@siteed/expo-audio-stream 2.1.0 → 2.2.0

package/build/ExpoAudioStreamModule.js.map

@@ -1 +0,0 @@
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crc32 from 'crc-32'\nimport { requireNativeModule } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n    ExtractAudioDataOptions,\n    ExtractedAudioData,\n    BitDepth,\n    TrimAudioOptions,\n    TrimAudioResult,\n} from './ExpoAudioStream.types'\nimport {\n    ExpoAudioStreamWeb,\n    ExpoAudioStreamWebProps,\n} from './ExpoAudioStream.web'\nimport { processAudioBuffer } from './utils/audioProcessing'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nlet ExpoAudioStreamModule: any\n\nif (Platform.OS === 'web') {\n    let instance: ExpoAudioStreamWeb | null = null\n\n    ExpoAudioStreamModule = (webProps: ExpoAudioStreamWebProps) => {\n        if (!instance) {\n            instance = new ExpoAudioStreamWeb(webProps)\n        }\n        return instance\n    }\n    ExpoAudioStreamModule.requestPermissionsAsync = async () => {\n        try {\n            const stream = await navigator.mediaDevices.getUserMedia({\n                audio: true,\n            })\n            stream.getTracks().forEach((track) => track.stop())\n            return {\n                status: 'granted',\n                expires: 'never',\n                canAskAgain: true,\n                granted: true,\n            }\n        } catch {\n            return {\n                status: 'denied',\n                expires: 'never',\n                canAskAgain: true,\n                granted: false,\n            }\n        }\n    }\n    ExpoAudioStreamModule.getPermissionsAsync = async () => {\n        let maybeStatus: string | null = null\n\n        if (navigator?.permissions?.query) {\n            try {\n                const { state } = await navigator.permissions.query({\n                    name: 'microphone' as PermissionName,\n                })\n                maybeStatus = state\n            } catch {\n                maybeStatus = null\n            }\n        }\n\n        switch (maybeStatus) {\n            case 'granted':\n                return {\n                    status: 'granted',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: true,\n                }\n            case 'denied':\n                return {\n                    status: 'denied',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: false,\n                }\n            default:\n                return await ExpoAudioStreamModule.requestPermissionsAsync()\n        }\n    }\n    ExpoAudioStreamModule.extractAudioData = async (\n        options: ExtractAudioDataOptions\n    ): Promise<ExtractedAudioData> => {\n        try {\n            const {\n                fileUri,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                decodingOptions,\n                includeNormalizedData,\n                includeBase64Data,\n                includeWavHeader = false,\n                logger,\n            } = options\n\n            logger?.debug('EXTRACT AUDIO - Step 1: Initial request', {\n                fileUri,\n                extractionParams: {\n                    position,\n                    length,\n                    startTimeMs,\n                    endTimeMs,\n                },\n                decodingOptions: {\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    targetBitDepth: decodingOptions?.targetBitDepth ?? 16,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                },\n                outputOptions: {\n                    includeNormalizedData,\n                    includeBase64Data,\n                    includeWavHeader,\n                },\n            })\n\n            // Process the audio using shared helper function\n            const processedBuffer = await processAudioBuffer({\n                fileUri,\n                targetSampleRate: decodingOptions?.targetSampleRate ?? 16000,\n                targetChannels: decodingOptions?.targetChannels ?? 1,\n                normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                logger,\n            })\n\n            logger?.debug('EXTRACT AUDIO - Step 2: Audio processing complete', {\n                processedData: {\n                    samples: processedBuffer.samples,\n                    sampleRate: processedBuffer.sampleRate,\n                    channels: processedBuffer.channels,\n                    durationMs: processedBuffer.durationMs,\n                },\n            })\n\n            const channelData = processedBuffer.channelData\n            const bitDepth = (decodingOptions?.targetBitDepth ?? 16) as BitDepth\n            const bytesPerSample = bitDepth / 8\n            const numSamples = processedBuffer.samples\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion setup', {\n                channelData: {\n                    length: channelData.length,\n                    first: channelData[0],\n                    last: channelData[channelData.length - 1],\n                },\n                calculation: {\n                    bitDepth,\n                    bytesPerSample,\n                    numSamples,\n                    expectedBytes: numSamples * bytesPerSample,\n                },\n            })\n\n            // Create PCM data with correct length based on original byte length\n            const pcmData = new Uint8Array(numSamples * bytesPerSample)\n            let offset = 0\n\n            // Convert Float32 samples to PCM format\n            for (let i = 0; i < numSamples; i++) {\n                const sample = channelData[i]\n                const value = Math.max(-1, Math.min(1, sample))\n                // Convert to 16-bit signed integer\n                let intValue = Math.round(value * 32767)\n\n                // Handle negative values correctly\n                if (intValue < 0) {\n                    intValue = 65536 + intValue\n                }\n\n                // Write as little-endian\n                pcmData[offset++] = intValue & 255 // Low byte\n                pcmData[offset++] = (intValue >> 8) & 255 // High byte\n            }\n\n            const durationMs = Math.round(\n                (numSamples / processedBuffer.sampleRate) * 1000\n            )\n\n            logger?.debug('EXTRACT AUDIO - Step 4: Final output', {\n                pcmData: {\n                    length: pcmData.length,\n                    first: pcmData[0],\n                    last: pcmData[pcmData.length - 1],\n                },\n                timing: {\n                    numSamples,\n                    sampleRate: processedBuffer.sampleRate,\n                    durationMs,\n                    shouldBe3000ms: endTimeMs\n                        ? endTimeMs - (startTimeMs ?? 0) === 3000\n                        : undefined,\n                },\n            })\n\n            const result: ExtractedAudioData = {\n                pcmData: new Uint8Array(pcmData.buffer),\n                sampleRate: processedBuffer.sampleRate,\n                channels: processedBuffer.channels,\n                bitDepth,\n                durationMs,\n                format: `pcm_${bitDepth}bit` as const,\n                samples: numSamples,\n            }\n\n            // Add WAV header if requested\n            if (includeWavHeader) {\n                logger?.debug('EXTRACT AUDIO - Step 4: Adding WAV header', {\n                    originalLength: pcmData.length,\n                    newLength: result.pcmData.length,\n                    firstBytes: Array.from(result.pcmData.slice(0, 44)), // WAV header is 44 bytes\n                })\n                const wavBuffer = writeWavHeader({\n                    buffer: pcmData.buffer.slice(0, pcmData.length),\n                    sampleRate: processedBuffer.sampleRate,\n                    numChannels: processedBuffer.channels,\n                    bitDepth,\n                })\n                result.pcmData = new Uint8Array(wavBuffer)\n                result.hasWavHeader = true\n            }\n\n            if (includeNormalizedData) {\n                // // Simple approach: Create normalized data directly from the PCM data\n                // // Just convert to -1 to 1 range without any amplification\n                // const normalizedData = new Float32Array(numSamples)\n\n                // // Convert the PCM data to float values\n                // for (let i = 0; i < numSamples; i++) {\n                //     // Get the 16-bit PCM value (little endian)\n                //     const lowByte = pcmData[i * 2]\n                //     const highByte = pcmData[i * 2 + 1]\n                //     const pcmValue = (highByte << 8) | lowByte\n\n                //     // Convert to signed 16-bit value\n                //     const signedValue =\n                //         pcmValue > 32767 ? pcmValue - 65536 : pcmValue\n\n                //     // Normalize to float between -1 and 1\n                //     normalizedData[i] = signedValue / 32768.0\n                // }\n                // Store the normalized data in the result\n                result.normalizedData = channelData\n            }\n\n            if (includeBase64Data) {\n                // Convert the PCM data to a base64 string\n                const binary = Array.from(new Uint8Array(pcmData.buffer))\n                    .map((b) => String.fromCharCode(b))\n                    .join('')\n                result.base64Data = btoa(binary)\n            }\n\n            if (options.computeChecksum) {\n                result.checksum = crc32.buf(pcmData)\n            }\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion complete', {\n                pcmStats: {\n                    length: pcmData.length,\n                    bytesPerSample,\n                    totalSamples: numSamples,\n                    firstBytes: Array.from(pcmData.slice(0, 16)),\n                    lastBytes: Array.from(pcmData.slice(-16)),\n                },\n            })\n\n            return result\n        } catch (error) {\n            options.logger?.error('EXTRACT AUDIO - Error:', error)\n            throw error\n        }\n    }\n\n    ExpoAudioStreamModule.trimAudio = async (\n        options: TrimAudioOptions\n    ): Promise<TrimAudioResult> => {\n        try {\n            const startTime = performance.now()\n            const {\n                fileUri,\n                mode = 'single',\n                startTimeMs,\n                endTimeMs,\n                ranges,\n                outputFileName,\n                outputFormat,\n            } = options\n\n            // Validate inputs\n            if (!fileUri) {\n                throw new Error('fileUri is required')\n            }\n\n            if (\n                mode === 'single' &&\n                startTimeMs === undefined &&\n                endTimeMs === undefined\n            ) {\n                throw new Error(\n                    'At least one of startTimeMs or endTimeMs must be provided in single mode'\n                )\n            }\n\n            if (\n                (mode === 'keep' || mode === 'remove') &&\n                (!ranges || ranges.length === 0)\n            ) {\n                throw new Error(\n                    'ranges must be provided and non-empty for keep or remove modes'\n                )\n            }\n\n            // Create AudioContext\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n\n            // First, load the entire audio file to get its properties\n            const response = await fetch(fileUri)\n            const arrayBuffer = await response.arrayBuffer()\n            const originalAudioBuffer =\n                await audioContext.decodeAudioData(arrayBuffer)\n\n            // Get original audio properties\n            const originalSampleRate = originalAudioBuffer.sampleRate\n            const originalChannels = originalAudioBuffer.numberOfChannels\n\n            // Add more detailed logging\n            console.log(`Original audio details:`, {\n                sampleRate: originalSampleRate,\n                channels: originalChannels,\n                duration: originalAudioBuffer.duration,\n                length: originalAudioBuffer.length,\n                // Log a few samples to verify content\n                firstSamples: Array.from(\n                    originalAudioBuffer.getChannelData(0).slice(0, 5)\n                ),\n            })\n\n            // Determine output format - use original values as defaults if not specified\n            let format = outputFormat?.format || 'wav'\n            const targetSampleRate =\n                outputFormat?.sampleRate || originalSampleRate\n            const targetChannels = outputFormat?.channels || originalChannels\n            const targetBitDepth = outputFormat?.bitDepth || 16\n\n            // Get file info from the URL\n            const filename =\n                outputFileName ||\n                fileUri.split('/').pop() ||\n                'trimmed-audio.wav'\n\n            // Process based on mode\n            let resultBuffer: AudioBuffer\n\n            // Report initial progress\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 10,\n            })\n\n            if (mode === 'single') {\n                // Single mode: extract a single range\n                // Use original sample rate and channels for extraction to preserve quality\n                const { buffer } = await processAudioBuffer({\n                    fileUri,\n                    targetSampleRate, // Use the requested sample rate\n                    targetChannels,\n                    normalizeAudio: false,\n                    startTimeMs,\n                    endTimeMs,\n                    audioContext,\n                })\n\n                console.log(`Processed buffer details:`, {\n                    sampleRate: buffer.sampleRate,\n                    channels: buffer.numberOfChannels,\n                    duration: buffer.duration,\n                    length: buffer.length,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        buffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                resultBuffer = buffer\n\n                // If we need to change sample rate or channels, do it after extraction\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        buffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            } else {\n                // For keep or remove modes\n                const fullDuration = originalAudioBuffer.duration * 1000 // in ms\n\n                type ProcessSegment = {\n                    startTimeMs: number\n                    endTimeMs: number\n                }\n\n                let segmentsToProcess: ProcessSegment[] = []\n\n                if (mode === 'keep') {\n                    // For keep mode, use the ranges directly\n                    segmentsToProcess = ranges!\n                } else {\n                    // mode === 'remove'\n                    // For remove mode, invert the ranges\n                    const sortedRanges = [...ranges!].sort(\n                        (a, b) => a.startTimeMs - b.startTimeMs\n                    )\n\n                    // Add segment from start to first range if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[0].startTimeMs > 0\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs: 0,\n                            endTimeMs: sortedRanges[0].startTimeMs,\n                        })\n                    }\n\n                    // Add segments between ranges\n                    for (let i = 0; i < sortedRanges.length - 1; i++) {\n                        segmentsToProcess.push({\n                            startTimeMs: sortedRanges[i].endTimeMs,\n                            endTimeMs: sortedRanges[i + 1].startTimeMs,\n                        })\n                    }\n\n                    // Add segment from last range to end if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[sortedRanges.length - 1].endTimeMs <\n                            fullDuration\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs:\n                                sortedRanges[sortedRanges.length - 1].endTimeMs,\n                            endTimeMs: fullDuration,\n                        })\n                    }\n                }\n\n                // Filter out empty or invalid segments\n                segmentsToProcess = segmentsToProcess.filter(\n                    (segment) =>\n                        segment.startTimeMs < segment.endTimeMs &&\n                        segment.endTimeMs - segment.startTimeMs > 1\n                ) // 1ms minimum\n\n                if (segmentsToProcess.length === 0) {\n                    throw new Error(\n                        'No valid segments to process after filtering ranges'\n                    )\n                }\n\n                // Process each segment using original sample rate and channels\n                const segmentBuffers: AudioBuffer[] = []\n\n                for (let i = 0; i < segmentsToProcess.length; i++) {\n                    const segment = segmentsToProcess[i]\n\n                    // Report progress for each segment\n                    ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                        progress:\n                            10 +\n                            Math.round((i / segmentsToProcess.length) * 40),\n                    })\n\n                    // Use processAudioBuffer to extract this segment\n                    const { buffer: segmentBuffer } = await processAudioBuffer({\n                        fileUri,\n                        targetSampleRate: originalSampleRate, // Use original sample rate\n                        targetChannels: originalChannels, // Use original channels\n                        normalizeAudio: false,\n                        startTimeMs: segment.startTimeMs,\n                        endTimeMs: segment.endTimeMs,\n                        audioContext,\n                    })\n\n                    segmentBuffers.push(segmentBuffer)\n                }\n\n                // Concatenate all segments\n                const totalSamples = segmentBuffers.reduce(\n                    (sum, buffer) => sum + buffer.length,\n                    0\n                )\n\n                // Create buffer with original properties first\n                const concatenatedBuffer = audioContext.createBuffer(\n                    originalChannels,\n                    totalSamples,\n                    originalSampleRate\n                )\n\n                let offset = 0\n                for (const segmentBuffer of segmentBuffers) {\n                    for (\n                        let channel = 0;\n                        channel < originalChannels;\n                        channel++\n                    ) {\n                        const outputData =\n                            concatenatedBuffer.getChannelData(channel)\n                        const segmentData =\n                            segmentBuffer.getChannelData(channel)\n\n                        for (let i = 0; i < segmentBuffer.length; i++) {\n                            outputData[offset + i] = segmentData[i]\n                        }\n                    }\n                    offset += segmentBuffer.length\n                }\n\n                resultBuffer = concatenatedBuffer\n\n                // If we need to change sample rate or channels, do it after concatenation\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling concatenated buffer from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        concatenatedBuffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            }\n\n            // Report progress (50% - processing complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 50,\n            })\n\n            // Encode the result based on the requested format\n            let outputData: ArrayBuffer\n            let outputMimeType: string\n            let compressionInfo: any = null\n\n            // Check if AAC was requested on web and show a warning\n            if (format === 'aac' && Platform.OS === 'web') {\n                console.warn(\n                    'AAC format is not supported on web platforms. Falling back to OPUS format.'\n                )\n                format = 'opus'\n            }\n\n            if (format === 'wav') {\n                // Create a properly interleaved buffer for WAV format\n                // For WAV, we need to convert Float32Array to Int16Array (for 16-bit audio)\n                const numSamples =\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                const interleavedData = new Int16Array(numSamples)\n\n                // Log detailed information about the buffer before encoding\n                console.log(`Creating WAV file:`, {\n                    bufferSampleRate: resultBuffer.sampleRate,\n                    bufferChannels: resultBuffer.numberOfChannels,\n                    bufferLength: resultBuffer.length,\n                    targetSampleRate,\n                    targetChannels,\n                    targetBitDepth,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        resultBuffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                // Interleave channels properly\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        // Convert float (-1.0 to 1.0) to int16 (-32768 to 32767)\n                        const floatSample =\n                            resultBuffer.getChannelData(channel)[i]\n                        // Clamp the value to -1.0 to 1.0\n                        const clampedSample = Math.max(\n                            -1.0,\n                            Math.min(1.0, floatSample)\n                        )\n                        // Convert to int16\n                        const intSample = Math.round(clampedSample * 32767)\n                        // Store in interleaved buffer\n                        interleavedData[\n                            i * resultBuffer.numberOfChannels + channel\n                        ] = intSample\n                    }\n                }\n\n                // Convert Int16Array to ArrayBuffer for WAV header\n                const rawBuffer = interleavedData.buffer\n\n                // IMPORTANT: Make sure we're using the ACTUAL sample rate of the buffer\n                // not just what was requested in the options\n                console.log(\n                    `Creating WAV with ${resultBuffer.numberOfChannels} channels at ${resultBuffer.sampleRate}Hz`\n                )\n\n                outputData = writeWavHeader({\n                    buffer: rawBuffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate, // Use the actual buffer's sample rate\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            } else if (format === 'opus' || format === 'aac') {\n                try {\n                    // Try to use MediaRecorder for compressed formats\n                    const { data, bitrate } = await encodeCompressedAudio(\n                        resultBuffer,\n                        format,\n                        outputFormat?.bitrate\n                    )\n\n                    outputData = data\n                    outputMimeType =\n                        format === 'opus' ? 'audio/webm' : 'audio/aac'\n                    compressionInfo = {\n                        format,\n                        bitrate,\n                        size: data.byteLength,\n                    }\n                } catch (error) {\n                    console.warn(\n                        `Failed to encode to ${format}, falling back to WAV: ${error}`\n                    )\n\n                    // Same WAV encoding as above\n                    const wavData = new Float32Array(\n                        resultBuffer.length * resultBuffer.numberOfChannels\n                    )\n\n                    for (let i = 0; i < resultBuffer.length; i++) {\n                        for (\n                            let channel = 0;\n                            channel < resultBuffer.numberOfChannels;\n                            channel++\n                        ) {\n                            wavData[\n                                i * resultBuffer.numberOfChannels + channel\n                            ] = resultBuffer.getChannelData(channel)[i]\n                        }\n                    }\n\n                    outputData = writeWavHeader({\n                        buffer: wavData.buffer as ArrayBuffer,\n                        sampleRate: resultBuffer.sampleRate,\n                        numChannels: resultBuffer.numberOfChannels,\n                        bitDepth: targetBitDepth as BitDepth,\n                    })\n                    outputMimeType = 'audio/wav'\n                }\n            } else {\n                // Default to WAV for unsupported formats\n                console.warn(\n                    `Format ${format} not supported on web, using WAV instead`\n                )\n\n                // Same WAV encoding as above\n                const wavData = new Float32Array(\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                )\n\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        wavData[i * resultBuffer.numberOfChannels + channel] =\n                            resultBuffer.getChannelData(channel)[i]\n                    }\n                }\n\n                outputData = writeWavHeader({\n                    buffer: wavData.buffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate,\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            }\n\n            // Report progress (90% - encoding complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 90,\n            })\n\n            // Create a blob and URL for the result\n            const blob = new Blob([outputData], { type: outputMimeType })\n            const outputUri = URL.createObjectURL(blob)\n\n            // Calculate processing time\n            const processingTimeMs = performance.now() - startTime\n\n            // Report progress (100% - complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 100,\n            })\n\n            // Create result object\n            const result: TrimAudioResult = {\n                uri: outputUri,\n                filename,\n                durationMs: Math.round(resultBuffer.duration * 1000),\n                size: outputData.byteLength,\n                sampleRate: resultBuffer.sampleRate,\n                channels: resultBuffer.numberOfChannels,\n                bitDepth: targetBitDepth,\n                mimeType: outputMimeType,\n                processingInfo: {\n                    durationMs: processingTimeMs,\n                },\n            }\n\n            // Add compression info if available\n            if (compressionInfo) {\n                result.compression = compressionInfo\n            }\n\n            return result\n        } catch (error) {\n            console.error('Error in trimAudio:', error)\n            throw error\n        }\n    }\n\n    // Add a sendEvent method for web\n    ExpoAudioStreamModule.sendEvent = (eventName: string, params: any) => {\n        // This will be picked up by the LegacyEventEmitter in trimAudio.ts\n        if (\n            ExpoAudioStreamModule.listeners &&\n            ExpoAudioStreamModule.listeners[eventName]\n        ) {\n            ExpoAudioStreamModule.listeners[eventName].forEach(\n                (listener: Function) => {\n                    listener(params)\n                }\n            )\n        }\n    }\n\n    // Initialize listeners object\n    ExpoAudioStreamModule.listeners = {}\n\n    // Add methods for event listeners that LegacyEventEmitter will use\n    ExpoAudioStreamModule.addListener = (\n        eventName: string,\n        listener: Function\n    ) => {\n        if (!ExpoAudioStreamModule.listeners[eventName]) {\n            ExpoAudioStreamModule.listeners[eventName] = []\n        }\n        ExpoAudioStreamModule.listeners[eventName].push(listener)\n\n        // Return an object with a remove method\n        return {\n            remove: () => {\n                const index =\n                    ExpoAudioStreamModule.listeners[eventName].indexOf(listener)\n                if (index !== -1) {\n                    ExpoAudioStreamModule.listeners[eventName].splice(index, 1)\n                }\n            },\n        }\n    }\n\n    ExpoAudioStreamModule.removeAllListeners = (eventName: string) => {\n        if (ExpoAudioStreamModule.listeners[eventName]) {\n            delete ExpoAudioStreamModule.listeners[eventName]\n        }\n    }\n}\n\n// Move the encodeCompressedAudio function outside the if block to fix the ESLint error\nasync function encodeCompressedAudio(\n    buffer: AudioBuffer,\n    format: 'opus' | 'aac',\n    bitrate?: number\n): Promise<{ data: ArrayBuffer; bitrate: number }> {\n    return new Promise((resolve, reject) => {\n        try {\n            // On web, always use opus if aac is requested\n            const actualFormat =\n                Platform.OS === 'web' && format === 'aac' ? 'opus' : format\n\n            // Check if MediaRecorder supports the requested format\n            const mimeType =\n                actualFormat === 'opus' ? 'audio/webm;codecs=opus' : 'audio/aac'\n            if (!MediaRecorder.isTypeSupported(mimeType)) {\n                throw new Error(`MediaRecorder does not support ${mimeType}`)\n            }\n\n            // Create a new AudioContext and source\n            const ctx = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n            const source = ctx.createBufferSource()\n            source.buffer = buffer\n\n            // Create a MediaStreamDestination to capture the audio\n            const destination = ctx.createMediaStreamDestination()\n            source.connect(destination)\n\n            // Create a MediaRecorder with the requested format\n            const recorder = new MediaRecorder(destination.stream, {\n                mimeType,\n                audioBitsPerSecond:\n                    bitrate || (actualFormat === 'opus' ? 32000 : 64000),\n            })\n\n            const chunks: Blob[] = []\n\n            recorder.ondataavailable = (e) => {\n                if (e.data.size > 0) {\n                    chunks.push(e.data)\n                }\n            }\n\n            recorder.onstop = async () => {\n                try {\n                    const blob = new Blob(chunks, { type: mimeType })\n                    const arrayBuffer = await blob.arrayBuffer()\n\n                    // Get the actual bitrate used\n                    const actualBitrate = Math.round(\n                        (arrayBuffer.byteLength * 8) / buffer.duration\n                    )\n\n                    resolve({\n                        data: arrayBuffer,\n                        bitrate: actualBitrate / 1000, // Convert to kbps\n                    })\n\n                    // Clean up\n                    ctx.close()\n                } catch (error) {\n                    reject(error)\n                }\n            }\n\n            // Start recording and playback\n            recorder.start()\n            source.start(0)\n\n            // Stop recording when the buffer finishes playing\n            setTimeout(() => {\n                recorder.stop()\n                source.stop()\n            }, buffer.duration * 1000)\n        } catch (error) {\n            reject(error)\n        }\n    })\n}\n\n// Improved resampleAudioBuffer function\nasync function resampleAudioBuffer(\n    context: AudioContext,\n    buffer: AudioBuffer,\n    targetSampleRate: number,\n    targetChannels: number\n): Promise<AudioBuffer> {\n    // If no change needed, return the original buffer\n    if (\n        buffer.sampleRate === targetSampleRate &&\n        buffer.numberOfChannels === targetChannels\n    ) {\n        return buffer\n    }\n\n    console.log(\n        `Resampling: ${buffer.sampleRate}Hz → ${targetSampleRate}Hz, ${buffer.numberOfChannels} → ${targetChannels} channels`\n    )\n\n    // Calculate the new length based on the sample rate change\n    const newLength = Math.round(\n        (buffer.length * targetSampleRate) / buffer.sampleRate\n    )\n\n    // Create an offline context for resampling\n    const offlineContext = new OfflineAudioContext(\n        targetChannels,\n        newLength,\n        targetSampleRate\n    )\n\n    // Create a source node\n    const source = offlineContext.createBufferSource()\n    source.buffer = buffer\n\n    // If we need to change channel count\n    if (buffer.numberOfChannels !== targetChannels) {\n        if (targetChannels === 1 && buffer.numberOfChannels > 1) {\n            // Downmix to mono\n            const merger = offlineContext.createChannelMerger(1)\n\n            // Create a gain node to reduce volume when downmixing to prevent clipping\n            const gainNode = offlineContext.createGain()\n            gainNode.gain.value = 1.0 / buffer.numberOfChannels\n\n            source.connect(gainNode)\n            gainNode.connect(merger)\n            merger.connect(offlineContext.destination)\n        } else if (targetChannels === 2 && buffer.numberOfChannels === 1) {\n            // Upmix mono to stereo (duplicate the channel)\n            const splitter = offlineContext.createChannelSplitter(1)\n            const merger = offlineContext.createChannelMerger(2)\n\n            source.connect(splitter)\n            splitter.connect(merger, 0, 0)\n            splitter.connect(merger, 0, 1)\n            merger.connect(offlineContext.destination)\n        } else {\n            // For other cases, just connect and let the system handle it\n            source.connect(offlineContext.destination)\n        }\n    } else {\n        // No channel conversion needed\n        source.connect(offlineContext.destination)\n    }\n\n    // Start rendering\n    source.start(0)\n    const resampledBuffer = await offlineContext.startRendering()\n\n    console.log(\n        `Resampling complete: ${resampledBuffer.length} samples at ${resampledBuffer.sampleRate}Hz`\n    )\n\n    return resampledBuffer\n}\n\nif (Platform.OS !== 'web') {\n    ExpoAudioStreamModule = requireNativeModule('ExpoAudioStream')\n}\n\nexport default ExpoAudioStreamModule\n"]}

package/build/WebRecorder.web.d.ts

@@ -1,119 +0,0 @@
-import { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types';
-import { ConsoleLike, RecordingConfig } from './ExpoAudioStream.types';
-import { EmitAudioAnalysisFunction, EmitAudioEventFunction } from './ExpoAudioStream.web';
-interface AudioFeaturesEvent {
-    data: {
-        command: string;
-        result: AudioAnalysis;
-    };
-}
-export declare class WebRecorder {
-    private audioContext;
-    private audioWorkletNode;
-    private featureExtractorWorker?;
-    private source;
-    private emitAudioEventCallback;
-    private emitAudioAnalysisCallback;
-    private config;
-    private position;
-    private numberOfChannels;
-    private bitDepth;
-    private exportBitDepth;
-    private audioAnalysisData;
-    private packetCount;
-    private logger?;
-    private compressedMediaRecorder;
-    private compressedChunks;
-    private compressedSize;
-    private pendingCompressedChunk;
-    private readonly wavMimeType;
-    private dataPointIdCounter;
-    /**
-     * Initializes a new WebRecorder instance for audio recording and processing
-     * @param audioContext - The AudioContext to use for recording
-     * @param source - The MediaStreamAudioSourceNode providing the audio input
-     * @param recordingConfig - Configuration options for the recording
-     * @param emitAudioEventCallback - Callback function for audio data events
-     * @param emitAudioAnalysisCallback - Callback function for audio analysis events
-     * @param logger - Optional logger for debugging information
-     */
-    constructor({ audioContext, source, recordingConfig, emitAudioEventCallback, emitAudioAnalysisCallback, logger, }: {
-        audioContext: AudioContext;
-        source: MediaStreamAudioSourceNode;
-        recordingConfig: RecordingConfig;
-        emitAudioEventCallback: EmitAudioEventFunction;
-        emitAudioAnalysisCallback: EmitAudioAnalysisFunction;
-        logger?: ConsoleLike;
-    });
-    /**
-     * Initializes the audio worklet using an inline script
-     * Creates and connects the audio processing pipeline
-     */
-    init(): Promise<void>;
-    /**
-     * Initializes the feature extractor worker for audio analysis
-     * Creates an inline worker from a blob for audio feature extraction
-     */
-    initFeatureExtractorWorker(): void;
-    /**
-     * Processes audio analysis results from the feature extractor worker
-     * Updates the audio analysis data and emits events
-     * @param event - The event containing audio analysis results
-     */
-    handleFeatureExtractorMessage(event: AudioFeaturesEvent): void;
-    /**
-     * Resets the data point ID counter
-     * Used when starting a new recording
-     */
-    resetDataPointCounter(): void;
-    /**
-     * Starts the audio recording process
-     * Connects the audio nodes and begins capturing audio data
-     */
-    start(): void;
-    /**
-     * Stops the audio recording process and returns the recorded data
-     * @returns Promise resolving to an object containing PCM data and optional compressed blob
-     */
-    stop(): Promise<{
-        pcmData: Float32Array;
-        compressedBlob?: Blob;
-    }>;
-    /**
-     * Cleans up resources when recording is stopped
-     * Closes audio context and disconnects nodes
-     */
-    private cleanup;
-    /**
-     * Pauses the audio recording process
-     * Disconnects audio nodes and pauses the media recorder
-     */
-    pause(): void;
-    /**
-     * Stops all media stream tracks to release hardware resources
-     * Ensures recording indicators (like microphone icon) are turned off
-     */
-    stopMediaStreamTracks(): void;
-    /**
-     * Determines the audio format capabilities of the current audio context
-     * @param sampleRate - The sample rate to check
-     * @returns Object containing format information (sample rate, bit depth, channels)
-     */
-    private checkAudioContextFormat;
-    /**
-     * Resumes a paused recording
-     * Reconnects audio nodes and resumes the media recorder
-     */
-    resume(): void;
-    /**
-     * Initializes the compressed media recorder if compression is enabled
-     * Sets up event handlers for compressed audio data
-     */
-    private initializeCompressedRecorder;
-    /**
-     * Processes features if enabled
-     */
-    processFeatures(chunk: Float32Array, sampleRate: number, chunkPosition: number, startPosition: number, endPosition: number, samples: number): void;
-}
-export {};
-//# sourceMappingURL=WebRecorder.web.d.ts.map

package/build/WebRecorder.web.d.ts.map

@@ -1 +0,0 @@
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