@siteed/audio-studio 3.0.2 → 3.0.3

package/build/cjs/AudioStudioModule.js

@@ -1,22 +1,14 @@
 "use strict";
-var __importDefault = (this && this.__importDefault) || function (mod) {
-    return (mod && mod.__esModule) ? mod : { "default": mod };
-};
 Object.defineProperty(exports, "__esModule", { value: true });
 const expo_modules_core_1 = require("expo-modules-core");
 const react_native_1 = require("react-native");
 const AudioStudio_web_1 = require("./AudioStudio.web");
-const audioProcessing_1 = require("./utils/audioProcessing");
-const crc32_1 = __importDefault(require("./utils/crc32"));
-const writeWavHeader_1 = require("./utils/writeWavHeader");
 // eslint-disable-next-line @typescript-eslint/no-explicit-any
 let AudioStudioModule;
 if (react_native_1.Platform.OS === 'web') {
     let instance = null;
     AudioStudioModule = (webProps) => {
-        if (!instance) {
-            instance = new AudioStudio_web_1.AudioStudioWeb(webProps);
-        }
+        instance ??= new AudioStudio_web_1.AudioStudioWeb(webProps);
         return instance;
     };
     AudioStudioModule.requestPermissionsAsync = async () => {
@@ -73,487 +65,12 @@ if (react_native_1.Platform.OS === 'web') {
                 return await AudioStudioModule.requestPermissionsAsync();
         }
     };
-    AudioStudioModule.extractAudioData = async (options) => {
-        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 (0, audioProcessing_1.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);
-            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 = {
-                pcmData: new Uint8Array(pcmData.buffer),
-                sampleRate: processedBuffer.sampleRate,
-                channels: processedBuffer.channels,
-                bitDepth,
-                durationMs,
-                format: `pcm_${bitDepth}bit`,
-                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 = (0, writeWavHeader_1.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_1.default.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) => {
-        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.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;
-            // 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 (0, audioProcessing_1.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
-                let segmentsToProcess = [];
-                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 = [];
-                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 (0, audioProcessing_1.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;
-            let outputMimeType;
-            let compressionInfo = null;
-            // Check if AAC was requested on web and show a warning
-            if (format === 'aac' && react_native_1.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 = (0, writeWavHeader_1.writeWavHeader)({
-                    buffer: rawBuffer,
-                    sampleRate: resultBuffer.sampleRate, // Use the actual buffer's sample rate
-                    numChannels: resultBuffer.numberOfChannels,
-                    bitDepth: targetBitDepth,
-                });
-                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 = (0, writeWavHeader_1.writeWavHeader)({
-                        buffer: wavData.buffer,
-                        sampleRate: resultBuffer.sampleRate,
-                        numChannels: resultBuffer.numberOfChannels,
-                        bitDepth: targetBitDepth,
-                    });
-                    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 = (0, writeWavHeader_1.writeWavHeader)({
-                    buffer: wavData.buffer,
-                    sampleRate: resultBuffer.sampleRate,
-                    numChannels: resultBuffer.numberOfChannels,
-                    bitDepth: targetBitDepth,
-                });
-                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 = {
-                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, params) => {
         // This will be picked up by the LegacyEventEmitter in trimAudio.ts
-        if (AudioStudioModule.listeners &&
-            AudioStudioModule.listeners[eventName]) {
-            AudioStudioModule.listeners[eventName].forEach((listener) => {
-                listener(params);
-            });
-        }
+        AudioStudioModule.listeners?.[eventName]?.forEach((listener) => {
+            listener(params);
+        });
     };
     // Initialize listeners object
     AudioStudioModule.listeners = {};
@@ -599,118 +116,6 @@ if (react_native_1.Platform.OS === 'web') {
         }
     };
 }
-// Move the encodeCompressedAudio function outside the if block to fix the ESLint error
-async function encodeCompressedAudio(buffer, format, bitrate) {
-    return new Promise((resolve, reject) => {
-        try {
-            // On web, always use opus if aac is requested
-            const actualFormat = react_native_1.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.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 = [];
-            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, buffer, targetSampleRate, targetChannels) {
-    // 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 (react_native_1.Platform.OS !== 'web') {
     AudioStudioModule = (0, expo_modules_core_1.requireNativeModule)('AudioStudio');
 }