });
-        return {
-            buffer: processedBuffer,
-            channelData,
-            samples: channelData.length,
-            durationMs,
-            sampleRate: targetSampleRate,
-            channels: processedBuffer.numberOfChannels,
-        };
-    }
-    catch (error) {
-        logger?.error('Failed to process audio buffer:', {
-            error,
-            position,
-            length,
-            startTimeMs,
-            endTimeMs,
-            bufferLength: buffer?.length,
-        });
-        throw error;
-    }
-    finally {
-        if (!audioContext && ctx) {
-            await ctx.close();
-        }
-    }
-}
-//# sourceMappingURL=audioProcessing.js.map

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packages/expo-audio-stream/src/utils/audioProcessing.ts\nimport { Platform } from 'react-native'\n\nimport { ConsoleLike } from '../ExpoAudioStream.types'\n\nexport interface ProcessAudioBufferOptions {\n    arrayBuffer?: ArrayBuffer\n    fileUri?: string\n    targetSampleRate: number\n    targetChannels: number\n    normalizeAudio: boolean\n    startTimeMs?: number\n    endTimeMs?: number\n    position?: number\n    length?: number\n    audioContext?: AudioContext\n    logger?: ConsoleLike\n}\n\nexport interface ProcessedAudioData {\n    channelData: Float32Array\n    samples: number\n    durationMs: number\n    sampleRate: number\n    channels: number\n    buffer: AudioBuffer\n}\n\nexport async function processAudioBuffer({\n    arrayBuffer,\n    fileUri,\n    targetSampleRate,\n    targetChannels,\n    normalizeAudio,\n    startTimeMs,\n    endTimeMs,\n    position,\n    length,\n    audioContext,\n    logger,\n}: ProcessAudioBufferOptions): Promise<ProcessedAudioData> {\n    if (Platform.OS !== 'web') {\n        throw new Error('processAudioBuffer is only supported on web')\n    }\n\n    let ctx: AudioContext | undefined\n    let buffer: AudioBuffer | undefined\n\n    try {\n        // Log initial parameters\n        logger?.debug('Process audio buffer - Initial params:', {\n            hasArrayBuffer: !!arrayBuffer,\n            fileUri,\n            targetSampleRate,\n            targetChannels,\n            normalizeAudio,\n            startTimeMs,\n            endTimeMs,\n            position,\n            length,\n        })\n\n        // Get the audio data\n        let audioData: ArrayBuffer\n        if (arrayBuffer) {\n            audioData = arrayBuffer\n        } else if (fileUri) {\n            const response = await fetch(fileUri)\n            if (!response.ok) {\n                throw new Error(\n                    `Failed to fetch fileUri: ${response.statusText}`\n                )\n            }\n            audioData = await response.arrayBuffer()\n        } else {\n            throw new Error('Either arrayBuffer or fileUri must be provided')\n        }\n\n        logger?.debug('Audio data loaded:', {\n            byteLength: audioData.byteLength,\n            firstBytes: Array.from(new Uint8Array(audioData.slice(0, 16))),\n        })\n\n        // Create context at original sample rate first\n        ctx =\n            audioContext ||\n            new (window.AudioContext || (window as any).webkitAudioContext)()\n        buffer = await ctx.decodeAudioData(audioData)\n\n        logger?.debug('Decoded audio buffer:', {\n            originalChannels: buffer.numberOfChannels,\n            originalSampleRate: buffer.sampleRate,\n            originalDuration: buffer.duration,\n            originalLength: buffer.length,\n        })\n\n        // Calculate time range\n        const startSample =\n            startTimeMs !== undefined\n                ? Math.floor((startTimeMs / 1000) * buffer.sampleRate)\n                : position !== undefined\n                  ? Math.floor(position / 2)\n                  : 0\n\n        // Fix: Adjust position calculation based on original sample rate\n        // When position is provided in bytes, we need to account for the original sample rate\n        const bytesPerSample = 2 // 16-bit audio = 2 bytes per sample\n        const adjustedStartSample =\n            position !== undefined\n                ? Math.floor(\n                      (position / bytesPerSample) *\n                          (buffer.sampleRate / targetSampleRate)\n                  )\n                : startSample\n\n        const samplesNeeded =\n            length !== undefined\n                ? Math.floor(\n                      (length / bytesPerSample) *\n                          (buffer.sampleRate / targetSampleRate)\n                  )\n                : endTimeMs !== undefined && startTimeMs !== undefined\n                  ? Math.floor(\n                        ((endTimeMs - startTimeMs) / 1000) * buffer.sampleRate\n                    )\n                  : buffer.length - adjustedStartSample\n\n        logger?.debug('Sample calculations (adjusted):', {\n            originalStartSample: startSample,\n            adjustedStartSample,\n            samplesNeeded,\n            originalSampleRate: buffer.sampleRate,\n            targetSampleRate,\n            conversionRatio: buffer.sampleRate / targetSampleRate,\n            expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,\n        })\n\n        // Create temporary buffer for the segment\n        const segmentBuffer = ctx.createBuffer(\n            buffer.numberOfChannels,\n            samplesNeeded,\n            buffer.sampleRate\n        )\n\n        // Copy the segment\n        for (let channel = 0; channel < buffer.numberOfChannels; channel++) {\n            const channelData = buffer.getChannelData(channel)\n            const segmentData = segmentBuffer.getChannelData(channel)\n            for (let i = 0; i < samplesNeeded; i++) {\n                segmentData[i] = channelData[adjustedStartSample + i]\n            }\n        }\n\n        // Create offline context for resampling\n        const offlineCtx = new OfflineAudioContext(\n            targetChannels,\n            Math.ceil((samplesNeeded * targetSampleRate) / buffer.sampleRate),\n            targetSampleRate\n        )\n\n        // Create source and connect\n        const source = offlineCtx.createBufferSource()\n        source.buffer = segmentBuffer\n        source.connect(offlineCtx.destination)\n\n        // Render at new sample rate\n        source.start()\n        const processedBuffer = await offlineCtx.startRendering()\n\n        // Get the final audio data\n        const channelData = processedBuffer.getChannelData(0)\n        const durationMs = Math.round(\n            (samplesNeeded / buffer.sampleRate) * 1000\n        )\n\n        logger?.debug('Final processed audio:', {\n            outputSamples: channelData.length,\n            outputSampleRate: targetSampleRate,\n            durationMs,\n        })\n\n        return {\n            buffer: processedBuffer,\n            channelData,\n            samples: channelData.length,\n            durationMs,\n            sampleRate: targetSampleRate,\n            channels: processedBuffer.numberOfChannels,\n        }\n    } catch (error) {\n        logger?.error('Failed to process audio buffer:', {\n            error,\n            position,\n            length,\n            startTimeMs,\n            endTimeMs,\n            bufferLength: buffer?.length,\n        })\n        throw error\n    } finally {\n        if (!audioContext && ctx) {\n            await ctx.close()\n        }\n    }\n}\n"]}

package/build/utils/concatenateBuffers.d.ts

@@ -1,8 +0,0 @@
-/**
- * Concatenates an array of ArrayBuffers into a single ArrayBuffer.
- *
- * @param buffers - An array of ArrayBuffers to be concatenated.
- * @returns A single ArrayBuffer containing the concatenated data.
- */
-export declare const concatenateBuffers: (buffers: ArrayBuffer[]) => ArrayBuffer;
-//# sourceMappingURL=concatenateBuffers.d.ts.map

package/build/utils/concatenateBuffers.d.ts.map

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package/build/utils/concatenateBuffers.js

@@ -1,21 +0,0 @@
-/**
- * Concatenates an array of ArrayBuffers into a single ArrayBuffer.
- *
- * @param buffers - An array of ArrayBuffers to be concatenated.
- * @returns A single ArrayBuffer containing the concatenated data.
- */
-export const concatenateBuffers = (buffers) => {
-    // Filter out any undefined or null buffers
-    const validBuffers = buffers.filter((buffer) => buffer);
-    const totalLength = validBuffers.reduce((sum, buffer) => sum + buffer.byteLength, 0);
-    // Create a new Uint8Array to hold the concatenated result
-    const result = new Uint8Array(totalLength);
-    // Offset to keep track of the current position in the result array
-    let offset = 0;
-    for (const buffer of validBuffers) {
-        result.set(new Uint8Array(buffer), offset);
-        offset += buffer.byteLength;
-    }
-    return result.buffer;
-};
-//# sourceMappingURL=concatenateBuffers.js.map

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package/build/utils/convertPCMToFloat32.d.ts

@@ -1,13 +0,0 @@
-import { ConsoleLike } from '../ExpoAudioStream.types';
-export declare const WAV_HEADER_SIZE = 44;
-export declare const convertPCMToFloat32: ({ bitDepth, buffer, skipWavHeader, logger, }: {
-    buffer: ArrayBuffer;
-    bitDepth: number;
-    skipWavHeader?: boolean | undefined;
-    logger?: ConsoleLike | undefined;
-}) => Promise<{
-    pcmValues: Float32Array;
-    min: number;
-    max: number;
-}>;
-//# sourceMappingURL=convertPCMToFloat32.d.ts.map

package/build/utils/convertPCMToFloat32.d.ts.map

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package/build/utils/convertPCMToFloat32.js

@@ -1,120 +0,0 @@
-import { Platform } from 'react-native';
-import { getWavFileInfo } from './getWavFileInfo';
-export const WAV_HEADER_SIZE = 44;
-const convertSample = (dataView, offset, bitDepth) => {
-    switch (bitDepth) {
-        case 8:
-            return (dataView.getUint8(offset) - 128) / 128;
-        case 16:
-            return dataView.getInt16(offset, true) / 32768;
-        case 24:
-            return (((dataView.getUint8(offset) |
-                (dataView.getUint8(offset + 1) << 8) |
-                (dataView.getUint8(offset + 2) << 16)) /
-                8388608) *
-                2 -
-                1);
-        case 32:
-            return dataView.getFloat32(offset, true);
-        default:
-            throw new Error(`Unsupported bit depth: ${bitDepth}`);
-    }
-};
-const convertSampleNative = (array, startIndex, bitDepth) => {
-    switch (bitDepth) {
-        case 8:
-            return (array[startIndex] - 128) / 128;
-        case 16: {
-            // Handle 16-bit PCM using Uint8Array directly
-            const low = array[startIndex];
-            const high = array[startIndex + 1];
-            const value = (high << 8) | low;
-            // Convert to signed 16-bit
-            return (value > 32767 ? value - 65536 : value) / 32768;
-        }
-        case 24: {
-            const byte1 = array[startIndex];
-            const byte2 = array[startIndex + 1];
-            const byte3 = array[startIndex + 2];
-            const value = (byte3 << 16) | (byte2 << 8) | byte1;
-            return (((value > 8388607 ? value - 16777216 : value) / 8388608) * 2 - 1);
-        }
-        case 32: {
-            // Assuming 32-bit float
-            const view = new DataView(array.buffer, startIndex, 4);
-            return view.getFloat32(0, true);
-        }
-        default:
-            throw new Error(`Unsupported bit depth: ${bitDepth}`);
-    }
-};
-export const convertPCMToFloat32 = async ({ bitDepth, buffer, skipWavHeader = false, logger, }) => {
-    try {
-        logger?.debug(`Converting PCM to Float32: bitDepth: ${bitDepth}, buffer.byteLength: ${buffer.byteLength}`);
-        let headerOffset = 0;
-        if (!skipWavHeader) {
-            const wavFileInfo = await getWavFileInfo(buffer);
-            headerOffset = wavFileInfo.dataChunkOffset;
-            logger?.debug(`Using WAV header offset: ${headerOffset}`);
-        }
-        // Convert ArrayBuffer to Uint8Array for more efficient native handling
-        const uint8Array = new Uint8Array(buffer);
-        const dataLength = buffer.byteLength - headerOffset;
-        const bytesPerSample = bitDepth / 8;
-        const sampleLength = Math.floor(dataLength / bytesPerSample);
-        // Create result array using SharedArrayBuffer for better memory handling
-        let float32Array;
-        try {
-            // Try using SharedArrayBuffer first
-            const sharedBuffer = new SharedArrayBuffer(sampleLength * 4);
-            float32Array = new Float32Array(sharedBuffer);
-        }
-        catch (e) {
-            // Fallback to regular ArrayBuffer if SharedArrayBuffer is not available
-            float32Array = new Float32Array(sampleLength);
-        }
-        let min = Infinity;
-        let max = -Infinity;
-        // Process in smaller chunks
-        const CHUNK_SIZE = Platform.OS === 'web' ? sampleLength : 4000; // Smaller chunks for native
-        const numChunks = Math.ceil(sampleLength / CHUNK_SIZE);
-        for (let chunk = 0; chunk < numChunks; chunk++) {
-            const startSample = chunk * CHUNK_SIZE;
-            const endSample = Math.min((chunk + 1) * CHUNK_SIZE, sampleLength);
-            // Process chunk
-            for (let i = startSample; i < endSample; i++) {
-                const startIndex = headerOffset + i * bytesPerSample;
-                if (startIndex + bytesPerSample <= uint8Array.length) {
-                    const value = Platform.OS === 'web'
-                        ? convertSample(new DataView(buffer), startIndex, bitDepth)
-                        : convertSampleNative(uint8Array, startIndex, bitDepth);
-                    if (isFinite(value)) {
-                        float32Array[i] = value;
-                        min = Math.min(min, value);
-                        max = Math.max(max, value);
-                    }
-                }
-            }
-            // Allow garbage collection between chunks on native
-            if (Platform.OS !== 'web' && chunk < numChunks - 1) {
-                await new Promise((resolve) => setTimeout(resolve, 0));
-            }
-        }
-        logger?.debug(`Conversion complete. Length: ${float32Array.length}, Range: [${min}, ${max}]`);
-        // Only log a small sample of values to avoid memory issues
-        if (logger?.debug) {
-            const sampleValues = Array.from(float32Array.slice(0, 5));
-            logger.debug('Sample values:', sampleValues);
-        }
-        return {
-            pcmValues: float32Array,
-            min,
-            max,
-        };
-    }
-    catch (error) {
-        logger?.error(`Error converting PCM to Float32`, error);
-        throw error;
-    }
-};
-//# sourceMappingURL=convertPCMToFloat32.js.map

package/build/utils/convertPCMToFloat32.js.map

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{ Platform } from 'react-native'\n\nimport { ConsoleLike } from '../ExpoAudioStream.types'\nimport { getWavFileInfo, WavFileInfo } from './getWavFileInfo'\n\nexport const WAV_HEADER_SIZE = 44\n\nconst convertSample = (\n    dataView: DataView,\n    offset: number,\n    bitDepth: number\n): number => {\n    switch (bitDepth) {\n        case 8:\n            return (dataView.getUint8(offset) - 128) / 128\n        case 16:\n            return dataView.getInt16(offset, true) / 32768\n        case 24:\n            return (\n                ((dataView.getUint8(offset) |\n                    (dataView.getUint8(offset + 1) << 8) |\n                    (dataView.getUint8(offset + 2) << 16)) /\n                    8388608) *\n                    2 -\n                1\n            )\n        case 32:\n            return dataView.getFloat32(offset, true)\n        default:\n            throw new Error(`Unsupported bit depth: ${bitDepth}`)\n    }\n}\n\nconst convertSampleNative = (\n    array: Uint8Array,\n    startIndex: number,\n    bitDepth: number\n): number => {\n    switch (bitDepth) {\n        case 8:\n            return (array[startIndex] - 128) / 128\n        case 16: {\n            // Handle 16-bit PCM using Uint8Array directly\n            const low = array[startIndex]\n            const high = array[startIndex + 1]\n            const value = (high << 8) | low\n            // Convert to signed 16-bit\n            return (value > 32767 ? value - 65536 : value) / 32768\n        }\n        case 24: {\n            const byte1 = array[startIndex]\n            const byte2 = array[startIndex + 1]\n            const byte3 = array[startIndex + 2]\n            const value = (byte3 << 16) | (byte2 << 8) | byte1\n            return (\n                ((value > 8388607 ? value - 16777216 : value) / 8388608) * 2 - 1\n            )\n        }\n        case 32: {\n            // Assuming 32-bit float\n            const view = new DataView(array.buffer, startIndex, 4)\n            return view.getFloat32(0, true)\n        }\n        default:\n            throw new Error(`Unsupported bit depth: ${bitDepth}`)\n    }\n}\n\nexport const convertPCMToFloat32 = async ({\n    bitDepth,\n    buffer,\n    skipWavHeader = false,\n    logger,\n}: {\n    buffer: ArrayBuffer\n    bitDepth: number\n    skipWavHeader?: boolean\n    logger?: ConsoleLike\n}): Promise<{ pcmValues: Float32Array; min: number; max: number }> => {\n    try {\n        logger?.debug(\n            `Converting PCM to Float32: bitDepth: ${bitDepth}, buffer.byteLength: ${buffer.byteLength}`\n        )\n\n        let headerOffset = 0\n        if (!skipWavHeader) {\n            const wavFileInfo: WavFileInfo = await getWavFileInfo(buffer)\n            headerOffset = wavFileInfo.dataChunkOffset\n            logger?.debug(`Using WAV header offset: ${headerOffset}`)\n        }\n\n        // Convert ArrayBuffer to Uint8Array for more efficient native handling\n        const uint8Array = new Uint8Array(buffer)\n        const dataLength = buffer.byteLength - headerOffset\n        const bytesPerSample = bitDepth / 8\n        const sampleLength = Math.floor(dataLength / bytesPerSample)\n\n        // Create result array using SharedArrayBuffer for better memory handling\n        let float32Array: Float32Array\n        try {\n            // Try using SharedArrayBuffer first\n            const sharedBuffer = new SharedArrayBuffer(sampleLength * 4)\n            float32Array = new Float32Array(sharedBuffer)\n        } catch (e) {\n            // Fallback to regular ArrayBuffer if SharedArrayBuffer is not available\n            float32Array = new Float32Array(sampleLength)\n        }\n\n        let min = Infinity\n        let max = -Infinity\n\n        // Process in smaller chunks\n        const CHUNK_SIZE = Platform.OS === 'web' ? sampleLength : 4000 // Smaller chunks for native\n        const numChunks = Math.ceil(sampleLength / CHUNK_SIZE)\n\n        for (let chunk = 0; chunk < numChunks; chunk++) {\n            const startSample = chunk * CHUNK_SIZE\n            const endSample = Math.min((chunk + 1) * CHUNK_SIZE, sampleLength)\n\n            // Process chunk\n            for (let i = startSample; i < endSample; i++) {\n                const startIndex = headerOffset + i * bytesPerSample\n                if (startIndex + bytesPerSample <= uint8Array.length) {\n                    const value =\n                        Platform.OS === 'web'\n                            ? convertSample(\n                                  new DataView(buffer),\n                                  startIndex,\n                                  bitDepth\n                              )\n                            : convertSampleNative(\n                                  uint8Array,\n                                  startIndex,\n                                  bitDepth\n                              )\n\n                    if (isFinite(value)) {\n                        float32Array[i] = value\n                        min = Math.min(min, value)\n                        max = Math.max(max, value)\n                    }\n                }\n            }\n\n            // Allow garbage collection between chunks on native\n            if (Platform.OS !== 'web' && chunk < numChunks - 1) {\n                await new Promise((resolve) => setTimeout(resolve, 0))\n            }\n        }\n\n        logger?.debug(\n            `Conversion complete. Length: ${float32Array.length}, Range: [${min}, ${max}]`\n        )\n\n        // Only log a small sample of values to avoid memory issues\n        if (logger?.debug) {\n            const sampleValues = Array.from(float32Array.slice(0, 5))\n            logger.debug('Sample values:', sampleValues)\n        }\n\n        return {\n            pcmValues: float32Array,\n            min,\n            max,\n        }\n    } catch (error: unknown) {\n        logger?.error(`Error converting PCM to Float32`, error)\n        throw error\n    }\n}\n"]}

package/build/utils/encodingToBitDepth.d.ts

@@ -1,5 +0,0 @@
-import { BitDepth, EncodingType } from '../ExpoAudioStream.types';
-export declare const encodingToBitDepth: ({ encoding, }: {
-    encoding: EncodingType;
-}) => BitDepth;
-//# sourceMappingURL=encodingToBitDepth.d.ts.map

package/build/utils/encodingToBitDepth.d.ts.map

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package/build/utils/encodingToBitDepth.js

@@ -1,13 +0,0 @@
-export const encodingToBitDepth = ({ encoding, }) => {
-    switch (encoding) {
-        case 'pcm_32bit':
-            return 32;
-        case 'pcm_16bit':
-            return 16;
-        case 'pcm_8bit':
-            return 8;
-        default:
-            throw new Error(`Unsupported encoding type: ${encoding}`);
-    }
-};
-//# sourceMappingURL=encodingToBitDepth.js.map

package/build/utils/encodingToBitDepth.js.map

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package/build/utils/getWavFileInfo.d.ts

@@ -1,26 +0,0 @@
-import { BitDepth, SampleRate } from '../ExpoAudioStream.types';
-/**
- * Interface representing the metadata of a WAV file.
- */
-export interface WavFileInfo {
-    sampleRate: SampleRate;
-    numChannels: number;
-    bitDepth: BitDepth;
-    size: number;
-    durationMs: number;
-    audioFormatDescription: string;
-    byteRate: number;
-    blockAlign: number;
-    creationDateTime?: string;
-    comments?: string;
-    compressionType?: string;
-    dataChunkOffset: number;
-}
-/**
- * Extracts metadata from a WAV buffer.
- *
- * @param arrayBuffer - The array buffer containing the WAV data.
- * @returns A promise that resolves to the extracted metadata.
- */
-export declare const getWavFileInfo: (arrayBuffer: ArrayBuffer) => Promise<WavFileInfo>;
-//# sourceMappingURL=getWavFileInfo.d.ts.map

package/build/utils/getWavFileInfo.d.ts.map

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package/build/utils/getWavFileInfo.js

@@ -1,92 +0,0 @@
-// packages/expo-audio-stream/src/utils/getWavFileInfo.ts
-import { DATA_CHUNK_ID, DEFAULT_BIT_DEPTH, DEFAULT_SAMPLE_RATE, FMT_CHUNK_ID, INFO_CHUNK_ID, RIFF_HEADER, WAVE_HEADER, } from '../constants';
-// Audio format descriptions
-const AUDIO_FORMATS = {
-    1: 'PCM',
-    3: 'IEEE float',
-    6: '8-bit ITU-T G.711 A-law',
-    7: '8-bit ITU-T G.711 µ-law',
-    65534: 'WAVE_FORMAT_EXTENSIBLE',
-};
-/**
- * Extracts metadata from a WAV buffer.
- *
- * @param arrayBuffer - The array buffer containing the WAV data.
- * @returns A promise that resolves to the extracted metadata.
- */
-export const getWavFileInfo = async (arrayBuffer) => {
-    const view = new DataView(arrayBuffer);
-    // Check if the file is a valid RIFF/WAVE file
-    const riffHeader = view.getUint32(0, false);
-    const waveHeader = view.getUint32(8, false);
-    if (riffHeader !== RIFF_HEADER || waveHeader !== WAVE_HEADER) {
-        throw new Error('Invalid WAV file');
-    }
-    // Initialize variables for the metadata
-    let fmtChunkOffset = 12;
-    let sampleRate = DEFAULT_SAMPLE_RATE;
-    let numChannels = 0;
-    let bitDepth = DEFAULT_BIT_DEPTH;
-    let dataChunkSize = 0;
-    let audioFormat = 0;
-    let byteRate = 0;
-    let blockAlign = 0;
-    let creationDateTime = '';
-    let comments = '';
-    let dataChunkOffset = 0;
-    // Parse chunks to find the "fmt " and "data" chunks
-    while (fmtChunkOffset < view.byteLength) {
-        const chunkId = view.getUint32(fmtChunkOffset, false);
-        const chunkSize = view.getUint32(fmtChunkOffset + 4, true);
-        if (chunkId === FMT_CHUNK_ID) {
-            // "fmt "
-            audioFormat = view.getUint16(fmtChunkOffset + 8, true);
-            if (!AUDIO_FORMATS[audioFormat]) {
-                throw new Error('Unsupported WAV file format');
-            }
-            numChannels = view.getUint16(fmtChunkOffset + 10, true);
-            sampleRate = view.getUint32(fmtChunkOffset + 12, true);
-            byteRate = view.getUint32(fmtChunkOffset + 16, true);
-            blockAlign = view.getUint16(fmtChunkOffset + 20, true);
-            bitDepth = view.getUint16(fmtChunkOffset + 22, true);
-        }
-        else if (chunkId === DATA_CHUNK_ID) {
-            // "data"
-            dataChunkSize = chunkSize;
-            dataChunkOffset = fmtChunkOffset + 8; // Position after chunk header
-            break;
-        }
-        else if (chunkId === INFO_CHUNK_ID) {
-            // "INFO"
-            // Read INFO chunk (assuming it contains a text-based creation date/time and comments)
-            const infoStart = fmtChunkOffset + 8;
-            const infoText = new TextDecoder().decode(new Uint8Array(arrayBuffer.slice(infoStart, infoStart + chunkSize)));
-            const infoParts = infoText.split('\0');
-            creationDateTime = infoParts[0];
-            comments = infoParts[1];
-        }
-        fmtChunkOffset += 8 + chunkSize;
-    }
-    if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {
-        throw new Error('Incomplete WAV file information');
-    }
-    // Calculate duration
-    const bytesPerSample = bitDepth / 8;
-    const numSamples = dataChunkSize / (numChannels * bytesPerSample);
-    const durationMs = (numSamples / sampleRate) * 1000;
-    return {
-        sampleRate,
-        numChannels,
-        bitDepth,
-        size: arrayBuffer.byteLength,
-        durationMs,
-        audioFormatDescription: AUDIO_FORMATS[audioFormat],
-        byteRate,
-        blockAlign,
-        creationDateTime: creationDateTime || undefined,
-        comments: comments || undefined,
-        compressionType: audioFormat === 1 ? 'None' : AUDIO_FORMATS[audioFormat],
-        dataChunkOffset,
-    };
-};
-//# sourceMappingURL=getWavFileInfo.js.map

package/build/utils/getWavFileInfo.js.map

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The array buffer containing the WAV data.\n * @returns A promise that resolves to the extracted metadata.\n */\nexport const getWavFileInfo = async (\n    arrayBuffer: ArrayBuffer\n): Promise<WavFileInfo> => {\n    const view = new DataView(arrayBuffer)\n\n    // Check if the file is a valid RIFF/WAVE file\n    const riffHeader = view.getUint32(0, false)\n    const waveHeader = view.getUint32(8, false)\n    if (riffHeader !== RIFF_HEADER || waveHeader !== WAVE_HEADER) {\n        throw new Error('Invalid WAV file')\n    }\n\n    // Initialize variables for the metadata\n    let fmtChunkOffset = 12\n    let sampleRate: SampleRate = DEFAULT_SAMPLE_RATE\n    let numChannels = 0\n    let bitDepth: BitDepth = DEFAULT_BIT_DEPTH\n    let dataChunkSize = 0\n    let audioFormat = 0\n    let byteRate = 0\n    let blockAlign = 0\n    let creationDateTime = ''\n    let comments = ''\n    let dataChunkOffset = 0\n\n    // Parse chunks to find the \"fmt \" and \"data\" chunks\n    while (fmtChunkOffset < view.byteLength) {\n        const chunkId = view.getUint32(fmtChunkOffset, false)\n        const chunkSize = view.getUint32(fmtChunkOffset + 4, true)\n        if (chunkId === FMT_CHUNK_ID) {\n            // \"fmt \"\n            audioFormat = view.getUint16(fmtChunkOffset + 8, true)\n            if (!AUDIO_FORMATS[audioFormat]) {\n                throw new Error('Unsupported WAV file format')\n            }\n            numChannels = view.getUint16(fmtChunkOffset + 10, true)\n            sampleRate = view.getUint32(fmtChunkOffset + 12, true) as SampleRate\n            byteRate = view.getUint32(fmtChunkOffset + 16, true)\n            blockAlign = view.getUint16(fmtChunkOffset + 20, true)\n            bitDepth = view.getUint16(fmtChunkOffset + 22, true) as BitDepth\n        } else if (chunkId === DATA_CHUNK_ID) {\n            // \"data\"\n            dataChunkSize = chunkSize\n            dataChunkOffset = fmtChunkOffset + 8 // Position after chunk header\n            break\n        } else if (chunkId === INFO_CHUNK_ID) {\n            // \"INFO\"\n            // Read INFO chunk (assuming it contains a text-based creation date/time and comments)\n            const infoStart = fmtChunkOffset + 8\n            const infoText = new TextDecoder().decode(\n                new Uint8Array(\n                    arrayBuffer.slice(infoStart, infoStart + chunkSize)\n                )\n            )\n            const infoParts = infoText.split('\\0')\n            creationDateTime = infoParts[0]\n            comments = infoParts[1]\n        }\n        fmtChunkOffset += 8 + chunkSize\n    }\n\n    if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {\n        throw new Error('Incomplete WAV file information')\n    }\n\n    // Calculate duration\n    const bytesPerSample = bitDepth / 8\n    const numSamples = dataChunkSize / (numChannels * bytesPerSample)\n    const durationMs = (numSamples / sampleRate) * 1000\n\n    return {\n        sampleRate,\n        numChannels,\n        bitDepth,\n        size: arrayBuffer.byteLength,\n        durationMs,\n        audioFormatDescription: AUDIO_FORMATS[audioFormat],\n        byteRate,\n        blockAlign,\n        creationDateTime: creationDateTime || undefined,\n        comments: comments || undefined,\n        compressionType:\n            audioFormat === 1 ? 'None' : AUDIO_FORMATS[audioFormat],\n        dataChunkOffset,\n    }\n}\n"]}

package/build/utils/writeWavHeader.d.ts

@@ -1,49 +0,0 @@
-/**
- * Options for creating a WAV header.
- */
-export interface WavHeaderOptions {
-    /** Optional buffer containing audio data. If provided, it will be combined with the header. */
-    buffer?: ArrayBuffer;
-    /** The sample rate of the audio in Hz (e.g., 44100). */
-    sampleRate: number;
-    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */
-    numChannels: number;
-    /** The bit depth of the audio (e.g., 16, 24, or 32). */
-    bitDepth: number;
-}
-/**
- * Writes or updates a WAV (RIFF) header based on the provided options.
- *
- * This function can be used in three ways:
- * 1. To create a standalone WAV header (when no buffer is provided).
- * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
- * 3. To update an existing WAV header in the provided buffer.
- *
- * For streaming audio where the final size is unknown, this function sets the size fields
- * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
- * `updateWavHeaderSize` function once the final size is known.
- *
- * @param options - The options for creating or updating the WAV header.
- * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
- *
- * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
- *
- * @example
- * // Create a standalone WAV header
- * const header = writeWavHeader({
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- *
- * @example
- * // Create a WAV header and combine it with audio data
- * const completeWav = writeWavHeader({
- *   buffer: audioData,
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- */
-export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, }: WavHeaderOptions) => ArrayBuffer;
-//# sourceMappingURL=writeWavHeader.d.ts.map

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