@siteed/audio-studio 3.0.2 → 3.0.3

package/build/esm/AudioStudioModule.js.map

@@ -1 +1 @@
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{ requireNativeModule } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n    ExtractAudioDataOptions,\n    ExtractedAudioData,\n    BitDepth,\n    TrimAudioOptions,\n    TrimAudioResult,\n} from './AudioStudio.types'\nimport { AudioStudioWeb, AudioStudioWebProps } from './AudioStudio.web'\nimport { processAudioBuffer } from './utils/audioProcessing'\nimport crc32 from './utils/crc32'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nlet AudioStudioModule: any\n\nif (Platform.OS === 'web') {\n    let instance: AudioStudioWeb | null = null\n\n    AudioStudioModule = (webProps: AudioStudioWebProps) => {\n        if (!instance) {\n            instance = new AudioStudioWeb(webProps)\n        }\n        return instance\n    }\n    AudioStudioModule.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    AudioStudioModule.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 AudioStudioModule.requestPermissionsAsync()\n        }\n    }\n    AudioStudioModule.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    AudioStudioModule.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            AudioStudioModule.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                    AudioStudioModule.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            AudioStudioModule.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            AudioStudioModule.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            AudioStudioModule.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    AudioStudioModule.sendEvent = (eventName: string, params: any) => {\n        // This will be picked up by the LegacyEventEmitter in trimAudio.ts\n        if (\n            AudioStudioModule.listeners &&\n            AudioStudioModule.listeners[eventName]\n        ) {\n            AudioStudioModule.listeners[eventName].forEach(\n                (listener: Function) => {\n                    listener(params)\n                }\n            )\n        }\n    }\n\n    // Initialize listeners object\n    AudioStudioModule.listeners = {}\n\n    // Add methods for event listeners that LegacyEventEmitter will use\n    AudioStudioModule.addListener = (eventName: string, listener: Function) => {\n        if (!AudioStudioModule.listeners[eventName]) {\n            AudioStudioModule.listeners[eventName] = []\n        }\n        AudioStudioModule.listeners[eventName].push(listener)\n\n        // Return an object with a remove method\n        return {\n            remove: () => {\n                const index =\n                    AudioStudioModule.listeners[eventName].indexOf(listener)\n                if (index !== -1) {\n                    AudioStudioModule.listeners[eventName].splice(index, 1)\n                }\n            },\n        }\n    }\n\n    AudioStudioModule.removeAllListeners = (eventName: string) => {\n        if (AudioStudioModule.listeners[eventName]) {\n            delete AudioStudioModule.listeners[eventName]\n        }\n    }\n\n    AudioStudioModule.prepareRecording = async (options: any) => {\n        // For web platform, we'll implement a simplified version that just checks permissions\n        // and does minimal setup. The actual recording setup will still happen in startRecording.\n        try {\n            // Check for microphone permissions\n            const permissionsResult =\n                await AudioStudioModule.getPermissionsAsync()\n            if (!permissionsResult.granted) {\n                throw new Error('Microphone permission not granted')\n            }\n\n            // If using a web instance, call its prepareRecording method\n            if (instance) {\n                return await instance.prepareRecording(options)\n            }\n\n            return true\n        } catch (error) {\n            console.error('Error preparing recording:', error)\n            throw error\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    AudioStudioModule = requireNativeModule('AudioStudio')\n}\n\nexport default AudioStudioModule\n"]}
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{ requireNativeModule } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport { AudioStudioWeb, AudioStudioWebProps } from './AudioStudio.web'\n\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nlet AudioStudioModule: any\n\nif (Platform.OS === 'web') {\n    let instance: AudioStudioWeb | null = null\n\n    AudioStudioModule = (webProps: AudioStudioWebProps) => {\n        instance ??= new AudioStudioWeb(webProps)\n        return instance\n    }\n    AudioStudioModule.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    AudioStudioModule.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 AudioStudioModule.requestPermissionsAsync()\n        }\n    }\n    // Add a sendEvent method for web\n    AudioStudioModule.sendEvent = (eventName: string, params: any) => {\n        // This will be picked up by the LegacyEventEmitter in trimAudio.ts\n        AudioStudioModule.listeners?.[eventName]?.forEach(\n            (listener: Function) => {\n                listener(params)\n            }\n        )\n    }\n\n    // Initialize listeners object\n    AudioStudioModule.listeners = {}\n\n    // Add methods for event listeners that LegacyEventEmitter will use\n    AudioStudioModule.addListener = (eventName: string, listener: Function) => {\n        if (!AudioStudioModule.listeners[eventName]) {\n            AudioStudioModule.listeners[eventName] = []\n        }\n        AudioStudioModule.listeners[eventName].push(listener)\n\n        // Return an object with a remove method\n        return {\n            remove: () => {\n                const index =\n                    AudioStudioModule.listeners[eventName].indexOf(listener)\n                if (index !== -1) {\n                    AudioStudioModule.listeners[eventName].splice(index, 1)\n                }\n            },\n        }\n    }\n\n    AudioStudioModule.removeAllListeners = (eventName: string) => {\n        if (AudioStudioModule.listeners[eventName]) {\n            delete AudioStudioModule.listeners[eventName]\n        }\n    }\n\n    AudioStudioModule.prepareRecording = async (options: any) => {\n        // For web platform, we'll implement a simplified version that just checks permissions\n        // and does minimal setup. The actual recording setup will still happen in startRecording.\n        try {\n            // Check for microphone permissions\n            const permissionsResult =\n                await AudioStudioModule.getPermissionsAsync()\n            if (!permissionsResult.granted) {\n                throw new Error('Microphone permission not granted')\n            }\n\n            // If using a web instance, call its prepareRecording method\n            if (instance) {\n                return await instance.prepareRecording(options)\n            }\n\n            return true\n        } catch (error) {\n            console.error('Error preparing recording:', error)\n            throw error\n        }\n    }\n}\n\nif (Platform.OS !== 'web') {\n    AudioStudioModule = requireNativeModule('AudioStudio')\n}\n\nexport default AudioStudioModule\n"]}

package/build/esm/trimAudio.js

@@ -1,8 +1,44 @@
 import { LegacyEventEmitter } from 'expo-modules-core';
 import AudioStudioModule from './AudioStudioModule';
+import { isWeb } from './constants';
+import { processAudioBuffer } from './utils/audioProcessing';
 import { cleanNativeOptions } from './utils/cleanNativeOptions';
+import { encodeCompressedAudio } from './utils/encodeCompressedAudio.web';
+import { resampleAudioBuffer } from './utils/resampleAudioBuffer.web';
+import { writeWavHeader } from './utils/writeWavHeader';
 // Create a single emitter instance
 const emitter = new LegacyEventEmitter(AudioStudioModule);
+function sliceAudioBuffer(src, ctx, startMs, endMs) {
+    const sr = src.sampleRate;
+    const start = Math.floor((startMs / 1000) * sr);
+    const end = Math.min(Math.ceil((endMs / 1000) * sr), src.length);
+    const length = Math.max(0, end - start);
+    const out = ctx.createBuffer(src.numberOfChannels, length, sr);
+    for (let c = 0; c < src.numberOfChannels; c++) {
+        out.getChannelData(c).set(src.getChannelData(c).subarray(start, end));
+    }
+    return out;
+}
+function encodeBufferToWav(buffer, bitDepth) {
+    const { length, numberOfChannels, sampleRate } = buffer;
+    const channels = [];
+    for (let c = 0; c < numberOfChannels; c++) {
+        channels.push(buffer.getChannelData(c));
+    }
+    const interleavedData = new Int16Array(length * numberOfChannels);
+    for (let i = 0; i < length; i++) {
+        for (let c = 0; c < numberOfChannels; c++) {
+            const clamped = Math.max(-1, Math.min(1, channels[c][i]));
+            interleavedData[i * numberOfChannels + c] = Math.round(clamped * 32767);
+        }
+    }
+    return writeWavHeader({
+        buffer: interleavedData.buffer,
+        sampleRate,
+        numChannels: numberOfChannels,
+        bitDepth,
+    });
+}
 /**
  * Trims an audio file based on the provided options.
  *
@@ -34,6 +70,197 @@ export async function trimAudio(options, progressCallback) {
     else {
         throw new Error(`Invalid mode: ${mode}. Must be 'single', 'keep', or 'remove'`);
     }
+    if (isWeb) {
+        try {
+            const startTime = performance.now();
+            const { fileUri, startTimeMs, endTimeMs, ranges, outputFileName, outputFormat, } = options;
+            // 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;
+            // 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
+            progressCallback?.({ 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,
+                });
+                resultBuffer = buffer;
+                // If we need to change sample rate or channels, do it after extraction
+                if (targetSampleRate !== originalSampleRate ||
+                    targetChannels !== originalChannels) {
+                    resultBuffer = await resampleAudioBuffer(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
+                    progressCallback?.({
+                        progress: 10 +
+                            Math.round((i / segmentsToProcess.length) * 40),
+                    });
+                    // Slice from the already-decoded buffer (avoids N re-fetches)
+                    const segmentBuffer = sliceAudioBuffer(originalAudioBuffer, audioContext, segment.startTimeMs, segment.endTimeMs);
+                    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) {
+                    resultBuffer = await resampleAudioBuffer(concatenatedBuffer, targetSampleRate, targetChannels);
+                }
+            }
+            // Report progress (50% - processing complete)
+            progressCallback?.({ progress: 50 });
+            // Encode the result based on the requested format
+            let outputData;
+            let outputMimeType;
+            let compressionInfo = undefined;
+            // AAC is not reliably supported in browsers; fall back to opus
+            if (format === 'aac') {
+                console.warn('AAC format is not supported on web platforms. Falling back to OPUS format.');
+                format = 'opus';
+            }
+            if (format === 'wav') {
+                outputData = encodeBufferToWav(resultBuffer, targetBitDepth);
+                outputMimeType = 'audio/wav';
+            }
+            else if (format === 'opus') {
+                try {
+                    const { data, bitrate } = await encodeCompressedAudio(resultBuffer, format, outputFormat?.bitrate);
+                    outputData = data;
+                    outputMimeType = 'audio/webm';
+                    compressionInfo = { format, bitrate, size: data.byteLength };
+                }
+                catch (error) {
+                    console.warn(`Failed to encode to ${format}, falling back to WAV: ${error}`);
+                    outputData = encodeBufferToWav(resultBuffer, targetBitDepth);
+                    outputMimeType = 'audio/wav';
+                }
+            }
+            else {
+                // Default to WAV for unsupported formats
+                console.warn(`Format ${format} not supported on web, using WAV instead`);
+                outputData = encodeBufferToWav(resultBuffer, targetBitDepth);
+                outputMimeType = 'audio/wav';
+            }
+            // Report progress (90% - encoding complete)
+            progressCallback?.({ 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)
+            progressCallback?.({ 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,
+                },
+            };
+            if (compressionInfo) {
+                result.compression = compressionInfo;
+            }
+            return result;
+        }
+        catch (error) {
+            console.error('Error in trimAudio:', error);
+            throw error;
+        }
+    }
     // Set up progress event listener if callback is provided
     let subscription;
     if (progressCallback) {

package/build/esm/trimAudio.js.map

@@ -1 +1 @@
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{ LegacyEventEmitter, type EventSubscription } from 'expo-modules-core'\n\nimport {\n    BitDepth,\n    TrimAudioOptions,\n    TrimAudioResult,\n    TrimProgressEvent,\n} from './AudioStudio.types'\nimport AudioStudioModule from './AudioStudioModule'\nimport { isWeb } from './constants'\nimport { processAudioBuffer } from './utils/audioProcessing'\nimport { cleanNativeOptions } from './utils/cleanNativeOptions'\nimport { encodeCompressedAudio } from './utils/encodeCompressedAudio.web'\nimport { resampleAudioBuffer } from './utils/resampleAudioBuffer.web'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\n// Create a single emitter instance\nconst emitter = new LegacyEventEmitter(AudioStudioModule)\n\nfunction sliceAudioBuffer(\n    src: AudioBuffer,\n    ctx: AudioContext,\n    startMs: number,\n    endMs: number\n): AudioBuffer {\n    const sr = src.sampleRate\n    const start = Math.floor((startMs / 1000) * sr)\n    const end = Math.min(Math.ceil((endMs / 1000) * sr), src.length)\n    const length = Math.max(0, end - start)\n    const out = ctx.createBuffer(src.numberOfChannels, length, sr)\n    for (let c = 0; c < src.numberOfChannels; c++) {\n        out.getChannelData(c).set(src.getChannelData(c).subarray(start, end))\n    }\n    return out\n}\n\nfunction encodeBufferToWav(buffer: AudioBuffer, bitDepth: BitDepth): ArrayBuffer {\n    const { length, numberOfChannels, sampleRate } = buffer\n    const channels: Float32Array[] = []\n    for (let c = 0; c < numberOfChannels; c++) {\n        channels.push(buffer.getChannelData(c))\n    }\n    const interleavedData = new Int16Array(length * numberOfChannels)\n    for (let i = 0; i < length; i++) {\n        for (let c = 0; c < numberOfChannels; c++) {\n            const clamped = Math.max(-1, Math.min(1, channels[c][i]))\n            interleavedData[i * numberOfChannels + c] = Math.round(clamped * 32767)\n        }\n    }\n    return writeWavHeader({\n        buffer: interleavedData.buffer as ArrayBuffer,\n        sampleRate,\n        numChannels: numberOfChannels,\n        bitDepth,\n    })\n}\n\n/**\n * Trims an audio file based on the provided options.\n *\n * @experimental This API is experimental and not fully optimized for production use.\n * Performance may vary based on file size and device capabilities.\n * Future versions may include breaking changes.\n *\n * @param options Configuration options for the trimming operation\n * @param progressCallback Optional callback to receive progress updates\n * @returns Promise resolving to the trimmed audio file information, including processing time\n */\nexport async function trimAudio(\n    options: TrimAudioOptions,\n    progressCallback?: (event: TrimProgressEvent) => void\n): Promise<TrimAudioResult> {\n    // Validation\n    if (!options.fileUri) {\n        throw new Error('fileUri is required')\n    }\n    const mode = options.mode ?? 'single'\n    if (mode === 'single') {\n        if (\n            options.startTimeMs === undefined &&\n            options.endTimeMs === undefined\n        ) {\n            throw new Error(\n                'At least one of startTimeMs or endTimeMs must be provided in single mode'\n            )\n        }\n    } else if (mode === 'keep' || mode === 'remove') {\n        if (!options.ranges || options.ranges.length === 0) {\n            throw new Error(\n                'ranges must be provided and non-empty for keep or remove modes'\n            )\n        }\n    } else {\n        throw new Error(\n            `Invalid mode: ${mode}. Must be 'single', 'keep', or 'remove'`\n        )\n    }\n\n    if (isWeb) {\n        try {\n            const startTime = performance.now()\n            const {\n                fileUri,\n                startTimeMs,\n                endTimeMs,\n                ranges,\n                outputFileName,\n                outputFormat,\n            } = options\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            // 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            progressCallback?.({ progress: 10 })\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                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                    resultBuffer = await resampleAudioBuffer(\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                    progressCallback?.({\n                        progress:\n                            10 +\n                            Math.round((i / segmentsToProcess.length) * 40),\n                    })\n\n                    // Slice from the already-decoded buffer (avoids N re-fetches)\n                    const segmentBuffer = sliceAudioBuffer(\n                        originalAudioBuffer,\n                        audioContext,\n                        segment.startTimeMs,\n                        segment.endTimeMs\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                    resultBuffer = await resampleAudioBuffer(\n                        concatenatedBuffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            }\n\n            // Report progress (50% - processing complete)\n            progressCallback?.({ progress: 50 })\n\n            // Encode the result based on the requested format\n            let outputData: ArrayBuffer\n            let outputMimeType: string\n            let compressionInfo: TrimAudioResult['compression'] = undefined\n\n            // AAC is not reliably supported in browsers; fall back to opus\n            if (format === 'aac') {\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                outputData = encodeBufferToWav(resultBuffer, targetBitDepth as BitDepth)\n                outputMimeType = 'audio/wav'\n            } else if (format === 'opus') {\n                try {\n                    const { data, bitrate } = await encodeCompressedAudio(\n                        resultBuffer,\n                        format,\n                        outputFormat?.bitrate\n                    )\n                    outputData = data\n                    outputMimeType = 'audio/webm'\n                    compressionInfo = { format, bitrate, size: data.byteLength }\n                } catch (error) {\n                    console.warn(\n                        `Failed to encode to ${format}, falling back to WAV: ${error}`\n                    )\n                    outputData = encodeBufferToWav(resultBuffer, targetBitDepth as BitDepth)\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                outputData = encodeBufferToWav(resultBuffer, targetBitDepth as BitDepth)\n                outputMimeType = 'audio/wav'\n            }\n\n            // Report progress (90% - encoding complete)\n            progressCallback?.({ progress: 90 })\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            progressCallback?.({ progress: 100 })\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            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    // Set up progress event listener if callback is provided\n    let subscription: EventSubscription | undefined\n    if (progressCallback) {\n        subscription = emitter.addListener(\n            'TrimProgress',\n            (event: TrimProgressEvent) => {\n                progressCallback(event)\n            }\n        )\n    }\n\n    try {\n        // Clean non-serializable/undefined values to avoid Android Kotlin bridge crash\n        const result = await AudioStudioModule.trimAudio(\n            cleanNativeOptions(options)\n        )\n        return result\n    } finally {\n        if (subscription) {\n            subscription.remove()\n        }\n    }\n}\n\n/**\n * Simplified version of trimAudio that returns only the URI of the trimmed file.\n *\n * @experimental This API is experimental and not fully optimized for production use.\n * Performance may vary based on file size and device capabilities.\n * Future versions may include breaking changes.\n *\n * @param options Configuration options for the trimming operation\n * @returns Promise resolving to the URI of the trimmed audio file\n */\nexport async function trimAudioSimple(\n    options: TrimAudioOptions\n): Promise<string> {\n    const result = await trimAudio(options)\n    return result.uri\n}\n"]}

package/build/esm/utils/encodeCompressedAudio.web.js

@@ -0,0 +1,62 @@
+/**
+ * NOTE: Encodes audio in real-time via AudioContext + MediaRecorder playback.
+ * A 60-second clip takes ~60 seconds to encode — this is a known Web Audio API
+ * limitation; there is no offline API for Opus/AAC encoding in browsers.
+ */
+export function encodeCompressedAudio(buffer, format, bitrate) {
+    return new Promise((resolve, reject) => {
+        try {
+            // On web, always use opus if aac is requested (browser aac support is rare)
+            const actualFormat = 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; stop recorder when playback ends
+            recorder.start();
+            source.onended = () => recorder.stop();
+            source.start(0);
+        }
+        catch (error) {
+            reject(error);
+        }
+    });
+}
+//# sourceMappingURL=encodeCompressedAudio.web.js.map

package/build/esm/utils/encodeCompressedAudio.web.js.map

@@ -0,0 +1 @@
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'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; stop recorder when playback ends\n            recorder.start()\n            source.onended = () => recorder.stop()\n            source.start(0)\n        } catch (error) {\n            reject(error)\n        }\n    })\n}\n"]}

package/build/esm/utils/resampleAudioBuffer.web.js

@@ -0,0 +1,22 @@
+export async function resampleAudioBuffer(buffer, targetSampleRate, targetChannels) {
+    // If no change needed, return the original buffer
+    if (buffer.sampleRate === targetSampleRate &&
+        buffer.numberOfChannels === targetChannels) {
+        return buffer;
+    }
+    // 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;
+    // The OfflineAudioContext was created with targetChannels; the Web Audio API
+    // applies its built-in speaker downmix/upmix rules automatically.
+    source.connect(offlineContext.destination);
+    // Start rendering
+    source.start(0);
+    const resampledBuffer = await offlineContext.startRendering();
+    return resampledBuffer;
+}
+//# sourceMappingURL=resampleAudioBuffer.web.js.map

package/build/esm/utils/resampleAudioBuffer.web.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"resampleAudioBuffer.web.js","sourceRoot":"","sources":["../../../src/utils/resampleAudioBuffer.web.ts"],"names":[],"mappings":"AAAA,MAAM,CAAC,KAAK,UAAU,mBAAmB,CACrC,MAAmB,EACnB,gBAAwB,EACxB,cAAsB;IAEtB,kDAAkD;IAClD,IACI,MAAM,CAAC,UAAU,KAAK,gBAAgB;QACtC,MAAM,CAAC,gBAAgB,KAAK,cAAc,EAC5C,CAAC;QACC,OAAO,MAAM,CAAA;IACjB,CAAC;IAED,2DAA2D;IAC3D,MAAM,SAAS,GAAG,IAAI,CAAC,KAAK,CACxB,CAAC,MAAM,CAAC,MAAM,GAAG,gBAAgB,CAAC,GAAG,MAAM,CAAC,UAAU,CACzD,CAAA;IAED,2CAA2C;IAC3C,MAAM,cAAc,GAAG,IAAI,mBAAmB,CAC1C,cAAc,EACd,SAAS,EACT,gBAAgB,CACnB,CAAA;IAED,uBAAuB;IACvB,MAAM,MAAM,GAAG,cAAc,CAAC,kBAAkB,EAAE,CAAA;IAClD,MAAM,CAAC,MAAM,GAAG,MAAM,CAAA;IAEtB,6EAA6E;IAC7E,kEAAkE;IAClE,MAAM,CAAC,OAAO,CAAC,cAAc,CAAC,WAAW,CAAC,CAAA;IAE1C,kBAAkB;IAClB,MAAM,CAAC,KAAK,CAAC,CAAC,CAAC,CAAA;IACf,MAAM,eAAe,GAAG,MAAM,cAAc,CAAC,cAAc,EAAE,CAAA;IAE7D,OAAO,eAAe,CAAA;AAC1B,CAAC","sourcesContent":["export async function resampleAudioBuffer(\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    // 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    // The OfflineAudioContext was created with targetChannels; the Web Audio API\n    // applies its built-in speaker downmix/upmix rules automatically.\n    source.connect(offlineContext.destination)\n\n    // Start rendering\n    source.start(0)\n    const resampledBuffer = await offlineContext.startRendering()\n\n    return resampledBuffer\n}\n"]}

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

@@ -182,6 +182,17 @@ export interface ExtractMelSpectrogramOptions {
     endTimeMs?: number;
     logger?: ConsoleLike;
 }
+/**
+ * Result type for WASM-based audio feature extraction.
+ */
+export interface AudioFeaturesWasmResult {
+    spectralCentroid: number;
+    spectralFlatness: number;
+    spectralRolloff: number;
+    spectralBandwidth: number;
+    mfcc: number[];
+    chromagram: number[];
+}
 /**
  * Return type for mel spectrogram extraction
  *