@siteed/audio-studio 3.2.0-beta.1 → 3.2.0

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 (85) hide show
  1. package/CHANGELOG.md +356 -5
  2. package/android/src/main/java/net/siteed/audiostudio/AudioStreamDecoder.kt +306 -94
  3. package/android/src/main/java/net/siteed/audiostudio/AudioStudioModule.kt +39 -6
  4. package/build/cjs/errors/AudioStreamError.js +9 -0
  5. package/build/cjs/errors/AudioStreamError.js.map +1 -1
  6. package/build/cjs/errors/AudioStreamError.test.js +22 -1
  7. package/build/cjs/errors/AudioStreamError.test.js.map +1 -1
  8. package/build/cjs/streamAudioData.js +99 -32
  9. package/build/cjs/streamAudioData.js.map +1 -1
  10. package/build/cjs/utils/audioProcessing.js +14 -10
  11. package/build/cjs/utils/audioProcessing.js.map +1 -1
  12. package/build/esm/errors/AudioStreamError.js +9 -0
  13. package/build/esm/errors/AudioStreamError.js.map +1 -1
  14. package/build/esm/errors/AudioStreamError.test.js +22 -1
  15. package/build/esm/errors/AudioStreamError.test.js.map +1 -1
  16. package/build/esm/streamAudioData.js +99 -32
  17. package/build/esm/streamAudioData.js.map +1 -1
  18. package/build/esm/utils/audioProcessing.js +14 -10
  19. package/build/esm/utils/audioProcessing.js.map +1 -1
  20. package/build/types/errors/AudioStreamError.d.ts.map +1 -1
  21. package/build/types/streamAudioData.d.ts +5 -0
  22. package/build/types/streamAudioData.d.ts.map +1 -1
  23. package/build/types/utils/audioProcessing.d.ts +2 -2
  24. package/build/types/utils/audioProcessing.d.ts.map +1 -1
  25. package/ios/AudioStreamDecoder.swift +191 -100
  26. package/ios/AudioStudioModule.swift +48 -9
  27. package/package.json +163 -146
  28. package/scripts/README.md +58 -0
  29. package/src/errors/AudioStreamError.test.ts +29 -2
  30. package/src/errors/AudioStreamError.ts +14 -0
  31. package/src/streamAudioData.ts +146 -42
  32. package/src/utils/audioProcessing.ts +25 -14
  33. package/android/src/androidTest/assets/chorus.wav +0 -0
  34. package/android/src/androidTest/assets/jfk.wav +0 -0
  35. package/android/src/androidTest/assets/osr_us_000_0010_8k.wav +0 -0
  36. package/android/src/androidTest/assets/recorder_hello_world.wav +0 -0
  37. package/android/src/androidTest/java/net/siteed/audiostudio/AudioFinalMetadataContractInstrumentedTest.kt +0 -190
  38. package/android/src/androidTest/java/net/siteed/audiostudio/AudioProcessorInstrumentedTest.kt +0 -197
  39. package/android/src/androidTest/java/net/siteed/audiostudio/AudioRecorderInstrumentedTest.kt +0 -487
  40. package/android/src/androidTest/java/net/siteed/audiostudio/AudioRecorderPerformanceInstrumentedTest.kt +0 -250
  41. package/android/src/androidTest/java/net/siteed/audiostudio/OpusRangeDecodeRegressionInstrumentedTest.kt +0 -186
  42. package/android/src/androidTest/java/net/siteed/audiostudio/integration/AudioFocusStrategyIntegrationTest.kt +0 -332
  43. package/android/src/androidTest/java/net/siteed/audiostudio/integration/BufferDurationIntegrationTest.kt +0 -324
  44. package/android/src/androidTest/java/net/siteed/audiostudio/integration/CompressedOnlyOutputTest.kt +0 -253
  45. package/android/src/androidTest/java/net/siteed/audiostudio/integration/DeviceDisconnectionFallbackTest.kt +0 -218
  46. package/android/src/androidTest/java/net/siteed/audiostudio/integration/EventEmissionIntervalTest.kt +0 -120
  47. package/android/src/androidTest/java/net/siteed/audiostudio/integration/M4aFormatTest.kt +0 -345
  48. package/android/src/androidTest/java/net/siteed/audiostudio/integration/OutputControlIntegrationTest.kt +0 -340
  49. package/android/src/androidTest/java/net/siteed/audiostudio/integration/PcmStreamingDurationTest.kt +0 -252
  50. package/android/src/androidTest/java/net/siteed/audiostudio/integration/README.md +0 -95
  51. package/android/src/androidTest/java/net/siteed/audiostudio/integration/run_integration_tests.sh +0 -43
  52. package/android/src/test/java/net/siteed/audiostudio/AndroidCallStateTest.kt +0 -37
  53. package/android/src/test/java/net/siteed/audiostudio/AndroidEventEmitterTest.kt +0 -28
  54. package/android/src/test/java/net/siteed/audiostudio/AudioFileHandlerTest.kt +0 -279
  55. package/android/src/test/java/net/siteed/audiostudio/AudioFocusStrategyTest.kt +0 -249
  56. package/android/src/test/java/net/siteed/audiostudio/AudioFormatTest.kt +0 -151
  57. package/android/src/test/java/net/siteed/audiostudio/AudioFormatUtilsTest.kt +0 -273
  58. package/android/src/test/java/net/siteed/audiostudio/DeviceDisconnectionFallbackUnitTest.kt +0 -140
  59. package/android/src/test/java/net/siteed/audiostudio/InterruptionAutoResumePolicyTest.kt +0 -49
  60. package/android/src/test/resources/chorus.wav +0 -0
  61. package/android/src/test/resources/generate_test_audio.py +0 -94
  62. package/android/src/test/resources/jfk.wav +0 -0
  63. package/android/src/test/resources/osr_us_000_0010_8k.wav +0 -0
  64. package/android/src/test/resources/recorder_hello_world.wav +0 -0
  65. package/ios/AudioStudioTests/AudioFileHandlerTests.swift +0 -338
  66. package/ios/AudioStudioTests/AudioFormatUtilsTests.swift +0 -331
  67. package/ios/AudioStudioTests/AudioStreamDecoderTests.swift +0 -128
  68. package/ios/AudioStudioTests/AudioTestHelpers.swift +0 -130
  69. package/ios/AudioStudioTests/CompressedOnlyOutputTests.swift +0 -334
  70. package/ios/AudioStudioTests/EventEmissionIntervalTests.swift +0 -105
  71. package/ios/AudioStudioTests/Info.plist +0 -22
  72. package/ios/AudioStudioTests/README.md +0 -39
  73. package/ios/AudioStudioTests/SimpleAudioTest.swift +0 -98
  74. package/ios/AudioStudioTests/TestAudioGenerator.swift +0 -75
  75. package/ios/tests/README.md +0 -41
  76. package/ios/tests/integration/buffer_and_fallback_test.swift +0 -178
  77. package/ios/tests/integration/buffer_duration_test.swift +0 -185
  78. package/ios/tests/integration/compressed_only_output_test.swift +0 -271
  79. package/ios/tests/integration/output_control_test.swift +0 -322
  80. package/ios/tests/integration/run_integration_tests.sh +0 -37
  81. package/ios/tests/opus_support_test_macos.swift +0 -154
  82. package/ios/tests/standalone/audio_processing_test.swift +0 -144
  83. package/ios/tests/standalone/audio_recording_test.swift +0 -277
  84. package/ios/tests/standalone/audio_streaming_test.swift +0 -249
  85. package/ios/tests/standalone/standalone_test.swift +0 -144
package/build/cjs/streamAudioData.js.map CHANGED
@@ -1 +1 @@
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{ LegacyEventEmitter, type EventSubscription } from 'expo-modules-core'\n\nimport AudioStudioModule from './AudioStudioModule'\nimport { isWeb } from './constants'\nimport {\n AudioStreamError,\n AudioStreamErrorCode,\n mapStreamError,\n} from './errors/AudioStreamError'\nimport { processAudioBuffer } from './utils/audioProcessing'\n\n/**\n * High-level API: stream decoded audio from a stored file as bounded Float32\n * chunks without materializing the full PCM range in memory.\n *\n * See `docs/STREAM_AUDIO_DATA.md` for the full contract and rollout notes.\n */\nexport interface StreamAudioDataOptions {\n /** URI of the audio file to decode. */\n fileUri: string\n /** Start time in milliseconds (default: 0). */\n startTimeMs?: number\n /** End time in milliseconds (default: end-of-file). */\n endTimeMs?: number\n /**\n * Source sample rate hint. Ignored if `targetSampleRate` is set; native\n * decoders read the actual rate from the file.\n */\n sampleRate?: number\n /** Output sample rate. Native resamples when this differs from the file. */\n targetSampleRate?: number\n /** Output channel count (1 = mono downmix, 2 = stereo passthrough). */\n channels?: number\n /** Clamp samples to [-1, 1] and replace non-finite values with 0. */\n normalizeAudio?: boolean\n /** Target chunk duration in ms (default: 1000, min: 10, max: 60000). */\n chunkDurationMs?: number\n /** Soft cap on chunk size in bytes (Float32 = 4 bytes/sample). */\n maxChunkBytes?: number\n /** Max chunks queued in native before JS ack pauses decode (default: 4). */\n maxBufferedChunks?: number\n /** Output PCM format; only `'float32'` supported today. */\n streamFormat?: 'float32'\n /** Abort the in-flight request. Resolves promise with `cancelled: true`. */\n signal?: AbortSignal\n}\n\nexport interface StreamAudioDataChunk {\n /** Native request id; constant across all chunks of one call. */\n requestId: string\n /** Zero-based monotonic chunk index. */\n chunkIndex: number\n /** Start time in output-rate ms (rounded to nearest sample). */\n startTimeMs: number\n /** End time in output-rate ms. */\n endTimeMs: number\n /** Duration in ms (`endTimeMs - startTimeMs`). */\n durationMs: number\n /** First sample index in the output timeline. */\n startSample: number\n /** Sample count in `samples` (interleaved if channels > 1). */\n sampleCount: number\n /** Output sample rate. */\n sampleRate: number\n /** Output channel count. */\n channels: number\n /** Interleaved Float32 samples in [-1, 1]. */\n samples: Float32Array\n /** True for the last chunk of a non-cancelled run. */\n isFinal: boolean\n}\n\nexport interface StreamAudioDataProgress {\n requestId: string\n processedMs: number\n durationMs: number\n progress: number\n emittedChunks: number\n bufferedChunks?: number\n}\n\nexport interface StreamAudioDataResult {\n requestId: string\n durationMs: number\n sampleRate: number\n channels: number\n chunks: number\n samples: number\n cancelled: boolean\n}\n\nexport interface StreamAudioDataCallbacks {\n /**\n * Called with each decoded chunk. If this returns a Promise, native decode\n * pauses until it resolves (backpressure). Throwing aborts the stream with\n * `ERR_AUDIO_STREAM_DECODE_FAILED`.\n */\n onChunk: (chunk: StreamAudioDataChunk) => void | Promise<void>\n /** Called whenever native reports progress. */\n onProgress?: (progress: StreamAudioDataProgress) => void\n}\n\nexport interface AudioDecodeCapabilities {\n platform: 'ios' | 'android' | 'web'\n supportedInputFormats: string[]\n supportedOutputFormats: Array<'float32'>\n supportsCancellation: boolean\n supportsBackpressure: boolean\n supportsTimeRange: boolean\n supportsTargetSampleRate: boolean\n supportsChannelMixing: boolean\n knownLimitations?: string[]\n}\n\nconst CHUNK_EVENT = 'AudioDataStreamChunk'\nconst PROGRESS_EVENT = 'AudioDataStreamProgress'\nconst COMPLETE_EVENT = 'AudioDataStreamComplete'\nconst ERROR_EVENT = 'AudioDataStreamError'\n\nlet cachedEmitter: LegacyEventEmitter | null = null\nfunction getEmitter(): LegacyEventEmitter {\n if (!cachedEmitter) {\n cachedEmitter = new LegacyEventEmitter(AudioStudioModule)\n }\n return cachedEmitter\n}\n\nfunction generateRequestId(): string {\n const g = globalThis as { crypto?: { randomUUID?: () => string } }\n if (typeof g.crypto?.randomUUID === 'function') {\n try {\n return g.crypto.randomUUID()\n } catch {\n // fall through\n }\n }\n return `asd_${Date.now().toString(36)}_${Math.random()\n .toString(36)\n .slice(2, 10)}`\n}\n\nfunction toFloat32(samples: unknown): Float32Array {\n if (samples instanceof Float32Array) return samples\n if (Array.isArray(samples)) {\n const out = new Float32Array(samples.length)\n for (let i = 0; i < samples.length; i++) {\n const v = Number(samples[i])\n out[i] = Number.isFinite(v) ? v : 0\n }\n return out\n }\n if (typeof samples === 'string') {\n const bytes = base64ToBytes(samples)\n const aligned =\n bytes.byteOffset % 4 === 0\n ? new Float32Array(\n bytes.buffer,\n bytes.byteOffset,\n bytes.byteLength / 4\n )\n : new Float32Array(bytes.buffer.slice(bytes.byteOffset))\n return new Float32Array(aligned)\n }\n if (samples && typeof samples === 'object' && 'length' in samples) {\n // ArrayLike fallback\n const arr = samples as ArrayLike<number>\n const out = new Float32Array(arr.length)\n for (let i = 0; i < arr.length; i++) {\n const v = Number(arr[i])\n out[i] = Number.isFinite(v) ? v : 0\n }\n return out\n }\n return new Float32Array(0)\n}\n\nfunction base64ToBytes(input: string): Uint8Array {\n const g = globalThis as { atob?: (s: string) => string }\n if (typeof g.atob !== 'function') {\n // Buffer path for environments without atob; React Native has atob.\n const Buf = (globalThis as { Buffer?: { from: Function } }).Buffer\n if (Buf) return new Uint8Array(Buf.from(input, 'base64'))\n return new Uint8Array(0)\n }\n const bin = g.atob(input)\n const out = new Uint8Array(bin.length)\n for (let i = 0; i < bin.length; i++) out[i] = bin.charCodeAt(i)\n return out\n}\n\nfunction validateOptions(options: StreamAudioDataOptions): void {\n if (!options.fileUri) {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n message: 'fileUri is required',\n recoverable: false,\n })\n }\n if (\n options.startTimeMs !== undefined &&\n options.endTimeMs !== undefined &&\n options.startTimeMs >= options.endTimeMs\n ) {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n message: 'startTimeMs must be < endTimeMs',\n recoverable: false,\n })\n }\n if (\n options.chunkDurationMs !== undefined &&\n (options.chunkDurationMs < 10 || options.chunkDurationMs > 60000)\n ) {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n message: 'chunkDurationMs must be in [10, 60000]',\n recoverable: false,\n })\n }\n if (\n options.streamFormat !== undefined &&\n options.streamFormat !== 'float32'\n ) {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',\n message: `Unsupported streamFormat: ${options.streamFormat}`,\n recoverable: false,\n })\n }\n}\n\n/**\n * Stream decoded audio from a stored file as bounded Float32 PCM chunks.\n *\n * Memory bound:\n * `chunkDurationMs * sampleRate * channels * 4 * maxBufferedChunks` +\n * native decoder buffers.\n *\n * Cancellation: pass `options.signal` and call `abort()`. The returned promise\n * resolves with `cancelled: true` (it does not reject) when cancellation wins.\n *\n * Backpressure: if `onChunk` returns a Promise, native decode is paused until\n * it resolves; if it throws, the stream is aborted with a `decode_failed` error.\n */\nexport async function streamAudioData(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n validateOptions(options)\n\n if (isWeb) {\n return streamAudioDataWeb(options, callbacks)\n }\n\n return streamAudioDataNative(options, callbacks)\n}\n\n/** Discover what the running platform supports. */\nexport async function getAudioDecodeCapabilities(): Promise<AudioDecodeCapabilities> {\n if (isWeb) {\n return {\n platform: 'web',\n supportedInputFormats: [\n 'audio/wav',\n 'audio/mpeg',\n 'audio/mp4',\n 'audio/aac',\n 'audio/ogg',\n 'audio/webm',\n ],\n supportedOutputFormats: ['float32'],\n supportsCancellation: true,\n supportsBackpressure: true,\n supportsTimeRange: true,\n supportsTargetSampleRate: true,\n supportsChannelMixing: true,\n knownLimitations: [\n 'Web decodes the entire file via AudioContext.decodeAudioData before chunking; very long files may exceed browser memory.',\n ],\n }\n }\n if (typeof AudioStudioModule.getAudioDecodeCapabilities === 'function') {\n try {\n const caps = await AudioStudioModule.getAudioDecodeCapabilities()\n return caps as AudioDecodeCapabilities\n } catch (err) {\n throw mapStreamError(err, undefined, 'native')\n }\n }\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n message: 'getAudioDecodeCapabilities is not available on this build',\n recoverable: false,\n })\n}\n\nasync function streamAudioDataNative(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n if (typeof AudioStudioModule.streamAudioData !== 'function') {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n message:\n 'streamAudioData native method missing — rebuild the host app with the latest @siteed/audio-studio.',\n recoverable: false,\n })\n }\n\n const requestId = generateRequestId()\n const emitter = getEmitter()\n const subs: EventSubscription[] = []\n let chunkCount = 0\n let sampleCount = 0\n let processingChain: Promise<void> = Promise.resolve()\n let settled = false\n let abortListener: (() => void) | null = null\n\n const finalize = () => {\n for (const sub of subs) {\n try {\n sub.remove()\n } catch {\n /* noop */\n }\n }\n subs.length = 0\n if (abortListener && options.signal) {\n options.signal.removeEventListener('abort', abortListener)\n }\n }\n\n return new Promise<StreamAudioDataResult>((resolve, reject) => {\n const settle = (\n fn: () => void,\n mode: 'resolve' | 'reject',\n value: unknown\n ) => {\n if (settled) return\n settled = true\n finalize()\n fn()\n if (mode === 'resolve') {\n resolve(value as StreamAudioDataResult)\n } else {\n reject(value as Error)\n }\n }\n\n const handleError = (err: unknown) => {\n settle(\n () => {\n try {\n AudioStudioModule.cancelStreamAudioData?.(requestId)\n } catch {\n /* noop */\n }\n },\n 'reject',\n mapStreamError(err, options.fileUri, 'native')\n )\n }\n\n subs.push(\n emitter.addListener(CHUNK_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n chunkIndex: number\n startTimeMs: number\n endTimeMs: number\n startSample: number\n sampleCount: number\n sampleRate: number\n channels: number\n samples: Float32Array | number[] | string\n isFinal: boolean\n }\n if (evt.requestId !== requestId) return\n const chunk: StreamAudioDataChunk = {\n requestId: evt.requestId,\n chunkIndex: evt.chunkIndex,\n startTimeMs: evt.startTimeMs,\n endTimeMs: evt.endTimeMs,\n durationMs: evt.endTimeMs - evt.startTimeMs,\n startSample: evt.startSample,\n sampleCount: evt.sampleCount,\n sampleRate: evt.sampleRate,\n channels: evt.channels,\n samples: toFloat32(evt.samples),\n isFinal: evt.isFinal,\n }\n chunkCount += 1\n sampleCount += chunk.sampleCount\n\n processingChain = processingChain\n .then(async () => {\n await callbacks.onChunk(chunk)\n try {\n AudioStudioModule.acknowledgeStreamAudioChunk?.(\n requestId,\n chunk.chunkIndex\n )\n } catch {\n /* noop */\n }\n })\n .catch(handleError)\n })\n )\n\n if (callbacks.onProgress) {\n subs.push(\n emitter.addListener(PROGRESS_EVENT, (raw: unknown) => {\n const evt = raw as StreamAudioDataProgress\n if (evt.requestId !== requestId) return\n callbacks.onProgress!(evt)\n })\n )\n }\n\n subs.push(\n emitter.addListener(COMPLETE_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n durationMs: number\n sampleRate: number\n channels: number\n chunks?: number\n samples?: number\n cancelled: boolean\n }\n if (evt.requestId !== requestId) return\n // Wait for in-flight onChunk callbacks before resolving.\n processingChain\n .then(() => {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs: evt.durationMs,\n sampleRate: evt.sampleRate,\n channels: evt.channels,\n chunks: evt.chunks ?? chunkCount,\n samples: evt.samples ?? sampleCount,\n cancelled: evt.cancelled,\n } satisfies StreamAudioDataResult)\n })\n .catch(handleError)\n })\n )\n\n subs.push(\n emitter.addListener(ERROR_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n code?: string\n message?: string\n nativeMessage?: string\n }\n if (evt.requestId !== requestId) return\n if (evt.code === 'ERR_AUDIO_STREAM_CANCELLED') {\n processingChain\n .catch(() => {})\n .then(() => {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs: 0,\n sampleRate:\n options.targetSampleRate ??\n options.sampleRate ??\n 0,\n channels: options.channels ?? 1,\n chunks: chunkCount,\n samples: sampleCount,\n cancelled: true,\n } satisfies StreamAudioDataResult)\n })\n return\n }\n handleError(\n new AudioStreamError({\n code:\n (evt.code as AudioStreamErrorCode) ??\n 'ERR_AUDIO_STREAM_UNKNOWN',\n message: evt.message ?? 'native stream error',\n nativeMessage: evt.nativeMessage,\n fileUri: options.fileUri,\n platform: 'native',\n recoverable: false,\n })\n )\n })\n )\n\n if (options.signal) {\n if (options.signal.aborted) {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs: 0,\n sampleRate:\n options.targetSampleRate ?? options.sampleRate ?? 0,\n channels: options.channels ?? 1,\n chunks: 0,\n samples: 0,\n cancelled: true,\n } satisfies StreamAudioDataResult)\n return\n }\n abortListener = () => {\n try {\n AudioStudioModule.cancelStreamAudioData?.(requestId)\n } catch {\n /* noop */\n }\n }\n options.signal.addEventListener('abort', abortListener)\n }\n\n const { signal: _signal, ...nativeOptions } = options\n AudioStudioModule.streamAudioData({\n ...nativeOptions,\n requestId,\n streamFormat: options.streamFormat ?? 'float32',\n chunkDurationMs: options.chunkDurationMs ?? 1000,\n maxBufferedChunks: options.maxBufferedChunks ?? 4,\n normalizeAudio: options.normalizeAudio ?? true,\n }).catch(handleError)\n })\n}\n\nasync function streamAudioDataWeb(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n const requestId = generateRequestId()\n const cancelled = () => options.signal?.aborted === true\n if (cancelled()) {\n return {\n requestId,\n durationMs: 0,\n sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,\n channels: options.channels ?? 1,\n chunks: 0,\n samples: 0,\n cancelled: true,\n }\n }\n\n try {\n const processed = await processAudioBuffer({\n fileUri: options.fileUri,\n targetSampleRate: options.targetSampleRate ?? 16000,\n targetChannels: options.channels ?? 1,\n normalizeAudio: options.normalizeAudio ?? true,\n startTimeMs: options.startTimeMs,\n endTimeMs: options.endTimeMs,\n })\n\n const sampleRate = processed.sampleRate\n const channels = processed.channels\n const durationMs = processed.durationMs\n const chunkDurationMs = options.chunkDurationMs ?? 1000\n let samplesPerChunk = Math.max(\n 1,\n Math.floor((chunkDurationMs / 1000) * sampleRate) * channels\n )\n if (options.maxChunkBytes) {\n const maxSamples = Math.floor(options.maxChunkBytes / 4)\n samplesPerChunk = Math.min(samplesPerChunk, maxSamples)\n }\n\n const all = sanitizeFloat32(processed.channelData, options.normalizeAudio ?? true)\n\n let chunkIndex = 0\n let emittedSamples = 0\n for (let off = 0; off < all.length; off += samplesPerChunk) {\n if (cancelled()) {\n return {\n requestId,\n durationMs,\n sampleRate,\n channels,\n chunks: chunkIndex,\n samples: emittedSamples,\n cancelled: true,\n }\n }\n const end = Math.min(off + samplesPerChunk, all.length)\n const slice = all.slice(off, end)\n const startSample = off / channels\n const endSample = end / channels\n const chunk: StreamAudioDataChunk = {\n requestId,\n chunkIndex,\n startTimeMs: Math.round((startSample / sampleRate) * 1000),\n endTimeMs: Math.round((endSample / sampleRate) * 1000),\n durationMs: Math.round(\n ((endSample - startSample) / sampleRate) * 1000\n ),\n startSample,\n sampleCount: slice.length,\n sampleRate,\n channels,\n samples: slice,\n isFinal: end >= all.length,\n }\n await callbacks.onChunk(chunk)\n callbacks.onProgress?.({\n requestId,\n processedMs: chunk.endTimeMs,\n durationMs,\n progress: durationMs > 0 ? chunk.endTimeMs / durationMs : 1,\n emittedChunks: chunkIndex + 1,\n })\n chunkIndex += 1\n emittedSamples += slice.length\n }\n\n return {\n requestId,\n durationMs,\n sampleRate,\n channels,\n chunks: chunkIndex,\n samples: emittedSamples,\n cancelled: false,\n }\n } catch (err) {\n throw mapStreamError(err, options.fileUri, 'web')\n }\n}\n\nfunction sanitizeFloat32(\n input: Float32Array,\n clamp: boolean\n): Float32Array {\n if (!clamp) {\n // still need NaN/Inf sanitation\n for (let i = 0; i < input.length; i++) {\n const v = input[i]\n if (!Number.isFinite(v)) input[i] = 0\n }\n return input\n }\n for (let i = 0; i < input.length; i++) {\n const v = input[i]\n if (!Number.isFinite(v)) {\n input[i] = 0\n } else if (v > 1) {\n input[i] = 1\n } else if (v < -1) {\n input[i] = -1\n }\n }\n return input\n}\n"]}
1
+ 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Ignored if `targetSampleRate` is set; native\n * decoders read the actual rate from the file.\n */\n sampleRate?: number\n /** Output sample rate. Native resamples when this differs from the file. */\n targetSampleRate?: number\n /** Output channel count (1 = mono downmix, 2 = stereo passthrough). */\n channels?: number\n /** Clamp samples to [-1, 1] and replace non-finite values with 0. */\n normalizeAudio?: boolean\n /** Target chunk duration in ms (default: 1000, min: 10, max: 60000). */\n chunkDurationMs?: number\n /** Soft cap on chunk size in bytes (Float32 = 4 bytes/sample). */\n maxChunkBytes?: number\n /** Max chunks queued in native before JS ack pauses decode (default: 4). */\n maxBufferedChunks?: number\n /**\n * Optional timeout for a chunk acknowledgement while backpressure is active.\n * Undefined/0 disables timeout so long transcription callbacks can run.\n */\n backpressureTimeoutMs?: number\n /** Output PCM format; only `'float32'` supported today. */\n streamFormat?: 'float32'\n /** Abort the in-flight request. Resolves promise with `cancelled: true`. */\n signal?: AbortSignal\n}\n\nexport interface StreamAudioDataChunk {\n /** Native request id; constant across all chunks of one call. */\n requestId: string\n /** Zero-based monotonic chunk index. */\n chunkIndex: number\n /** Start time in output-rate ms (rounded to nearest sample). */\n startTimeMs: number\n /** End time in output-rate ms. */\n endTimeMs: number\n /** Duration in ms (`endTimeMs - startTimeMs`). */\n durationMs: number\n /** First sample index in the output timeline. */\n startSample: number\n /** Sample count in `samples` (interleaved if channels > 1). */\n sampleCount: number\n /** Output sample rate. */\n sampleRate: number\n /** Output channel count. */\n channels: number\n /** Interleaved Float32 samples in [-1, 1]. */\n samples: Float32Array\n /** True for the last chunk of a non-cancelled run. */\n isFinal: boolean\n}\n\nexport interface StreamAudioDataProgress {\n requestId: string\n processedMs: number\n durationMs: number\n progress: number\n emittedChunks: number\n bufferedChunks?: number\n}\n\nexport interface StreamAudioDataResult {\n requestId: string\n durationMs: number\n sampleRate: number\n channels: number\n chunks: number\n samples: number\n cancelled: boolean\n}\n\nexport interface StreamAudioDataCallbacks {\n /**\n * Called with each decoded chunk. If this returns a Promise, native decode\n * pauses until it resolves (backpressure). Throwing aborts the stream with\n * `ERR_AUDIO_STREAM_DECODE_FAILED`.\n */\n onChunk: (chunk: StreamAudioDataChunk) => void | Promise<void>\n /** Called whenever native reports progress. */\n onProgress?: (progress: StreamAudioDataProgress) => void\n}\n\nexport interface AudioDecodeCapabilities {\n platform: 'ios' | 'android' | 'web'\n supportedInputFormats: string[]\n supportedOutputFormats: Array<'float32'>\n supportsCancellation: boolean\n supportsBackpressure: boolean\n supportsTimeRange: boolean\n supportsTargetSampleRate: boolean\n supportsChannelMixing: boolean\n knownLimitations?: string[]\n}\n\nconst CHUNK_EVENT = 'AudioDataStreamChunk'\nconst PROGRESS_EVENT = 'AudioDataStreamProgress'\nconst COMPLETE_EVENT = 'AudioDataStreamComplete'\nconst ERROR_EVENT = 'AudioDataStreamError'\n\nlet cachedEmitter: LegacyEventEmitter | null = null\nfunction getEmitter(): LegacyEventEmitter {\n if (!cachedEmitter) {\n cachedEmitter = new LegacyEventEmitter(AudioStudioModule)\n }\n return cachedEmitter\n}\n\nfunction generateRequestId(): string {\n const g = globalThis as { crypto?: { randomUUID?: () => string } }\n if (typeof g.crypto?.randomUUID === 'function') {\n try {\n return g.crypto.randomUUID()\n } catch {\n // fall through\n }\n }\n return `asd_${Date.now().toString(36)}_${Math.random()\n .toString(36)\n .slice(2, 10)}`\n}\n\nfunction toFloat32(samples: unknown): Float32Array {\n if (samples instanceof Float32Array) return samples\n if (Array.isArray(samples)) {\n const out = new Float32Array(samples.length)\n for (let i = 0; i < samples.length; i++) {\n const v = Number(samples[i])\n out[i] = Number.isFinite(v) ? v : 0\n }\n return out\n }\n if (typeof samples === 'string') {\n const bytes = base64ToBytes(samples)\n const floatLength = Math.floor(bytes.byteLength / 4)\n if (bytes.byteOffset % 4 === 0) {\n return new Float32Array(bytes.buffer, bytes.byteOffset, floatLength)\n }\n const sliced = bytes.buffer.slice(\n bytes.byteOffset,\n bytes.byteOffset + bytes.byteLength\n )\n return new Float32Array(sliced, 0, Math.floor(sliced.byteLength / 4))\n }\n if (samples && typeof samples === 'object' && 'length' in samples) {\n // ArrayLike fallback\n const arr = samples as ArrayLike<number>\n const out = new Float32Array(arr.length)\n for (let i = 0; i < arr.length; i++) {\n const v = Number(arr[i])\n out[i] = Number.isFinite(v) ? v : 0\n }\n return out\n }\n return new Float32Array(0)\n}\n\nfunction base64ToBytes(input: string): Uint8Array {\n const g = globalThis as { atob?: (s: string) => string }\n if (typeof g.atob !== 'function') {\n // Buffer path for environments without atob; React Native has atob.\n const Buf = (\n globalThis as {\n Buffer?: {\n from: (input: string, encoding: string) => Uint8Array\n }\n }\n ).Buffer\n if (Buf) return new Uint8Array(Buf.from(input, 'base64'))\n return new Uint8Array(0)\n }\n const bin = g.atob(input)\n const out = new Uint8Array(bin.length)\n for (let i = 0; i < bin.length; i++) out[i] = bin.charCodeAt(i)\n return out\n}\n\nfunction rejectInvalidRange(message: string): never {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n message,\n recoverable: false,\n })\n}\n\nfunction assertPositiveFiniteOption(\n value: number | undefined,\n name: string,\n integer = false\n): void {\n if (value === undefined) return\n if (\n !Number.isFinite(value) ||\n value <= 0 ||\n (integer && !Number.isInteger(value))\n ) {\n rejectInvalidRange(\n `${name} must be a positive${integer ? ' integer' : ''}`\n )\n }\n}\n\nfunction validateOptions(options: StreamAudioDataOptions): void {\n if (!options.fileUri) {\n rejectInvalidRange('fileUri is required')\n }\n if (\n options.startTimeMs !== undefined &&\n options.endTimeMs !== undefined &&\n options.startTimeMs >= options.endTimeMs\n ) {\n rejectInvalidRange('startTimeMs must be < endTimeMs')\n }\n if (options.endTimeMs !== undefined && options.endTimeMs <= 0) {\n rejectInvalidRange('endTimeMs must be > 0')\n }\n if (options.startTimeMs !== undefined && options.startTimeMs < 0) {\n rejectInvalidRange('startTimeMs must be >= 0')\n }\n if (\n options.chunkDurationMs !== undefined &&\n (options.chunkDurationMs < 10 || options.chunkDurationMs > 60000)\n ) {\n rejectInvalidRange('chunkDurationMs must be in [10, 60000]')\n }\n if (\n options.backpressureTimeoutMs !== undefined &&\n options.backpressureTimeoutMs < 0\n ) {\n rejectInvalidRange('backpressureTimeoutMs must be >= 0')\n }\n assertPositiveFiniteOption(options.targetSampleRate, 'targetSampleRate')\n assertPositiveFiniteOption(options.sampleRate, 'sampleRate')\n assertPositiveFiniteOption(options.channels, 'channels', true)\n assertPositiveFiniteOption(\n options.maxBufferedChunks,\n 'maxBufferedChunks',\n true\n )\n assertPositiveFiniteOption(options.maxChunkBytes, 'maxChunkBytes', true)\n\n if (\n options.streamFormat !== undefined &&\n options.streamFormat !== 'float32'\n ) {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',\n message: `Unsupported streamFormat: ${options.streamFormat}`,\n recoverable: false,\n })\n }\n}\n\n/**\n * Stream decoded audio from a stored file as bounded Float32 PCM chunks.\n *\n * Memory bound:\n * `chunkDurationMs * sampleRate * channels * 4 * maxBufferedChunks` +\n * native decoder buffers.\n *\n * Cancellation: pass `options.signal` and call `abort()`. The returned promise\n * resolves with `cancelled: true` (it does not reject) when cancellation wins.\n *\n * Backpressure: if `onChunk` returns a Promise, native decode is paused until\n * it resolves; if it throws, the stream is aborted with a `decode_failed` error.\n */\nexport async function streamAudioData(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n validateOptions(options)\n\n if (isWeb) {\n return streamAudioDataWeb(options, callbacks)\n }\n\n return streamAudioDataNative(options, callbacks)\n}\n\n/** Discover what the running platform supports. */\nexport async function getAudioDecodeCapabilities(): Promise<AudioDecodeCapabilities> {\n if (isWeb) {\n return {\n platform: 'web',\n supportedInputFormats: [\n 'audio/wav',\n 'audio/mpeg',\n 'audio/mp4',\n 'audio/aac',\n 'audio/ogg',\n 'audio/webm',\n ],\n supportedOutputFormats: ['float32'],\n supportsCancellation: true,\n supportsBackpressure: true,\n supportsTimeRange: true,\n supportsTargetSampleRate: true,\n supportsChannelMixing: true,\n knownLimitations: [\n 'Web decodes the entire file via AudioContext.decodeAudioData before chunking; very long files may exceed browser memory.',\n ],\n }\n }\n if (typeof AudioStudioModule.getAudioDecodeCapabilities === 'function') {\n try {\n const caps = await AudioStudioModule.getAudioDecodeCapabilities()\n return caps as AudioDecodeCapabilities\n } catch (err) {\n throw mapStreamError(err, undefined, 'native')\n }\n }\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n message: 'getAudioDecodeCapabilities is not available on this build',\n recoverable: false,\n })\n}\n\nasync function streamAudioDataNative(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n if (typeof AudioStudioModule.streamAudioData !== 'function') {\n throw new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n message:\n 'streamAudioData native method missing — rebuild the host app with the latest @siteed/audio-studio.',\n recoverable: false,\n })\n }\n\n const requestId = generateRequestId()\n const emitter = getEmitter()\n const subs: EventSubscription[] = []\n let chunkCount = 0\n let sampleCount = 0\n let processingChain: Promise<void> = Promise.resolve()\n let settled = false\n let abortListener: (() => void) | null = null\n let lastProgress: StreamAudioDataProgress | null = null\n\n const finalize = () => {\n for (const sub of subs) {\n try {\n sub.remove()\n } catch {\n /* noop */\n }\n }\n subs.length = 0\n if (abortListener && options.signal) {\n options.signal.removeEventListener('abort', abortListener)\n }\n }\n\n return new Promise<StreamAudioDataResult>((resolve, reject) => {\n const settle = (\n fn: () => void,\n mode: 'resolve' | 'reject',\n value: unknown\n ) => {\n if (settled) return\n settled = true\n finalize()\n fn()\n if (mode === 'resolve') {\n resolve(value as StreamAudioDataResult)\n } else {\n reject(value as Error)\n }\n }\n\n const handleError = (err: unknown) => {\n settle(\n () => {\n try {\n AudioStudioModule.cancelStreamAudioData?.(requestId)\n } catch {\n /* noop */\n }\n },\n 'reject',\n mapStreamError(err, options.fileUri, 'native')\n )\n }\n\n subs.push(\n emitter.addListener(CHUNK_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n chunkIndex: number\n startTimeMs: number\n endTimeMs: number\n startSample: number\n sampleCount: number\n sampleRate: number\n channels: number\n samples: Float32Array | number[] | string\n isFinal: boolean\n }\n if (evt.requestId !== requestId) return\n const chunk: StreamAudioDataChunk = {\n requestId: evt.requestId,\n chunkIndex: evt.chunkIndex,\n startTimeMs: evt.startTimeMs,\n endTimeMs: evt.endTimeMs,\n durationMs: evt.endTimeMs - evt.startTimeMs,\n startSample: evt.startSample,\n sampleCount: evt.sampleCount,\n sampleRate: evt.sampleRate,\n channels: evt.channels,\n samples: toFloat32(evt.samples),\n isFinal: evt.isFinal,\n }\n chunkCount += 1\n sampleCount += chunk.sampleCount\n\n processingChain = processingChain\n .then(async () => {\n await callbacks.onChunk(chunk)\n try {\n AudioStudioModule.acknowledgeStreamAudioChunk?.(\n requestId,\n chunk.chunkIndex\n )\n } catch {\n /* noop */\n }\n })\n .catch(handleError)\n })\n )\n\n if (callbacks.onProgress) {\n subs.push(\n emitter.addListener(PROGRESS_EVENT, (raw: unknown) => {\n const evt = raw as StreamAudioDataProgress\n if (evt.requestId !== requestId) return\n lastProgress = evt\n callbacks.onProgress!(evt)\n })\n )\n }\n\n subs.push(\n emitter.addListener(COMPLETE_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n durationMs: number\n sampleRate: number\n channels: number\n chunks?: number\n samples?: number\n cancelled: boolean\n }\n if (evt.requestId !== requestId) return\n // Wait for in-flight onChunk callbacks before resolving.\n processingChain\n .then(() => {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs:\n evt.durationMs > 0\n ? evt.durationMs\n : (lastProgress?.durationMs ?? 0),\n sampleRate: evt.sampleRate,\n channels: evt.channels,\n chunks: evt.chunks ?? chunkCount,\n samples: evt.samples ?? sampleCount,\n cancelled: evt.cancelled,\n } satisfies StreamAudioDataResult)\n })\n .catch(handleError)\n })\n )\n\n subs.push(\n emitter.addListener(ERROR_EVENT, (raw: unknown) => {\n const evt = raw as {\n requestId: string\n code?: string\n message?: string\n nativeMessage?: string\n }\n if (evt.requestId !== requestId) return\n if (evt.code === 'ERR_AUDIO_STREAM_CANCELLED') {\n processingChain\n .catch(() => {})\n .then(() => {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs: lastProgress?.durationMs ?? 0,\n sampleRate:\n options.targetSampleRate ??\n options.sampleRate ??\n 0,\n channels: options.channels ?? 1,\n chunks: chunkCount,\n samples: sampleCount,\n cancelled: true,\n } satisfies StreamAudioDataResult)\n })\n return\n }\n handleError(\n new AudioStreamError({\n code:\n (evt.code as AudioStreamErrorCode) ??\n 'ERR_AUDIO_STREAM_UNKNOWN',\n message: evt.message ?? 'native stream error',\n nativeMessage: evt.nativeMessage,\n fileUri: options.fileUri,\n platform: 'native',\n recoverable: false,\n })\n )\n })\n )\n\n if (options.signal) {\n if (options.signal.aborted) {\n settle(() => {}, 'resolve', {\n requestId,\n durationMs: lastProgress?.durationMs ?? 0,\n sampleRate:\n options.targetSampleRate ?? options.sampleRate ?? 0,\n channels: options.channels ?? 1,\n chunks: 0,\n samples: 0,\n cancelled: true,\n } satisfies StreamAudioDataResult)\n return\n }\n abortListener = () => {\n try {\n AudioStudioModule.cancelStreamAudioData?.(requestId)\n } catch {\n /* noop */\n }\n }\n options.signal.addEventListener('abort', abortListener)\n }\n\n const { signal: _signal, ...nativeOptions } = options\n AudioStudioModule.streamAudioData({\n ...nativeOptions,\n requestId,\n streamFormat: options.streamFormat ?? 'float32',\n chunkDurationMs: options.chunkDurationMs ?? 1000,\n maxBufferedChunks: options.maxBufferedChunks ?? 4,\n normalizeAudio: options.normalizeAudio ?? true,\n }).catch(handleError)\n })\n}\n\nasync function streamAudioDataWeb(\n options: StreamAudioDataOptions,\n callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n const requestId = generateRequestId()\n const cancelled = () => options.signal?.aborted === true\n if (cancelled()) {\n return {\n requestId,\n durationMs: 0,\n sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,\n channels: options.channels ?? 1,\n chunks: 0,\n samples: 0,\n cancelled: true,\n }\n }\n\n try {\n const processed = await processAudioBuffer({\n fileUri: options.fileUri,\n targetSampleRate: options.targetSampleRate,\n targetChannels: options.channels,\n normalizeAudio: options.normalizeAudio ?? true,\n startTimeMs: options.startTimeMs,\n endTimeMs: options.endTimeMs,\n })\n\n const sampleRate = processed.sampleRate\n const channels = processed.channels\n const durationMs = processed.durationMs\n const chunkDurationMs = options.chunkDurationMs ?? 1000\n let samplesPerChunk = Math.max(\n channels,\n Math.floor((chunkDurationMs / 1000) * sampleRate) * channels\n )\n if (options.maxChunkBytes) {\n // Round down to a multiple of `channels` so we never split an\n // interleaved frame across two chunks (that would produce a\n // fractional `startSample` for the next chunk).\n const rawMax = Math.floor(options.maxChunkBytes / 4)\n const maxSamples = Math.max(\n channels,\n Math.floor(rawMax / channels) * channels\n )\n samplesPerChunk = Math.max(\n channels,\n Math.min(samplesPerChunk, maxSamples)\n )\n }\n\n const all = sanitizeFloat32(\n interleaveBuffer(processed.buffer, channels),\n options.normalizeAudio ?? true\n )\n\n // Chunk timestamps are absolute (range start + offset) on every\n // platform; progress is *elapsed within the range* so the\n // `processedMs / durationMs` fraction stays in [0, 1] regardless of\n // `startTimeMs`. The native decoders use the same split.\n const rangeStartMs = options.startTimeMs ?? 0\n let chunkIndex = 0\n let emittedSamples = 0\n for (let off = 0; off < all.length; off += samplesPerChunk) {\n if (cancelled()) {\n return {\n requestId,\n durationMs,\n sampleRate,\n channels,\n chunks: chunkIndex,\n samples: emittedSamples,\n cancelled: true,\n }\n }\n const end = Math.min(off + samplesPerChunk, all.length)\n const slice = all.slice(off, end)\n const startSample = off / channels\n const endSample = end / channels\n const startMs =\n Math.round((startSample / sampleRate) * 1000) + rangeStartMs\n const endMs =\n Math.round((endSample / sampleRate) * 1000) + rangeStartMs\n const chunk: StreamAudioDataChunk = {\n requestId,\n chunkIndex,\n startTimeMs: startMs,\n endTimeMs: endMs,\n durationMs: Math.round(\n ((endSample - startSample) / sampleRate) * 1000\n ),\n startSample,\n sampleCount: slice.length,\n sampleRate,\n channels,\n samples: slice,\n isFinal: end >= all.length,\n }\n await callbacks.onChunk(chunk)\n // Resample rounding (Math.ceil in processAudioBuffer) can push\n // elapsed past the source-rate-derived range duration on the tail\n // chunk. Cap so onProgress consumers always see a [0, 1] ratio,\n // matching the native `coerceIn(0, 1)` / `min(1, max(0, …))`\n // clamp.\n const rawElapsedMs = Math.round((endSample / sampleRate) * 1000)\n const elapsedMs =\n durationMs > 0\n ? Math.min(rawElapsedMs, durationMs)\n : rawElapsedMs\n callbacks.onProgress?.({\n requestId,\n processedMs: elapsedMs,\n durationMs,\n progress:\n durationMs > 0\n ? Math.min(1, Math.max(0, elapsedMs / durationMs))\n : 1,\n emittedChunks: chunkIndex + 1,\n })\n chunkIndex += 1\n emittedSamples += slice.length\n }\n\n return {\n requestId,\n durationMs,\n sampleRate,\n channels,\n chunks: chunkIndex,\n samples: emittedSamples,\n cancelled: false,\n }\n } catch (err) {\n throw mapStreamError(err, options.fileUri, 'web')\n }\n}\n\nfunction interleaveBuffer(buffer: AudioBuffer, channels: number): Float32Array {\n const numCh = Math.max(1, Math.min(channels, buffer.numberOfChannels))\n const framesPerCh = buffer.length\n if (numCh === 1) {\n // Cheap path: clone channel 0 so downstream mutation doesn't touch the\n // underlying AudioBuffer storage.\n return new Float32Array(buffer.getChannelData(0))\n }\n const out = new Float32Array(framesPerCh * numCh)\n const channelData: Float32Array[] = []\n for (let c = 0; c < numCh; c++) {\n channelData.push(buffer.getChannelData(c))\n }\n for (let f = 0; f < framesPerCh; f++) {\n for (let c = 0; c < numCh; c++) {\n out[f * numCh + c] = channelData[c][f]\n }\n }\n return out\n}\n\nfunction sanitizeFloat32(input: Float32Array, clamp: boolean): Float32Array {\n if (!clamp) {\n // still need NaN/Inf sanitation\n for (let i = 0; i < input.length; i++) {\n const v = input[i]\n if (!Number.isFinite(v)) input[i] = 0\n }\n return input\n }\n for (let i = 0; i < input.length; i++) {\n const v = input[i]\n if (!Number.isFinite(v)) {\n input[i] = 0\n } else if (v > 1) {\n input[i] = 1\n } else if (v < -1) {\n input[i] = -1\n }\n }\n return input\n}\n"]}
package/build/cjs/utils/audioProcessing.js CHANGED
@@ -44,8 +44,11 @@ async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targ
44
44
  // Create context at original sample rate first
45
45
  ctx =
46
46
  audioContext ||
47
- new (window.AudioContext || window.webkitAudioContext)();
47
+ new (window.AudioContext ||
48
+ window.webkitAudioContext)();
48
49
  buffer = await ctx.decodeAudioData(audioData);
50
+ const effectiveTargetSampleRate = targetSampleRate ?? buffer.sampleRate;
51
+ const effectiveTargetChannels = targetChannels ?? buffer.numberOfChannels;
49
52
  logger?.debug('Decoded audio buffer:', {
50
53
  originalChannels: buffer.numberOfChannels,
51
54
  originalSampleRate: buffer.sampleRate,
@@ -63,21 +66,22 @@ async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targ
63
66
  const bytesPerSample = 2; // 16-bit audio = 2 bytes per sample
64
67
  const adjustedStartSample = position !== undefined
65
68
  ? Math.floor((position / bytesPerSample) *
66
- (buffer.sampleRate / targetSampleRate))
69
+ (buffer.sampleRate / effectiveTargetSampleRate))
67
70
  : startSample;
68
71
  const samplesNeeded = length !== undefined
69
72
  ? Math.floor((length / bytesPerSample) *
70
- (buffer.sampleRate / targetSampleRate))
71
- : endTimeMs !== undefined && startTimeMs !== undefined
72
- ? Math.floor(((endTimeMs - startTimeMs) / 1000) * buffer.sampleRate)
73
+ (buffer.sampleRate / effectiveTargetSampleRate))
74
+ : endTimeMs !== undefined
75
+ ? Math.floor(((endTimeMs - (startTimeMs ?? 0)) / 1000) *
76
+ buffer.sampleRate)
73
77
  : buffer.length - adjustedStartSample;
74
78
  logger?.debug('Sample calculations (adjusted):', {
75
79
  originalStartSample: startSample,
76
80
  adjustedStartSample,
77
81
  samplesNeeded,
78
82
  originalSampleRate: buffer.sampleRate,
79
- targetSampleRate,
80
- conversionRatio: buffer.sampleRate / targetSampleRate,
83
+ targetSampleRate: effectiveTargetSampleRate,
84
+ conversionRatio: buffer.sampleRate / effectiveTargetSampleRate,
81
85
  expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,
82
86
  });
83
87
  // Create temporary buffer for the segment
@@ -91,7 +95,7 @@ async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targ
91
95
  }
92
96
  }
93
97
  // Create offline context for resampling
94
- const offlineCtx = new OfflineAudioContext(targetChannels, Math.ceil((samplesNeeded * targetSampleRate) / buffer.sampleRate), targetSampleRate);
98
+ const offlineCtx = new OfflineAudioContext(effectiveTargetChannels, Math.ceil((samplesNeeded * effectiveTargetSampleRate) / buffer.sampleRate), effectiveTargetSampleRate);
95
99
  // Create source and connect
96
100
  const source = offlineCtx.createBufferSource();
97
101
  source.buffer = segmentBuffer;
@@ -104,7 +108,7 @@ async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targ
104
108
  const durationMs = Math.round((samplesNeeded / buffer.sampleRate) * 1000);
105
109
  logger?.debug('Final processed audio:', {
106
110
  outputSamples: channelData.length,
107
- outputSampleRate: targetSampleRate,
111
+ outputSampleRate: effectiveTargetSampleRate,
108
112
  durationMs,
109
113
  });
110
114
  return {
@@ -112,7 +116,7 @@ async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRate, targ
112
116
  channelData,
113
117
  samples: channelData.length,
114
118
  durationMs,
115
- sampleRate: targetSampleRate,
119
+ sampleRate: effectiveTargetSampleRate,
116
120
  channels: processedBuffer.numberOfChannels,
117
121
  };
118
122
  }
package/build/cjs/utils/audioProcessing.js.map CHANGED
@@ -1 +1 @@
1
- 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packages/audio-studio/src/utils/audioProcessing.ts\nimport { Platform } from 'react-native'\n\nimport { ConsoleLike } from '../AudioStudio.types'\n\nexport interface ProcessAudioBufferOptions {\n arrayBuffer?: ArrayBuffer\n fileUri?: string\n targetSampleRate: number\n targetChannels: number\n normalizeAudio: boolean\n startTimeMs?: number\n endTimeMs?: number\n position?: number\n length?: number\n audioContext?: AudioContext\n logger?: ConsoleLike\n}\n\nexport interface ProcessedAudioData {\n channelData: Float32Array\n samples: number\n durationMs: number\n sampleRate: number\n channels: number\n buffer: AudioBuffer\n}\n\nexport async function processAudioBuffer({\n arrayBuffer,\n fileUri,\n targetSampleRate,\n targetChannels,\n normalizeAudio,\n startTimeMs,\n endTimeMs,\n position,\n length,\n audioContext,\n logger,\n}: ProcessAudioBufferOptions): Promise<ProcessedAudioData> {\n if (Platform.OS !== 'web') {\n throw new Error('processAudioBuffer is only supported on web')\n }\n\n let ctx: AudioContext | undefined\n let buffer: AudioBuffer | undefined\n\n try {\n // Log initial parameters\n logger?.debug('Process audio buffer - Initial params:', {\n hasArrayBuffer: !!arrayBuffer,\n fileUri,\n targetSampleRate,\n targetChannels,\n normalizeAudio,\n startTimeMs,\n endTimeMs,\n position,\n length,\n })\n\n // Get the audio data\n let audioData: ArrayBuffer\n if (arrayBuffer) {\n audioData = arrayBuffer\n } else if (fileUri) {\n const response = await fetch(fileUri)\n if (!response.ok) {\n throw new Error(\n `Failed to fetch fileUri: ${response.statusText}`\n )\n }\n audioData = await response.arrayBuffer()\n } else {\n throw new Error('Either arrayBuffer or fileUri must be provided')\n }\n\n logger?.debug('Audio data loaded:', {\n byteLength: audioData.byteLength,\n firstBytes: Array.from(new Uint8Array(audioData.slice(0, 16))),\n })\n\n // Create context at original sample rate first\n ctx =\n audioContext ||\n new (window.AudioContext || (window as any).webkitAudioContext)()\n buffer = await ctx.decodeAudioData(audioData)\n\n logger?.debug('Decoded audio buffer:', {\n originalChannels: buffer.numberOfChannels,\n originalSampleRate: buffer.sampleRate,\n originalDuration: buffer.duration,\n originalLength: buffer.length,\n })\n\n // Calculate time range\n const startSample =\n startTimeMs !== undefined\n ? Math.floor((startTimeMs / 1000) * buffer.sampleRate)\n : position !== undefined\n ? Math.floor(position / 2)\n : 0\n\n // Fix: Adjust position calculation based on original sample rate\n // When position is provided in bytes, we need to account for the original sample rate\n const bytesPerSample = 2 // 16-bit audio = 2 bytes per sample\n const adjustedStartSample =\n position !== undefined\n ? Math.floor(\n (position / bytesPerSample) *\n (buffer.sampleRate / targetSampleRate)\n )\n : startSample\n\n const samplesNeeded =\n length !== undefined\n ? Math.floor(\n (length / bytesPerSample) *\n (buffer.sampleRate / targetSampleRate)\n )\n : endTimeMs !== undefined && startTimeMs !== undefined\n ? Math.floor(\n ((endTimeMs - startTimeMs) / 1000) * buffer.sampleRate\n )\n : buffer.length - adjustedStartSample\n\n logger?.debug('Sample calculations (adjusted):', {\n originalStartSample: startSample,\n adjustedStartSample,\n samplesNeeded,\n originalSampleRate: buffer.sampleRate,\n targetSampleRate,\n conversionRatio: buffer.sampleRate / targetSampleRate,\n expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,\n })\n\n // Create temporary buffer for the segment\n const segmentBuffer = ctx.createBuffer(\n buffer.numberOfChannels,\n samplesNeeded,\n buffer.sampleRate\n )\n\n // Copy the segment\n for (let channel = 0; channel < buffer.numberOfChannels; channel++) {\n const channelData = buffer.getChannelData(channel)\n const segmentData = segmentBuffer.getChannelData(channel)\n for (let i = 0; i < samplesNeeded; i++) {\n segmentData[i] = channelData[adjustedStartSample + i]\n }\n }\n\n // Create offline context for resampling\n const offlineCtx = new OfflineAudioContext(\n targetChannels,\n Math.ceil((samplesNeeded * targetSampleRate) / buffer.sampleRate),\n targetSampleRate\n )\n\n // Create source and connect\n const source = offlineCtx.createBufferSource()\n source.buffer = segmentBuffer\n source.connect(offlineCtx.destination)\n\n // Render at new sample rate\n source.start()\n const processedBuffer = await offlineCtx.startRendering()\n\n // Get the final audio data\n const channelData = processedBuffer.getChannelData(0)\n const durationMs = Math.round(\n (samplesNeeded / buffer.sampleRate) * 1000\n )\n\n logger?.debug('Final processed audio:', {\n outputSamples: channelData.length,\n outputSampleRate: targetSampleRate,\n durationMs,\n })\n\n return {\n buffer: processedBuffer,\n channelData,\n samples: channelData.length,\n durationMs,\n sampleRate: targetSampleRate,\n channels: processedBuffer.numberOfChannels,\n }\n } catch (error) {\n logger?.error('Failed to process audio buffer:', {\n error,\n position,\n length,\n startTimeMs,\n endTimeMs,\n bufferLength: buffer?.length,\n })\n throw error\n } finally {\n if (!audioContext && ctx) {\n await ctx.close()\n }\n }\n}\n"]}
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packages/audio-studio/src/utils/audioProcessing.ts\nimport { Platform } from 'react-native'\n\nimport { ConsoleLike } from '../AudioStudio.types'\n\nexport interface ProcessAudioBufferOptions {\n arrayBuffer?: ArrayBuffer\n fileUri?: string\n targetSampleRate?: number\n targetChannels?: number\n normalizeAudio: boolean\n startTimeMs?: number\n endTimeMs?: number\n position?: number\n length?: number\n audioContext?: AudioContext\n logger?: ConsoleLike\n}\n\nexport interface ProcessedAudioData {\n channelData: Float32Array\n samples: number\n durationMs: number\n sampleRate: number\n channels: number\n buffer: AudioBuffer\n}\n\nexport async function processAudioBuffer({\n arrayBuffer,\n fileUri,\n targetSampleRate,\n targetChannels,\n normalizeAudio,\n startTimeMs,\n endTimeMs,\n position,\n length,\n audioContext,\n logger,\n}: ProcessAudioBufferOptions): Promise<ProcessedAudioData> {\n if (Platform.OS !== 'web') {\n throw new Error('processAudioBuffer is only supported on web')\n }\n\n let ctx: AudioContext | undefined\n let buffer: AudioBuffer | undefined\n\n try {\n // Log initial parameters\n logger?.debug('Process audio buffer - Initial params:', {\n hasArrayBuffer: !!arrayBuffer,\n fileUri,\n targetSampleRate,\n targetChannels,\n normalizeAudio,\n startTimeMs,\n endTimeMs,\n position,\n length,\n })\n\n // Get the audio data\n let audioData: ArrayBuffer\n if (arrayBuffer) {\n audioData = arrayBuffer\n } else if (fileUri) {\n const response = await fetch(fileUri)\n if (!response.ok) {\n throw new Error(\n `Failed to fetch fileUri: ${response.statusText}`\n )\n }\n audioData = await response.arrayBuffer()\n } else {\n throw new Error('Either arrayBuffer or fileUri must be provided')\n }\n\n logger?.debug('Audio data loaded:', {\n byteLength: audioData.byteLength,\n firstBytes: Array.from(new Uint8Array(audioData.slice(0, 16))),\n })\n\n // Create context at original sample rate first\n ctx =\n audioContext ||\n new (window.AudioContext ||\n (\n window as unknown as {\n webkitAudioContext?: typeof AudioContext\n }\n ).webkitAudioContext)()\n buffer = await ctx.decodeAudioData(audioData)\n\n const effectiveTargetSampleRate = targetSampleRate ?? buffer.sampleRate\n const effectiveTargetChannels = targetChannels ?? buffer.numberOfChannels\n\n logger?.debug('Decoded audio buffer:', {\n originalChannels: buffer.numberOfChannels,\n originalSampleRate: buffer.sampleRate,\n originalDuration: buffer.duration,\n originalLength: buffer.length,\n })\n\n // Calculate time range\n const startSample =\n startTimeMs !== undefined\n ? Math.floor((startTimeMs / 1000) * buffer.sampleRate)\n : position !== undefined\n ? Math.floor(position / 2)\n : 0\n\n // Fix: Adjust position calculation based on original sample rate\n // When position is provided in bytes, we need to account for the original sample rate\n const bytesPerSample = 2 // 16-bit audio = 2 bytes per sample\n const adjustedStartSample =\n position !== undefined\n ? Math.floor(\n (position / bytesPerSample) *\n (buffer.sampleRate / effectiveTargetSampleRate)\n )\n : startSample\n\n const samplesNeeded =\n length !== undefined\n ? Math.floor(\n (length / bytesPerSample) *\n (buffer.sampleRate / effectiveTargetSampleRate)\n )\n : endTimeMs !== undefined\n ? Math.floor(\n ((endTimeMs - (startTimeMs ?? 0)) / 1000) *\n buffer.sampleRate\n )\n : buffer.length - adjustedStartSample\n\n logger?.debug('Sample calculations (adjusted):', {\n originalStartSample: startSample,\n adjustedStartSample,\n samplesNeeded,\n originalSampleRate: buffer.sampleRate,\n targetSampleRate: effectiveTargetSampleRate,\n conversionRatio: buffer.sampleRate / effectiveTargetSampleRate,\n expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,\n })\n\n // Create temporary buffer for the segment\n const segmentBuffer = ctx.createBuffer(\n buffer.numberOfChannels,\n samplesNeeded,\n buffer.sampleRate\n )\n\n // Copy the segment\n for (let channel = 0; channel < buffer.numberOfChannels; channel++) {\n const channelData = buffer.getChannelData(channel)\n const segmentData = segmentBuffer.getChannelData(channel)\n for (let i = 0; i < samplesNeeded; i++) {\n segmentData[i] = channelData[adjustedStartSample + i]\n }\n }\n\n // Create offline context for resampling\n const offlineCtx = new OfflineAudioContext(\n effectiveTargetChannels,\n Math.ceil(\n (samplesNeeded * effectiveTargetSampleRate) / buffer.sampleRate\n ),\n effectiveTargetSampleRate\n )\n\n // Create source and connect\n const source = offlineCtx.createBufferSource()\n source.buffer = segmentBuffer\n source.connect(offlineCtx.destination)\n\n // Render at new sample rate\n source.start()\n const processedBuffer = await offlineCtx.startRendering()\n\n // Get the final audio data\n const channelData = processedBuffer.getChannelData(0)\n const durationMs = Math.round(\n (samplesNeeded / buffer.sampleRate) * 1000\n )\n\n logger?.debug('Final processed audio:', {\n outputSamples: channelData.length,\n outputSampleRate: effectiveTargetSampleRate,\n durationMs,\n })\n\n return {\n buffer: processedBuffer,\n channelData,\n samples: channelData.length,\n durationMs,\n sampleRate: effectiveTargetSampleRate,\n channels: processedBuffer.numberOfChannels,\n }\n } catch (error) {\n logger?.error('Failed to process audio buffer:', {\n error,\n position,\n length,\n startTimeMs,\n endTimeMs,\n bufferLength: buffer?.length,\n })\n throw error\n } finally {\n if (!audioContext && ctx) {\n await ctx.close()\n }\n }\n}\n"]}
package/build/esm/errors/AudioStreamError.js CHANGED
@@ -53,6 +53,12 @@ function getNativeCode(err) {
53
53
  }
54
54
  return undefined;
55
55
  }
56
+ function isUnknownAudioStreamCode(raw) {
57
+ if (!raw)
58
+ return false;
59
+ return (raw.toUpperCase().startsWith('ERR_AUDIO_STREAM_') &&
60
+ normalizeCode(raw) === null);
61
+ }
56
62
  function normalizeCode(raw) {
57
63
  if (!raw)
58
64
  return null;
@@ -113,6 +119,9 @@ export function mapStreamError(err, fileUri, platform) {
113
119
  const nativeMessage = getNativeMessage(err);
114
120
  const nativeCode = getNativeCode(err);
115
121
  const lower = nativeMessage.toLowerCase();
122
+ if (isUnknownAudioStreamCode(nativeCode)) {
123
+ console.warn(`[AudioStreamError] Unknown native audio stream error code: ${nativeCode}`);
124
+ }
116
125
  let code = normalizeCode(nativeCode) ??
117
126
  normalizeCode(nativeMessage) ??
118
127
  'ERR_AUDIO_STREAM_UNKNOWN';
package/build/esm/errors/AudioStreamError.js.map CHANGED
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Callers can switch on `code`.\n */\nexport type AudioStreamErrorCode =\n | 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n | 'ERR_AUDIO_STREAM_INVALID_RANGE'\n | 'ERR_AUDIO_STREAM_DECODE_FAILED'\n | 'ERR_AUDIO_STREAM_CANCELLED'\n | 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n | 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n | 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT'\n | 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE'\n | 'ERR_AUDIO_STREAM_BUSY'\n | 'ERR_AUDIO_STREAM_UNKNOWN'\n\nexport interface AudioStreamErrorPayload {\n code: AudioStreamErrorCode\n message: string\n recoverable: boolean\n fileUri?: string\n platform?: string\n nativeCode?: string\n nativeMessage?: string\n}\n\nconst RECOVERABLE: AudioStreamErrorCode[] = [\n 'ERR_AUDIO_STREAM_CANCELLED',\n 'ERR_AUDIO_STREAM_BUSY',\n 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',\n 'ERR_AUDIO_STREAM_PERMISSION_DENIED',\n]\n\nexport class AudioStreamError extends Error {\n readonly code: AudioStreamErrorCode\n readonly recoverable: boolean\n readonly fileUri?: string\n readonly platform?: string\n readonly nativeCode?: string\n readonly nativeMessage?: string\n\n constructor(payload: AudioStreamErrorPayload) {\n super(payload.message)\n this.name = 'AudioStreamError'\n this.code = payload.code\n this.recoverable = payload.recoverable\n this.fileUri = payload.fileUri\n this.platform = payload.platform\n this.nativeCode = payload.nativeCode\n this.nativeMessage = payload.nativeMessage\n }\n\n toJSON(): AudioStreamErrorPayload {\n return {\n code: this.code,\n message: this.message,\n recoverable: this.recoverable,\n fileUri: this.fileUri,\n platform: this.platform,\n nativeCode: this.nativeCode,\n nativeMessage: this.nativeMessage,\n }\n }\n}\n\nfunction getNativeMessage(err: unknown): string {\n if (err instanceof Error) return err.message\n if (typeof err === 'string') return err\n try {\n return JSON.stringify(err) ?? String(err)\n } catch {\n return String(err)\n }\n}\n\nfunction getNativeCode(err: unknown): string | undefined {\n if (err && typeof err === 'object' && 'code' in err) {\n const code = (err as { code?: unknown }).code\n if (typeof code === 'string') return code\n }\n return undefined\n}\n\nfunction normalizeCode(raw: string | undefined): AudioStreamErrorCode | null {\n if (!raw) return null\n const upper = raw.toUpperCase()\n if (upper.startsWith('ERR_AUDIO_STREAM_')) {\n const known: AudioStreamErrorCode[] = [\n 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',\n 'ERR_AUDIO_STREAM_INVALID_RANGE',\n 'ERR_AUDIO_STREAM_DECODE_FAILED',\n 'ERR_AUDIO_STREAM_CANCELLED',\n 'ERR_AUDIO_STREAM_PERMISSION_DENIED',\n 'ERR_AUDIO_STREAM_FILE_NOT_FOUND',\n 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',\n 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n 'ERR_AUDIO_STREAM_BUSY',\n 'ERR_AUDIO_STREAM_UNKNOWN',\n ]\n if ((known as string[]).includes(upper)) {\n return upper as AudioStreamErrorCode\n }\n }\n if (upper.includes('FILE_NOT_FOUND') || upper === 'ENOENT') {\n return 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n }\n if (upper.includes('PERMISSION') || upper === 'EACCES') {\n return 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n }\n if (\n upper.includes('UNSUPPORTED') ||\n upper.includes('NO_SUITABLE_CODEC') ||\n upper.includes('NO SUITABLE CODEC') ||\n upper.includes('NOT SUPPORTED')\n ) {\n return 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n }\n if (\n upper.includes('INVALID_RANGE') ||\n upper.includes('OUT_OF_RANGE') ||\n upper.includes('INVALID_TIME')\n ) {\n return 'ERR_AUDIO_STREAM_INVALID_RANGE'\n }\n if (upper.includes('CANCELLED') || upper.includes('CANCELED')) {\n return 'ERR_AUDIO_STREAM_CANCELLED'\n }\n if (upper.includes('BUSY')) {\n return 'ERR_AUDIO_STREAM_BUSY'\n }\n if (upper.includes('BACKPRESSURE')) {\n return 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT'\n }\n if (\n upper.includes('DECODE') ||\n upper.includes('CODEC') ||\n upper.includes('MALFORMED')\n ) {\n return 'ERR_AUDIO_STREAM_DECODE_FAILED'\n }\n return null\n}\n\nexport function mapStreamError(\n err: unknown,\n fileUri?: string,\n platform?: string\n): AudioStreamError {\n if (err instanceof AudioStreamError) return err\n\n const nativeMessage = getNativeMessage(err)\n const nativeCode = getNativeCode(err)\n const lower = nativeMessage.toLowerCase()\n\n let code =\n normalizeCode(nativeCode) ??\n normalizeCode(nativeMessage) ??\n 'ERR_AUDIO_STREAM_UNKNOWN'\n\n if (code === 'ERR_AUDIO_STREAM_UNKNOWN') {\n if (lower.includes('not found') || lower.includes('does not exist')) {\n code = 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n } else if (\n lower.includes('unsupported') ||\n lower.includes('no suitable codec')\n ) {\n code = 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n } else if (lower.includes('permission') || lower.includes('denied')) {\n code = 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n } else if (lower.includes('decode') || lower.includes('codec')) {\n code = 'ERR_AUDIO_STREAM_DECODE_FAILED'\n } else if (lower.includes('cancel')) {\n code = 'ERR_AUDIO_STREAM_CANCELLED'\n }\n }\n\n return new AudioStreamError({\n code,\n message: `Audio stream failed (${code}): ${nativeMessage}`,\n recoverable: RECOVERABLE.includes(code),\n fileUri,\n platform,\n nativeCode,\n nativeMessage,\n })\n}\n"]}
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Callers can switch on `code`.\n */\nexport type AudioStreamErrorCode =\n | 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n | 'ERR_AUDIO_STREAM_INVALID_RANGE'\n | 'ERR_AUDIO_STREAM_DECODE_FAILED'\n | 'ERR_AUDIO_STREAM_CANCELLED'\n | 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n | 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n | 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT'\n | 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE'\n | 'ERR_AUDIO_STREAM_BUSY'\n | 'ERR_AUDIO_STREAM_UNKNOWN'\n\nexport interface AudioStreamErrorPayload {\n code: AudioStreamErrorCode\n message: string\n recoverable: boolean\n fileUri?: string\n platform?: string\n nativeCode?: string\n nativeMessage?: string\n}\n\nconst RECOVERABLE: AudioStreamErrorCode[] = [\n 'ERR_AUDIO_STREAM_CANCELLED',\n 'ERR_AUDIO_STREAM_BUSY',\n 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',\n 'ERR_AUDIO_STREAM_PERMISSION_DENIED',\n]\n\nexport class AudioStreamError extends Error {\n readonly code: AudioStreamErrorCode\n readonly recoverable: boolean\n readonly fileUri?: string\n readonly platform?: string\n readonly nativeCode?: string\n readonly nativeMessage?: string\n\n constructor(payload: AudioStreamErrorPayload) {\n super(payload.message)\n this.name = 'AudioStreamError'\n this.code = payload.code\n this.recoverable = payload.recoverable\n this.fileUri = payload.fileUri\n this.platform = payload.platform\n this.nativeCode = payload.nativeCode\n this.nativeMessage = payload.nativeMessage\n }\n\n toJSON(): AudioStreamErrorPayload {\n return {\n code: this.code,\n message: this.message,\n recoverable: this.recoverable,\n fileUri: this.fileUri,\n platform: this.platform,\n nativeCode: this.nativeCode,\n nativeMessage: this.nativeMessage,\n }\n }\n}\n\nfunction getNativeMessage(err: unknown): string {\n if (err instanceof Error) return err.message\n if (typeof err === 'string') return err\n try {\n return JSON.stringify(err) ?? String(err)\n } catch {\n return String(err)\n }\n}\n\nfunction getNativeCode(err: unknown): string | undefined {\n if (err && typeof err === 'object' && 'code' in err) {\n const code = (err as { code?: unknown }).code\n if (typeof code === 'string') return code\n }\n return undefined\n}\n\nfunction isUnknownAudioStreamCode(raw: string | undefined): boolean {\n if (!raw) return false\n return (\n raw.toUpperCase().startsWith('ERR_AUDIO_STREAM_') &&\n normalizeCode(raw) === null\n )\n}\n\nfunction normalizeCode(raw: string | undefined): AudioStreamErrorCode | null {\n if (!raw) return null\n const upper = raw.toUpperCase()\n if (upper.startsWith('ERR_AUDIO_STREAM_')) {\n const known: AudioStreamErrorCode[] = [\n 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',\n 'ERR_AUDIO_STREAM_INVALID_RANGE',\n 'ERR_AUDIO_STREAM_DECODE_FAILED',\n 'ERR_AUDIO_STREAM_CANCELLED',\n 'ERR_AUDIO_STREAM_PERMISSION_DENIED',\n 'ERR_AUDIO_STREAM_FILE_NOT_FOUND',\n 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',\n 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n 'ERR_AUDIO_STREAM_BUSY',\n 'ERR_AUDIO_STREAM_UNKNOWN',\n ]\n if ((known as string[]).includes(upper)) {\n return upper as AudioStreamErrorCode\n }\n }\n if (upper.includes('FILE_NOT_FOUND') || upper === 'ENOENT') {\n return 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n }\n if (upper.includes('PERMISSION') || upper === 'EACCES') {\n return 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n }\n if (\n upper.includes('UNSUPPORTED') ||\n upper.includes('NO_SUITABLE_CODEC') ||\n upper.includes('NO SUITABLE CODEC') ||\n upper.includes('NOT SUPPORTED')\n ) {\n return 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n }\n if (\n upper.includes('INVALID_RANGE') ||\n upper.includes('OUT_OF_RANGE') ||\n upper.includes('INVALID_TIME')\n ) {\n return 'ERR_AUDIO_STREAM_INVALID_RANGE'\n }\n if (upper.includes('CANCELLED') || upper.includes('CANCELED')) {\n return 'ERR_AUDIO_STREAM_CANCELLED'\n }\n if (upper.includes('BUSY')) {\n return 'ERR_AUDIO_STREAM_BUSY'\n }\n if (upper.includes('BACKPRESSURE')) {\n return 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT'\n }\n if (\n upper.includes('DECODE') ||\n upper.includes('CODEC') ||\n upper.includes('MALFORMED')\n ) {\n return 'ERR_AUDIO_STREAM_DECODE_FAILED'\n }\n return null\n}\n\nexport function mapStreamError(\n err: unknown,\n fileUri?: string,\n platform?: string\n): AudioStreamError {\n if (err instanceof AudioStreamError) return err\n\n const nativeMessage = getNativeMessage(err)\n const nativeCode = getNativeCode(err)\n const lower = nativeMessage.toLowerCase()\n\n if (isUnknownAudioStreamCode(nativeCode)) {\n console.warn(\n `[AudioStreamError] Unknown native audio stream error code: ${nativeCode}`\n )\n }\n\n let code =\n normalizeCode(nativeCode) ??\n normalizeCode(nativeMessage) ??\n 'ERR_AUDIO_STREAM_UNKNOWN'\n\n if (code === 'ERR_AUDIO_STREAM_UNKNOWN') {\n if (lower.includes('not found') || lower.includes('does not exist')) {\n code = 'ERR_AUDIO_STREAM_FILE_NOT_FOUND'\n } else if (\n lower.includes('unsupported') ||\n lower.includes('no suitable codec')\n ) {\n code = 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT'\n } else if (lower.includes('permission') || lower.includes('denied')) {\n code = 'ERR_AUDIO_STREAM_PERMISSION_DENIED'\n } else if (lower.includes('decode') || lower.includes('codec')) {\n code = 'ERR_AUDIO_STREAM_DECODE_FAILED'\n } else if (lower.includes('cancel')) {\n code = 'ERR_AUDIO_STREAM_CANCELLED'\n }\n }\n\n return new AudioStreamError({\n code,\n message: `Audio stream failed (${code}): ${nativeMessage}`,\n recoverable: RECOVERABLE.includes(code),\n fileUri,\n platform,\n nativeCode,\n nativeMessage,\n })\n}\n"]}
package/build/esm/errors/AudioStreamError.test.js CHANGED
@@ -9,7 +9,10 @@ describe('AudioStreamError', () => {
9
9
  expect(mapStreamError(original)).toBe(original);
10
10
  });
11
11
  it('maps native FILE_NOT_FOUND code', () => {
12
- const mapped = mapStreamError({ code: 'FILE_NOT_FOUND', message: 'gone' });
12
+ const mapped = mapStreamError({
13
+ code: 'FILE_NOT_FOUND',
14
+ message: 'gone',
15
+ });
13
16
  expect(mapped.code).toBe('ERR_AUDIO_STREAM_FILE_NOT_FOUND');
14
17
  expect(mapped.recoverable).toBe(false);
15
18
  });
@@ -25,10 +28,28 @@ describe('AudioStreamError', () => {
25
28
  expect(mapped.code).toBe('ERR_AUDIO_STREAM_CANCELLED');
26
29
  expect(mapped.recoverable).toBe(true);
27
30
  });
31
+ it('maps backpressure timeout as recoverable', () => {
32
+ const mapped = mapStreamError({
33
+ code: 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',
34
+ message: 'ack timed out',
35
+ });
36
+ expect(mapped.code).toBe('ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT');
37
+ expect(mapped.recoverable).toBe(true);
38
+ });
28
39
  it('falls back to UNKNOWN', () => {
29
40
  const mapped = mapStreamError({});
30
41
  expect(mapped.code).toBe('ERR_AUDIO_STREAM_UNKNOWN');
31
42
  });
43
+ it('warns when native returns an unknown audio stream code', () => {
44
+ const warn = jest.spyOn(console, 'warn').mockImplementation(() => { });
45
+ const mapped = mapStreamError({
46
+ code: 'ERR_AUDIO_STREAM_FOOBAR',
47
+ message: 'new native code',
48
+ });
49
+ expect(mapped.code).toBe('ERR_AUDIO_STREAM_UNKNOWN');
50
+ expect(warn).toHaveBeenCalledWith('[AudioStreamError] Unknown native audio stream error code: ERR_AUDIO_STREAM_FOOBAR');
51
+ warn.mockRestore();
52
+ });
32
53
  it('preserves nativeCode and nativeMessage', () => {
33
54
  const mapped = mapStreamError({
34
55
  code: 'WEIRD_NATIVE_CODE',
package/build/esm/errors/AudioStreamError.test.js.map CHANGED
@@ -1 +1 @@
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1
+ 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{ AudioStreamError, mapStreamError } from './AudioStreamError'\n\ndescribe('AudioStreamError', () => {\n it('passes through an existing AudioStreamError unchanged', () => {\n const original = new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_CANCELLED',\n message: 'aborted',\n recoverable: true,\n })\n expect(mapStreamError(original)).toBe(original)\n })\n\n it('maps native FILE_NOT_FOUND code', () => {\n const mapped = mapStreamError({\n code: 'FILE_NOT_FOUND',\n message: 'gone',\n })\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_FILE_NOT_FOUND')\n expect(mapped.recoverable).toBe(false)\n })\n\n it('maps unsupported codec text', () => {\n const mapped = mapStreamError(\n new Error('No suitable codec for audio/opus')\n )\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT')\n })\n\n it('marks cancellation as recoverable', () => {\n const mapped = mapStreamError({\n code: 'ERR_AUDIO_STREAM_CANCELLED',\n message: 'user cancelled',\n })\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_CANCELLED')\n expect(mapped.recoverable).toBe(true)\n })\n\n it('maps backpressure timeout as recoverable', () => {\n const mapped = mapStreamError({\n code: 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT',\n message: 'ack timed out',\n })\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT')\n expect(mapped.recoverable).toBe(true)\n })\n\n it('falls back to UNKNOWN', () => {\n const mapped = mapStreamError({})\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_UNKNOWN')\n })\n\n it('warns when native returns an unknown audio stream code', () => {\n const warn = jest.spyOn(console, 'warn').mockImplementation(() => {})\n const mapped = mapStreamError({\n code: 'ERR_AUDIO_STREAM_FOOBAR',\n message: 'new native code',\n })\n expect(mapped.code).toBe('ERR_AUDIO_STREAM_UNKNOWN')\n expect(warn).toHaveBeenCalledWith(\n '[AudioStreamError] Unknown native audio stream error code: ERR_AUDIO_STREAM_FOOBAR'\n )\n warn.mockRestore()\n })\n\n it('preserves nativeCode and nativeMessage', () => {\n const mapped = mapStreamError({\n code: 'WEIRD_NATIVE_CODE',\n message: 'something went wrong on the bridge',\n })\n expect(mapped.nativeCode).toBe('WEIRD_NATIVE_CODE')\n expect(mapped.nativeMessage).toContain('bridge')\n })\n\n it('serialises to a stable JSON payload', () => {\n const err = new AudioStreamError({\n code: 'ERR_AUDIO_STREAM_DECODE_FAILED',\n message: 'decoder bust',\n recoverable: false,\n fileUri: 'file:///a.m4a',\n platform: 'ios',\n })\n expect(err.toJSON()).toEqual({\n code: 'ERR_AUDIO_STREAM_DECODE_FAILED',\n message: 'decoder bust',\n recoverable: false,\n fileUri: 'file:///a.m4a',\n platform: 'ios',\n nativeCode: undefined,\n nativeMessage: undefined,\n })\n })\n})\n"]}