npm - @siteed/audio-studio - Versions diffs - 3.0.3 → 3.0.4 - Mend

@siteed/audio-studio 3.0.3 → 3.0.4

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.

Files changed (19) hide show

package/CHANGELOG.md +11 -1
package/android/src/main/java/net/siteed/audiostudio/AudioRecorderManager.kt +41 -35
package/android/src/main/java/net/siteed/audiostudio/AudioStudioModule.kt +500 -485
package/build/cjs/AudioAnalysis/wasmConfig.js.map +1 -1
package/build/cjs/AudioAnalysis/wasmLoader.web.js +2 -1
package/build/cjs/AudioAnalysis/wasmLoader.web.js.map +1 -1
package/build/cjs/trimAudio.js.map +1 -1
package/build/esm/AudioAnalysis/wasmConfig.js.map +1 -1
package/build/esm/AudioAnalysis/wasmLoader.web.js +2 -1
package/build/esm/AudioAnalysis/wasmLoader.web.js.map +1 -1
package/build/esm/trimAudio.js.map +1 -1
package/build/types/AudioAnalysis/wasmLoader.web.d.ts.map +1 -1
package/build/types/trimAudio.d.ts.map +1 -1
package/ios/AudioStreamManager.swift +135 -89
package/ios/AudioStudioModule.swift +239 -216
package/package.json +1 -1
package/src/AudioAnalysis/wasmConfig.ts +1 -1
package/src/AudioAnalysis/wasmLoader.web.ts +8 -3
package/src/trimAudio.ts +19 -5

package/build/cjs/AudioAnalysis/wasmConfig.js.map CHANGED Viewed

	@@ -1 +1 @@
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package/build/cjs/AudioAnalysis/wasmLoader.web.js CHANGED Viewed

@@ -56,7 +56,8 @@ function getWasmModule() {
         modulePromise = (async () => {
             const url = (0, wasmConfig_1.getMelSpectrogramWasmUrl)();
             // Try ESM import first; fall back to <script> tag for UMD modules
-            const mod = await Promise.resolve(`${url}`).then(s => __importStar(require(s)));
+            const mod = await Promise.resolve(`${
+            /* webpackIgnore: true */ /* @vite-ignore */ url}`).then(s => __importStar(require(s)));
             let factory = mod.default ?? mod;
             if (typeof factory !== 'function') {
                 // UMD fallback: load via <script> tag so the top-level `var` becomes a global and

package/build/cjs/AudioAnalysis/wasmLoader.web.js.map CHANGED Viewed

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package/build/cjs/trimAudio.js.map CHANGED Viewed

	@@ -1 +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"]}
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(\n buffer: AudioBuffer,\n bitDepth: BitDepth\n): 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(\n clamped * 32767\n )\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(\n resultBuffer,\n targetBitDepth as BitDepth\n )\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(\n resultBuffer,\n 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 outputData = encodeBufferToWav(\n resultBuffer,\n targetBitDepth as BitDepth\n )\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/AudioAnalysis/wasmConfig.js.map CHANGED Viewed

	@@ -1 +1 @@
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package/build/esm/AudioAnalysis/wasmLoader.web.js CHANGED Viewed

@@ -20,7 +20,8 @@ export function getWasmModule() {
         modulePromise = (async () => {
             const url = getMelSpectrogramWasmUrl();
             // Try ESM import first; fall back to <script> tag for UMD modules
-            const mod = await import(/* webpackIgnore: true */ /* @vite-ignore */ url);
+            const mod = await import(
+            /* webpackIgnore: true */ /* @vite-ignore */ url);
             let factory = mod.default ?? mod;
             if (typeof factory !== 'function') {
                 // UMD fallback: load via <script> tag so the top-level `var` becomes a global and

package/build/esm/AudioAnalysis/wasmLoader.web.js.map CHANGED Viewed

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package/build/esm/trimAudio.js.map CHANGED Viewed

	@@ -1 +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"]}
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(\n buffer: AudioBuffer,\n bitDepth: BitDepth\n): 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(\n clamped * 32767\n )\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(\n resultBuffer,\n targetBitDepth as BitDepth\n )\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(\n resultBuffer,\n 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 outputData = encodeBufferToWav(\n resultBuffer,\n targetBitDepth as BitDepth\n )\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"]}

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