@mihe-dev/js 0.0.0-stage → 0.2.0

package/LICENSE

@@ -0,0 +1,21 @@
+MIT License
+
+Copyright (c) 2026 Neek Digital Solutions Ltd
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

@@ -1,3 +1,115 @@
-# Temporary Holding Version
+# @mihe-dev/js
 
-This version is a temporary placeholder for this package. An operational version to replace this has been submitted for review and is awaiting a staged release.
+Mihe is an assistant you embed in your web app. It knows your screens,
+components and actions, and it operates the app for your users, under your
+own permission model: it finds things, opens screens, fills in and points
+at forms, and runs actions, asking for approval wherever you say it must.
+Users can type or talk to it.
+
+This is the framework-free package: plain JavaScript or TypeScript, for Vue,
+Svelte, Angular, server-rendered pages, or no framework at all. For React,
+use `@mihe-dev/react`, which is built on this one.
+
+## Install
+
+```bash
+npm install @mihe-dev/js
+```
+
+The package is ES modules with TypeScript types, and has no dependencies
+beyond `@mihe-dev/protocol`.
+
+No build step? It also ships as one self-contained file, for WordPress, a
+server-rendered page or a plain `static/` folder:
+
+```html
+<script src="https://cdn.jsdelivr.net/npm/@mihe-dev/js@0.1.0/dist/mihe.iife.min.js"></script>
+<script>
+  const mihe = Mihe.createMihe({ url: "wss://runtime.mihe.dev/v1/ws", publicKey: "pk_live_…", getUserToken });
+  Mihe.mountWidget(mihe);
+</script>
+```
+
+`dist/mihe.min.js` is the same as an ES module. Pin the version in the URL,
+or copy the file into your site.
+
+## A first integration
+
+```ts
+import { createMihe } from "@mihe-dev/js";
+
+const mihe = createMihe({
+  url: "wss://runtime.mihe.dev/v1/ws",
+  publicKey: "pk_live_…",
+  // Your server mints this token for the signed-in user, signed with your
+  // app secret. It carries who they are and what they may do.
+  getUserToken: () => fetch("/api/mihe/token").then((r) => r.text()),
+  navigate: (route) => { location.hash = route; },
+  // Refetch what the page shows after the assistant changes data.
+  onActionComplete: () => refresh(),
+});
+
+// On each page: tell the assistant where the user is and what they can see.
+// Keep the returned function and call it when they leave.
+const leave = mihe.setScreen({
+  screenId: "orders.list",
+  route: "/orders",
+  title: "Orders",
+  visibleData: { pending: 16 },
+});
+```
+
+Every `set…` and `register…` method returns a function that removes what it
+declared. Calling one again with the same id updates the declaration rather
+than adding a second.
+
+```ts
+mihe.registerComponent(button, "orders.new", { type: "button", label: "New order" });
+mihe.registerAction("print_receipt", ({ orderId }) => printReceipt(orderId));
+mihe.registerTable({
+  id: "orders.table",
+  label: "Orders",
+  columns: [{ key: "number", label: "Order" }, { key: "total", label: "Total" }],
+  rows: orders,
+  rowId: (order) => order.id,
+  rowIdName: "orderId",
+});
+mihe.bindTrigger(document.querySelector("#ask")!);
+```
+
+The widget is `<mihe-assistant>`, a web component in its own shadow root,
+so your page's CSS can't reach it:
+
+```ts
+import { mountWidget } from "@mihe-dev/js";
+
+mountWidget(mihe);                                           // the orb
+mountWidget(mihe, { form: "classic" });                      // launcher and panel
+mountWidget(mihe, { layout: "inline", container: sidebar }); // the panel, in your layout
+```
+
+Theme it with `--mihe-accent`, `--mihe-radius` and `--mihe-font` on the
+`mihe-assistant` element. If your page already has a floating widget in the
+same corner, such as a support chat, lift Mihe above it:
+`mountWidget(mihe, { offset: { bottom: 92 } })`.
+
+Or draw your own UI. State is a snapshot you subscribe to:
+
+```ts
+mihe.subscribe(() => render(mihe.getState()));
+mihe.send("How many orders are pending?");
+```
+
+Create an app and its keys in the [dashboard](https://app.mihe.dev), then
+follow the [quickstart](https://mihe.dev/docs/quickstart).
+
+## Documentation
+
+- [Quickstart](https://mihe.dev/docs/quickstart)
+- [SDK reference](https://mihe.dev/docs/sdk-reference)
+- [Security](https://mihe.dev/docs/security): what the assistant can and
+  cannot do, and why
+
+## License
+
+MIT © Neek Digital Solutions Ltd

package/dist/audio.d.ts

@@ -0,0 +1,121 @@
+/**
+ * Microphone capture and speaker playback for voice sessions.
+ *
+ * One format everywhere: **16-bit PCM, 24 kHz, mono, little endian**, matching
+ * the voice socket and the provider behind it. Nothing on this path transcodes,
+ * which is the point — a resampler in a latency-critical loop buys nothing when
+ * every participant already agrees on the rate.
+ *
+ * Two browser realities shape the code:
+ *
+ * - **iOS Safari will not start audio without a user gesture.** Push-to-talk
+ *   makes that free: the button press *is* the gesture. `resume()` is called on
+ *   press rather than at construction for exactly this reason, and calling it
+ *   from anywhere else will produce a context stuck in `suspended` with no
+ *   error.
+ * - **`AudioContext` may refuse the sample rate we ask for.** Chrome, Firefox,
+ *   and current Safari honour it; older Safari does not. So capture resamples
+ *   when it has to rather than assuming, and playback hands the browser a
+ *   buffer tagged 24 kHz and lets the graph resample it.
+ */
+/** Float samples in [-1, 1] to little-endian 16-bit PCM. */
+export declare function toPcm16(samples: Float32Array): ArrayBuffer;
+/** Little-endian 16-bit PCM back to float samples. */
+export declare function fromPcm16(buffer: ArrayBuffer): Float32Array;
+/**
+ * How loud a chunk of speech is, from 0 (silence) to 1 (as loud as speech gets).
+ *
+ * RMS rather than peak, because a peak jumps on every consonant and a ring
+ * drawn from it flickers instead of breathing. Scaled by eye: ordinary speech
+ * into a laptop microphone sits around 0.05 to 0.15 RMS, so a square root and
+ * a gain put a normal voice in the middle of the range rather than at its foot.
+ */
+export declare function pcmLevel(pcm: ArrayBuffer): number;
+/**
+ * Linear resample, used only when the browser refused our sample rate.
+ *
+ * Linear rather than anything better on purpose: the input is a voice at
+ * 44.1 or 48 kHz going to 24 kHz, the output is fed to a speech model rather
+ * than to a listener, and a windowed-sinc filter would cost more than it could
+ * possibly be worth here. When the rates already match this is not called at
+ * all.
+ */
+export declare function resample(samples: Float32Array, fromRate: number, toRate: number): Float32Array;
+/**
+ * The microphone, as a stream of PCM frames.
+ *
+ * Capture runs continuously once started and is *gated* rather than stopped
+ * between utterances: `getUserMedia` takes hundreds of milliseconds and shows a
+ * permission prompt the first time, so tearing it down on every button release
+ * would clip the beginning of the next sentence — which is exactly where the
+ * word the model needs most usually is.
+ */
+export declare class MicrophoneCapture {
+    private readonly onFrame;
+    private context;
+    private stream;
+    private node;
+    private source;
+    private open;
+    constructor(onFrame: (pcm: ArrayBuffer) => void);
+    /** True while frames are being delivered. */
+    get capturing(): boolean;
+    /**
+     * Ask for the microphone and build the graph. Must be called from a user
+     * gesture on iOS Safari.
+     */
+    start(): Promise<void>;
+    /** Begin delivering frames. */
+    unmute(): void;
+    /** Stop delivering frames, without releasing the microphone. */
+    mute(): void;
+    /** Release the device and tear down the graph. */
+    stop(): Promise<void>;
+}
+/**
+ * The assistant's voice, as a queue of scheduled buffers.
+ *
+ * Chunks are scheduled back to back against the audio clock rather than played
+ * on arrival, because `AudioBufferSourceNode.start()` with no time argument
+ * plays *now* — and forty of those a second produce overlapping speech. The
+ * running `nextStart` cursor is what turns a stream of chunks into continuous
+ * audio.
+ */
+export declare class SpeakerQueue {
+    private readonly onLevel?;
+    /**
+     * Called with each chunk's level at the moment it starts playing, not when
+     * it arrives. The provider streams seconds ahead of the speaker, so a level
+     * taken on arrival would animate words the user has not heard yet.
+     */
+    constructor(onLevel?: ((level: number) => void) | undefined);
+    private levelTimers;
+    private context;
+    private gain;
+    private sources;
+    /** Audio-clock time the next chunk should begin at. */
+    private nextStart;
+    /** Audio-clock time the current run of speech began. */
+    private startedAt;
+    /** Must be called from a user gesture on iOS Safari. */
+    start(): Promise<void>;
+    /** Queue a chunk of assistant speech. */
+    enqueue(pcm: ArrayBuffer): void;
+    /**
+     * How much of the current utterance the user has actually heard, in
+     * milliseconds.
+     *
+     * This is the number the runtime needs to truncate the model's history on a
+     * barge-in, and it is not the same as how much has been *received*: the
+     * provider streams several seconds ahead of the speaker, so trusting bytes
+     * delivered would tell the model the user heard a sentence they were
+     * interrupted three words into.
+     */
+    heardMs(): number;
+    /** True while anything is scheduled or playing. */
+    get speaking(): boolean;
+    /** Drop everything queued — an interruption, or the end of a session. */
+    flush(): void;
+    stop(): Promise<void>;
+}
+//# sourceMappingURL=audio.d.ts.map

package/dist/audio.d.ts.map

@@ -0,0 +1 @@
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package/dist/audio.js

@@ -0,0 +1,377 @@
+/**
+ * Microphone capture and speaker playback for voice sessions.
+ *
+ * One format everywhere: **16-bit PCM, 24 kHz, mono, little endian**, matching
+ * the voice socket and the provider behind it. Nothing on this path transcodes,
+ * which is the point — a resampler in a latency-critical loop buys nothing when
+ * every participant already agrees on the rate.
+ *
+ * Two browser realities shape the code:
+ *
+ * - **iOS Safari will not start audio without a user gesture.** Push-to-talk
+ *   makes that free: the button press *is* the gesture. `resume()` is called on
+ *   press rather than at construction for exactly this reason, and calling it
+ *   from anywhere else will produce a context stuck in `suspended` with no
+ *   error.
+ * - **`AudioContext` may refuse the sample rate we ask for.** Chrome, Firefox,
+ *   and current Safari honour it; older Safari does not. So capture resamples
+ *   when it has to rather than assuming, and playback hands the browser a
+ *   buffer tagged 24 kHz and lets the graph resample it.
+ */
+import { VOICE_SAMPLE_RATE } from "@mihe-dev/protocol";
+/**
+ * Frames sent up per second. 50 works out to 20 ms per frame — small enough
+ * that the end of an utterance is not padded by a buffer waiting to fill, and
+ * large enough that the socket is not doing a send per millisecond.
+ */
+const FRAME_MS = 20;
+/**
+ * Playback runs this far ahead of the speaker.
+ *
+ * Without it every chunk that arrives a few milliseconds late produces an
+ * audible click as the graph runs dry between buffers. With it, network jitter
+ * up to this much is inaudible, at the cost of the same delay before the first
+ * word. 200 ms is well under the threshold where a reply feels slow.
+ */
+const JITTER_BUFFER_MS = 200;
+/** The AudioWorklet processor, injected as a Blob so the package ships one file. */
+const CAPTURE_WORKLET = `
+class MiheCapture extends AudioWorkletProcessor {
+  constructor(options) {
+    super();
+    // A render quantum is 128 samples — about 5 ms at 24 kHz — so posting one
+    // per call means roughly 190 messages a second, each becoming its own
+    // WebSocket frame and its own JSON envelope upstream. Under push-to-talk
+    // that was merely wasteful; with the microphone open continuously it
+    // saturated the relay and audio started being dropped, which reads to the
+    // user as the assistant simply not answering.
+    // Defaulted rather than assumed. An absent option would make \`target\`
+    // undefined, every arithmetic result NaN, and the fill loop exit without
+    // ever posting — total silence, no error, and nothing in any log to say
+    // the microphone had stopped working.
+    this.target = options?.processorOptions?.frameSamples || 480;
+    this.buffer = new Float32Array(this.target);
+    this.filled = 0;
+  }
+
+  process(inputs) {
+    const channel = inputs[0] && inputs[0][0];
+    // No input yet, or the track ended. Keep the node alive either way:
+    // returning false here permanently removes it from the graph, and the
+    // user is only between utterances.
+    if (!channel) return true;
+
+    let offset = 0;
+    while (offset < channel.length) {
+      const take = Math.min(this.target - this.filled, channel.length - offset);
+      this.buffer.set(channel.subarray(offset, offset + take), this.filled);
+      this.filled += take;
+      offset += take;
+
+      if (this.filled === this.target) {
+        this.port.postMessage(this.buffer.slice());
+        this.filled = 0;
+      }
+    }
+
+    return true;
+  }
+}
+registerProcessor('mihe-capture', MiheCapture);
+`;
+/** Float samples in [-1, 1] to little-endian 16-bit PCM. */
+export function toPcm16(samples) {
+    const buffer = new ArrayBuffer(samples.length * 2);
+    const view = new DataView(buffer);
+    for (let i = 0; i < samples.length; i += 1) {
+        // Clamped before scaling: a sample above 1 would wrap to a large negative
+        // number, which is heard as a loud click rather than as clipping.
+        const clamped = Math.max(-1, Math.min(1, samples[i] ?? 0));
+        view.setInt16(i * 2, clamped * 0x7fff, true);
+    }
+    return buffer;
+}
+/** Little-endian 16-bit PCM back to float samples. */
+export function fromPcm16(buffer) {
+    const view = new DataView(buffer);
+    const samples = new Float32Array(buffer.byteLength / 2);
+    for (let i = 0; i < samples.length; i += 1) {
+        samples[i] = view.getInt16(i * 2, true) / 0x7fff;
+    }
+    return samples;
+}
+/**
+ * How loud a chunk of speech is, from 0 (silence) to 1 (as loud as speech gets).
+ *
+ * RMS rather than peak, because a peak jumps on every consonant and a ring
+ * drawn from it flickers instead of breathing. Scaled by eye: ordinary speech
+ * into a laptop microphone sits around 0.05 to 0.15 RMS, so a square root and
+ * a gain put a normal voice in the middle of the range rather than at its foot.
+ */
+export function pcmLevel(pcm) {
+    const samples = fromPcm16(pcm);
+    if (samples.length === 0)
+        return 0;
+    let sum = 0;
+    for (const sample of samples)
+        sum += sample * sample;
+    const rms = Math.sqrt(sum / samples.length);
+    return Math.min(1, Math.sqrt(rms) * 1.8);
+}
+/**
+ * Linear resample, used only when the browser refused our sample rate.
+ *
+ * Linear rather than anything better on purpose: the input is a voice at
+ * 44.1 or 48 kHz going to 24 kHz, the output is fed to a speech model rather
+ * than to a listener, and a windowed-sinc filter would cost more than it could
+ * possibly be worth here. When the rates already match this is not called at
+ * all.
+ */
+export function resample(samples, fromRate, toRate) {
+    if (fromRate === toRate)
+        return samples;
+    const ratio = fromRate / toRate;
+    const length = Math.floor(samples.length / ratio);
+    const out = new Float32Array(length);
+    for (let i = 0; i < length; i += 1) {
+        const position = i * ratio;
+        const index = Math.floor(position);
+        const next = Math.min(index + 1, samples.length - 1);
+        const fraction = position - index;
+        out[i] = (samples[index] ?? 0) * (1 - fraction) + (samples[next] ?? 0) * fraction;
+    }
+    return out;
+}
+/**
+ * The microphone, as a stream of PCM frames.
+ *
+ * Capture runs continuously once started and is *gated* rather than stopped
+ * between utterances: `getUserMedia` takes hundreds of milliseconds and shows a
+ * permission prompt the first time, so tearing it down on every button release
+ * would clip the beginning of the next sentence — which is exactly where the
+ * word the model needs most usually is.
+ */
+export class MicrophoneCapture {
+    onFrame;
+    context = null;
+    stream = null;
+    node = null;
+    source = null;
+    open = false;
+    constructor(onFrame) {
+        this.onFrame = onFrame;
+    }
+    /** True while frames are being delivered. */
+    get capturing() {
+        return this.open;
+    }
+    /**
+     * Ask for the microphone and build the graph. Must be called from a user
+     * gesture on iOS Safari.
+     */
+    async start() {
+        if (this.context) {
+            await this.context.resume();
+            return;
+        }
+        this.stream = await navigator.mediaDevices.getUserMedia({
+            audio: {
+                channelCount: 1,
+                // All three on: the assistant's own voice comes out of the same device
+                // in a hands-free session, and without cancellation the model hears
+                // itself and answers its own last sentence.
+                echoCancellation: true,
+                noiseSuppression: true,
+                autoGainControl: true,
+            },
+        });
+        const context = new AudioContext({ sampleRate: VOICE_SAMPLE_RATE });
+        this.context = context;
+        this.source = context.createMediaStreamSource(this.stream);
+        const deliver = (samples) => {
+            if (!this.open)
+                return;
+            this.onFrame(toPcm16(resample(samples, context.sampleRate, VOICE_SAMPLE_RATE)));
+        };
+        try {
+            const url = URL.createObjectURL(new Blob([CAPTURE_WORKLET], { type: "application/javascript" }));
+            await context.audioWorklet.addModule(url);
+            URL.revokeObjectURL(url);
+            const worklet = new AudioWorkletNode(context, "mihe-capture", {
+                processorOptions: {
+                    // Measured in the context's own samples: the resample to 24 kHz
+                    // happens after this, on the main thread.
+                    frameSamples: Math.round((context.sampleRate * FRAME_MS) / 1000),
+                },
+            });
+            worklet.port.onmessage = (event) => deliver(event.data);
+            this.source.connect(worklet);
+            // Worklets are not pulled unless something downstream asks for their
+            // output. A zero-gain node is the standard way to run the graph without
+            // routing the microphone to the speaker.
+            const sink = context.createGain();
+            sink.gain.value = 0;
+            worklet.connect(sink).connect(context.destination);
+            this.node = worklet;
+        }
+        catch {
+            // Older Safari, and any browser that blocks Blob worklet modules under a
+            // strict CSP. Deprecated but universally present, and it runs on the
+            // main thread, which for one mono voice track is fine.
+            const frames = Math.max(256, 2 ** Math.round(Math.log2((context.sampleRate * FRAME_MS) / 1000)));
+            const processor = context.createScriptProcessor(frames, 1, 1);
+            processor.onaudioprocess = (event) => deliver(new Float32Array(event.inputBuffer.getChannelData(0)));
+            this.source.connect(processor);
+            processor.connect(context.destination);
+            this.node = processor;
+        }
+        await context.resume();
+    }
+    /** Begin delivering frames. */
+    unmute() {
+        this.open = true;
+    }
+    /** Stop delivering frames, without releasing the microphone. */
+    mute() {
+        this.open = false;
+    }
+    /** Release the device and tear down the graph. */
+    async stop() {
+        this.open = false;
+        this.node?.disconnect();
+        this.source?.disconnect();
+        this.stream?.getTracks().forEach((track) => track.stop());
+        await this.context?.close().catch(() => undefined);
+        this.node = null;
+        this.source = null;
+        this.stream = null;
+        this.context = null;
+    }
+}
+/**
+ * The assistant's voice, as a queue of scheduled buffers.
+ *
+ * Chunks are scheduled back to back against the audio clock rather than played
+ * on arrival, because `AudioBufferSourceNode.start()` with no time argument
+ * plays *now* — and forty of those a second produce overlapping speech. The
+ * running `nextStart` cursor is what turns a stream of chunks into continuous
+ * audio.
+ */
+export class SpeakerQueue {
+    onLevel;
+    /**
+     * Called with each chunk's level at the moment it starts playing, not when
+     * it arrives. The provider streams seconds ahead of the speaker, so a level
+     * taken on arrival would animate words the user has not heard yet.
+     */
+    constructor(onLevel) {
+        this.onLevel = onLevel;
+    }
+    levelTimers = new Set();
+    context = null;
+    gain = null;
+    sources = new Set();
+    /** Audio-clock time the next chunk should begin at. */
+    nextStart = 0;
+    /** Audio-clock time the current run of speech began. */
+    startedAt = 0;
+    /** Must be called from a user gesture on iOS Safari. */
+    async start() {
+        if (this.context) {
+            await this.context.resume();
+            return;
+        }
+        const context = new AudioContext({ sampleRate: VOICE_SAMPLE_RATE });
+        this.gain = context.createGain();
+        this.gain.connect(context.destination);
+        this.context = context;
+        await context.resume();
+    }
+    /** Queue a chunk of assistant speech. */
+    enqueue(pcm) {
+        const context = this.context;
+        if (!context || !this.gain)
+            return;
+        const samples = fromPcm16(pcm);
+        if (samples.length === 0)
+            return;
+        // Tagged 24 kHz regardless of the context's own rate, so a browser that
+        // refused our request resamples on playback instead of playing the
+        // assistant at the wrong pitch.
+        const buffer = context.createBuffer(1, samples.length, VOICE_SAMPLE_RATE);
+        buffer.getChannelData(0).set(samples);
+        const source = context.createBufferSource();
+        source.buffer = buffer;
+        source.connect(this.gain);
+        const now = context.currentTime;
+        if (this.nextStart < now) {
+            // Either the first chunk of an utterance or a gap the network opened up.
+            // Restarting the cursor ahead of `now` is what the jitter buffer is:
+            // scheduling at `now` exactly means anything arriving late is already
+            // too late.
+            this.nextStart = now + JITTER_BUFFER_MS / 1000;
+            this.startedAt = this.nextStart;
+        }
+        source.start(this.nextStart);
+        if (this.onLevel) {
+            const level = pcmLevel(pcm);
+            const report = this.onLevel;
+            const timer = setTimeout(() => {
+                this.levelTimers.delete(timer);
+                report(level);
+            }, Math.max(0, (this.nextStart - now) * 1000));
+            this.levelTimers.add(timer);
+        }
+        this.nextStart += buffer.duration;
+        this.sources.add(source);
+        source.onended = () => this.sources.delete(source);
+    }
+    /**
+     * How much of the current utterance the user has actually heard, in
+     * milliseconds.
+     *
+     * This is the number the runtime needs to truncate the model's history on a
+     * barge-in, and it is not the same as how much has been *received*: the
+     * provider streams several seconds ahead of the speaker, so trusting bytes
+     * delivered would tell the model the user heard a sentence they were
+     * interrupted three words into.
+     */
+    heardMs() {
+        if (!this.context || this.startedAt === 0)
+            return 0;
+        const heard = this.context.currentTime - this.startedAt;
+        return Math.max(0, Math.round(heard * 1000));
+    }
+    /** True while anything is scheduled or playing. */
+    get speaking() {
+        return this.sources.size > 0;
+    }
+    /** Drop everything queued — an interruption, or the end of a session. */
+    flush() {
+        for (const source of this.sources) {
+            try {
+                source.stop();
+            }
+            catch {
+                // Already ended. `stop()` on a finished node throws in some browsers.
+            }
+        }
+        this.sources.clear();
+        this.nextStart = 0;
+        this.startedAt = 0;
+        // Levels for audio that will now never play, and a last zero so a ring
+        // following them settles instead of freezing mid-word.
+        for (const timer of this.levelTimers)
+            clearTimeout(timer);
+        this.levelTimers.clear();
+        this.onLevel?.(0);
+    }
+    async stop() {
+        this.flush();
+        this.gain?.disconnect();
+        await this.context?.close().catch(() => undefined);
+        this.gain = null;
+        this.context = null;
+    }
+}
+//# sourceMappingURL=audio.js.map

package/dist/audio.js.map

@@ -0,0 +1 @@
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