@siteed/audio-studio 3.0.2 → 3.0.3

package/src/AudioStudioModule.ts

@@ -1,17 +1,7 @@
 import { requireNativeModule } from 'expo-modules-core'
 import { Platform } from 'react-native'
 
-import {
-    ExtractAudioDataOptions,
-    ExtractedAudioData,
-    BitDepth,
-    TrimAudioOptions,
-    TrimAudioResult,
-} from './AudioStudio.types'
 import { AudioStudioWeb, AudioStudioWebProps } from './AudioStudio.web'
-import { processAudioBuffer } from './utils/audioProcessing'
-import crc32 from './utils/crc32'
-import { writeWavHeader } from './utils/writeWavHeader'
 
 // eslint-disable-next-line @typescript-eslint/no-explicit-any
 let AudioStudioModule: any
@@ -20,9 +10,7 @@ if (Platform.OS === 'web') {
     let instance: AudioStudioWeb | null = null
 
     AudioStudioModule = (webProps: AudioStudioWebProps) => {
-        if (!instance) {
-            instance = new AudioStudioWeb(webProps)
-        }
+        instance ??= new AudioStudioWeb(webProps)
         return instance
     }
     AudioStudioModule.requestPermissionsAsync = async () => {
@@ -79,692 +67,14 @@ if (Platform.OS === 'web') {
                 return await AudioStudioModule.requestPermissionsAsync()
         }
     }
-    AudioStudioModule.extractAudioData = async (
-        options: ExtractAudioDataOptions
-    ): Promise<ExtractedAudioData> => {
-        try {
-            const {
-                fileUri,
-                position,
-                length,
-                startTimeMs,
-                endTimeMs,
-                decodingOptions,
-                includeNormalizedData,
-                includeBase64Data,
-                includeWavHeader = false,
-                logger,
-            } = options
-
-            logger?.debug('EXTRACT AUDIO - Step 1: Initial request', {
-                fileUri,
-                extractionParams: {
-                    position,
-                    length,
-                    startTimeMs,
-                    endTimeMs,
-                },
-                decodingOptions: {
-                    targetSampleRate:
-                        decodingOptions?.targetSampleRate ?? 16000,
-                    targetChannels: decodingOptions?.targetChannels ?? 1,
-                    targetBitDepth: decodingOptions?.targetBitDepth ?? 16,
-                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,
-                },
-                outputOptions: {
-                    includeNormalizedData,
-                    includeBase64Data,
-                    includeWavHeader,
-                },
-            })
-
-            // Process the audio using shared helper function
-            const processedBuffer = await processAudioBuffer({
-                fileUri,
-                targetSampleRate: decodingOptions?.targetSampleRate ?? 16000,
-                targetChannels: decodingOptions?.targetChannels ?? 1,
-                normalizeAudio: decodingOptions?.normalizeAudio ?? false,
-                position,
-                length,
-                startTimeMs,
-                endTimeMs,
-                logger,
-            })
-
-            logger?.debug('EXTRACT AUDIO - Step 2: Audio processing complete', {
-                processedData: {
-                    samples: processedBuffer.samples,
-                    sampleRate: processedBuffer.sampleRate,
-                    channels: processedBuffer.channels,
-                    durationMs: processedBuffer.durationMs,
-                },
-            })
-
-            const channelData = processedBuffer.channelData
-            const bitDepth = (decodingOptions?.targetBitDepth ?? 16) as BitDepth
-            const bytesPerSample = bitDepth / 8
-            const numSamples = processedBuffer.samples
-
-            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion setup', {
-                channelData: {
-                    length: channelData.length,
-                    first: channelData[0],
-                    last: channelData[channelData.length - 1],
-                },
-                calculation: {
-                    bitDepth,
-                    bytesPerSample,
-                    numSamples,
-                    expectedBytes: numSamples * bytesPerSample,
-                },
-            })
-
-            // Create PCM data with correct length based on original byte length
-            const pcmData = new Uint8Array(numSamples * bytesPerSample)
-            let offset = 0
-
-            // Convert Float32 samples to PCM format
-            for (let i = 0; i < numSamples; i++) {
-                const sample = channelData[i]
-                const value = Math.max(-1, Math.min(1, sample))
-                // Convert to 16-bit signed integer
-                let intValue = Math.round(value * 32767)
-
-                // Handle negative values correctly
-                if (intValue < 0) {
-                    intValue = 65536 + intValue
-                }
-
-                // Write as little-endian
-                pcmData[offset++] = intValue & 255 // Low byte
-                pcmData[offset++] = (intValue >> 8) & 255 // High byte
-            }
-
-            const durationMs = Math.round(
-                (numSamples / processedBuffer.sampleRate) * 1000
-            )
-
-            logger?.debug('EXTRACT AUDIO - Step 4: Final output', {
-                pcmData: {
-                    length: pcmData.length,
-                    first: pcmData[0],
-                    last: pcmData[pcmData.length - 1],
-                },
-                timing: {
-                    numSamples,
-                    sampleRate: processedBuffer.sampleRate,
-                    durationMs,
-                    shouldBe3000ms: endTimeMs
-                        ? endTimeMs - (startTimeMs ?? 0) === 3000
-                        : undefined,
-                },
-            })
-
-            const result: ExtractedAudioData = {
-                pcmData: new Uint8Array(pcmData.buffer),
-                sampleRate: processedBuffer.sampleRate,
-                channels: processedBuffer.channels,
-                bitDepth,
-                durationMs,
-                format: `pcm_${bitDepth}bit` as const,
-                samples: numSamples,
-            }
-
-            // Add WAV header if requested
-            if (includeWavHeader) {
-                logger?.debug('EXTRACT AUDIO - Step 4: Adding WAV header', {
-                    originalLength: pcmData.length,
-                    newLength: result.pcmData.length,
-                    firstBytes: Array.from(result.pcmData.slice(0, 44)), // WAV header is 44 bytes
-                })
-                const wavBuffer = writeWavHeader({
-                    buffer: pcmData.buffer.slice(0, pcmData.length),
-                    sampleRate: processedBuffer.sampleRate,
-                    numChannels: processedBuffer.channels,
-                    bitDepth,
-                })
-                result.pcmData = new Uint8Array(wavBuffer)
-                result.hasWavHeader = true
-            }
-
-            if (includeNormalizedData) {
-                // // Simple approach: Create normalized data directly from the PCM data
-                // // Just convert to -1 to 1 range without any amplification
-                // const normalizedData = new Float32Array(numSamples)
-
-                // // Convert the PCM data to float values
-                // for (let i = 0; i < numSamples; i++) {
-                //     // Get the 16-bit PCM value (little endian)
-                //     const lowByte = pcmData[i * 2]
-                //     const highByte = pcmData[i * 2 + 1]
-                //     const pcmValue = (highByte << 8) | lowByte
-
-                //     // Convert to signed 16-bit value
-                //     const signedValue =
-                //         pcmValue > 32767 ? pcmValue - 65536 : pcmValue
-
-                //     // Normalize to float between -1 and 1
-                //     normalizedData[i] = signedValue / 32768.0
-                // }
-                // Store the normalized data in the result
-                result.normalizedData = channelData
-            }
-
-            if (includeBase64Data) {
-                // Convert the PCM data to a base64 string
-                const binary = Array.from(new Uint8Array(pcmData.buffer))
-                    .map((b) => String.fromCharCode(b))
-                    .join('')
-                result.base64Data = btoa(binary)
-            }
-
-            if (options.computeChecksum) {
-                result.checksum = crc32.buf(pcmData)
-            }
-
-            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion complete', {
-                pcmStats: {
-                    length: pcmData.length,
-                    bytesPerSample,
-                    totalSamples: numSamples,
-                    firstBytes: Array.from(pcmData.slice(0, 16)),
-                    lastBytes: Array.from(pcmData.slice(-16)),
-                },
-            })
-
-            return result
-        } catch (error) {
-            options.logger?.error('EXTRACT AUDIO - Error:', error)
-            throw error
-        }
-    }
-
-    AudioStudioModule.trimAudio = async (
-        options: TrimAudioOptions
-    ): Promise<TrimAudioResult> => {
-        try {
-            const startTime = performance.now()
-            const {
-                fileUri,
-                mode = 'single',
-                startTimeMs,
-                endTimeMs,
-                ranges,
-                outputFileName,
-                outputFormat,
-            } = options
-
-            // Validate inputs
-            if (!fileUri) {
-                throw new Error('fileUri is required')
-            }
-
-            if (
-                mode === 'single' &&
-                startTimeMs === undefined &&
-                endTimeMs === undefined
-            ) {
-                throw new Error(
-                    'At least one of startTimeMs or endTimeMs must be provided in single mode'
-                )
-            }
-
-            if (
-                (mode === 'keep' || mode === 'remove') &&
-                (!ranges || ranges.length === 0)
-            ) {
-                throw new Error(
-                    'ranges must be provided and non-empty for keep or remove modes'
-                )
-            }
-
-            // Create AudioContext
-            const audioContext = new (window.AudioContext ||
-                (window as any).webkitAudioContext)()
-
-            // First, load the entire audio file to get its properties
-            const response = await fetch(fileUri)
-            const arrayBuffer = await response.arrayBuffer()
-            const originalAudioBuffer =
-                await audioContext.decodeAudioData(arrayBuffer)
-
-            // Get original audio properties
-            const originalSampleRate = originalAudioBuffer.sampleRate
-            const originalChannels = originalAudioBuffer.numberOfChannels
-
-            // Add more detailed logging
-            console.log(`Original audio details:`, {
-                sampleRate: originalSampleRate,
-                channels: originalChannels,
-                duration: originalAudioBuffer.duration,
-                length: originalAudioBuffer.length,
-                // Log a few samples to verify content
-                firstSamples: Array.from(
-                    originalAudioBuffer.getChannelData(0).slice(0, 5)
-                ),
-            })
-
-            // Determine output format - use original values as defaults if not specified
-            let format = outputFormat?.format || 'wav'
-            const targetSampleRate =
-                outputFormat?.sampleRate || originalSampleRate
-            const targetChannels = outputFormat?.channels || originalChannels
-            const targetBitDepth = outputFormat?.bitDepth || 16
-
-            // Get file info from the URL
-            const filename =
-                outputFileName ||
-                fileUri.split('/').pop() ||
-                'trimmed-audio.wav'
-
-            // Process based on mode
-            let resultBuffer: AudioBuffer
-
-            // Report initial progress
-            AudioStudioModule.sendEvent('TrimProgress', {
-                progress: 10,
-            })
-
-            if (mode === 'single') {
-                // Single mode: extract a single range
-                // Use original sample rate and channels for extraction to preserve quality
-                const { buffer } = await processAudioBuffer({
-                    fileUri,
-                    targetSampleRate, // Use the requested sample rate
-                    targetChannels,
-                    normalizeAudio: false,
-                    startTimeMs,
-                    endTimeMs,
-                    audioContext,
-                })
-
-                console.log(`Processed buffer details:`, {
-                    sampleRate: buffer.sampleRate,
-                    channels: buffer.numberOfChannels,
-                    duration: buffer.duration,
-                    length: buffer.length,
-                    // Log a few samples to verify content
-                    firstSamples: Array.from(
-                        buffer.getChannelData(0).slice(0, 5)
-                    ),
-                })
-
-                resultBuffer = buffer
-
-                // If we need to change sample rate or channels, do it after extraction
-                if (
-                    targetSampleRate !== originalSampleRate ||
-                    targetChannels !== originalChannels
-                ) {
-                    console.log(
-                        `Resampling from ${originalSampleRate}Hz to ${targetSampleRate}Hz`
-                    )
-                    resultBuffer = await resampleAudioBuffer(
-                        audioContext,
-                        buffer,
-                        targetSampleRate,
-                        targetChannels
-                    )
-                }
-            } else {
-                // For keep or remove modes
-                const fullDuration = originalAudioBuffer.duration * 1000 // in ms
-
-                type ProcessSegment = {
-                    startTimeMs: number
-                    endTimeMs: number
-                }
-
-                let segmentsToProcess: ProcessSegment[] = []
-
-                if (mode === 'keep') {
-                    // For keep mode, use the ranges directly
-                    segmentsToProcess = ranges!
-                } else {
-                    // mode === 'remove'
-                    // For remove mode, invert the ranges
-                    const sortedRanges = [...ranges!].sort(
-                        (a, b) => a.startTimeMs - b.startTimeMs
-                    )
-
-                    // Add segment from start to first range if needed
-                    if (
-                        sortedRanges.length > 0 &&
-                        sortedRanges[0].startTimeMs > 0
-                    ) {
-                        segmentsToProcess.push({
-                            startTimeMs: 0,
-                            endTimeMs: sortedRanges[0].startTimeMs,
-                        })
-                    }
-
-                    // Add segments between ranges
-                    for (let i = 0; i < sortedRanges.length - 1; i++) {
-                        segmentsToProcess.push({
-                            startTimeMs: sortedRanges[i].endTimeMs,
-                            endTimeMs: sortedRanges[i + 1].startTimeMs,
-                        })
-                    }
-
-                    // Add segment from last range to end if needed
-                    if (
-                        sortedRanges.length > 0 &&
-                        sortedRanges[sortedRanges.length - 1].endTimeMs <
-                            fullDuration
-                    ) {
-                        segmentsToProcess.push({
-                            startTimeMs:
-                                sortedRanges[sortedRanges.length - 1].endTimeMs,
-                            endTimeMs: fullDuration,
-                        })
-                    }
-                }
-
-                // Filter out empty or invalid segments
-                segmentsToProcess = segmentsToProcess.filter(
-                    (segment) =>
-                        segment.startTimeMs < segment.endTimeMs &&
-                        segment.endTimeMs - segment.startTimeMs > 1
-                ) // 1ms minimum
-
-                if (segmentsToProcess.length === 0) {
-                    throw new Error(
-                        'No valid segments to process after filtering ranges'
-                    )
-                }
-
-                // Process each segment using original sample rate and channels
-                const segmentBuffers: AudioBuffer[] = []
-
-                for (let i = 0; i < segmentsToProcess.length; i++) {
-                    const segment = segmentsToProcess[i]
-
-                    // Report progress for each segment
-                    AudioStudioModule.sendEvent('TrimProgress', {
-                        progress:
-                            10 +
-                            Math.round((i / segmentsToProcess.length) * 40),
-                    })
-
-                    // Use processAudioBuffer to extract this segment
-                    const { buffer: segmentBuffer } = await processAudioBuffer({
-                        fileUri,
-                        targetSampleRate: originalSampleRate, // Use original sample rate
-                        targetChannels: originalChannels, // Use original channels
-                        normalizeAudio: false,
-                        startTimeMs: segment.startTimeMs,
-                        endTimeMs: segment.endTimeMs,
-                        audioContext,
-                    })
-
-                    segmentBuffers.push(segmentBuffer)
-                }
-
-                // Concatenate all segments
-                const totalSamples = segmentBuffers.reduce(
-                    (sum, buffer) => sum + buffer.length,
-                    0
-                )
-
-                // Create buffer with original properties first
-                const concatenatedBuffer = audioContext.createBuffer(
-                    originalChannels,
-                    totalSamples,
-                    originalSampleRate
-                )
-
-                let offset = 0
-                for (const segmentBuffer of segmentBuffers) {
-                    for (
-                        let channel = 0;
-                        channel < originalChannels;
-                        channel++
-                    ) {
-                        const outputData =
-                            concatenatedBuffer.getChannelData(channel)
-                        const segmentData =
-                            segmentBuffer.getChannelData(channel)
-
-                        for (let i = 0; i < segmentBuffer.length; i++) {
-                            outputData[offset + i] = segmentData[i]
-                        }
-                    }
-                    offset += segmentBuffer.length
-                }
-
-                resultBuffer = concatenatedBuffer
-
-                // If we need to change sample rate or channels, do it after concatenation
-                if (
-                    targetSampleRate !== originalSampleRate ||
-                    targetChannels !== originalChannels
-                ) {
-                    console.log(
-                        `Resampling concatenated buffer from ${originalSampleRate}Hz to ${targetSampleRate}Hz`
-                    )
-                    resultBuffer = await resampleAudioBuffer(
-                        audioContext,
-                        concatenatedBuffer,
-                        targetSampleRate,
-                        targetChannels
-                    )
-                }
-            }
-
-            // Report progress (50% - processing complete)
-            AudioStudioModule.sendEvent('TrimProgress', {
-                progress: 50,
-            })
-
-            // Encode the result based on the requested format
-            let outputData: ArrayBuffer
-            let outputMimeType: string
-            let compressionInfo: any = null
-
-            // Check if AAC was requested on web and show a warning
-            if (format === 'aac' && Platform.OS === 'web') {
-                console.warn(
-                    'AAC format is not supported on web platforms. Falling back to OPUS format.'
-                )
-                format = 'opus'
-            }
-
-            if (format === 'wav') {
-                // Create a properly interleaved buffer for WAV format
-                // For WAV, we need to convert Float32Array to Int16Array (for 16-bit audio)
-                const numSamples =
-                    resultBuffer.length * resultBuffer.numberOfChannels
-                const interleavedData = new Int16Array(numSamples)
-
-                // Log detailed information about the buffer before encoding
-                console.log(`Creating WAV file:`, {
-                    bufferSampleRate: resultBuffer.sampleRate,
-                    bufferChannels: resultBuffer.numberOfChannels,
-                    bufferLength: resultBuffer.length,
-                    targetSampleRate,
-                    targetChannels,
-                    targetBitDepth,
-                    // Log a few samples to verify content
-                    firstSamples: Array.from(
-                        resultBuffer.getChannelData(0).slice(0, 5)
-                    ),
-                })
-
-                // Interleave channels properly
-                for (let i = 0; i < resultBuffer.length; i++) {
-                    for (
-                        let channel = 0;
-                        channel < resultBuffer.numberOfChannels;
-                        channel++
-                    ) {
-                        // Convert float (-1.0 to 1.0) to int16 (-32768 to 32767)
-                        const floatSample =
-                            resultBuffer.getChannelData(channel)[i]
-                        // Clamp the value to -1.0 to 1.0
-                        const clampedSample = Math.max(
-                            -1.0,
-                            Math.min(1.0, floatSample)
-                        )
-                        // Convert to int16
-                        const intSample = Math.round(clampedSample * 32767)
-                        // Store in interleaved buffer
-                        interleavedData[
-                            i * resultBuffer.numberOfChannels + channel
-                        ] = intSample
-                    }
-                }
-
-                // Convert Int16Array to ArrayBuffer for WAV header
-                const rawBuffer = interleavedData.buffer
-
-                // IMPORTANT: Make sure we're using the ACTUAL sample rate of the buffer
-                // not just what was requested in the options
-                console.log(
-                    `Creating WAV with ${resultBuffer.numberOfChannels} channels at ${resultBuffer.sampleRate}Hz`
-                )
-
-                outputData = writeWavHeader({
-                    buffer: rawBuffer as ArrayBuffer,
-                    sampleRate: resultBuffer.sampleRate, // Use the actual buffer's sample rate
-                    numChannels: resultBuffer.numberOfChannels,
-                    bitDepth: targetBitDepth as BitDepth,
-                })
-                outputMimeType = 'audio/wav'
-            } else if (format === 'opus' || format === 'aac') {
-                try {
-                    // Try to use MediaRecorder for compressed formats
-                    const { data, bitrate } = await encodeCompressedAudio(
-                        resultBuffer,
-                        format,
-                        outputFormat?.bitrate
-                    )
-
-                    outputData = data
-                    outputMimeType =
-                        format === 'opus' ? 'audio/webm' : 'audio/aac'
-                    compressionInfo = {
-                        format,
-                        bitrate,
-                        size: data.byteLength,
-                    }
-                } catch (error) {
-                    console.warn(
-                        `Failed to encode to ${format}, falling back to WAV: ${error}`
-                    )
-
-                    // Same WAV encoding as above
-                    const wavData = new Float32Array(
-                        resultBuffer.length * resultBuffer.numberOfChannels
-                    )
-
-                    for (let i = 0; i < resultBuffer.length; i++) {
-                        for (
-                            let channel = 0;
-                            channel < resultBuffer.numberOfChannels;
-                            channel++
-                        ) {
-                            wavData[
-                                i * resultBuffer.numberOfChannels + channel
-                            ] = resultBuffer.getChannelData(channel)[i]
-                        }
-                    }
-
-                    outputData = writeWavHeader({
-                        buffer: wavData.buffer as ArrayBuffer,
-                        sampleRate: resultBuffer.sampleRate,
-                        numChannels: resultBuffer.numberOfChannels,
-                        bitDepth: targetBitDepth as BitDepth,
-                    })
-                    outputMimeType = 'audio/wav'
-                }
-            } else {
-                // Default to WAV for unsupported formats
-                console.warn(
-                    `Format ${format} not supported on web, using WAV instead`
-                )
-
-                // Same WAV encoding as above
-                const wavData = new Float32Array(
-                    resultBuffer.length * resultBuffer.numberOfChannels
-                )
-
-                for (let i = 0; i < resultBuffer.length; i++) {
-                    for (
-                        let channel = 0;
-                        channel < resultBuffer.numberOfChannels;
-                        channel++
-                    ) {
-                        wavData[i * resultBuffer.numberOfChannels + channel] =
-                            resultBuffer.getChannelData(channel)[i]
-                    }
-                }
-
-                outputData = writeWavHeader({
-                    buffer: wavData.buffer as ArrayBuffer,
-                    sampleRate: resultBuffer.sampleRate,
-                    numChannels: resultBuffer.numberOfChannels,
-                    bitDepth: targetBitDepth as BitDepth,
-                })
-                outputMimeType = 'audio/wav'
-            }
-
-            // Report progress (90% - encoding complete)
-            AudioStudioModule.sendEvent('TrimProgress', {
-                progress: 90,
-            })
-
-            // Create a blob and URL for the result
-            const blob = new Blob([outputData], { type: outputMimeType })
-            const outputUri = URL.createObjectURL(blob)
-
-            // Calculate processing time
-            const processingTimeMs = performance.now() - startTime
-
-            // Report progress (100% - complete)
-            AudioStudioModule.sendEvent('TrimProgress', {
-                progress: 100,
-            })
-
-            // Create result object
-            const result: TrimAudioResult = {
-                uri: outputUri,
-                filename,
-                durationMs: Math.round(resultBuffer.duration * 1000),
-                size: outputData.byteLength,
-                sampleRate: resultBuffer.sampleRate,
-                channels: resultBuffer.numberOfChannels,
-                bitDepth: targetBitDepth,
-                mimeType: outputMimeType,
-                processingInfo: {
-                    durationMs: processingTimeMs,
-                },
-            }
-
-            // Add compression info if available
-            if (compressionInfo) {
-                result.compression = compressionInfo
-            }
-
-            return result
-        } catch (error) {
-            console.error('Error in trimAudio:', error)
-            throw error
-        }
-    }
-
     // Add a sendEvent method for web
     AudioStudioModule.sendEvent = (eventName: string, params: any) => {
         // This will be picked up by the LegacyEventEmitter in trimAudio.ts
-        if (
-            AudioStudioModule.listeners &&
-            AudioStudioModule.listeners[eventName]
-        ) {
-            AudioStudioModule.listeners[eventName].forEach(
-                (listener: Function) => {
-                    listener(params)
-                }
-            )
-        }
+        AudioStudioModule.listeners?.[eventName]?.forEach(
+            (listener: Function) => {
+                listener(params)
+            }
+        )
     }
 
     // Initialize listeners object
@@ -819,164 +129,6 @@ if (Platform.OS === 'web') {
     }
 }
 
-// Move the encodeCompressedAudio function outside the if block to fix the ESLint error
-async function encodeCompressedAudio(
-    buffer: AudioBuffer,
-    format: 'opus' | 'aac',
-    bitrate?: number
-): Promise<{ data: ArrayBuffer; bitrate: number }> {
-    return new Promise((resolve, reject) => {
-        try {
-            // On web, always use opus if aac is requested
-            const actualFormat =
-                Platform.OS === 'web' && format === 'aac' ? 'opus' : format
-
-            // Check if MediaRecorder supports the requested format
-            const mimeType =
-                actualFormat === 'opus' ? 'audio/webm;codecs=opus' : 'audio/aac'
-            if (!MediaRecorder.isTypeSupported(mimeType)) {
-                throw new Error(`MediaRecorder does not support ${mimeType}`)
-            }
-
-            // Create a new AudioContext and source
-            const ctx = new (window.AudioContext ||
-                (window as any).webkitAudioContext)()
-            const source = ctx.createBufferSource()
-            source.buffer = buffer
-
-            // Create a MediaStreamDestination to capture the audio
-            const destination = ctx.createMediaStreamDestination()
-            source.connect(destination)
-
-            // Create a MediaRecorder with the requested format
-            const recorder = new MediaRecorder(destination.stream, {
-                mimeType,
-                audioBitsPerSecond:
-                    bitrate || (actualFormat === 'opus' ? 32000 : 64000),
-            })
-
-            const chunks: Blob[] = []
-
-            recorder.ondataavailable = (e) => {
-                if (e.data.size > 0) {
-                    chunks.push(e.data)
-                }
-            }
-
-            recorder.onstop = async () => {
-                try {
-                    const blob = new Blob(chunks, { type: mimeType })
-                    const arrayBuffer = await blob.arrayBuffer()
-
-                    // Get the actual bitrate used
-                    const actualBitrate = Math.round(
-                        (arrayBuffer.byteLength * 8) / buffer.duration
-                    )
-
-                    resolve({
-                        data: arrayBuffer,
-                        bitrate: actualBitrate / 1000, // Convert to kbps
-                    })
-
-                    // Clean up
-                    ctx.close()
-                } catch (error) {
-                    reject(error)
-                }
-            }
-
-            // Start recording and playback
-            recorder.start()
-            source.start(0)
-
-            // Stop recording when the buffer finishes playing
-            setTimeout(() => {
-                recorder.stop()
-                source.stop()
-            }, buffer.duration * 1000)
-        } catch (error) {
-            reject(error)
-        }
-    })
-}
-
-// Improved resampleAudioBuffer function
-async function resampleAudioBuffer(
-    context: AudioContext,
-    buffer: AudioBuffer,
-    targetSampleRate: number,
-    targetChannels: number
-): Promise<AudioBuffer> {
-    // If no change needed, return the original buffer
-    if (
-        buffer.sampleRate === targetSampleRate &&
-        buffer.numberOfChannels === targetChannels
-    ) {
-        return buffer
-    }
-
-    console.log(
-        `Resampling: ${buffer.sampleRate}Hz → ${targetSampleRate}Hz, ${buffer.numberOfChannels} → ${targetChannels} channels`
-    )
-
-    // Calculate the new length based on the sample rate change
-    const newLength = Math.round(
-        (buffer.length * targetSampleRate) / buffer.sampleRate
-    )
-
-    // Create an offline context for resampling
-    const offlineContext = new OfflineAudioContext(
-        targetChannels,
-        newLength,
-        targetSampleRate
-    )
-
-    // Create a source node
-    const source = offlineContext.createBufferSource()
-    source.buffer = buffer
-
-    // If we need to change channel count
-    if (buffer.numberOfChannels !== targetChannels) {
-        if (targetChannels === 1 && buffer.numberOfChannels > 1) {
-            // Downmix to mono
-            const merger = offlineContext.createChannelMerger(1)
-
-            // Create a gain node to reduce volume when downmixing to prevent clipping
-            const gainNode = offlineContext.createGain()
-            gainNode.gain.value = 1.0 / buffer.numberOfChannels
-
-            source.connect(gainNode)
-            gainNode.connect(merger)
-            merger.connect(offlineContext.destination)
-        } else if (targetChannels === 2 && buffer.numberOfChannels === 1) {
-            // Upmix mono to stereo (duplicate the channel)
-            const splitter = offlineContext.createChannelSplitter(1)
-            const merger = offlineContext.createChannelMerger(2)
-
-            source.connect(splitter)
-            splitter.connect(merger, 0, 0)
-            splitter.connect(merger, 0, 1)
-            merger.connect(offlineContext.destination)
-        } else {
-            // For other cases, just connect and let the system handle it
-            source.connect(offlineContext.destination)
-        }
-    } else {
-        // No channel conversion needed
-        source.connect(offlineContext.destination)
-    }
-
-    // Start rendering
-    source.start(0)
-    const resampledBuffer = await offlineContext.startRendering()
-
-    console.log(
-        `Resampling complete: ${resampledBuffer.length} samples at ${resampledBuffer.sampleRate}Hz`
-    )
-
-    return resampledBuffer
-}
-
 if (Platform.OS !== 'web') {
     AudioStudioModule = requireNativeModule('AudioStudio')
 }