@siteed/audio-studio 3.1.0 → 3.2.0-beta.1

package/build/cjs/streamAudioData.js

@@ -0,0 +1,467 @@
+"use strict";
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.streamAudioData = streamAudioData;
+exports.getAudioDecodeCapabilities = getAudioDecodeCapabilities;
+const expo_modules_core_1 = require("expo-modules-core");
+const AudioStudioModule_1 = __importDefault(require("./AudioStudioModule"));
+const constants_1 = require("./constants");
+const AudioStreamError_1 = require("./errors/AudioStreamError");
+const audioProcessing_1 = require("./utils/audioProcessing");
+const CHUNK_EVENT = 'AudioDataStreamChunk';
+const PROGRESS_EVENT = 'AudioDataStreamProgress';
+const COMPLETE_EVENT = 'AudioDataStreamComplete';
+const ERROR_EVENT = 'AudioDataStreamError';
+let cachedEmitter = null;
+function getEmitter() {
+    if (!cachedEmitter) {
+        cachedEmitter = new expo_modules_core_1.LegacyEventEmitter(AudioStudioModule_1.default);
+    }
+    return cachedEmitter;
+}
+function generateRequestId() {
+    const g = globalThis;
+    if (typeof g.crypto?.randomUUID === 'function') {
+        try {
+            return g.crypto.randomUUID();
+        }
+        catch {
+            // fall through
+        }
+    }
+    return `asd_${Date.now().toString(36)}_${Math.random()
+        .toString(36)
+        .slice(2, 10)}`;
+}
+function toFloat32(samples) {
+    if (samples instanceof Float32Array)
+        return samples;
+    if (Array.isArray(samples)) {
+        const out = new Float32Array(samples.length);
+        for (let i = 0; i < samples.length; i++) {
+            const v = Number(samples[i]);
+            out[i] = Number.isFinite(v) ? v : 0;
+        }
+        return out;
+    }
+    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);
+    }
+    if (samples && typeof samples === 'object' && 'length' in samples) {
+        // ArrayLike fallback
+        const arr = samples;
+        const out = new Float32Array(arr.length);
+        for (let i = 0; i < arr.length; i++) {
+            const v = Number(arr[i]);
+            out[i] = Number.isFinite(v) ? v : 0;
+        }
+        return out;
+    }
+    return new Float32Array(0);
+}
+function base64ToBytes(input) {
+    const g = globalThis;
+    if (typeof g.atob !== 'function') {
+        // Buffer path for environments without atob; React Native has atob.
+        const Buf = globalThis.Buffer;
+        if (Buf)
+            return new Uint8Array(Buf.from(input, 'base64'));
+        return new Uint8Array(0);
+    }
+    const bin = g.atob(input);
+    const out = new Uint8Array(bin.length);
+    for (let i = 0; i < bin.length; i++)
+        out[i] = bin.charCodeAt(i);
+    return out;
+}
+function validateOptions(options) {
+    if (!options.fileUri) {
+        throw new AudioStreamError_1.AudioStreamError({
+            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
+            message: 'fileUri is required',
+            recoverable: false,
+        });
+    }
+    if (options.startTimeMs !== undefined &&
+        options.endTimeMs !== undefined &&
+        options.startTimeMs >= options.endTimeMs) {
+        throw new AudioStreamError_1.AudioStreamError({
+            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
+            message: 'startTimeMs must be < endTimeMs',
+            recoverable: false,
+        });
+    }
+    if (options.chunkDurationMs !== undefined &&
+        (options.chunkDurationMs < 10 || options.chunkDurationMs > 60000)) {
+        throw new AudioStreamError_1.AudioStreamError({
+            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',
+            message: 'chunkDurationMs must be in [10, 60000]',
+            recoverable: false,
+        });
+    }
+    if (options.streamFormat !== undefined &&
+        options.streamFormat !== 'float32') {
+        throw new AudioStreamError_1.AudioStreamError({
+            code: 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',
+            message: `Unsupported streamFormat: ${options.streamFormat}`,
+            recoverable: false,
+        });
+    }
+}
+/**
+ * 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.
+ */
+async function streamAudioData(options, callbacks) {
+    validateOptions(options);
+    if (constants_1.isWeb) {
+        return streamAudioDataWeb(options, callbacks);
+    }
+    return streamAudioDataNative(options, callbacks);
+}
+/** Discover what the running platform supports. */
+async function getAudioDecodeCapabilities() {
+    if (constants_1.isWeb) {
+        return {
+            platform: 'web',
+            supportedInputFormats: [
+                'audio/wav',
+                'audio/mpeg',
+                'audio/mp4',
+                'audio/aac',
+                'audio/ogg',
+                'audio/webm',
+            ],
+            supportedOutputFormats: ['float32'],
+            supportsCancellation: true,
+            supportsBackpressure: true,
+            supportsTimeRange: true,
+            supportsTargetSampleRate: true,
+            supportsChannelMixing: true,
+            knownLimitations: [
+                'Web decodes the entire file via AudioContext.decodeAudioData before chunking; very long files may exceed browser memory.',
+            ],
+        };
+    }
+    if (typeof AudioStudioModule_1.default.getAudioDecodeCapabilities === 'function') {
+        try {
+            const caps = await AudioStudioModule_1.default.getAudioDecodeCapabilities();
+            return caps;
+        }
+        catch (err) {
+            throw (0, AudioStreamError_1.mapStreamError)(err, undefined, 'native');
+        }
+    }
+    throw new AudioStreamError_1.AudioStreamError({
+        code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',
+        message: 'getAudioDecodeCapabilities is not available on this build',
+        recoverable: false,
+    });
+}
+async function streamAudioDataNative(options, callbacks) {
+    if (typeof AudioStudioModule_1.default.streamAudioData !== 'function') {
+        throw new AudioStreamError_1.AudioStreamError({
+            code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',
+            message: 'streamAudioData native method missing — rebuild the host app with the latest @siteed/audio-studio.',
+            recoverable: false,
+        });
+    }
+    const requestId = generateRequestId();
+    const emitter = getEmitter();
+    const subs = [];
+    let chunkCount = 0;
+    let sampleCount = 0;
+    let processingChain = Promise.resolve();
+    let settled = false;
+    let abortListener = null;
+    const finalize = () => {
+        for (const sub of subs) {
+            try {
+                sub.remove();
+            }
+            catch {
+                /* noop */
+            }
+        }
+        subs.length = 0;
+        if (abortListener && options.signal) {
+            options.signal.removeEventListener('abort', abortListener);
+        }
+    };
+    return new Promise((resolve, reject) => {
+        const settle = (fn, mode, value) => {
+            if (settled)
+                return;
+            settled = true;
+            finalize();
+            fn();
+            if (mode === 'resolve') {
+                resolve(value);
+            }
+            else {
+                reject(value);
+            }
+        };
+        const handleError = (err) => {
+            settle(() => {
+                try {
+                    AudioStudioModule_1.default.cancelStreamAudioData?.(requestId);
+                }
+                catch {
+                    /* noop */
+                }
+            }, 'reject', (0, AudioStreamError_1.mapStreamError)(err, options.fileUri, 'native'));
+        };
+        subs.push(emitter.addListener(CHUNK_EVENT, (raw) => {
+            const evt = raw;
+            if (evt.requestId !== requestId)
+                return;
+            const chunk = {
+                requestId: evt.requestId,
+                chunkIndex: evt.chunkIndex,
+                startTimeMs: evt.startTimeMs,
+                endTimeMs: evt.endTimeMs,
+                durationMs: evt.endTimeMs - evt.startTimeMs,
+                startSample: evt.startSample,
+                sampleCount: evt.sampleCount,
+                sampleRate: evt.sampleRate,
+                channels: evt.channels,
+                samples: toFloat32(evt.samples),
+                isFinal: evt.isFinal,
+            };
+            chunkCount += 1;
+            sampleCount += chunk.sampleCount;
+            processingChain = processingChain
+                .then(async () => {
+                await callbacks.onChunk(chunk);
+                try {
+                    AudioStudioModule_1.default.acknowledgeStreamAudioChunk?.(requestId, chunk.chunkIndex);
+                }
+                catch {
+                    /* noop */
+                }
+            })
+                .catch(handleError);
+        }));
+        if (callbacks.onProgress) {
+            subs.push(emitter.addListener(PROGRESS_EVENT, (raw) => {
+                const evt = raw;
+                if (evt.requestId !== requestId)
+                    return;
+                callbacks.onProgress(evt);
+            }));
+        }
+        subs.push(emitter.addListener(COMPLETE_EVENT, (raw) => {
+            const evt = raw;
+            if (evt.requestId !== requestId)
+                return;
+            // Wait for in-flight onChunk callbacks before resolving.
+            processingChain
+                .then(() => {
+                settle(() => { }, 'resolve', {
+                    requestId,
+                    durationMs: evt.durationMs,
+                    sampleRate: evt.sampleRate,
+                    channels: evt.channels,
+                    chunks: evt.chunks ?? chunkCount,
+                    samples: evt.samples ?? sampleCount,
+                    cancelled: evt.cancelled,
+                });
+            })
+                .catch(handleError);
+        }));
+        subs.push(emitter.addListener(ERROR_EVENT, (raw) => {
+            const evt = raw;
+            if (evt.requestId !== requestId)
+                return;
+            if (evt.code === 'ERR_AUDIO_STREAM_CANCELLED') {
+                processingChain
+                    .catch(() => { })
+                    .then(() => {
+                    settle(() => { }, 'resolve', {
+                        requestId,
+                        durationMs: 0,
+                        sampleRate: options.targetSampleRate ??
+                            options.sampleRate ??
+                            0,
+                        channels: options.channels ?? 1,
+                        chunks: chunkCount,
+                        samples: sampleCount,
+                        cancelled: true,
+                    });
+                });
+                return;
+            }
+            handleError(new AudioStreamError_1.AudioStreamError({
+                code: evt.code ??
+                    'ERR_AUDIO_STREAM_UNKNOWN',
+                message: evt.message ?? 'native stream error',
+                nativeMessage: evt.nativeMessage,
+                fileUri: options.fileUri,
+                platform: 'native',
+                recoverable: false,
+            }));
+        }));
+        if (options.signal) {
+            if (options.signal.aborted) {
+                settle(() => { }, 'resolve', {
+                    requestId,
+                    durationMs: 0,
+                    sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,
+                    channels: options.channels ?? 1,
+                    chunks: 0,
+                    samples: 0,
+                    cancelled: true,
+                });
+                return;
+            }
+            abortListener = () => {
+                try {
+                    AudioStudioModule_1.default.cancelStreamAudioData?.(requestId);
+                }
+                catch {
+                    /* noop */
+                }
+            };
+            options.signal.addEventListener('abort', abortListener);
+        }
+        const { signal: _signal, ...nativeOptions } = options;
+        AudioStudioModule_1.default.streamAudioData({
+            ...nativeOptions,
+            requestId,
+            streamFormat: options.streamFormat ?? 'float32',
+            chunkDurationMs: options.chunkDurationMs ?? 1000,
+            maxBufferedChunks: options.maxBufferedChunks ?? 4,
+            normalizeAudio: options.normalizeAudio ?? true,
+        }).catch(handleError);
+    });
+}
+async function streamAudioDataWeb(options, callbacks) {
+    const requestId = generateRequestId();
+    const cancelled = () => options.signal?.aborted === true;
+    if (cancelled()) {
+        return {
+            requestId,
+            durationMs: 0,
+            sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,
+            channels: options.channels ?? 1,
+            chunks: 0,
+            samples: 0,
+            cancelled: true,
+        };
+    }
+    try {
+        const processed = await (0, audioProcessing_1.processAudioBuffer)({
+            fileUri: options.fileUri,
+            targetSampleRate: options.targetSampleRate ?? 16000,
+            targetChannels: options.channels ?? 1,
+            normalizeAudio: options.normalizeAudio ?? true,
+            startTimeMs: options.startTimeMs,
+            endTimeMs: options.endTimeMs,
+        });
+        const sampleRate = processed.sampleRate;
+        const channels = processed.channels;
+        const durationMs = processed.durationMs;
+        const chunkDurationMs = options.chunkDurationMs ?? 1000;
+        let samplesPerChunk = Math.max(1, Math.floor((chunkDurationMs / 1000) * sampleRate) * channels);
+        if (options.maxChunkBytes) {
+            const maxSamples = Math.floor(options.maxChunkBytes / 4);
+            samplesPerChunk = Math.min(samplesPerChunk, maxSamples);
+        }
+        const all = sanitizeFloat32(processed.channelData, options.normalizeAudio ?? true);
+        let chunkIndex = 0;
+        let emittedSamples = 0;
+        for (let off = 0; off < all.length; off += samplesPerChunk) {
+            if (cancelled()) {
+                return {
+                    requestId,
+                    durationMs,
+                    sampleRate,
+                    channels,
+                    chunks: chunkIndex,
+                    samples: emittedSamples,
+                    cancelled: true,
+                };
+            }
+            const end = Math.min(off + samplesPerChunk, all.length);
+            const slice = all.slice(off, end);
+            const startSample = off / channels;
+            const endSample = end / channels;
+            const chunk = {
+                requestId,
+                chunkIndex,
+                startTimeMs: Math.round((startSample / sampleRate) * 1000),
+                endTimeMs: Math.round((endSample / sampleRate) * 1000),
+                durationMs: Math.round(((endSample - startSample) / sampleRate) * 1000),
+                startSample,
+                sampleCount: slice.length,
+                sampleRate,
+                channels,
+                samples: slice,
+                isFinal: end >= all.length,
+            };
+            await callbacks.onChunk(chunk);
+            callbacks.onProgress?.({
+                requestId,
+                processedMs: chunk.endTimeMs,
+                durationMs,
+                progress: durationMs > 0 ? chunk.endTimeMs / durationMs : 1,
+                emittedChunks: chunkIndex + 1,
+            });
+            chunkIndex += 1;
+            emittedSamples += slice.length;
+        }
+        return {
+            requestId,
+            durationMs,
+            sampleRate,
+            channels,
+            chunks: chunkIndex,
+            samples: emittedSamples,
+            cancelled: false,
+        };
+    }
+    catch (err) {
+        throw (0, AudioStreamError_1.mapStreamError)(err, options.fileUri, 'web');
+    }
+}
+function sanitizeFloat32(input, clamp) {
+    if (!clamp) {
+        // still need NaN/Inf sanitation
+        for (let i = 0; i < input.length; i++) {
+            const v = input[i];
+            if (!Number.isFinite(v))
+                input[i] = 0;
+        }
+        return input;
+    }
+    for (let i = 0; i < input.length; i++) {
+        const v = input[i];
+        if (!Number.isFinite(v)) {
+            input[i] = 0;
+        }
+        else if (v > 1) {
+            input[i] = 1;
+        }
+        else if (v < -1) {
+            input[i] = -1;
+        }
+    }
+    return input;
+}
+//# sourceMappingURL=streamAudioData.js.map

package/build/cjs/streamAudioData.js.map

@@ -0,0 +1 @@
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{ LegacyEventEmitter, type EventSubscription } from 'expo-modules-core'\n\nimport AudioStudioModule from './AudioStudioModule'\nimport { isWeb } from './constants'\nimport {\n    AudioStreamError,\n    AudioStreamErrorCode,\n    mapStreamError,\n} from './errors/AudioStreamError'\nimport { processAudioBuffer } from './utils/audioProcessing'\n\n/**\n * High-level API: stream decoded audio from a stored file as bounded Float32\n * chunks without materializing the full PCM range in memory.\n *\n * See `docs/STREAM_AUDIO_DATA.md` for the full contract and rollout notes.\n */\nexport interface StreamAudioDataOptions {\n    /** URI of the audio file to decode. */\n    fileUri: string\n    /** Start time in milliseconds (default: 0). */\n    startTimeMs?: number\n    /** End time in milliseconds (default: end-of-file). */\n    endTimeMs?: number\n    /**\n     * Source sample rate hint. Ignored if `targetSampleRate` is set; native\n     * decoders read the actual rate from the file.\n     */\n    sampleRate?: number\n    /** Output sample rate. Native resamples when this differs from the file. */\n    targetSampleRate?: number\n    /** Output channel count (1 = mono downmix, 2 = stereo passthrough). */\n    channels?: number\n    /** Clamp samples to [-1, 1] and replace non-finite values with 0. */\n    normalizeAudio?: boolean\n    /** Target chunk duration in ms (default: 1000, min: 10, max: 60000). */\n    chunkDurationMs?: number\n    /** Soft cap on chunk size in bytes (Float32 = 4 bytes/sample). */\n    maxChunkBytes?: number\n    /** Max chunks queued in native before JS ack pauses decode (default: 4). */\n    maxBufferedChunks?: number\n    /** Output PCM format; only `'float32'` supported today. */\n    streamFormat?: 'float32'\n    /** Abort the in-flight request. Resolves promise with `cancelled: true`. */\n    signal?: AbortSignal\n}\n\nexport interface StreamAudioDataChunk {\n    /** Native request id; constant across all chunks of one call. */\n    requestId: string\n    /** Zero-based monotonic chunk index. */\n    chunkIndex: number\n    /** Start time in output-rate ms (rounded to nearest sample). */\n    startTimeMs: number\n    /** End time in output-rate ms. */\n    endTimeMs: number\n    /** Duration in ms (`endTimeMs - startTimeMs`). */\n    durationMs: number\n    /** First sample index in the output timeline. */\n    startSample: number\n    /** Sample count in `samples` (interleaved if channels > 1). */\n    sampleCount: number\n    /** Output sample rate. */\n    sampleRate: number\n    /** Output channel count. */\n    channels: number\n    /** Interleaved Float32 samples in [-1, 1]. */\n    samples: Float32Array\n    /** True for the last chunk of a non-cancelled run. */\n    isFinal: boolean\n}\n\nexport interface StreamAudioDataProgress {\n    requestId: string\n    processedMs: number\n    durationMs: number\n    progress: number\n    emittedChunks: number\n    bufferedChunks?: number\n}\n\nexport interface StreamAudioDataResult {\n    requestId: string\n    durationMs: number\n    sampleRate: number\n    channels: number\n    chunks: number\n    samples: number\n    cancelled: boolean\n}\n\nexport interface StreamAudioDataCallbacks {\n    /**\n     * Called with each decoded chunk. If this returns a Promise, native decode\n     * pauses until it resolves (backpressure). Throwing aborts the stream with\n     * `ERR_AUDIO_STREAM_DECODE_FAILED`.\n     */\n    onChunk: (chunk: StreamAudioDataChunk) => void | Promise<void>\n    /** Called whenever native reports progress. */\n    onProgress?: (progress: StreamAudioDataProgress) => void\n}\n\nexport interface AudioDecodeCapabilities {\n    platform: 'ios' | 'android' | 'web'\n    supportedInputFormats: string[]\n    supportedOutputFormats: Array<'float32'>\n    supportsCancellation: boolean\n    supportsBackpressure: boolean\n    supportsTimeRange: boolean\n    supportsTargetSampleRate: boolean\n    supportsChannelMixing: boolean\n    knownLimitations?: string[]\n}\n\nconst CHUNK_EVENT = 'AudioDataStreamChunk'\nconst PROGRESS_EVENT = 'AudioDataStreamProgress'\nconst COMPLETE_EVENT = 'AudioDataStreamComplete'\nconst ERROR_EVENT = 'AudioDataStreamError'\n\nlet cachedEmitter: LegacyEventEmitter | null = null\nfunction getEmitter(): LegacyEventEmitter {\n    if (!cachedEmitter) {\n        cachedEmitter = new LegacyEventEmitter(AudioStudioModule)\n    }\n    return cachedEmitter\n}\n\nfunction generateRequestId(): string {\n    const g = globalThis as { crypto?: { randomUUID?: () => string } }\n    if (typeof g.crypto?.randomUUID === 'function') {\n        try {\n            return g.crypto.randomUUID()\n        } catch {\n            // fall through\n        }\n    }\n    return `asd_${Date.now().toString(36)}_${Math.random()\n        .toString(36)\n        .slice(2, 10)}`\n}\n\nfunction toFloat32(samples: unknown): Float32Array {\n    if (samples instanceof Float32Array) return samples\n    if (Array.isArray(samples)) {\n        const out = new Float32Array(samples.length)\n        for (let i = 0; i < samples.length; i++) {\n            const v = Number(samples[i])\n            out[i] = Number.isFinite(v) ? v : 0\n        }\n        return out\n    }\n    if (typeof samples === 'string') {\n        const bytes = base64ToBytes(samples)\n        const aligned =\n            bytes.byteOffset % 4 === 0\n                ? new Float32Array(\n                      bytes.buffer,\n                      bytes.byteOffset,\n                      bytes.byteLength / 4\n                  )\n                : new Float32Array(bytes.buffer.slice(bytes.byteOffset))\n        return new Float32Array(aligned)\n    }\n    if (samples && typeof samples === 'object' && 'length' in samples) {\n        // ArrayLike fallback\n        const arr = samples as ArrayLike<number>\n        const out = new Float32Array(arr.length)\n        for (let i = 0; i < arr.length; i++) {\n            const v = Number(arr[i])\n            out[i] = Number.isFinite(v) ? v : 0\n        }\n        return out\n    }\n    return new Float32Array(0)\n}\n\nfunction base64ToBytes(input: string): Uint8Array {\n    const g = globalThis as { atob?: (s: string) => string }\n    if (typeof g.atob !== 'function') {\n        // Buffer path for environments without atob; React Native has atob.\n        const Buf = (globalThis as { Buffer?: { from: Function } }).Buffer\n        if (Buf) return new Uint8Array(Buf.from(input, 'base64'))\n        return new Uint8Array(0)\n    }\n    const bin = g.atob(input)\n    const out = new Uint8Array(bin.length)\n    for (let i = 0; i < bin.length; i++) out[i] = bin.charCodeAt(i)\n    return out\n}\n\nfunction validateOptions(options: StreamAudioDataOptions): void {\n    if (!options.fileUri) {\n        throw new AudioStreamError({\n            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n            message: 'fileUri is required',\n            recoverable: false,\n        })\n    }\n    if (\n        options.startTimeMs !== undefined &&\n        options.endTimeMs !== undefined &&\n        options.startTimeMs >= options.endTimeMs\n    ) {\n        throw new AudioStreamError({\n            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n            message: 'startTimeMs must be < endTimeMs',\n            recoverable: false,\n        })\n    }\n    if (\n        options.chunkDurationMs !== undefined &&\n        (options.chunkDurationMs < 10 || options.chunkDurationMs > 60000)\n    ) {\n        throw new AudioStreamError({\n            code: 'ERR_AUDIO_STREAM_INVALID_RANGE',\n            message: 'chunkDurationMs must be in [10, 60000]',\n            recoverable: false,\n        })\n    }\n    if (\n        options.streamFormat !== undefined &&\n        options.streamFormat !== 'float32'\n    ) {\n        throw new AudioStreamError({\n            code: 'ERR_AUDIO_STREAM_UNSUPPORTED_FORMAT',\n            message: `Unsupported streamFormat: ${options.streamFormat}`,\n            recoverable: false,\n        })\n    }\n}\n\n/**\n * Stream decoded audio from a stored file as bounded Float32 PCM chunks.\n *\n * Memory bound:\n *   `chunkDurationMs * sampleRate * channels * 4 * maxBufferedChunks` +\n *   native decoder buffers.\n *\n * Cancellation: pass `options.signal` and call `abort()`. The returned promise\n * resolves with `cancelled: true` (it does not reject) when cancellation wins.\n *\n * Backpressure: if `onChunk` returns a Promise, native decode is paused until\n * it resolves; if it throws, the stream is aborted with a `decode_failed` error.\n */\nexport async function streamAudioData(\n    options: StreamAudioDataOptions,\n    callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n    validateOptions(options)\n\n    if (isWeb) {\n        return streamAudioDataWeb(options, callbacks)\n    }\n\n    return streamAudioDataNative(options, callbacks)\n}\n\n/** Discover what the running platform supports. */\nexport async function getAudioDecodeCapabilities(): Promise<AudioDecodeCapabilities> {\n    if (isWeb) {\n        return {\n            platform: 'web',\n            supportedInputFormats: [\n                'audio/wav',\n                'audio/mpeg',\n                'audio/mp4',\n                'audio/aac',\n                'audio/ogg',\n                'audio/webm',\n            ],\n            supportedOutputFormats: ['float32'],\n            supportsCancellation: true,\n            supportsBackpressure: true,\n            supportsTimeRange: true,\n            supportsTargetSampleRate: true,\n            supportsChannelMixing: true,\n            knownLimitations: [\n                'Web decodes the entire file via AudioContext.decodeAudioData before chunking; very long files may exceed browser memory.',\n            ],\n        }\n    }\n    if (typeof AudioStudioModule.getAudioDecodeCapabilities === 'function') {\n        try {\n            const caps = await AudioStudioModule.getAudioDecodeCapabilities()\n            return caps as AudioDecodeCapabilities\n        } catch (err) {\n            throw mapStreamError(err, undefined, 'native')\n        }\n    }\n    throw new AudioStreamError({\n        code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n        message: 'getAudioDecodeCapabilities is not available on this build',\n        recoverable: false,\n    })\n}\n\nasync function streamAudioDataNative(\n    options: StreamAudioDataOptions,\n    callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n    if (typeof AudioStudioModule.streamAudioData !== 'function') {\n        throw new AudioStreamError({\n            code: 'ERR_AUDIO_STREAM_NATIVE_UNAVAILABLE',\n            message:\n                'streamAudioData native method missing — rebuild the host app with the latest @siteed/audio-studio.',\n            recoverable: false,\n        })\n    }\n\n    const requestId = generateRequestId()\n    const emitter = getEmitter()\n    const subs: EventSubscription[] = []\n    let chunkCount = 0\n    let sampleCount = 0\n    let processingChain: Promise<void> = Promise.resolve()\n    let settled = false\n    let abortListener: (() => void) | null = null\n\n    const finalize = () => {\n        for (const sub of subs) {\n            try {\n                sub.remove()\n            } catch {\n                /* noop */\n            }\n        }\n        subs.length = 0\n        if (abortListener && options.signal) {\n            options.signal.removeEventListener('abort', abortListener)\n        }\n    }\n\n    return new Promise<StreamAudioDataResult>((resolve, reject) => {\n        const settle = (\n            fn: () => void,\n            mode: 'resolve' | 'reject',\n            value: unknown\n        ) => {\n            if (settled) return\n            settled = true\n            finalize()\n            fn()\n            if (mode === 'resolve') {\n                resolve(value as StreamAudioDataResult)\n            } else {\n                reject(value as Error)\n            }\n        }\n\n        const handleError = (err: unknown) => {\n            settle(\n                () => {\n                    try {\n                        AudioStudioModule.cancelStreamAudioData?.(requestId)\n                    } catch {\n                        /* noop */\n                    }\n                },\n                'reject',\n                mapStreamError(err, options.fileUri, 'native')\n            )\n        }\n\n        subs.push(\n            emitter.addListener(CHUNK_EVENT, (raw: unknown) => {\n                const evt = raw as {\n                    requestId: string\n                    chunkIndex: number\n                    startTimeMs: number\n                    endTimeMs: number\n                    startSample: number\n                    sampleCount: number\n                    sampleRate: number\n                    channels: number\n                    samples: Float32Array | number[] | string\n                    isFinal: boolean\n                }\n                if (evt.requestId !== requestId) return\n                const chunk: StreamAudioDataChunk = {\n                    requestId: evt.requestId,\n                    chunkIndex: evt.chunkIndex,\n                    startTimeMs: evt.startTimeMs,\n                    endTimeMs: evt.endTimeMs,\n                    durationMs: evt.endTimeMs - evt.startTimeMs,\n                    startSample: evt.startSample,\n                    sampleCount: evt.sampleCount,\n                    sampleRate: evt.sampleRate,\n                    channels: evt.channels,\n                    samples: toFloat32(evt.samples),\n                    isFinal: evt.isFinal,\n                }\n                chunkCount += 1\n                sampleCount += chunk.sampleCount\n\n                processingChain = processingChain\n                    .then(async () => {\n                        await callbacks.onChunk(chunk)\n                        try {\n                            AudioStudioModule.acknowledgeStreamAudioChunk?.(\n                                requestId,\n                                chunk.chunkIndex\n                            )\n                        } catch {\n                            /* noop */\n                        }\n                    })\n                    .catch(handleError)\n            })\n        )\n\n        if (callbacks.onProgress) {\n            subs.push(\n                emitter.addListener(PROGRESS_EVENT, (raw: unknown) => {\n                    const evt = raw as StreamAudioDataProgress\n                    if (evt.requestId !== requestId) return\n                    callbacks.onProgress!(evt)\n                })\n            )\n        }\n\n        subs.push(\n            emitter.addListener(COMPLETE_EVENT, (raw: unknown) => {\n                const evt = raw as {\n                    requestId: string\n                    durationMs: number\n                    sampleRate: number\n                    channels: number\n                    chunks?: number\n                    samples?: number\n                    cancelled: boolean\n                }\n                if (evt.requestId !== requestId) return\n                // Wait for in-flight onChunk callbacks before resolving.\n                processingChain\n                    .then(() => {\n                        settle(() => {}, 'resolve', {\n                            requestId,\n                            durationMs: evt.durationMs,\n                            sampleRate: evt.sampleRate,\n                            channels: evt.channels,\n                            chunks: evt.chunks ?? chunkCount,\n                            samples: evt.samples ?? sampleCount,\n                            cancelled: evt.cancelled,\n                        } satisfies StreamAudioDataResult)\n                    })\n                    .catch(handleError)\n            })\n        )\n\n        subs.push(\n            emitter.addListener(ERROR_EVENT, (raw: unknown) => {\n                const evt = raw as {\n                    requestId: string\n                    code?: string\n                    message?: string\n                    nativeMessage?: string\n                }\n                if (evt.requestId !== requestId) return\n                if (evt.code === 'ERR_AUDIO_STREAM_CANCELLED') {\n                    processingChain\n                        .catch(() => {})\n                        .then(() => {\n                            settle(() => {}, 'resolve', {\n                                requestId,\n                                durationMs: 0,\n                                sampleRate:\n                                    options.targetSampleRate ??\n                                    options.sampleRate ??\n                                    0,\n                                channels: options.channels ?? 1,\n                                chunks: chunkCount,\n                                samples: sampleCount,\n                                cancelled: true,\n                            } satisfies StreamAudioDataResult)\n                        })\n                    return\n                }\n                handleError(\n                    new AudioStreamError({\n                        code:\n                            (evt.code as AudioStreamErrorCode) ??\n                            'ERR_AUDIO_STREAM_UNKNOWN',\n                        message: evt.message ?? 'native stream error',\n                        nativeMessage: evt.nativeMessage,\n                        fileUri: options.fileUri,\n                        platform: 'native',\n                        recoverable: false,\n                    })\n                )\n            })\n        )\n\n        if (options.signal) {\n            if (options.signal.aborted) {\n                settle(() => {}, 'resolve', {\n                    requestId,\n                    durationMs: 0,\n                    sampleRate:\n                        options.targetSampleRate ?? options.sampleRate ?? 0,\n                    channels: options.channels ?? 1,\n                    chunks: 0,\n                    samples: 0,\n                    cancelled: true,\n                } satisfies StreamAudioDataResult)\n                return\n            }\n            abortListener = () => {\n                try {\n                    AudioStudioModule.cancelStreamAudioData?.(requestId)\n                } catch {\n                    /* noop */\n                }\n            }\n            options.signal.addEventListener('abort', abortListener)\n        }\n\n        const { signal: _signal, ...nativeOptions } = options\n        AudioStudioModule.streamAudioData({\n            ...nativeOptions,\n            requestId,\n            streamFormat: options.streamFormat ?? 'float32',\n            chunkDurationMs: options.chunkDurationMs ?? 1000,\n            maxBufferedChunks: options.maxBufferedChunks ?? 4,\n            normalizeAudio: options.normalizeAudio ?? true,\n        }).catch(handleError)\n    })\n}\n\nasync function streamAudioDataWeb(\n    options: StreamAudioDataOptions,\n    callbacks: StreamAudioDataCallbacks\n): Promise<StreamAudioDataResult> {\n    const requestId = generateRequestId()\n    const cancelled = () => options.signal?.aborted === true\n    if (cancelled()) {\n        return {\n            requestId,\n            durationMs: 0,\n            sampleRate: options.targetSampleRate ?? options.sampleRate ?? 0,\n            channels: options.channels ?? 1,\n            chunks: 0,\n            samples: 0,\n            cancelled: true,\n        }\n    }\n\n    try {\n        const processed = await processAudioBuffer({\n            fileUri: options.fileUri,\n            targetSampleRate: options.targetSampleRate ?? 16000,\n            targetChannels: options.channels ?? 1,\n            normalizeAudio: options.normalizeAudio ?? true,\n            startTimeMs: options.startTimeMs,\n            endTimeMs: options.endTimeMs,\n        })\n\n        const sampleRate = processed.sampleRate\n        const channels = processed.channels\n        const durationMs = processed.durationMs\n        const chunkDurationMs = options.chunkDurationMs ?? 1000\n        let samplesPerChunk = Math.max(\n            1,\n            Math.floor((chunkDurationMs / 1000) * sampleRate) * channels\n        )\n        if (options.maxChunkBytes) {\n            const maxSamples = Math.floor(options.maxChunkBytes / 4)\n            samplesPerChunk = Math.min(samplesPerChunk, maxSamples)\n        }\n\n        const all = sanitizeFloat32(processed.channelData, options.normalizeAudio ?? true)\n\n        let chunkIndex = 0\n        let emittedSamples = 0\n        for (let off = 0; off < all.length; off += samplesPerChunk) {\n            if (cancelled()) {\n                return {\n                    requestId,\n                    durationMs,\n                    sampleRate,\n                    channels,\n                    chunks: chunkIndex,\n                    samples: emittedSamples,\n                    cancelled: true,\n                }\n            }\n            const end = Math.min(off + samplesPerChunk, all.length)\n            const slice = all.slice(off, end)\n            const startSample = off / channels\n            const endSample = end / channels\n            const chunk: StreamAudioDataChunk = {\n                requestId,\n                chunkIndex,\n                startTimeMs: Math.round((startSample / sampleRate) * 1000),\n                endTimeMs: Math.round((endSample / sampleRate) * 1000),\n                durationMs: Math.round(\n                    ((endSample - startSample) / sampleRate) * 1000\n                ),\n                startSample,\n                sampleCount: slice.length,\n                sampleRate,\n                channels,\n                samples: slice,\n                isFinal: end >= all.length,\n            }\n            await callbacks.onChunk(chunk)\n            callbacks.onProgress?.({\n                requestId,\n                processedMs: chunk.endTimeMs,\n                durationMs,\n                progress: durationMs > 0 ? chunk.endTimeMs / durationMs : 1,\n                emittedChunks: chunkIndex + 1,\n            })\n            chunkIndex += 1\n            emittedSamples += slice.length\n        }\n\n        return {\n            requestId,\n            durationMs,\n            sampleRate,\n            channels,\n            chunks: chunkIndex,\n            samples: emittedSamples,\n            cancelled: false,\n        }\n    } catch (err) {\n        throw mapStreamError(err, options.fileUri, 'web')\n    }\n}\n\nfunction sanitizeFloat32(\n    input: Float32Array,\n    clamp: boolean\n): Float32Array {\n    if (!clamp) {\n        // still need NaN/Inf sanitation\n        for (let i = 0; i < input.length; i++) {\n            const v = input[i]\n            if (!Number.isFinite(v)) input[i] = 0\n        }\n        return input\n    }\n    for (let i = 0; i < input.length; i++) {\n        const v = input[i]\n        if (!Number.isFinite(v)) {\n            input[i] = 0\n        } else if (v > 1) {\n            input[i] = 1\n        } else if (v < -1) {\n            input[i] = -1\n        }\n    }\n    return input\n}\n"]}

package/build/esm/AudioAnalysis/AudioAnalysis.types.js.map

@@ -1 +1 @@
-{"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,iEAAiE","sourcesContent":["// packages/audio-studio/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../AudioStudio.types'\n\n/**\n * Represents the configuration for decoding audio data.\n */\nexport interface DecodingConfig {\n    /** Target sample rate for decoded audio (Android and Web) */\n    targetSampleRate?: number\n    /** Target number of channels (Android and Web) */\n    targetChannels?: number\n    /** Target bit depth (Android and Web) */\n    targetBitDepth?: BitDepth\n    /** Whether to normalize audio levels (Android and Web) */\n    normalizeAudio?: boolean\n}\n\n/**\n * Represents speech-related features extracted from audio.\n */\nexport interface SpeechFeatures {\n    isActive: boolean // Whether speech is detected in this segment\n    speakerId?: number // Optional speaker identification\n    // Could add more speech-related features here like:\n    // confidence: number\n    // language?: string\n    // sentiment?: number\n    // etc.\n}\n\n/**\n * Represents various audio features extracted from an audio signal.\n */\nexport interface AudioFeatures {\n    energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n    mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n    rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n    minAmplitude?: number // Minimum amplitude value in the audio signal.\n    maxAmplitude?: number // Maximum amplitude value in the audio signal.\n    zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n    spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n    spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n    spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n    spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n    chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n    tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n    hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n    melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n    spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n    tonnetz?: number[] // Tonal network features representing harmonic relationships.\n    pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n    crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/**\n * Options to specify which audio features to extract.\n * Note: Advanced features (spectral features, chromagram, pitch, etc.) are experimental,\n * especially during live recording, due to high processing requirements.\n */\nexport interface AudioFeaturesOptions {\n    // Basic features - well optimized\n    energy?: boolean\n    rms?: boolean\n    zcr?: boolean\n\n    // Advanced features - experimental, may impact performance in live recording\n    mfcc?: boolean\n    spectralCentroid?: boolean\n    spectralFlatness?: boolean\n    spectralRolloff?: boolean\n    spectralBandwidth?: boolean\n    chromagram?: boolean\n    tempo?: boolean\n    hnr?: boolean\n    melSpectrogram?: boolean\n    spectralContrast?: boolean\n    tonnetz?: boolean\n    pitch?: boolean\n\n    // Utility\n    crc32?: boolean\n}\n\n/**\n * Represents a single data point in the audio analysis.\n */\nexport interface DataPoint {\n    id: number\n    amplitude: number // Peak amplitude for the segment\n    rms: number // Root mean square value\n    dB: number // dBFS (decibels relative to full scale) computed from RMS value\n    silent: boolean // Always computed\n    features?: AudioFeatures\n    speech?: SpeechFeatures\n    startTime?: number\n    endTime?: number\n    // start / end position in bytes\n    startPosition?: number\n    endPosition?: number\n    // number of audio samples for this point (samples size depends on bit depth)\n    samples?: number\n}\n\n/**\n * Represents the complete data from the audio analysis.\n */\nexport interface AudioAnalysis {\n    segmentDurationMs: number // Duration of each segment in milliseconds\n    durationMs: number // Duration of the audio in milliseconds\n    /**\n     * Bit depth used for audio analysis processing.\n     *\n     * **Important**: This represents the internal processing bit depth, which may differ\n     * from the recording bit depth. Audio is typically converted to 32-bit float for\n     * analysis to ensure precision in calculations, regardless of the original recording format.\n     *\n     * Platform behavior:\n     * - iOS: Always 32 (float processing)\n     * - Android: Always 32 (float processing)\n     * - Web: Always 32 (Web Audio API standard)\n     *\n     * The actual recorded file will maintain the requested bit depth (8, 16, or 32).\n     */\n    bitDepth: number\n    samples: number // Size of the audio in bytes\n    numberOfChannels: number // Number of audio channels\n    sampleRate: number // Sample rate of the audio\n    dataPoints: DataPoint[] // Array of data points from the analysis.\n    amplitudeRange: {\n        min: number\n        max: number\n    }\n    rmsRange: {\n        min: number\n        max: number\n    }\n    extractionTimeMs: number // Time taken to extract/process the analysis in milliseconds\n    // TODO: speaker changes into a broader speech analysis section\n    speechAnalysis?: {\n        speakerChanges: {\n            timestamp: number\n            speakerId: number\n        }[]\n        // Could add more speech analysis data here like:\n        // dominantSpeaker?: number\n        // totalSpeechDuration?: number\n        // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n    }\n}\n\n/**\n * Options for specifying a time range within an audio file.\n */\nexport interface AudioRangeOptions {\n    /** Start time in milliseconds */\n    startTimeMs?: number\n    /** End time in milliseconds */\n    endTimeMs?: number\n}\n\n/**\n * Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n */\nexport interface PreviewOptions extends AudioRangeOptions {\n    /** URI of the audio file to analyze */\n    fileUri: string\n    /**\n     * Total number of points to generate for the preview.\n     * @default 100\n     */\n    numberOfPoints?: number\n    /**\n     * Optional logger for debugging.\n     */\n    logger?: ConsoleLike\n    /**\n     * Optional configuration for decoding the audio file.\n     * Defaults to:\n     * - targetSampleRate: undefined (keep original)\n     * - targetChannels: undefined (keep original)\n     * - targetBitDepth: 16\n     * - normalizeAudio: false\n     */\n    decodingOptions?: DecodingConfig\n}\n\n/**\n * Options for mel-spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface ExtractMelSpectrogramOptions {\n    fileUri?: string // Path to audio file\n    arrayBuffer?: ArrayBuffer // Raw audio buffer\n    windowSizeMs: number // Window size in ms (e.g., 25)\n    hopLengthMs: number // Hop length in ms (e.g., 10)\n    nMels: number // Number of mel filters (e.g., 60)\n    fMin?: number // Min frequency (default: 0)\n    fMax?: number // Max frequency (default: sampleRate / 2)\n    windowType?: 'hann' | 'hamming' // Window function (default: 'hann')\n    normalize?: boolean // Mean normalization (default: false)\n    logScale?: boolean // Log scaling of mel energies (default: true)\n    decodingOptions?: DecodingConfig // Audio decoding settings\n    /** Optional start time in ms. If neither startTimeMs nor endTimeMs is set, defaults to 0. */\n    startTimeMs?: number\n    /** Optional end time in ms. Clamped so that the range does not exceed MAX_DURATION_MS (30 s). */\n    endTimeMs?: number\n    logger?: ConsoleLike\n}\n\n/**\n * Result type for WASM-based audio feature extraction.\n */\nexport interface AudioFeaturesWasmResult {\n    spectralCentroid: number\n    spectralFlatness: number\n    spectralRolloff: number\n    spectralBandwidth: number\n    mfcc: number[]\n    chromagram: number[]\n}\n\n/**\n * Return type for mel spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n    spectrogram: number[][] // 2D array [time][mel]\n    sampleRate: number // Audio sample rate\n    nMels: number // Number of mel filters\n    timeSteps: number // Number of time frames\n    durationMs: number // Audio duration in ms\n}\n"]}
+{"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,iEAAiE","sourcesContent":["// packages/audio-studio/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../AudioStudio.types'\n\n/**\n * Represents the configuration for decoding audio data.\n */\nexport interface DecodingConfig {\n    /** Target sample rate for decoded audio (Android and Web) */\n    targetSampleRate?: number\n    /** Target number of channels (Android and Web) */\n    targetChannels?: number\n    /** Target bit depth (Android and Web) */\n    targetBitDepth?: BitDepth\n    /** Whether to normalize audio levels (Android and Web) */\n    normalizeAudio?: boolean\n    /**\n     * RMS threshold below which a segment is flagged silent.\n     * Range 0..1. Default 0.01.\n     * Currently applied as a JS post-process so the same behavior holds across iOS/Android/Web.\n     */\n    silenceRmsThreshold?: number\n}\n\n/**\n * Represents speech-related features extracted from audio.\n */\nexport interface SpeechFeatures {\n    isActive: boolean // Whether speech is detected in this segment\n    speakerId?: number // Optional speaker identification\n    // Could add more speech-related features here like:\n    // confidence: number\n    // language?: string\n    // sentiment?: number\n    // etc.\n}\n\n/**\n * Represents various audio features extracted from an audio signal.\n */\nexport interface AudioFeatures {\n    energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n    mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n    rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n    minAmplitude?: number // Minimum amplitude value in the audio signal.\n    maxAmplitude?: number // Maximum amplitude value in the audio signal.\n    zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n    spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n    spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n    spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n    spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n    chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n    tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n    hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n    melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n    spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n    tonnetz?: number[] // Tonal network features representing harmonic relationships.\n    pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n    crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/**\n * Options to specify which audio features to extract.\n * Note: Advanced features (spectral features, chromagram, pitch, etc.) are experimental,\n * especially during live recording, due to high processing requirements.\n */\nexport interface AudioFeaturesOptions {\n    // Basic features - well optimized\n    energy?: boolean\n    rms?: boolean\n    zcr?: boolean\n\n    // Advanced features - experimental, may impact performance in live recording\n    mfcc?: boolean\n    spectralCentroid?: boolean\n    spectralFlatness?: boolean\n    spectralRolloff?: boolean\n    spectralBandwidth?: boolean\n    chromagram?: boolean\n    tempo?: boolean\n    hnr?: boolean\n    melSpectrogram?: boolean\n    spectralContrast?: boolean\n    tonnetz?: boolean\n    pitch?: boolean\n\n    // Utility\n    crc32?: boolean\n}\n\n/**\n * Represents a single data point in the audio analysis.\n */\nexport interface DataPoint {\n    id: number\n    amplitude: number // Peak amplitude for the segment\n    rms: number // Root mean square value\n    dB: number // dBFS (decibels relative to full scale) computed from RMS value\n    silent: boolean // Always computed\n    features?: AudioFeatures\n    speech?: SpeechFeatures\n    startTime?: number\n    endTime?: number\n    // start / end position in bytes\n    startPosition?: number\n    endPosition?: number\n    // number of audio samples for this point (samples size depends on bit depth)\n    samples?: number\n}\n\n/**\n * Represents the complete data from the audio analysis.\n */\nexport interface AudioAnalysis {\n    segmentDurationMs: number // Duration of each segment in milliseconds\n    durationMs: number // Duration of the audio in milliseconds\n    /**\n     * Bit depth used for audio analysis processing.\n     *\n     * **Important**: This represents the internal processing bit depth, which may differ\n     * from the recording bit depth. Audio is typically converted to 32-bit float for\n     * analysis to ensure precision in calculations, regardless of the original recording format.\n     *\n     * Platform behavior:\n     * - iOS: Always 32 (float processing)\n     * - Android: Always 32 (float processing)\n     * - Web: Always 32 (Web Audio API standard)\n     *\n     * The actual recorded file will maintain the requested bit depth (8, 16, or 32).\n     */\n    bitDepth: number\n    samples: number // Size of the audio in bytes\n    numberOfChannels: number // Number of audio channels\n    sampleRate: number // Sample rate of the audio\n    dataPoints: DataPoint[] // Array of data points from the analysis.\n    amplitudeRange: {\n        min: number\n        max: number\n    }\n    rmsRange: {\n        min: number\n        max: number\n    }\n    extractionTimeMs: number // Time taken to extract/process the analysis in milliseconds\n    // TODO: speaker changes into a broader speech analysis section\n    speechAnalysis?: {\n        speakerChanges: {\n            timestamp: number\n            speakerId: number\n        }[]\n        // Could add more speech analysis data here like:\n        // dominantSpeaker?: number\n        // totalSpeechDuration?: number\n        // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n    }\n}\n\n/**\n * Options for specifying a time range within an audio file.\n */\nexport interface AudioRangeOptions {\n    /** Start time in milliseconds */\n    startTimeMs?: number\n    /** End time in milliseconds */\n    endTimeMs?: number\n}\n\n/**\n * Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n */\nexport interface PreviewBar {\n    /** Stable zero-based bar identifier. */\n    id: number\n    /** Peak amplitude for this bar, normalized to 0..1. */\n    amplitude: number\n    /** Root mean square amplitude for this bar, normalized to 0..1. */\n    rms: number\n    /** Whether this bar is below the configured silence RMS threshold. */\n    silent: boolean\n    /** Bar start time in milliseconds from the extracted range start. */\n    startTimeMs: number\n    /** Bar end time in milliseconds from the extracted range start. */\n    endTimeMs: number\n}\n\n/**\n * Compact preview-bars result for UI waveform rendering.\n * Unlike `AudioAnalysis`, this intentionally omits full `DataPoint` feature data.\n */\nexport interface PreviewBarsResult {\n    bars: PreviewBar[]\n    durationMs: number\n    sampleRate: number\n    numberOfChannels: number\n    bitDepth: number\n    samples: number\n    /** Requested bar count before native/platform clamping. */\n    requestedNumberOfBars: number\n    /** Approximate duration represented by each bar. */\n    barDurationMs: number\n    amplitudeRange: {\n        min: number\n        max: number\n    }\n    rmsRange: {\n        min: number\n        max: number\n    }\n    extractionTimeMs: number\n}\n\n/**\n * Options for extracting compact waveform preview bars for UI rendering.\n */\nexport interface PreviewBarsOptions extends AudioRangeOptions {\n    /** URI of the audio file to analyze */\n    fileUri: string\n    /**\n     * Total number of bars to generate for the preview.\n     * @default 100\n     */\n    numberOfBars?: number\n    /** Optional logger for debugging. */\n    logger?: ConsoleLike\n    /** Optional configuration for decoding the audio file. */\n    decodingOptions?: DecodingConfig\n    /**\n     * Optional callback fired once per compact bar after extraction resolves.\n     * Native progressive streaming is not implied by this callback.\n     */\n    onBarReady?: (bar: PreviewBar, index: number, total: number) => void\n}\n\nexport interface PreviewOptions extends AudioRangeOptions {\n    /** URI of the audio file to analyze */\n    fileUri: string\n    /**\n     * Total number of points to generate for the preview.\n     * @default 100\n     */\n    numberOfPoints?: number\n    /**\n     * Optional logger for debugging.\n     */\n    logger?: ConsoleLike\n    /**\n     * Optional configuration for decoding the audio file.\n     * Defaults to:\n     * - targetSampleRate: undefined (keep original)\n     * - targetChannels: undefined (keep original)\n     * - targetBitDepth: 16\n     * - normalizeAudio: false\n     */\n    decodingOptions?: DecodingConfig\n    /**\n     * Optional callback fired once per data point as the preview becomes available.\n     * Today the native module returns the full analysis in one shot; the points are then\n     * micro-batched on the JS side so consumers can render bars incrementally.\n     * Native progressive streaming is a future enhancement.\n     */\n    onPointReady?: (point: DataPoint, index: number, total: number) => void\n    /**\n     * Optional cancellation signal for JS-side progressive point emission.\n     * Aborting does not cancel native extraction after it has started, but it\n     * stops any queued `onPointReady` callbacks from an older request.\n     */\n    signal?: AbortSignal\n}\n\n/**\n * Options for mel-spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface ExtractMelSpectrogramOptions {\n    fileUri?: string // Path to audio file\n    arrayBuffer?: ArrayBuffer // Raw audio buffer\n    windowSizeMs: number // Window size in ms (e.g., 25)\n    hopLengthMs: number // Hop length in ms (e.g., 10)\n    nMels: number // Number of mel filters (e.g., 60)\n    fMin?: number // Min frequency (default: 0)\n    fMax?: number // Max frequency (default: sampleRate / 2)\n    windowType?: 'hann' | 'hamming' // Window function (default: 'hann')\n    normalize?: boolean // Mean normalization (default: false)\n    logScale?: boolean // Log scaling of mel energies (default: true)\n    decodingOptions?: DecodingConfig // Audio decoding settings\n    /** Optional start time in ms. If neither startTimeMs nor endTimeMs is set, defaults to 0. */\n    startTimeMs?: number\n    /** Optional end time in ms. Clamped so that the range does not exceed MAX_DURATION_MS (30 s). */\n    endTimeMs?: number\n    logger?: ConsoleLike\n}\n\n/**\n * Result type for WASM-based audio feature extraction.\n */\nexport interface AudioFeaturesWasmResult {\n    spectralCentroid: number\n    spectralFlatness: number\n    spectralRolloff: number\n    spectralBandwidth: number\n    mfcc: number[]\n    chromagram: number[]\n}\n\n/**\n * Return type for mel spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n    spectrogram: number[][] // 2D array [time][mel]\n    sampleRate: number // Audio sample rate\n    nMels: number // Number of mel filters\n    timeSteps: number // Number of time frames\n    durationMs: number // Audio duration in ms\n}\n"]}