@siteed/audio-studio 3.1.1 → 3.2.0

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

@@ -0,0 +1,25 @@
+/**
+ * Stable typed errors for `streamAudioData`. Callers can switch on `code`.
+ */
+export type AudioStreamErrorCode = 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT' | 'ERR_AUDIO_STREAM_INVALID_RANGE' | 'ERR_AUDIO_STREAM_DECODE_FAILED' | 'ERR_AUDIO_STREAM_CANCELLED' | 'ERR_AUDIO_STREAM_PERMISSION_DENIED' | 'ERR_AUDIO_STREAM_FILE_NOT_FOUND' | 'ERR_AUDIO_STREAM_BACKPRESSURE_TIMEOUT' | 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE' | 'ERR_AUDIO_STREAM_BUSY' | 'ERR_AUDIO_STREAM_UNKNOWN';
+export interface AudioStreamErrorPayload {
+    code: AudioStreamErrorCode;
+    message: string;
+    recoverable: boolean;
+    fileUri?: string;
+    platform?: string;
+    nativeCode?: string;
+    nativeMessage?: string;
+}
+export declare class AudioStreamError extends Error {
+    readonly code: AudioStreamErrorCode;
+    readonly recoverable: boolean;
+    readonly fileUri?: string;
+    readonly platform?: string;
+    readonly nativeCode?: string;
+    readonly nativeMessage?: string;
+    constructor(payload: AudioStreamErrorPayload);
+    toJSON(): AudioStreamErrorPayload;
+}
+export declare function mapStreamError(err: unknown, fileUri?: string, platform?: string): AudioStreamError;
+//# sourceMappingURL=AudioStreamError.d.ts.map

package/build/types/errors/AudioStreamError.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"AudioStreamError.d.ts","sourceRoot":"","sources":["../../../src/errors/AudioStreamError.ts"],"names":[],"mappings":"AAAA;;GAEG;AACH,MAAM,MAAM,oBAAoB,GAC1B,qCAAqC,GACrC,gCAAgC,GAChC,gCAAgC,GAChC,4BAA4B,GAC5B,oCAAoC,GACpC,iCAAiC,GACjC,uCAAuC,GACvC,qCAAqC,GACrC,uBAAuB,GACvB,0BAA0B,CAAA;AAEhC,MAAM,WAAW,uBAAuB;IACpC,IAAI,EAAE,oBAAoB,CAAA;IAC1B,OAAO,EAAE,MAAM,CAAA;IACf,WAAW,EAAE,OAAO,CAAA;IACpB,OAAO,CAAC,EAAE,MAAM,CAAA;IAChB,QAAQ,CAAC,EAAE,MAAM,CAAA;IACjB,UAAU,CAAC,EAAE,MAAM,CAAA;IACnB,aAAa,CAAC,EAAE,MAAM,CAAA;CACzB;AASD,qBAAa,gBAAiB,SAAQ,KAAK;IACvC,QAAQ,CAAC,IAAI,EAAE,oBAAoB,CAAA;IACnC,QAAQ,CAAC,WAAW,EAAE,OAAO,CAAA;IAC7B,QAAQ,CAAC,OAAO,CAAC,EAAE,MAAM,CAAA;IACzB,QAAQ,CAAC,QAAQ,CAAC,EAAE,MAAM,CAAA;IAC1B,QAAQ,CAAC,UAAU,CAAC,EAAE,MAAM,CAAA;IAC5B,QAAQ,CAAC,aAAa,CAAC,EAAE,MAAM,CAAA;gBAEnB,OAAO,EAAE,uBAAuB;IAW5C,MAAM,IAAI,uBAAuB;CAWpC;AAwFD,wBAAgB,cAAc,CAC1B,GAAG,EAAE,OAAO,EACZ,OAAO,CAAC,EAAE,MAAM,EAChB,QAAQ,CAAC,EAAE,MAAM,GAClB,gBAAgB,CA4ClB"}

package/build/types/errors/AudioStreamError.test.d.ts

@@ -0,0 +1,2 @@
+export {};
+//# sourceMappingURL=AudioStreamError.test.d.ts.map

package/build/types/errors/AudioStreamError.test.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"AudioStreamError.test.d.ts","sourceRoot":"","sources":["../../../src/errors/AudioStreamError.test.ts"],"names":[],"mappings":""}

package/build/types/index.d.ts

@@ -5,6 +5,7 @@ import { extractPreview } from './AudioAnalysis/extractPreview';
 import { initMelStreamingWasm, computeMelFrameWasm } from './AudioAnalysis/melSpectrogramWasm';
 import { AudioRecorderProvider, useSharedAudioRecorder } from './AudioRecorder.provider';
 import AudioStudioModule from './AudioStudioModule';
+import { getAudioDecodeCapabilities, streamAudioData } from './streamAudioData';
 import { trimAudio } from './trimAudio';
 import { useAudioRecorder } from './useAudioRecorder';
 export * from './utils/convertPCMToFloat32';
@@ -15,9 +16,12 @@ export { AudioDeviceManager, audioDeviceManager } from './AudioDeviceManager';
 export { useAudioDevices } from './hooks/useAudioDevices';
 export { setMelSpectrogramWasmUrl } from './AudioAnalysis/wasmConfig';
 export { extractPreviewBars } from './AudioAnalysis/extractPreviewBars';
-export { AudioRecorderProvider, AudioStudioModule, extractRawWavAnalysis, extractAudioAnalysis, extractPreview, trimAudio, extractAudioData, extractMelSpectrogram, initMelStreamingWasm, computeMelFrameWasm, MAX_DURATION_MS, useAudioRecorder, useSharedAudioRecorder, };
+export { AudioRecorderProvider, AudioStudioModule, extractRawWavAnalysis, extractAudioAnalysis, extractPreview, trimAudio, extractAudioData, streamAudioData, getAudioDecodeCapabilities, extractMelSpectrogram, initMelStreamingWasm, computeMelFrameWasm, MAX_DURATION_MS, useAudioRecorder, useSharedAudioRecorder, };
 export { AudioExtractionError, mapExtractionError, } from './errors/AudioExtractionError';
 export type { AudioExtractionErrorCode, AudioExtractionErrorPayload, } from './errors/AudioExtractionError';
+export { AudioStreamError, mapStreamError, } from './errors/AudioStreamError';
+export type { AudioStreamErrorCode, AudioStreamErrorPayload, } from './errors/AudioStreamError';
+export type { StreamAudioDataOptions, StreamAudioDataChunk, StreamAudioDataProgress, StreamAudioDataResult, StreamAudioDataCallbacks, AudioDecodeCapabilities, } from './streamAudioData';
 export type * from './AudioAnalysis/AudioAnalysis.types';
 export type * from './AudioStudio.types';
 /** @deprecated Use AudioStudioModule instead */

package/build/types/index.d.ts.map

@@ -1 +1 @@
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package/build/types/streamAudioData.d.ts

@@ -0,0 +1,119 @@
+/**
+ * High-level API: stream decoded audio from a stored file as bounded Float32
+ * chunks without materializing the full PCM range in memory.
+ *
+ * See `docs/STREAM_AUDIO_DATA.md` for the full contract and rollout notes.
+ */
+export interface StreamAudioDataOptions {
+    /** URI of the audio file to decode. */
+    fileUri: string;
+    /** Start time in milliseconds (default: 0). */
+    startTimeMs?: number;
+    /** End time in milliseconds (default: end-of-file). */
+    endTimeMs?: number;
+    /**
+     * Source sample rate hint. Ignored if `targetSampleRate` is set; native
+     * decoders read the actual rate from the file.
+     */
+    sampleRate?: number;
+    /** Output sample rate. Native resamples when this differs from the file. */
+    targetSampleRate?: number;
+    /** Output channel count (1 = mono downmix, 2 = stereo passthrough). */
+    channels?: number;
+    /** Clamp samples to [-1, 1] and replace non-finite values with 0. */
+    normalizeAudio?: boolean;
+    /** Target chunk duration in ms (default: 1000, min: 10, max: 60000). */
+    chunkDurationMs?: number;
+    /** Soft cap on chunk size in bytes (Float32 = 4 bytes/sample). */
+    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`. */
+    signal?: AbortSignal;
+}
+export interface StreamAudioDataChunk {
+    /** Native request id; constant across all chunks of one call. */
+    requestId: string;
+    /** Zero-based monotonic chunk index. */
+    chunkIndex: number;
+    /** Start time in output-rate ms (rounded to nearest sample). */
+    startTimeMs: number;
+    /** End time in output-rate ms. */
+    endTimeMs: number;
+    /** Duration in ms (`endTimeMs - startTimeMs`). */
+    durationMs: number;
+    /** First sample index in the output timeline. */
+    startSample: number;
+    /** Sample count in `samples` (interleaved if channels > 1). */
+    sampleCount: number;
+    /** Output sample rate. */
+    sampleRate: number;
+    /** Output channel count. */
+    channels: number;
+    /** Interleaved Float32 samples in [-1, 1]. */
+    samples: Float32Array;
+    /** True for the last chunk of a non-cancelled run. */
+    isFinal: boolean;
+}
+export interface StreamAudioDataProgress {
+    requestId: string;
+    processedMs: number;
+    durationMs: number;
+    progress: number;
+    emittedChunks: number;
+    bufferedChunks?: number;
+}
+export interface StreamAudioDataResult {
+    requestId: string;
+    durationMs: number;
+    sampleRate: number;
+    channels: number;
+    chunks: number;
+    samples: number;
+    cancelled: boolean;
+}
+export interface StreamAudioDataCallbacks {
+    /**
+     * Called with each decoded chunk. If this returns a Promise, native decode
+     * pauses until it resolves (backpressure). Throwing aborts the stream with
+     * `ERR_AUDIO_STREAM_DECODE_FAILED`.
+     */
+    onChunk: (chunk: StreamAudioDataChunk) => void | Promise<void>;
+    /** Called whenever native reports progress. */
+    onProgress?: (progress: StreamAudioDataProgress) => void;
+}
+export interface AudioDecodeCapabilities {
+    platform: 'ios' | 'android' | 'web';
+    supportedInputFormats: string[];
+    supportedOutputFormats: Array<'float32'>;
+    supportsCancellation: boolean;
+    supportsBackpressure: boolean;
+    supportsTimeRange: boolean;
+    supportsTargetSampleRate: boolean;
+    supportsChannelMixing: boolean;
+    knownLimitations?: string[];
+}
+/**
+ * Stream decoded audio from a stored file as bounded Float32 PCM chunks.
+ *
+ * Memory bound:
+ *   `chunkDurationMs * sampleRate * channels * 4 * maxBufferedChunks` +
+ *   native decoder buffers.
+ *
+ * Cancellation: pass `options.signal` and call `abort()`. The returned promise
+ * resolves with `cancelled: true` (it does not reject) when cancellation wins.
+ *
+ * Backpressure: if `onChunk` returns a Promise, native decode is paused until
+ * it resolves; if it throws, the stream is aborted with a `decode_failed` error.
+ */
+export declare function streamAudioData(options: StreamAudioDataOptions, callbacks: StreamAudioDataCallbacks): Promise<StreamAudioDataResult>;
+/** Discover what the running platform supports. */
+export declare function getAudioDecodeCapabilities(): Promise<AudioDecodeCapabilities>;
+//# sourceMappingURL=streamAudioData.d.ts.map

package/build/types/streamAudioData.d.ts.map

@@ -0,0 +1 @@
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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

@@ -1 +1 @@
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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"}

package/ios/AudioProcessingHelpers.swift

@@ -681,15 +681,20 @@ func extractRawAudioData(
         for channel in 0..<channels {
             let sample = floatData[channel][frame]
             
-            let normalizedSample = decodingConfig.normalizeAudio ? 
-                max(-1.0, min(1.0, sample)) : sample
-            
+            // Sanitize: replace NaN/Inf with 0 then clamp. Skip clamp when
+            // normalizeAudio=false so callers see the raw decoded magnitude,
+            // but always finite-check to avoid Swift's `Int16(_:)` /
+            // `Int32(_:)` trap on non-finite values.
+            let safeSample: Float = sample.isFinite ? sample : 0
+            let normalizedSample = decodingConfig.normalizeAudio ?
+                max(-1.0, min(1.0, safeSample)) : safeSample
+
             switch targetBitDepth {
             case 16:
-                let intValue = Int16(normalizedSample * Float(Int16.max))
+                let intValue = safeFloatToInt16(normalizedSample)
                 pcmData.append(contentsOf: withUnsafeBytes(of: intValue) { Array($0) })
             case 32:
-                let intValue = Int32(normalizedSample * Float(Int32.max))
+                let intValue = safeFloatToInt32(normalizedSample)
                 pcmData.append(contentsOf: withUnsafeBytes(of: intValue) { Array($0) })
             default:
                 throw NSError(domain: "AudioProcessing", code: -1, userInfo: [NSLocalizedDescriptionKey: "Unsupported bit depth \(targetBitDepth)"])