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

package/build/esm/streamAudioData.js

@@ -41,10 +41,12 @@ function toFloat32(samples) {
     }
     if (typeof samples === 'string') {
         const bytes = base64ToBytes(samples);
-        const aligned = bytes.byteOffset % 4 === 0
-            ? new Float32Array(bytes.buffer, bytes.byteOffset, bytes.byteLength / 4)
-            : new Float32Array(bytes.buffer.slice(bytes.byteOffset));
-        return new Float32Array(aligned);
+        const floatLength = Math.floor(bytes.byteLength / 4);
+        if (bytes.byteOffset % 4 === 0) {
+            return new Float32Array(bytes.buffer, bytes.byteOffset, floatLength);
+        }
+        const sliced = bytes.buffer.slice(bytes.byteOffset, bytes.byteOffset + bytes.byteLength);
+        return new Float32Array(sliced, 0, Math.floor(sliced.byteLength / 4));
     }
     if (samples && typeof samples === 'object' && 'length' in samples) {
         // ArrayLike fallback
@@ -73,31 +75,50 @@ function base64ToBytes(input) {
         out[i] = bin.charCodeAt(i);
     return out;
 }
+function rejectInvalidRange(message) {
+    throw new AudioStreamError({
+        code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
+        message,
+        recoverable: false,
+    });
+}
+function assertPositiveFiniteOption(value, name, integer = false) {
+    if (value === undefined)
+        return;
+    if (!Number.isFinite(value) ||
+        value <= 0 ||
+        (integer && !Number.isInteger(value))) {
+        rejectInvalidRange(`${name} must be a positive${integer ? ' integer' : ''}`);
+    }
+}
 function validateOptions(options) {
     if (!options.fileUri) {
-        throw new AudioStreamError({
-            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
-            message: 'fileUri is required',
-            recoverable: false,
-        });
+        rejectInvalidRange('fileUri is required');
     }
     if (options.startTimeMs !== undefined &&
         options.endTimeMs !== undefined &&
         options.startTimeMs >= options.endTimeMs) {
-        throw new AudioStreamError({
-            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
-            message: 'startTimeMs must be < endTimeMs',
-            recoverable: false,
-        });
+        rejectInvalidRange('startTimeMs must be < endTimeMs');
+    }
+    if (options.endTimeMs !== undefined && options.endTimeMs <= 0) {
+        rejectInvalidRange('endTimeMs must be > 0');
+    }
+    if (options.startTimeMs !== undefined && options.startTimeMs < 0) {
+        rejectInvalidRange('startTimeMs must be >= 0');
     }
     if (options.chunkDurationMs !== undefined &&
         (options.chunkDurationMs < 10 || options.chunkDurationMs > 60000)) {
-        throw new AudioStreamError({
-            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
-            message: 'chunkDurationMs must be in [10, 60000]',
-            recoverable: false,
-        });
+        rejectInvalidRange('chunkDurationMs must be in [10, 60000]');
+    }
+    if (options.backpressureTimeoutMs !== undefined &&
+        options.backpressureTimeoutMs < 0) {
+        rejectInvalidRange('backpressureTimeoutMs must be >= 0');
     }
+    assertPositiveFiniteOption(options.targetSampleRate, 'targetSampleRate');
+    assertPositiveFiniteOption(options.sampleRate, 'sampleRate');
+    assertPositiveFiniteOption(options.channels, 'channels', true);
+    assertPositiveFiniteOption(options.maxBufferedChunks, 'maxBufferedChunks', true);
+    assertPositiveFiniteOption(options.maxChunkBytes, 'maxChunkBytes', true);
     if (options.streamFormat !== undefined &&
         options.streamFormat !== 'float32') {
         throw new AudioStreamError({
@@ -182,6 +203,7 @@ async function streamAudioDataNative(options, callbacks) {
     let processingChain = Promise.resolve();
     let settled = false;
     let abortListener = null;
+    let lastProgress = null;
     const finalize = () => {
         for (const sub of subs) {
             try {
@@ -256,6 +278,7 @@ async function streamAudioDataNative(options, callbacks) {
                 const evt = raw;
                 if (evt.requestId !== requestId)
                     return;
+                lastProgress = evt;
                 callbacks.onProgress(evt);
             }));
         }
@@ -268,7 +291,9 @@ async function streamAudioDataNative(options, callbacks) {
                 .then(() => {
                 settle(() => { }, 'resolve', {
                     requestId,
-                    durationMs: evt.durationMs,
+                    durationMs: evt.durationMs > 0
+                        ? evt.durationMs
+                        : (lastProgress?.durationMs ?? 0),
                     sampleRate: evt.sampleRate,
                     channels: evt.channels,
                     chunks: evt.chunks ?? chunkCount,
@@ -288,7 +313,7 @@ async function streamAudioDataNative(options, callbacks) {
                     .then(() => {
                     settle(() => { }, 'resolve', {
                         requestId,
-                        durationMs: 0,
+                        durationMs: lastProgress?.durationMs ?? 0,
                         sampleRate: options.targetSampleRate ??
                             options.sampleRate ??
                             0,
@@ -314,7 +339,7 @@ async function streamAudioDataNative(options, callbacks) {
             if (options.signal.aborted) {
                 settle(() => { }, 'resolve', {
                     requestId,
-                    durationMs: 0,
+                    durationMs: lastProgress?.durationMs ?? 0,
                     sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,
                     channels: options.channels ?? 1,
                     chunks: 0,
@@ -361,8 +386,8 @@ async function streamAudioDataWeb(options, callbacks) {
     try {
         const processed = await processAudioBuffer({
             fileUri: options.fileUri,
-            targetSampleRate: options.targetSampleRate ?? 16000,
-            targetChannels: options.channels ?? 1,
+            targetSampleRate: options.targetSampleRate,
+            targetChannels: options.channels,
             normalizeAudio: options.normalizeAudio ?? true,
             startTimeMs: options.startTimeMs,
             endTimeMs: options.endTimeMs,
@@ -371,12 +396,21 @@ async function streamAudioDataWeb(options, callbacks) {
         const channels = processed.channels;
         const durationMs = processed.durationMs;
         const chunkDurationMs = options.chunkDurationMs ?? 1000;
-        let samplesPerChunk = Math.max(1, Math.floor((chunkDurationMs / 1000) * sampleRate) * channels);
+        let samplesPerChunk = Math.max(channels, Math.floor((chunkDurationMs / 1000) * sampleRate) * channels);
         if (options.maxChunkBytes) {
-            const maxSamples = Math.floor(options.maxChunkBytes / 4);
-            samplesPerChunk = Math.min(samplesPerChunk, maxSamples);
+            // Round down to a multiple of `channels` so we never split an
+            // interleaved frame across two chunks (that would produce a
+            // fractional `startSample` for the next chunk).
+            const rawMax = Math.floor(options.maxChunkBytes / 4);
+            const maxSamples = Math.max(channels, Math.floor(rawMax / channels) * channels);
+            samplesPerChunk = Math.max(channels, Math.min(samplesPerChunk, maxSamples));
         }
-        const all = sanitizeFloat32(processed.channelData, options.normalizeAudio ?? true);
+        const all = sanitizeFloat32(interleaveBuffer(processed.buffer, channels), options.normalizeAudio ?? true);
+        // Chunk timestamps are absolute (range start + offset) on every
+        // platform; progress is *elapsed within the range* so the
+        // `processedMs / durationMs` fraction stays in [0, 1] regardless of
+        // `startTimeMs`. The native decoders use the same split.
+        const rangeStartMs = options.startTimeMs ?? 0;
         let chunkIndex = 0;
         let emittedSamples = 0;
         for (let off = 0; off < all.length; off += samplesPerChunk) {
@@ -395,11 +429,13 @@ async function streamAudioDataWeb(options, callbacks) {
             const slice = all.slice(off, end);
             const startSample = off / channels;
             const endSample = end / channels;
+            const startMs = Math.round((startSample / sampleRate) * 1000) + rangeStartMs;
+            const endMs = Math.round((endSample / sampleRate) * 1000) + rangeStartMs;
             const chunk = {
                 requestId,
                 chunkIndex,
-                startTimeMs: Math.round((startSample / sampleRate) * 1000),
-                endTimeMs: Math.round((endSample / sampleRate) * 1000),
+                startTimeMs: startMs,
+                endTimeMs: endMs,
                 durationMs: Math.round(((endSample - startSample) / sampleRate) * 1000),
                 startSample,
                 sampleCount: slice.length,
@@ -409,11 +445,22 @@ async function streamAudioDataWeb(options, callbacks) {
                 isFinal: end >= all.length,
             };
             await callbacks.onChunk(chunk);
+            // Resample rounding (Math.ceil in processAudioBuffer) can push
+            // elapsed past the source-rate-derived range duration on the tail
+            // chunk. Cap so onProgress consumers always see a [0, 1] ratio,
+            // matching the native `coerceIn(0, 1)` / `min(1, max(0, …))`
+            // clamp.
+            const rawElapsedMs = Math.round((endSample / sampleRate) * 1000);
+            const elapsedMs = durationMs > 0
+                ? Math.min(rawElapsedMs, durationMs)
+                : rawElapsedMs;
             callbacks.onProgress?.({
                 requestId,
-                processedMs: chunk.endTimeMs,
+                processedMs: elapsedMs,
                 durationMs,
-                progress: durationMs > 0 ? chunk.endTimeMs / durationMs : 1,
+                progress: durationMs > 0
+                    ? Math.min(1, Math.max(0, elapsedMs / durationMs))
+                    : 1,
                 emittedChunks: chunkIndex + 1,
             });
             chunkIndex += 1;
@@ -433,6 +480,26 @@ async function streamAudioDataWeb(options, callbacks) {
         throw mapStreamError(err, options.fileUri, 'web');
     }
 }
+function interleaveBuffer(buffer, channels) {
+    const numCh = Math.max(1, Math.min(channels, buffer.numberOfChannels));
+    const framesPerCh = buffer.length;
+    if (numCh === 1) {
+        // Cheap path: clone channel 0 so downstream mutation doesn't touch the
+        // underlying AudioBuffer storage.
+        return new Float32Array(buffer.getChannelData(0));
+    }
+    const out = new Float32Array(framesPerCh * numCh);
+    const channelData = [];
+    for (let c = 0; c < numCh; c++) {
+        channelData.push(buffer.getChannelData(c));
+    }
+    for (let f = 0; f < framesPerCh; f++) {
+        for (let c = 0; c < numCh; c++) {
+            out[f * numCh + c] = channelData[c][f];
+        }
+    }
+    return out;
+}
 function sanitizeFloat32(input, clamp) {
     if (!clamp) {
         // still need NaN/Inf sanitation

package/build/esm/streamAudioData.js.map

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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"]}
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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    /**\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/esm/utils/audioProcessing.js

@@ -41,8 +41,11 @@ export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRat
         // Create context at original sample rate first
         ctx =
             audioContext ||
-                new (window.AudioContext || window.webkitAudioContext)();
+                new (window.AudioContext ||
+                    window.webkitAudioContext)();
         buffer = await ctx.decodeAudioData(audioData);
+        const effectiveTargetSampleRate = targetSampleRate ?? buffer.sampleRate;
+        const effectiveTargetChannels = targetChannels ?? buffer.numberOfChannels;
         logger?.debug('Decoded audio buffer:', {
             originalChannels: buffer.numberOfChannels,
             originalSampleRate: buffer.sampleRate,
@@ -60,21 +63,22 @@ export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRat
         const bytesPerSample = 2; // 16-bit audio = 2 bytes per sample
         const adjustedStartSample = position !== undefined
             ? Math.floor((position / bytesPerSample) *
-                (buffer.sampleRate / targetSampleRate))
+                (buffer.sampleRate / effectiveTargetSampleRate))
             : startSample;
         const samplesNeeded = length !== undefined
             ? Math.floor((length / bytesPerSample) *
-                (buffer.sampleRate / targetSampleRate))
-            : endTimeMs !== undefined && startTimeMs !== undefined
-                ? Math.floor(((endTimeMs - startTimeMs) / 1000) * buffer.sampleRate)
+                (buffer.sampleRate / effectiveTargetSampleRate))
+            : endTimeMs !== undefined
+                ? Math.floor(((endTimeMs - (startTimeMs ?? 0)) / 1000) *
+                    buffer.sampleRate)
                 : buffer.length - adjustedStartSample;
         logger?.debug('Sample calculations (adjusted):', {
             originalStartSample: startSample,
             adjustedStartSample,
             samplesNeeded,
             originalSampleRate: buffer.sampleRate,
-            targetSampleRate,
-            conversionRatio: buffer.sampleRate / targetSampleRate,
+            targetSampleRate: effectiveTargetSampleRate,
+            conversionRatio: buffer.sampleRate / effectiveTargetSampleRate,
             expectedDurationMs: (samplesNeeded / buffer.sampleRate) * 1000,
         });
         // Create temporary buffer for the segment
@@ -88,7 +92,7 @@ export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRat
             }
         }
         // Create offline context for resampling
-        const offlineCtx = new OfflineAudioContext(targetChannels, Math.ceil((samplesNeeded * targetSampleRate) / buffer.sampleRate), targetSampleRate);
+        const offlineCtx = new OfflineAudioContext(effectiveTargetChannels, Math.ceil((samplesNeeded * effectiveTargetSampleRate) / buffer.sampleRate), effectiveTargetSampleRate);
         // Create source and connect
         const source = offlineCtx.createBufferSource();
         source.buffer = segmentBuffer;
@@ -101,7 +105,7 @@ export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRat
         const durationMs = Math.round((samplesNeeded / buffer.sampleRate) * 1000);
         logger?.debug('Final processed audio:', {
             outputSamples: channelData.length,
-            outputSampleRate: targetSampleRate,
+            outputSampleRate: effectiveTargetSampleRate,
             durationMs,
         });
         return {
@@ -109,7 +113,7 @@ export async function processAudioBuffer({ arrayBuffer, fileUri, targetSampleRat
             channelData,
             samples: channelData.length,
             durationMs,
-            sampleRate: targetSampleRate,
+            sampleRate: effectiveTargetSampleRate,
             channels: processedBuffer.numberOfChannels,
         };
     }

package/build/esm/utils/audioProcessing.js.map

@@ -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/types/errors/AudioStreamError.d.ts.map

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package/build/types/streamAudioData.d.ts

@@ -28,6 +28,11 @@ export interface StreamAudioDataOptions {
     maxChunkBytes?: number;
     /** Max chunks queued in native before JS ack pauses decode (default: 4). */
     maxBufferedChunks?: number;
+    /**
+     * Optional timeout for a chunk acknowledgement while backpressure is active.
+     * Undefined/0 disables timeout so long transcription callbacks can run.
+     */
+    backpressureTimeoutMs?: number;
     /** Output PCM format; only `'float32'` supported today. */
     streamFormat?: 'float32';
     /** Abort the in-flight request. Resolves promise with `cancelled: true`. */

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package/build/types/utils/audioProcessing.d.ts

@@ -2,8 +2,8 @@ import { ConsoleLike } from '../AudioStudio.types';
 export interface ProcessAudioBufferOptions {
     arrayBuffer?: ArrayBuffer;
     fileUri?: string;
-    targetSampleRate: number;
-    targetChannels: number;
+    targetSampleRate?: number;
+    targetChannels?: number;
     normalizeAudio: boolean;
     startTimeMs?: number;
     endTimeMs?: number;

package/build/types/utils/audioProcessing.d.ts.map

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+{"version":3,"file":"audioProcessing.d.ts","sourceRoot":"","sources":["../../../src/utils/audioProcessing.ts"],"names":[],"mappings":"AAGA,OAAO,EAAE,WAAW,EAAE,MAAM,sBAAsB,CAAA;AAElD,MAAM,WAAW,yBAAyB;IACtC,WAAW,CAAC,EAAE,WAAW,CAAA;IACzB,OAAO,CAAC,EAAE,MAAM,CAAA;IAChB,gBAAgB,CAAC,EAAE,MAAM,CAAA;IACzB,cAAc,CAAC,EAAE,MAAM,CAAA;IACvB,cAAc,EAAE,OAAO,CAAA;IACvB,WAAW,CAAC,EAAE,MAAM,CAAA;IACpB,SAAS,CAAC,EAAE,MAAM,CAAA;IAClB,QAAQ,CAAC,EAAE,MAAM,CAAA;IACjB,MAAM,CAAC,EAAE,MAAM,CAAA;IACf,YAAY,CAAC,EAAE,YAAY,CAAA;IAC3B,MAAM,CAAC,EAAE,WAAW,CAAA;CACvB;AAED,MAAM,WAAW,kBAAkB;IAC/B,WAAW,EAAE,YAAY,CAAA;IACzB,OAAO,EAAE,MAAM,CAAA;IACf,UAAU,EAAE,MAAM,CAAA;IAClB,UAAU,EAAE,MAAM,CAAA;IAClB,QAAQ,EAAE,MAAM,CAAA;IAChB,MAAM,EAAE,WAAW,CAAA;CACtB;AAED,wBAAsB,kBAAkB,CAAC,EACrC,WAAW,EACX,OAAO,EACP,gBAAgB,EAChB,cAAc,EACd,cAAc,EACd,WAAW,EACX,SAAS,EACT,QAAQ,EACR,MAAM,EACN,YAAY,EACZ,MAAM,GACT,EAAE,yBAAyB,GAAG,OAAO,CAAC,kBAAkB,CAAC,CA+KzD"}