tempest-react-sdk 0.78.0 → 0.79.0

package/dist/audio/audio-bus.js.map

@@ -1 +1 @@
-{"version":3,"file":"audio-bus.js","names":[],"sources":["../../src/audio/audio-bus.ts"],"sourcesContent":["/**\n * @tempest-limits file-lines — the mixing graph, the per-source handle and the\n * output route are one object's lifetime: the limiter only makes sense after the\n * sum, and the element that carries `setSinkId` is the same one the mix leaves\n * through. Splitting them would hand the caller two halves that are useless apart.\n */\nimport { isAudioOutputSelectionSupported, setAudioOutput } from \"./audio-output\";\n\n/** Default ceiling for a per-source or master gain, as a multiplier. */\nexport const DEFAULT_MAX_GAIN = 3;\n\n/** Shape of the master limiter. Matches `DynamicsCompressorNode`'s params. */\nexport interface LimiterSettings {\n    /** dBFS above which the compressor starts working. */\n    threshold: number;\n    /** dB range over which the curve bends. `0` is a hard knee. */\n    knee: number;\n    /** Input/output ratio above the threshold. 20 is limiting, not compression. */\n    ratio: number;\n    /** Seconds to clamp a peak. */\n    attack: number;\n    /** Seconds to let go. */\n    release: number;\n}\n\n/**\n * A limiter, not a compressor: a hard knee at a 20:1 ratio with a 3 ms attack\n * catches the sum before it clips without audibly ducking anything below it.\n */\nconst DEFAULT_LIMITER: LimiterSettings = {\n    threshold: -8,\n    knee: 0,\n    ratio: 20,\n    attack: 0.003,\n    release: 0.25,\n};\n\n/** Options for {@link createAudioBus}. */\nexport interface AudioBusOptions {\n    /** Ceiling for every gain on this bus. Default {@link DEFAULT_MAX_GAIN}. */\n    maxGain?: number;\n    /** Override the master limiter, or pass `false` to run without one. */\n    limiter?: Partial<LimiterSettings> | false;\n    /**\n     * Reuse an existing `AudioContext` instead of creating one.\n     *\n     * Browsers cap the number of live contexts (Chrome allows around six), so a\n     * page that already has one — a level meter, a player — should hand it over\n     * rather than open a second.\n     */\n    context?: AudioContext;\n}\n\n/** One source attached to the bus. */\nexport interface AudioBusHandle {\n    /**\n     * Set this source's gain, where `1` is the level it arrived at.\n     *\n     * Values above `1` are the point of the whole graph: `element.volume` is\n     * clamped at `1`, so a quiet talker could only ever be attenuated — the one\n     * correction nobody needs.\n     */\n    setGain: (gain: number) => void;\n    /** Current gain, after clamping. */\n    readonly gain: number;\n    /** Detach this source and release its nodes. The bus stays up. */\n    stop: () => void;\n}\n\n/** A running mix. */\nexport interface AudioBus {\n    /**\n     * Play a stream through the shared mix.\n     *\n     * @param stream - The stream to play. Only its first audio track is used.\n     * @param options - `gain` is the initial multiplier, default `1`.\n     * @returns A handle to adjust or detach this source.\n     */\n    attach: (stream: MediaStream, options?: { gain?: number }) => AudioBusHandle;\n    /** Scale every source, on top of its own gain. */\n    setMasterGain: (gain: number) => void;\n    /** Current master gain, after clamping. */\n    readonly masterGain: number;\n    /**\n     * Route the whole mix to one output device.\n     *\n     * @param deviceId - Device id from `useMediaDevices().audioOutputs`, or `\"\"`\n     *   for the system default.\n     * @returns `false` when the engine cannot route audio, or the device is gone.\n     */\n    setOutputDevice: (deviceId: string) => Promise<boolean>;\n    /** Device id the mix is routed to. `\"\"` is the system default. */\n    readonly outputDevice: string;\n    /**\n     * Resume a context the browser started suspended.\n     *\n     * Autoplay policy suspends a context created outside a user gesture, and a\n     * suspended context is silent with no error anywhere. Call this from the\n     * click that starts playback.\n     */\n    resume: () => Promise<void>;\n    /**\n     * Detach everything and close the context this bus created.\n     *\n     * Idempotent, and the bus stays callable afterwards: `attach` hands back an\n     * inert handle instead of throwing. That is not politeness — creating a node\n     * on a closed `AudioContext` throws `InvalidStateError`, and the stream that\n     * arrives late is the normal case (a WebRTC `ontrack` firing after the\n     * component that owned the bus went away).\n     */\n    close: () => void;\n    /** Whether this browser gave us a Web Audio graph at all. */\n    readonly supported: boolean;\n    /** Whether this browser can route the mix to a chosen output device. */\n    readonly canSelectOutput: boolean;\n}\n\n/**\n * Clamp a gain into `0..max`, treating a non-finite value as unity.\n *\n * `NaN` reaches here from an empty input or a failed parse, and assigning it to\n * an `AudioParam` throws — losing the audio for a typo in a number field.\n */\nfunction clampGain(gain: number, max: number): number {\n    if (!Number.isFinite(gain)) return 1;\n    return Math.min(max, Math.max(0, gain));\n}\n\n/** The `AudioContext` constructor this engine exposes, prefixed or not. */\nfunction audioContextConstructor(): typeof AudioContext | undefined {\n    if (typeof AudioContext !== \"undefined\") return AudioContext;\n    return (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n}\n\n/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLAudioElement & { setSinkId?: (sinkId: string) => Promise<void> };\n\n/**\n * The stream to feed the graph: just the audio track, when this engine can build\n * one.\n *\n * Isolating the track keeps a camera's video out of the source node. `MediaStream`\n * is missing outside a browser (jsdom, a worker), and there the original stream is\n * the right answer anyway — `createMediaStreamSource` reads its first audio track.\n */\nfunction audioOnly(stream: MediaStream, track: MediaStreamTrack): MediaStream {\n    if (typeof MediaStream === \"undefined\") return stream;\n    return new MediaStream([track]);\n}\n\n/** Start playback without letting an environment that has no media pipeline throw. */\nfunction playQuietly(element: HTMLAudioElement): void {\n    try {\n        void element.play?.()?.catch(() => undefined);\n    } catch {\n        /* jsdom and any engine without a media pipeline — nothing to play anyway */\n    }\n}\n\n/**\n * Mix several streams into one output, with gain above 100% and a master limiter.\n *\n * Three things make this a graph instead of a few `<audio>` elements:\n *\n * 1. **`element.volume` is clamped at `1`.** A participant who speaks too quietly\n *    can only be turned *down* — the one correction nobody needs. A `GainNode`\n *    has no ceiling, so this bus takes one (`maxGain`, default 3).\n * 2. **Clipping is a property of the sum.** Three sources at 200% each are clean\n *    alone and distort the instant they play together. A per-source limiter\n *    cannot see that; the one after the mix can, which is where this puts it.\n * 3. **`setSinkId` lives on the element, not on the context.** Sending the mix to\n *    a headset while the rest of the system keeps the speakers is only reachable\n *    by leaving through a `MediaStreamAudioDestinationNode` into a real\n *    `<audio>`. `AudioContext.setSinkId` has far thinner support.\n *\n * @param options - See {@link AudioBusOptions}.\n * @returns The bus. On an engine with no Web Audio it is inert but callable, and\n *   `supported` is `false` — a page without sound beats a page that throws.\n *\n * @example\n * const bus = createAudioBus({ maxGain: 3 });\n * const handle = bus.attach(remoteStream, { gain: 1 });\n * handle.setGain(2.4);              // above 1 — the point of the whole thing\n * await bus.setOutputDevice(headsetId);\n */\nexport function createAudioBus({\n    maxGain = DEFAULT_MAX_GAIN,\n    limiter: limiterOptions,\n    context: injectedContext,\n}: AudioBusOptions = {}): AudioBus {\n    const Ctor = audioContextConstructor();\n    const context = injectedContext ?? (Ctor ? new Ctor() : null);\n\n    if (!context) {\n        return {\n            attach: () => ({ setGain: () => undefined, gain: 1, stop: () => undefined }),\n            setMasterGain: () => undefined,\n            masterGain: 1,\n            setOutputDevice: async () => false,\n            outputDevice: \"\",\n            resume: async () => undefined,\n            close: () => undefined,\n            supported: false,\n            canSelectOutput: false,\n        };\n    }\n\n    const graph = context;\n    const ownsContext = !injectedContext;\n    const master = graph.createGain();\n    const destination = graph.createMediaStreamDestination();\n\n    let limiterNode: DynamicsCompressorNode | null = null;\n    if (limiterOptions === false) {\n        master.connect(destination);\n    } else {\n        const limiter = graph.createDynamicsCompressor();\n        limiterNode = limiter;\n        const settings = { ...DEFAULT_LIMITER, ...limiterOptions };\n        limiter.threshold.value = settings.threshold;\n        limiter.knee.value = settings.knee;\n        limiter.ratio.value = settings.ratio;\n        limiter.attack.value = settings.attack;\n        limiter.release.value = settings.release;\n        master.connect(limiter);\n        limiter.connect(destination);\n    }\n\n    const element = new Audio();\n    element.autoplay = true;\n    element.srcObject = destination.stream;\n    playQuietly(element);\n\n    let sinkId = \"\";\n    let masterGain = 1;\n    let closed = false;\n    const handles = new Set<AudioBusHandle>();\n\n    /** The handle a bus hands out when there is nothing to play through. */\n    function inertHandle(): AudioBusHandle {\n        return { setGain: () => undefined, gain: 1, stop: () => undefined };\n    }\n\n    /**\n     * Play one stream through the mix.\n     *\n     * The muted `<audio>` anchor is not dead weight, and deleting it as unused is\n     * the mistake this comment exists to prevent: Chrome will not pull samples\n     * from a `MediaStreamAudioSourceNode` built over a **remote** WebRTC stream\n     * unless that same stream is also attached to a media element\n     * (crbug.com/687574). Without it the graph is visibly correct and completely\n     * silent — a day of debugging for anyone who does not know the bug. It is\n     * muted because the audible copy is the one leaving the bus.\n     */\n    function attach(stream: MediaStream, { gain = 1 }: { gain?: number } = {}): AudioBusHandle {\n        const track = stream.getAudioTracks()[0];\n        if (closed || !track) return inertHandle();\n\n        const anchor = new Audio();\n        anchor.muted = true;\n        anchor.autoplay = true;\n        anchor.srcObject = stream;\n        playQuietly(anchor);\n\n        const source = graph.createMediaStreamSource(audioOnly(stream, track));\n        const node = graph.createGain();\n        let current = clampGain(gain, maxGain);\n        node.gain.value = current;\n        source.connect(node);\n        node.connect(master);\n\n        const handle: AudioBusHandle = {\n            setGain(next: number): void {\n                current = clampGain(next, maxGain);\n                node.gain.value = current;\n            },\n            get gain(): number {\n                return current;\n            },\n            stop(): void {\n                if (!handles.has(handle)) return;\n                handles.delete(handle);\n                source.disconnect();\n                node.disconnect();\n                anchor.srcObject = null;\n                anchor.pause();\n            },\n        };\n        handles.add(handle);\n        return handle;\n    }\n\n    return {\n        attach,\n        setMasterGain(gain: number): void {\n            if (closed) return;\n            masterGain = clampGain(gain, maxGain);\n            master.gain.value = masterGain;\n        },\n        get masterGain(): number {\n            return masterGain;\n        },\n        async setOutputDevice(deviceId: string): Promise<boolean> {\n            const applied = await setAudioOutput(element as SinkCapableElement, deviceId);\n            if (applied) sinkId = deviceId;\n            return applied;\n        },\n        get outputDevice(): string {\n            return sinkId;\n        },\n        async resume(): Promise<void> {\n            await graph.resume?.().catch(() => undefined);\n        },\n        close(): void {\n            if (closed) return;\n            closed = true;\n            for (const handle of [...handles]) handle.stop();\n            element.srcObject = null;\n            element.pause();\n            master.disconnect();\n            limiterNode?.disconnect();\n            if (ownsContext) void graph.close().catch(() => undefined);\n        },\n        supported: true,\n        get canSelectOutput(): boolean {\n            return isAudioOutputSelectionSupported();\n        },\n    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+{"version":3,"file":"audio-bus.js","names":["DEFAULT_MAX_GAIN","DEFAULT_LIMITER","clampGain","gain","max","audioContextConstructor","audioOnly","stream","track","playQuietly","createAudioBus","limiterOptions","injectedContext","Ctor","context","graph","ownsContext","master","destination","limiterNode","limiter","settings","element","sinkId","masterGain","closed","handles","inertHandle","attach","anchor","source","node","current","maxGain","handle","next","applied","setAudioOutput","deviceId","isAudioOutputSelectionSupported"],"sources":["../../src/audio/audio-bus.ts"],"sourcesContent":["/**\n * @tempest-limits file-lines — the mixing graph, the per-source handle and the\n * output route are one object's lifetime: the limiter only makes sense after the\n * sum, and the element that carries `setSinkId` is the same one the mix leaves\n * through. Splitting them would hand the caller two halves that are useless apart.\n */\nimport { isAudioOutputSelectionSupported, setAudioOutput } from \"./audio-output\";\n\n/** Default ceiling for a per-source or master gain, as a multiplier. */\nexport const DEFAULT_MAX_GAIN = 3;\n\n/** Shape of the master limiter. Matches `DynamicsCompressorNode`'s params. */\nexport interface LimiterSettings {\n    /** dBFS above which the compressor starts working. */\n    threshold: number;\n    /** dB range over which the curve bends. `0` is a hard knee. */\n    knee: number;\n    /** Input/output ratio above the threshold. 20 is limiting, not compression. */\n    ratio: number;\n    /** Seconds to clamp a peak. */\n    attack: number;\n    /** Seconds to let go. */\n    release: number;\n}\n\n/**\n * A limiter, not a compressor: a hard knee at a 20:1 ratio with a 3 ms attack\n * catches the sum before it clips without audibly ducking anything below it.\n */\nconst DEFAULT_LIMITER: LimiterSettings = {\n    threshold: -8,\n    knee: 0,\n    ratio: 20,\n    attack: 0.003,\n    release: 0.25,\n};\n\n/** Options for {@link createAudioBus}. */\nexport interface AudioBusOptions {\n    /** Ceiling for every gain on this bus. Default {@link DEFAULT_MAX_GAIN}. */\n    maxGain?: number;\n    /** Override the master limiter, or pass `false` to run without one. */\n    limiter?: Partial<LimiterSettings> | false;\n    /**\n     * Reuse an existing `AudioContext` instead of creating one.\n     *\n     * Browsers cap the number of live contexts (Chrome allows around six), so a\n     * page that already has one — a level meter, a player — should hand it over\n     * rather than open a second.\n     */\n    context?: AudioContext;\n}\n\n/** One source attached to the bus. */\nexport interface AudioBusHandle {\n    /**\n     * Set this source's gain, where `1` is the level it arrived at.\n     *\n     * Values above `1` are the point of the whole graph: `element.volume` is\n     * clamped at `1`, so a quiet talker could only ever be attenuated — the one\n     * correction nobody needs.\n     */\n    setGain: (gain: number) => void;\n    /** Current gain, after clamping. */\n    readonly gain: number;\n    /** Detach this source and release its nodes. The bus stays up. */\n    stop: () => void;\n}\n\n/** A running mix. */\nexport interface AudioBus {\n    /**\n     * Play a stream through the shared mix.\n     *\n     * @param stream - The stream to play. Only its first audio track is used.\n     * @param options - `gain` is the initial multiplier, default `1`.\n     * @returns A handle to adjust or detach this source.\n     */\n    attach: (stream: MediaStream, options?: { gain?: number }) => AudioBusHandle;\n    /** Scale every source, on top of its own gain. */\n    setMasterGain: (gain: number) => void;\n    /** Current master gain, after clamping. */\n    readonly masterGain: number;\n    /**\n     * Route the whole mix to one output device.\n     *\n     * @param deviceId - Device id from `useMediaDevices().audioOutputs`, or `\"\"`\n     *   for the system default.\n     * @returns `false` when the engine cannot route audio, or the device is gone.\n     */\n    setOutputDevice: (deviceId: string) => Promise<boolean>;\n    /** Device id the mix is routed to. `\"\"` is the system default. */\n    readonly outputDevice: string;\n    /**\n     * Resume a context the browser started suspended.\n     *\n     * Autoplay policy suspends a context created outside a user gesture, and a\n     * suspended context is silent with no error anywhere. Call this from the\n     * click that starts playback.\n     */\n    resume: () => Promise<void>;\n    /**\n     * Detach everything and close the context this bus created.\n     *\n     * Idempotent, and the bus stays callable afterwards: `attach` hands back an\n     * inert handle instead of throwing. That is not politeness — creating a node\n     * on a closed `AudioContext` throws `InvalidStateError`, and the stream that\n     * arrives late is the normal case (a WebRTC `ontrack` firing after the\n     * component that owned the bus went away).\n     */\n    close: () => void;\n    /** Whether this browser gave us a Web Audio graph at all. */\n    readonly supported: boolean;\n    /** Whether this browser can route the mix to a chosen output device. */\n    readonly canSelectOutput: boolean;\n}\n\n/**\n * Clamp a gain into `0..max`, treating a non-finite value as unity.\n *\n * `NaN` reaches here from an empty input or a failed parse, and assigning it to\n * an `AudioParam` throws — losing the audio for a typo in a number field.\n */\nfunction clampGain(gain: number, max: number): number {\n    if (!Number.isFinite(gain)) return 1;\n    return Math.min(max, Math.max(0, gain));\n}\n\n/** The `AudioContext` constructor this engine exposes, prefixed or not. */\nfunction audioContextConstructor(): typeof AudioContext | undefined {\n    if (typeof AudioContext !== \"undefined\") return AudioContext;\n    return (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n}\n\n/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLAudioElement & { setSinkId?: (sinkId: string) => Promise<void> };\n\n/**\n * The stream to feed the graph: just the audio track, when this engine can build\n * one.\n *\n * Isolating the track keeps a camera's video out of the source node. `MediaStream`\n * is missing outside a browser (jsdom, a worker), and there the original stream is\n * the right answer anyway — `createMediaStreamSource` reads its first audio track.\n */\nfunction audioOnly(stream: MediaStream, track: MediaStreamTrack): MediaStream {\n    if (typeof MediaStream === \"undefined\") return stream;\n    return new MediaStream([track]);\n}\n\n/** Start playback without letting an environment that has no media pipeline throw. */\nfunction playQuietly(element: HTMLAudioElement): void {\n    try {\n        void element.play?.()?.catch(() => undefined);\n    } catch {\n        /* jsdom and any engine without a media pipeline — nothing to play anyway */\n    }\n}\n\n/**\n * Mix several streams into one output, with gain above 100% and a master limiter.\n *\n * Three things make this a graph instead of a few `<audio>` elements:\n *\n * 1. **`element.volume` is clamped at `1`.** A participant who speaks too quietly\n *    can only be turned *down* — the one correction nobody needs. A `GainNode`\n *    has no ceiling, so this bus takes one (`maxGain`, default 3).\n * 2. **Clipping is a property of the sum.** Three sources at 200% each are clean\n *    alone and distort the instant they play together. A per-source limiter\n *    cannot see that; the one after the mix can, which is where this puts it.\n * 3. **`setSinkId` lives on the element, not on the context.** Sending the mix to\n *    a headset while the rest of the system keeps the speakers is only reachable\n *    by leaving through a `MediaStreamAudioDestinationNode` into a real\n *    `<audio>`. `AudioContext.setSinkId` has far thinner support.\n *\n * @param options - See {@link AudioBusOptions}.\n * @returns The bus. On an engine with no Web Audio it is inert but callable, and\n *   `supported` is `false` — a page without sound beats a page that throws.\n *\n * @example\n * const bus = createAudioBus({ maxGain: 3 });\n * const handle = bus.attach(remoteStream, { gain: 1 });\n * handle.setGain(2.4);              // above 1 — the point of the whole thing\n * await bus.setOutputDevice(headsetId);\n */\nexport function createAudioBus({\n    maxGain = DEFAULT_MAX_GAIN,\n    limiter: limiterOptions,\n    context: injectedContext,\n}: AudioBusOptions = {}): AudioBus {\n    const Ctor = audioContextConstructor();\n    const context = injectedContext ?? (Ctor ? new Ctor() : null);\n\n    if (!context) {\n        return {\n            attach: () => ({ setGain: () => undefined, gain: 1, stop: () => undefined }),\n            setMasterGain: () => undefined,\n            masterGain: 1,\n            setOutputDevice: async () => false,\n            outputDevice: \"\",\n            resume: async () => undefined,\n            close: () => undefined,\n            supported: false,\n            canSelectOutput: false,\n        };\n    }\n\n    const graph = context;\n    const ownsContext = !injectedContext;\n    const master = graph.createGain();\n    const destination = graph.createMediaStreamDestination();\n\n    let limiterNode: DynamicsCompressorNode | null = null;\n    if (limiterOptions === false) {\n        master.connect(destination);\n    } else {\n        const limiter = graph.createDynamicsCompressor();\n        limiterNode = limiter;\n        const settings = { ...DEFAULT_LIMITER, ...limiterOptions };\n        limiter.threshold.value = settings.threshold;\n        limiter.knee.value = settings.knee;\n        limiter.ratio.value = settings.ratio;\n        limiter.attack.value = settings.attack;\n        limiter.release.value = settings.release;\n        master.connect(limiter);\n        limiter.connect(destination);\n    }\n\n    const element = new Audio();\n    element.autoplay = true;\n    element.srcObject = destination.stream;\n    playQuietly(element);\n\n    let sinkId = \"\";\n    let masterGain = 1;\n    let closed = false;\n    const handles = new Set<AudioBusHandle>();\n\n    /** The handle a bus hands out when there is nothing to play through. */\n    function inertHandle(): AudioBusHandle {\n        return { setGain: () => undefined, gain: 1, stop: () => undefined };\n    }\n\n    /**\n     * Play one stream through the mix.\n     *\n     * The muted `<audio>` anchor is not dead weight, and deleting it as unused is\n     * the mistake this comment exists to prevent: Chrome will not pull samples\n     * from a `MediaStreamAudioSourceNode` built over a **remote** WebRTC stream\n     * unless that same stream is also attached to a media element\n     * (crbug.com/687574). Without it the graph is visibly correct and completely\n     * silent — a day of debugging for anyone who does not know the bug. 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async resume(): Promise<void> {\n            await graph.resume?.().catch(() => undefined);\n        },\n        close(): void {\n            if (closed) return;\n            closed = true;\n            for (const handle of [...handles]) handle.stop();\n            element.srcObject = null;\n            element.pause();\n            master.disconnect();\n            limiterNode?.disconnect();\n            if (ownsContext) void graph.close().catch(() => undefined);\n        },\n        supported: true,\n        get canSelectOutput(): boolean {\n            return isAudioOutputSelectionSupported();\n        },\n    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package/dist/audio/audio-output.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"audio-output.cjs","names":[],"sources":["../../src/audio/audio-output.ts"],"sourcesContent":["/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLMediaElement & {\n    setSinkId?: (sinkId: string) => Promise<void>;\n    sinkId?: string;\n};\n\n/**\n * Whether this browser can route audio to a chosen output device.\n *\n * Chromium-only at the time of writing: Firefox has `setSinkId` behind a preference\n * that is off by default, and Safari does not implement it at all. Check this before\n * rendering an output picker, or the control is a lie on two of three engines.\n */\nexport function isAudioOutputSelectionSupported(): boolean {\n    return (\n        typeof HTMLMediaElement !== \"undefined\" &&\n        typeof (HTMLMediaElement.prototype as SinkCapableElement).setSinkId === \"function\"\n    );\n}\n\n/**\n * Send an element's audio to a specific output device.\n *\n * The `sinkId` comes from `useMediaDevices().audioOutputs` — and, like every device\n * label, that list is anonymous until a capture permission has been granted, so an\n * output picker realistically only works after the microphone has been allowed once.\n *\n * Returns `false` instead of throwing when the browser has no `setSinkId`, because\n * \"this engine cannot route audio\" is the common case and not an error the caller can\n * act on: the sound still plays, on the system default device. A rejected\n * `setSinkId` — a device that was unplugged between enumeration and playback is the\n * usual cause — also resolves `false`, and playback continues on the default.\n *\n * @param element - The `<audio>` or `<video>` element to route.\n * @param sinkId - Device id, or `\"\"` for the system default.\n * @returns `true` when the route was applied.\n *\n * @example\n * const ok = await setAudioOutput(audioRef.current, selectedSpeakerId);\n * if (!ok) toast(\"Este navegador não permite escolher a saída de som.\");\n */\nexport async function setAudioOutput(\n    element: HTMLMediaElement | null,\n    sinkId: string,\n): Promise<boolean> {\n    const target = element as SinkCapableElement | null;\n    if (!target || typeof target.setSinkId !== \"function\") return false;\n    try {\n        await target.setSinkId(sinkId);\n        return true;\n    } catch {\n        return false;\n    }\n}\n"],"mappings":"AAaA,SAAgB,GAA2C,CACvD,OACI,OAAO,iBAAqB,KAC5B,OAAQ,iBAAiB,UAAiC,WAAc,UAEhF,CAuBA,eAAsB,EAClB,EACA,EACgB,CAChB,IAAM,EAAS,EACf,GAAI,CAAC,GAAU,OAAO,EAAO,WAAc,WAAY,MAAO,GAC9D,GAAI,CAEA,OADA,MAAM,EAAO,UAAU,CAAM,EACtB,EACX,MAAQ,CACJ,MAAO,EACX,CACJ"}
+{"version":3,"file":"audio-output.cjs","names":["isAudioOutputSelectionSupported","setAudioOutput","target","element","sinkId"],"sources":["../../src/audio/audio-output.ts"],"sourcesContent":["/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLMediaElement & {\n    setSinkId?: (sinkId: string) => Promise<void>;\n    sinkId?: string;\n};\n\n/**\n * Whether this browser can route audio to a chosen output device.\n *\n * Chromium-only at the time of writing: Firefox has `setSinkId` behind a preference\n * that is off by default, and Safari does not implement it at all. Check this before\n * rendering an output picker, or the control is a lie on two of three engines.\n */\nexport function isAudioOutputSelectionSupported(): boolean {\n    return (\n        typeof HTMLMediaElement !== \"undefined\" &&\n        typeof (HTMLMediaElement.prototype as SinkCapableElement).setSinkId === \"function\"\n    );\n}\n\n/**\n * Send an element's audio to a specific output device.\n *\n * The `sinkId` comes from `useMediaDevices().audioOutputs` — and, like every device\n * label, that list is anonymous until a capture permission has been granted, so an\n * output picker realistically only works after the microphone has been allowed once.\n *\n * Returns `false` instead of throwing when the browser has no `setSinkId`, because\n * \"this engine cannot route audio\" is the common case and not an error the caller can\n * act on: the sound still plays, on the system default device. A rejected\n * `setSinkId` — a device that was unplugged between enumeration and playback is the\n * usual cause — also resolves `false`, and playback continues on the default.\n *\n * @param element - The `<audio>` or `<video>` element to route.\n * @param sinkId - Device id, or `\"\"` for the system default.\n * @returns `true` when the route was applied.\n *\n * @example\n * const ok = await setAudioOutput(audioRef.current, selectedSpeakerId);\n * if (!ok) toast(\"Este navegador não permite escolher a saída de som.\");\n */\nexport async function setAudioOutput(\n    element: HTMLMediaElement | null,\n    sinkId: string,\n): Promise<boolean> {\n    const target = element as SinkCapableElement | null;\n    if (!target || typeof target.setSinkId !== \"function\") return false;\n    try {\n        await target.setSinkId(sinkId);\n        return true;\n    } catch {\n        return false;\n    }\n}\n"],"mappings":"AAaA,SAAgBA,GAA2C,CACvD,OACI,OAAO,iBAAqB,KAC5B,OAAQ,iBAAiB,UAAiC,WAAc,UAEhF,CAuBA,eAAsBC,EAClB,EACA,EACgB,CAChB,IAAMC,EAASC,EACf,GAAI,CAACD,GAAU,OAAOA,EAAO,WAAc,WAAY,MAAO,GAC9D,GAAI,CAEA,OADA,MAAMA,EAAO,UAAUE,CAAM,EACtB,EACX,MAAQ,CACJ,MAAO,EACX,CACJ,C"}

package/dist/audio/audio-output.js.map

@@ -1 +1 @@
-{"version":3,"file":"audio-output.js","names":[],"sources":["../../src/audio/audio-output.ts"],"sourcesContent":["/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLMediaElement & {\n    setSinkId?: (sinkId: string) => Promise<void>;\n    sinkId?: string;\n};\n\n/**\n * Whether this browser can route audio to a chosen output device.\n *\n * Chromium-only at the time of writing: Firefox has `setSinkId` behind a preference\n * that is off by default, and Safari does not implement it at all. Check this before\n * rendering an output picker, or the control is a lie on two of three engines.\n */\nexport function isAudioOutputSelectionSupported(): boolean {\n    return (\n        typeof HTMLMediaElement !== \"undefined\" &&\n        typeof (HTMLMediaElement.prototype as SinkCapableElement).setSinkId === \"function\"\n    );\n}\n\n/**\n * Send an element's audio to a specific output device.\n *\n * The `sinkId` comes from `useMediaDevices().audioOutputs` — and, like every device\n * label, that list is anonymous until a capture permission has been granted, so an\n * output picker realistically only works after the microphone has been allowed once.\n *\n * Returns `false` instead of throwing when the browser has no `setSinkId`, because\n * \"this engine cannot route audio\" is the common case and not an error the caller can\n * act on: the sound still plays, on the system default device. A rejected\n * `setSinkId` — a device that was unplugged between enumeration and playback is the\n * usual cause — also resolves `false`, and playback continues on the default.\n *\n * @param element - The `<audio>` or `<video>` element to route.\n * @param sinkId - Device id, or `\"\"` for the system default.\n * @returns `true` when the route was applied.\n *\n * @example\n * const ok = await setAudioOutput(audioRef.current, selectedSpeakerId);\n * if (!ok) toast(\"Este navegador não permite escolher a saída de som.\");\n */\nexport async function setAudioOutput(\n    element: HTMLMediaElement | null,\n    sinkId: string,\n): Promise<boolean> {\n    const target = element as SinkCapableElement | null;\n    if (!target || typeof target.setSinkId !== \"function\") return false;\n    try {\n        await target.setSinkId(sinkId);\n        return true;\n    } catch {\n        return false;\n    }\n}\n"],"mappings":";AAaA,SAAgB,IAA2C;CACvD,OACI,OAAO,mBAAqB,OAC5B,OAAQ,iBAAiB,UAAiC,aAAc;AAEhF;AAuBA,eAAsB,EAClB,GACA,GACgB;CAChB,IAAM,IAAS;CACf,IAAI,CAAC,KAAU,OAAO,EAAO,aAAc,YAAY,OAAO;CAC9D,IAAI;EAEA,OADA,MAAM,EAAO,UAAU,CAAM,GACtB;CACX,QAAQ;EACJ,OAAO;CACX;AACJ"}
+{"version":3,"file":"audio-output.js","names":["isAudioOutputSelectionSupported","setAudioOutput","target","element","sinkId"],"sources":["../../src/audio/audio-output.ts"],"sourcesContent":["/** A media element that may support output routing. */\ntype SinkCapableElement = HTMLMediaElement & {\n    setSinkId?: (sinkId: string) => Promise<void>;\n    sinkId?: string;\n};\n\n/**\n * Whether this browser can route audio to a chosen output device.\n *\n * Chromium-only at the time of writing: Firefox has `setSinkId` behind a preference\n * that is off by default, and Safari does not implement it at all. Check this before\n * rendering an output picker, or the control is a lie on two of three engines.\n */\nexport function isAudioOutputSelectionSupported(): boolean {\n    return (\n        typeof HTMLMediaElement !== \"undefined\" &&\n        typeof (HTMLMediaElement.prototype as SinkCapableElement).setSinkId === \"function\"\n    );\n}\n\n/**\n * Send an element's audio to a specific output device.\n *\n * The `sinkId` comes from `useMediaDevices().audioOutputs` — and, like every device\n * label, that list is anonymous until a capture permission has been granted, so an\n * output picker realistically only works after the microphone has been allowed once.\n *\n * Returns `false` instead of throwing when the browser has no `setSinkId`, because\n * \"this engine cannot route audio\" is the common case and not an error the caller can\n * act on: the sound still plays, on the system default device. A rejected\n * `setSinkId` — a device that was unplugged between enumeration and playback is the\n * usual cause — also resolves `false`, and playback continues on the default.\n *\n * @param element - The `<audio>` or `<video>` element to route.\n * @param sinkId - Device id, or `\"\"` for the system default.\n * @returns `true` when the route was applied.\n *\n * @example\n * const ok = await setAudioOutput(audioRef.current, selectedSpeakerId);\n * if (!ok) toast(\"Este navegador não permite escolher a saída de som.\");\n */\nexport async function setAudioOutput(\n    element: HTMLMediaElement | null,\n    sinkId: string,\n): Promise<boolean> {\n    const target = element as SinkCapableElement | null;\n    if (!target || typeof target.setSinkId !== \"function\") return false;\n    try {\n        await target.setSinkId(sinkId);\n        return true;\n    } catch {\n        return false;\n    }\n}\n"],"mappings":";AAaA,SAAgBA,IAA2C;CACvD,OACI,OAAO,mBAAqB,OAC5B,OAAQ,iBAAiB,UAAiC,aAAc;AAEhF;AAuBA,eAAsBC,EAClB,GACA,GACgB;CAChB,IAAMC,IAASC;CACf,IAAI,CAACD,KAAU,OAAOA,EAAO,aAAc,YAAY,OAAO;CAC9D,IAAI;EAEA,OADA,MAAMA,EAAO,UAAUE,CAAM,GACtB;CACX,QAAQ;EACJ,OAAO;CACX;AACJ"}

package/dist/audio/audio-player.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"audio-player.cjs","names":[],"sources":["../../src/audio/audio-player.ts"],"sourcesContent":["import { setAudioOutput } from \"./audio-output\";\n\nexport interface PlayAudioOptions {\n    /** Volume between 0 and 1. Default: 1. */\n    volume?: number;\n    /** Loop the clip. Default: false. */\n    loop?: boolean;\n    /** Begin playback immediately. Default: true. */\n    autoplay?: boolean;\n    /** Stop the previous clip managed by this player. Default: false. */\n    stopPrevious?: boolean;\n    /**\n     * Output device to play on, from `useMediaDevices().audioOutputs`.\n     *\n     * Ignored where the browser has no output routing (everything but Chromium) —\n     * the clip still plays, on the system default. Useful for a chime that must land\n     * on a headset while the call audio stays on the speakers.\n     */\n    sinkId?: string;\n    /** Fired when playback ends naturally. */\n    onEnded?: () => void;\n    /** Fired on playback error. */\n    onError?: (error: unknown) => void;\n}\n\n/**\n * Imperative handle over one \"current\" clip.\n *\n * Named `*Handle` like the SDK's other imperative handles (`ChatComposerHandle`,\n * `SignaturePadHandle`) — it is a control surface, not the `<AudioPlayer>` component.\n */\nexport interface AudioPlayerHandle {\n    /** Play `src`. Returns the underlying element, or `null` when the browser blocked autoplay. */\n    play: (src: string, options?: PlayAudioOptions) => Promise<HTMLAudioElement | null>;\n    /** Stop the currently-playing clip and rewind it. */\n    stop: () => void;\n    /** Currently playing audio element, or `null`. */\n    current: () => HTMLAudioElement | null;\n}\n\n/**\n * Create an isolated audio player that tracks a single \"current\" clip.\n * Multiple players coexist independently; use this when several layers of UI\n * need their own playback state.\n */\nexport function createAudioPlayer(): AudioPlayerHandle {\n    let current: HTMLAudioElement | null = null;\n\n    async function play(\n        src: string,\n        {\n            volume = 1,\n            loop = false,\n            autoplay = true,\n            stopPrevious = false,\n            sinkId,\n            onEnded,\n            onError,\n        }: PlayAudioOptions = {},\n    ): Promise<HTMLAudioElement | null> {\n        try {\n            if (stopPrevious && current) {\n                current.pause();\n                current.currentTime = 0;\n            }\n            const audio = new Audio(src);\n            audio.volume = Math.max(0, Math.min(1, volume));\n            audio.loop = loop;\n            audio.preload = \"auto\";\n            if (onEnded) audio.onended = onEnded;\n            if (onError) audio.onerror = (event) => onError(event);\n            current = audio;\n            // Routed before `play()`: applying a sink to an element that is already\n            // playing restarts its audio pipeline and clips the first few ms.\n            if (sinkId !== undefined) await setAudioOutput(audio, sinkId);\n            if (autoplay) await audio.play();\n            return audio;\n        } catch (error) {\n            onError?.(error);\n            return null;\n        }\n    }\n\n    function stop(): void {\n        if (!current) return;\n        current.pause();\n        current.currentTime = 0;\n    }\n\n    return { play, stop, current: () => current };\n}\n\nlet defaultPlayer: AudioPlayerHandle | null = null;\n\nfunction getDefaultPlayer(): AudioPlayerHandle {\n    if (!defaultPlayer) defaultPlayer = createAudioPlayer();\n    return defaultPlayer;\n}\n\n/**\n * Convenience wrapper around a shared {@link AudioPlayerHandle}. Use this for\n * one-off notification sounds. For more complex flows (e.g. several\n * simultaneous channels), build a dedicated player with {@link createAudioPlayer}.\n */\nexport async function playAudio(\n    src: string,\n    options?: PlayAudioOptions,\n): Promise<HTMLAudioElement | null> {\n    return getDefaultPlayer().play(src, options);\n}\n\n/** Stop the clip currently playing on the shared default player. */\nexport function stopAudio(): void {\n    getDefaultPlayer().stop();\n}\n"],"mappings":"sCA6CA,SAAgB,GAAuC,CACnD,IAAI,EAAmC,KAEvC,eAAe,EACX,EACA,CACI,SAAS,EACT,OAAO,GACP,WAAW,GACX,eAAe,GACf,SACA,UACA,WACkB,CAAC,EACS,CAChC,GAAI,CACI,GAAgB,IAChB,EAAQ,MAAM,EACd,EAAQ,YAAc,GAE1B,IAAM,EAAQ,IAAI,MAAM,CAAG,EAW3B,MAVA,GAAM,OAAS,KAAK,IAAI,EAAG,KAAK,IAAI,EAAG,CAAM,CAAC,EAC9C,EAAM,KAAO,EACb,EAAM,QAAU,OACZ,IAAS,EAAM,QAAU,GACzB,IAAS,EAAM,QAAW,GAAU,EAAQ,CAAK,GACrD,EAAU,EAGN,IAAW,IAAA,IAAW,MAAM,EAAA,eAAe,EAAO,CAAM,EACxD,GAAU,MAAM,EAAM,KAAK,EACxB,CACX,OAAS,EAAO,CAEZ,OADA,IAAU,CAAK,EACR,IACX,CACJ,CAEA,SAAS,GAAa,CACb,IACL,EAAQ,MAAM,EACd,EAAQ,YAAc,EAC1B,CAEA,MAAO,CAAE,OAAM,OAAM,YAAe,CAAQ,CAChD,CAEA,IAAI,EAA0C,KAE9C,SAAS,GAAsC,CAE3C,MADA,CAAoB,IAAgB,EAAkB,EAC/C,CACX,CAOA,eAAsB,EAClB,EACA,EACgC,CAChC,OAAO,EAAiB,CAAC,CAAC,KAAK,EAAK,CAAO,CAC/C,CAGA,SAAgB,GAAkB,CAC9B,EAAiB,CAAC,CAAC,KAAK,CAC5B"}
+{"version":3,"file":"audio-player.cjs","names":["setAudioOutput","require_audio_output","createAudioPlayer","current","play","stopPrevious","audio","src","volume","loop","onEnded","onError","event","sinkId","autoplay","error","stop","defaultPlayer","getDefaultPlayer","playAudio","options","stopAudio"],"sources":["../../src/audio/audio-player.ts"],"sourcesContent":["import { setAudioOutput } from \"./audio-output\";\n\nexport interface PlayAudioOptions {\n    /** Volume between 0 and 1. Default: 1. */\n    volume?: number;\n    /** Loop the clip. Default: false. */\n    loop?: boolean;\n    /** Begin playback immediately. Default: true. */\n    autoplay?: boolean;\n    /** Stop the previous clip managed by this player. Default: false. */\n    stopPrevious?: boolean;\n    /**\n     * Output device to play on, from `useMediaDevices().audioOutputs`.\n     *\n     * Ignored where the browser has no output routing (everything but Chromium) —\n     * the clip still plays, on the system default. Useful for a chime that must land\n     * on a headset while the call audio stays on the speakers.\n     */\n    sinkId?: string;\n    /** Fired when playback ends naturally. */\n    onEnded?: () => void;\n    /** Fired on playback error. */\n    onError?: (error: unknown) => void;\n}\n\n/**\n * Imperative handle over one \"current\" clip.\n *\n * Named `*Handle` like the SDK's other imperative handles (`ChatComposerHandle`,\n * `SignaturePadHandle`) — it is a control surface, not the `<AudioPlayer>` component.\n */\nexport interface AudioPlayerHandle {\n    /** Play `src`. Returns the underlying element, or `null` when the browser blocked autoplay. */\n    play: (src: string, options?: PlayAudioOptions) => Promise<HTMLAudioElement | null>;\n    /** Stop the currently-playing clip and rewind it. */\n    stop: () => void;\n    /** Currently playing audio element, or `null`. */\n    current: () => HTMLAudioElement | null;\n}\n\n/**\n * Create an isolated audio player that tracks a single \"current\" clip.\n * Multiple players coexist independently; use this when several layers of UI\n * need their own playback state.\n */\nexport function createAudioPlayer(): AudioPlayerHandle {\n    let current: HTMLAudioElement | null = null;\n\n    async function play(\n        src: string,\n        {\n            volume = 1,\n            loop = false,\n            autoplay = true,\n            stopPrevious = false,\n            sinkId,\n            onEnded,\n            onError,\n        }: PlayAudioOptions = {},\n    ): Promise<HTMLAudioElement | null> {\n        try {\n            if (stopPrevious && current) {\n                current.pause();\n                current.currentTime = 0;\n            }\n            const audio = new Audio(src);\n            audio.volume = Math.max(0, Math.min(1, volume));\n            audio.loop = loop;\n            audio.preload = \"auto\";\n            if (onEnded) audio.onended = onEnded;\n            if (onError) audio.onerror = (event) => onError(event);\n            current = audio;\n            // Routed before `play()`: applying a sink to an element that is already\n            // playing restarts its audio pipeline and clips the first few ms.\n            if (sinkId !== undefined) await setAudioOutput(audio, sinkId);\n            if (autoplay) await audio.play();\n            return audio;\n        } catch (error) {\n            onError?.(error);\n            return null;\n        }\n    }\n\n    function stop(): void {\n        if (!current) return;\n        current.pause();\n        current.currentTime = 0;\n    }\n\n    return { play, stop, current: () => current };\n}\n\nlet defaultPlayer: AudioPlayerHandle | null = null;\n\nfunction getDefaultPlayer(): AudioPlayerHandle {\n    if (!defaultPlayer) defaultPlayer = createAudioPlayer();\n    return defaultPlayer;\n}\n\n/**\n * Convenience wrapper around a shared {@link AudioPlayerHandle}. Use this for\n * one-off notification sounds. For more complex flows (e.g. several\n * simultaneous channels), build a dedicated player with {@link createAudioPlayer}.\n */\nexport async function playAudio(\n    src: string,\n    options?: PlayAudioOptions,\n): Promise<HTMLAudioElement | null> {\n    return getDefaultPlayer().play(src, options);\n}\n\n/** Stop the clip currently playing on the shared default player. */\nexport function stopAudio(): void {\n    getDefaultPlayer().stop();\n}\n"],"mappings":"A,sCA6CA,SAAgBE,GAAuC,CACnD,IAAIC,EAAmC,KAEvC,eAAeC,EACX,EACA,CACI,SAAS,EACT,OAAO,GACP,WAAW,GACX,eAAe,GACf,SACA,UACA,WACkB,CAAC,EACS,CAChC,GAAI,CACIC,GAAgBF,IAChBA,EAAQ,MAAM,EACd,EAAQ,YAAc,GAE1B,IAAMG,EAAQ,IAAI,MAAMC,CAAG,EAW3B,MAVA,GAAM,OAAS,KAAK,IAAI,EAAG,KAAK,IAAI,EAAGC,CAAM,CAAC,EAC9C,EAAM,KAAOC,EACb,EAAM,QAAU,OACZC,IAAS,EAAM,QAAUA,GACzBC,IAAS,EAAM,QAAW,GAAUA,EAAQC,CAAK,GACrD,EAAUN,EAGNO,IAAW,IAAA,IAAW,MAAMb,EAAAA,eAAeM,EAAOO,CAAM,EACxDC,GAAU,MAAMR,EAAM,KAAK,EACxBA,CACX,OAASS,EAAO,CAEZ,OADAJ,IAAUI,CAAK,EACR,IACX,CACJ,CAEA,SAASC,GAAa,CACbb,IACLA,EAAQ,MAAM,EACd,EAAQ,YAAc,EAC1B,CAEA,MAAO,CAAE,OAAM,OAAM,YAAeA,CAAQ,CAChD,CAEA,IAAIc,EAA0C,KAE9C,SAASC,GAAsC,CAE3C,MADA,CAAoBD,IAAgBf,EAAkB,EAC/Ce,CACX,CAOA,eAAsBE,EAClB,EACA,EACgC,CAChC,OAAOD,EAAiB,CAAC,CAAC,KAAKX,EAAKa,CAAO,CAC/C,CAGA,SAAgBC,GAAkB,CAC9BH,EAAiB,CAAC,CAAC,KAAK,CAC5B,C"}

package/dist/audio/audio-player.js.map

@@ -1 +1 @@
-{"version":3,"file":"audio-player.js","names":[],"sources":["../../src/audio/audio-player.ts"],"sourcesContent":["import { setAudioOutput } from \"./audio-output\";\n\nexport interface PlayAudioOptions {\n    /** Volume between 0 and 1. Default: 1. */\n    volume?: number;\n    /** Loop the clip. Default: false. */\n    loop?: boolean;\n    /** Begin playback immediately. Default: true. */\n    autoplay?: boolean;\n    /** Stop the previous clip managed by this player. Default: false. */\n    stopPrevious?: boolean;\n    /**\n     * Output device to play on, from `useMediaDevices().audioOutputs`.\n     *\n     * Ignored where the browser has no output routing (everything but Chromium) —\n     * the clip still plays, on the system default. Useful for a chime that must land\n     * on a headset while the call audio stays on the speakers.\n     */\n    sinkId?: string;\n    /** Fired when playback ends naturally. */\n    onEnded?: () => void;\n    /** Fired on playback error. */\n    onError?: (error: unknown) => void;\n}\n\n/**\n * Imperative handle over one \"current\" clip.\n *\n * Named `*Handle` like the SDK's other imperative handles (`ChatComposerHandle`,\n * `SignaturePadHandle`) — it is a control surface, not the `<AudioPlayer>` component.\n */\nexport interface AudioPlayerHandle {\n    /** Play `src`. Returns the underlying element, or `null` when the browser blocked autoplay. */\n    play: (src: string, options?: PlayAudioOptions) => Promise<HTMLAudioElement | null>;\n    /** Stop the currently-playing clip and rewind it. */\n    stop: () => void;\n    /** Currently playing audio element, or `null`. */\n    current: () => HTMLAudioElement | null;\n}\n\n/**\n * Create an isolated audio player that tracks a single \"current\" clip.\n * Multiple players coexist independently; use this when several layers of UI\n * need their own playback state.\n */\nexport function createAudioPlayer(): AudioPlayerHandle {\n    let current: HTMLAudioElement | null = null;\n\n    async function play(\n        src: string,\n        {\n            volume = 1,\n            loop = false,\n            autoplay = true,\n            stopPrevious = false,\n            sinkId,\n            onEnded,\n            onError,\n        }: PlayAudioOptions = {},\n    ): Promise<HTMLAudioElement | null> {\n        try {\n            if (stopPrevious && current) {\n                current.pause();\n                current.currentTime = 0;\n            }\n            const audio = new Audio(src);\n            audio.volume = Math.max(0, Math.min(1, volume));\n            audio.loop = loop;\n            audio.preload = \"auto\";\n            if (onEnded) audio.onended = onEnded;\n            if (onError) audio.onerror = (event) => onError(event);\n            current = audio;\n            // Routed before `play()`: applying a sink to an element that is already\n            // playing restarts its audio pipeline and clips the first few ms.\n            if (sinkId !== undefined) await setAudioOutput(audio, sinkId);\n            if (autoplay) await audio.play();\n            return audio;\n        } catch (error) {\n            onError?.(error);\n            return null;\n        }\n    }\n\n    function stop(): void {\n        if (!current) return;\n        current.pause();\n        current.currentTime = 0;\n    }\n\n    return { play, stop, current: () => current };\n}\n\nlet defaultPlayer: AudioPlayerHandle | null = null;\n\nfunction getDefaultPlayer(): AudioPlayerHandle {\n    if (!defaultPlayer) defaultPlayer = createAudioPlayer();\n    return defaultPlayer;\n}\n\n/**\n * Convenience wrapper around a shared {@link AudioPlayerHandle}. Use this for\n * one-off notification sounds. For more complex flows (e.g. several\n * simultaneous channels), build a dedicated player with {@link createAudioPlayer}.\n */\nexport async function playAudio(\n    src: string,\n    options?: PlayAudioOptions,\n): Promise<HTMLAudioElement | null> {\n    return getDefaultPlayer().play(src, options);\n}\n\n/** Stop the clip currently playing on the shared default player. */\nexport function stopAudio(): void {\n    getDefaultPlayer().stop();\n}\n"],"mappings":";;AA6CA,SAAgB,IAAuC;CACnD,IAAI,IAAmC;CAEvC,eAAe,EACX,GACA,EACI,YAAS,GACT,UAAO,IACP,cAAW,IACX,kBAAe,IACf,WACA,YACA,eACkB,CAAC,GACS;EAChC,IAAI;GACA,AAAI,KAAgB,MAChB,EAAQ,MAAM,GACd,EAAQ,cAAc;GAE1B,IAAM,IAAQ,IAAI,MAAM,CAAG;GAW3B,OAVA,EAAM,SAAS,KAAK,IAAI,GAAG,KAAK,IAAI,GAAG,CAAM,CAAC,GAC9C,EAAM,OAAO,GACb,EAAM,UAAU,QACZ,MAAS,EAAM,UAAU,IACzB,MAAS,EAAM,WAAW,MAAU,EAAQ,CAAK,IACrD,IAAU,GAGN,MAAW,KAAA,KAAW,MAAM,EAAe,GAAO,CAAM,GACxD,KAAU,MAAM,EAAM,KAAK,GACxB;EACX,SAAS,GAAO;GAEZ,OADA,IAAU,CAAK,GACR;EACX;CACJ;CAEA,SAAS,IAAa;EACb,MACL,EAAQ,MAAM,GACd,EAAQ,cAAc;CAC1B;CAEA,OAAO;EAAE;EAAM;EAAM,eAAe;CAAQ;AAChD;AAEA,IAAI,IAA0C;AAE9C,SAAS,IAAsC;CAE3C,OADA,AAAoB,MAAgB,EAAkB,GAC/C;AACX;AAOA,eAAsB,EAClB,GACA,GACgC;CAChC,OAAO,EAAiB,CAAC,CAAC,KAAK,GAAK,CAAO;AAC/C;AAGA,SAAgB,IAAkB;CAC9B,EAAiB,CAAC,CAAC,KAAK;AAC5B"}
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Useful for a chime that must land\n     * on a headset while the call audio stays on the speakers.\n     */\n    sinkId?: string;\n    /** Fired when playback ends naturally. */\n    onEnded?: () => void;\n    /** Fired on playback error. */\n    onError?: (error: unknown) => void;\n}\n\n/**\n * Imperative handle over one \"current\" clip.\n *\n * Named `*Handle` like the SDK's other imperative handles (`ChatComposerHandle`,\n * `SignaturePadHandle`) — it is a control surface, not the `<AudioPlayer>` component.\n */\nexport interface AudioPlayerHandle {\n    /** Play `src`. 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package/dist/audio/audio-recorder.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"audio-recorder.cjs","names":[],"sources":["../../src/audio/audio-recorder.ts"],"sourcesContent":["import {\n    createMediaRecorder,\n    pickRecordingMimeType,\n    type MediaRecorderStatus,\n} from \"@/capture/media-recorder\";\n\n/** Lifecycle of a recording. Shared with the video recorder — the states are the same. */\nexport type AudioRecorderStatus = MediaRecorderStatus;\n\n/**\n * Container/codec candidates, best first.\n *\n * No browser supports all of these, and none supports MP3 or WAV from\n * `MediaRecorder` — see the note on {@link pickAudioMimeType}. Opus in WebM is the\n * first choice because it is the smallest at speech bitrates and is what Chromium\n * and Firefox produce natively; `audio/mp4` is here for Safari, which produces AAC\n * and nothing else.\n *\n * Ported from the intersection of what `MediaRecorder.isTypeSupported` accepts in\n * Chrome 120+, Firefox 115+ and Safari 17. Re-check against\n * <https://developer.mozilla.org/docs/Web/API/MediaRecorder/isTypeSupported> before\n * adding to the list.\n */\nexport const AUDIO_MIME_CANDIDATES: readonly string[] = [\n    \"audio/webm;codecs=opus\",\n    \"audio/webm\",\n    \"audio/ogg;codecs=opus\",\n    \"audio/mp4;codecs=mp4a.40.2\",\n    \"audio/mp4\",\n];\n\n/** A finished recording. */\nexport interface AudioRecording {\n    /** The audio. Wrap with `useObjectUrl` to play it, or POST it as-is. */\n    blob: Blob;\n    /** What the browser actually produced — not necessarily what you asked for. */\n    mimeType: string;\n    /** Recorded length, excluding time spent paused. */\n    durationMs: number;\n}\n\n/** Options for {@link createAudioRecorder}. */\nexport interface AudioRecorderOptions {\n    /**\n     * Force a container. Throws when the browser cannot produce it.\n     *\n     * Leave it out. The default negotiates from {@link AUDIO_MIME_CANDIDATES}, which\n     * is the only way one call site works on both Chromium and Safari.\n     */\n    mimeType?: string;\n    /** Target bitrate. 32000–64000 is plenty for speech in Opus. */\n    audioBitsPerSecond?: number;\n    /**\n     * Emit a chunk every N ms through `onChunk`, for streaming upload.\n     *\n     * Without it the whole recording is buffered in memory until `stop()` — fine for\n     * a voice note, not fine for an hour-long meeting.\n     */\n    timesliceMs?: number;\n    /** Receives each chunk when `timesliceMs` is set. Chunks are **not** independently playable. */\n    onChunk?: (chunk: Blob) => void;\n    /** Recorder-level failure (device unplugged mid-recording, encoder error). */\n    onError?: (error: unknown) => void;\n}\n\n/** Imperative recorder. */\nexport interface AudioRecorderHandle {\n    /** Begin recording. No-op when already recording or paused. */\n    start: () => void;\n    /** Pause. The clock stops; `durationMs` freezes. */\n    pause: () => void;\n    /** Resume after `pause()`. */\n    resume: () => void;\n    /** Stop and resolve with the assembled recording. */\n    stop: () => Promise<AudioRecording>;\n    /** Stop and throw the audio away. */\n    cancel: () => void;\n    status: () => AudioRecorderStatus;\n    /** Recorded length so far, excluding paused time. */\n    durationMs: () => number;\n    /** The negotiated container. */\n    mimeType: string;\n}\n\n/** Whether `MediaRecorder` exists and can produce at least one audio container. */\nexport function isAudioRecordingSupported(): boolean {\n    return typeof MediaRecorder !== \"undefined\" && pickAudioMimeType() !== null;\n}\n\n/**\n * First container in `preferred` the browser can actually produce, or `null`.\n *\n * **There is no MP3 or WAV here, and that is not an omission.** `MediaRecorder`\n * emits Opus (in WebM or Ogg) on Chromium and Firefox and AAC (in MP4) on Safari —\n * no engine implements an MP3 or WAV encoder for it. If a backend needs WAV,\n * {@link blobToWav} converts one client-side with no dependency; if it needs MP3,\n * transcode on the server. Shipping an MP3 encoder would mean a WASM build of the\n * order of 150 KB in every consumer's bundle to serve one format, which is the\n * trade this SDK does not make.\n *\n * @param preferred - Candidates, best first. Defaults to {@link AUDIO_MIME_CANDIDATES}.\n * @returns A supported MIME type, or `null` when none is.\n */\nexport function pickAudioMimeType(\n    preferred: readonly string[] = AUDIO_MIME_CANDIDATES,\n): string | null {\n    return pickRecordingMimeType(preferred);\n}\n\n/**\n * Wrap a `MediaStream` in a recorder with a real state machine and an honest clock.\n *\n * The engine is shared with the video recorder (`createMediaRecorder`), because the\n * subtle parts are not audio-specific: `MediaRecorder` reports **no duration** (WebM\n * it writes carries none in its header, which is why `<audio>` shows `Infinity` for a\n * fresh recording) so the clock is kept by hand and subtracts paused time; and\n * `stop()` returns *before* the last `dataavailable` event, so the blob can only be\n * assembled in `onstop`. What stays here is the part that genuinely differs — the\n * container list.\n *\n * The stream is **not** owned here: `stop()` leaves the microphone open so a retake\n * does not need a second permission round-trip. Release it with the owning\n * `useMicrophone().stop()`.\n *\n * @param stream - A live audio stream, usually from {@link useMicrophone}.\n * @param options - See {@link AudioRecorderOptions}.\n * @returns The imperative recorder.\n * @throws When `MediaRecorder` is unavailable, or an explicit `mimeType` is unsupported.\n */\nexport function createAudioRecorder(\n    stream: MediaStream,\n    options: AudioRecorderOptions = {},\n): AudioRecorderHandle {\n    return createMediaRecorder(stream, {\n        candidates: AUDIO_MIME_CANDIDATES,\n        kind: \"audio\",\n        mimeType: options.mimeType,\n        audioBitsPerSecond: options.audioBitsPerSecond,\n        timesliceMs: options.timesliceMs,\n        onChunk: options.onChunk,\n        onError: options.onError,\n    });\n}\n"],"mappings":"iDAuBA,IAAa,EAA2C,CACpD,yBACA,aACA,wBACA,6BACA,WACJ,EAwDA,SAAgB,GAAqC,CACjD,OAAO,OAAO,cAAkB,KAAe,EAAkB,IAAM,IAC3E,CAgBA,SAAgB,EACZ,EAA+B,EAClB,CACb,OAAO,EAAA,sBAAsB,CAAS,CAC1C,CAsBA,SAAgB,EACZ,EACA,EAAgC,CAAC,EACd,CACnB,OAAO,EAAA,oBAAoB,EAAQ,CAC/B,WAAY,EACZ,KAAM,QACN,SAAU,EAAQ,SAClB,mBAAoB,EAAQ,mBAC5B,YAAa,EAAQ,YACrB,QAAS,EAAQ,QACjB,QAAS,EAAQ,OACrB,CAAC,CACL"}
+{"version":3,"file":"audio-recorder.cjs","names":["pickRecordingMimeType","createMediaRecorder","require_media_recorder","AUDIO_MIME_CANDIDATES","isAudioRecordingSupported","pickAudioMimeType","preferred","createAudioRecorder","stream","options"],"sources":["../../src/audio/audio-recorder.ts"],"sourcesContent":["import {\n    createMediaRecorder,\n    pickRecordingMimeType,\n    type MediaRecorderStatus,\n} from \"@/capture/media-recorder\";\n\n/** Lifecycle of a recording. Shared with the video recorder — the states are the same. */\nexport type AudioRecorderStatus = MediaRecorderStatus;\n\n/**\n * Container/codec candidates, best first.\n *\n * No browser supports all of these, and none supports MP3 or WAV from\n * `MediaRecorder` — see the note on {@link pickAudioMimeType}. Opus in WebM is the\n * first choice because it is the smallest at speech bitrates and is what Chromium\n * and Firefox produce natively; `audio/mp4` is here for Safari, which produces AAC\n * and nothing else.\n *\n * Ported from the intersection of what `MediaRecorder.isTypeSupported` accepts in\n * Chrome 120+, Firefox 115+ and Safari 17. Re-check against\n * <https://developer.mozilla.org/docs/Web/API/MediaRecorder/isTypeSupported> before\n * adding to the list.\n */\nexport const AUDIO_MIME_CANDIDATES: readonly string[] = [\n    \"audio/webm;codecs=opus\",\n    \"audio/webm\",\n    \"audio/ogg;codecs=opus\",\n    \"audio/mp4;codecs=mp4a.40.2\",\n    \"audio/mp4\",\n];\n\n/** A finished recording. */\nexport interface AudioRecording {\n    /** The audio. Wrap with `useObjectUrl` to play it, or POST it as-is. */\n    blob: Blob;\n    /** What the browser actually produced — not necessarily what you asked for. */\n    mimeType: string;\n    /** Recorded length, excluding time spent paused. */\n    durationMs: number;\n}\n\n/** Options for {@link createAudioRecorder}. */\nexport interface AudioRecorderOptions {\n    /**\n     * Force a container. Throws when the browser cannot produce it.\n     *\n     * Leave it out. The default negotiates from {@link AUDIO_MIME_CANDIDATES}, which\n     * is the only way one call site works on both Chromium and Safari.\n     */\n    mimeType?: string;\n    /** Target bitrate. 32000–64000 is plenty for speech in Opus. */\n    audioBitsPerSecond?: number;\n    /**\n     * Emit a chunk every N ms through `onChunk`, for streaming upload.\n     *\n     * Without it the whole recording is buffered in memory until `stop()` — fine for\n     * a voice note, not fine for an hour-long meeting.\n     */\n    timesliceMs?: number;\n    /** Receives each chunk when `timesliceMs` is set. Chunks are **not** independently playable. */\n    onChunk?: (chunk: Blob) => void;\n    /** Recorder-level failure (device unplugged mid-recording, encoder error). */\n    onError?: (error: unknown) => void;\n}\n\n/** Imperative recorder. */\nexport interface AudioRecorderHandle {\n    /** Begin recording. No-op when already recording or paused. */\n    start: () => void;\n    /** Pause. The clock stops; `durationMs` freezes. */\n    pause: () => void;\n    /** Resume after `pause()`. */\n    resume: () => void;\n    /** Stop and resolve with the assembled recording. */\n    stop: () => Promise<AudioRecording>;\n    /** Stop and throw the audio away. */\n    cancel: () => void;\n    status: () => AudioRecorderStatus;\n    /** Recorded length so far, excluding paused time. */\n    durationMs: () => number;\n    /** The negotiated container. */\n    mimeType: string;\n}\n\n/** Whether `MediaRecorder` exists and can produce at least one audio container. */\nexport function isAudioRecordingSupported(): boolean {\n    return typeof MediaRecorder !== \"undefined\" && pickAudioMimeType() !== null;\n}\n\n/**\n * First container in `preferred` the browser can actually produce, or `null`.\n *\n * **There is no MP3 or WAV here, and that is not an omission.** `MediaRecorder`\n * emits Opus (in WebM or Ogg) on Chromium and Firefox and AAC (in MP4) on Safari —\n * no engine implements an MP3 or WAV encoder for it. If a backend needs WAV,\n * {@link blobToWav} converts one client-side with no dependency; if it needs MP3,\n * transcode on the server. Shipping an MP3 encoder would mean a WASM build of the\n * order of 150 KB in every consumer's bundle to serve one format, which is the\n * trade this SDK does not make.\n *\n * @param preferred - Candidates, best first. Defaults to {@link AUDIO_MIME_CANDIDATES}.\n * @returns A supported MIME type, or `null` when none is.\n */\nexport function pickAudioMimeType(\n    preferred: readonly string[] = AUDIO_MIME_CANDIDATES,\n): string | null {\n    return pickRecordingMimeType(preferred);\n}\n\n/**\n * Wrap a `MediaStream` in a recorder with a real state machine and an honest clock.\n *\n * The engine is shared with the video recorder (`createMediaRecorder`), because the\n * subtle parts are not audio-specific: `MediaRecorder` reports **no duration** (WebM\n * it writes carries none in its header, which is why `<audio>` shows `Infinity` for a\n * fresh recording) so the clock is kept by hand and subtracts paused time; and\n * `stop()` returns *before* the last `dataavailable` event, so the blob can only be\n * assembled in `onstop`. What stays here is the part that genuinely differs — the\n * container list.\n *\n * The stream is **not** owned here: `stop()` leaves the microphone open so a retake\n * does not need a second permission round-trip. Release it with the owning\n * `useMicrophone().stop()`.\n *\n * @param stream - A live audio stream, usually from {@link useMicrophone}.\n * @param options - See {@link AudioRecorderOptions}.\n * @returns The imperative recorder.\n * @throws When `MediaRecorder` is unavailable, or an explicit `mimeType` is unsupported.\n */\nexport function createAudioRecorder(\n    stream: MediaStream,\n    options: AudioRecorderOptions = {},\n): AudioRecorderHandle {\n    return createMediaRecorder(stream, {\n        candidates: AUDIO_MIME_CANDIDATES,\n        kind: \"audio\",\n        mimeType: options.mimeType,\n        audioBitsPerSecond: options.audioBitsPerSecond,\n        timesliceMs: options.timesliceMs,\n        onChunk: options.onChunk,\n        onError: options.onError,\n    });\n}\n"],"mappings":"A,iDAuBA,IAAaG,EAA2C,CACpD,yBACA,aACA,wBACA,6BACA,WACJ,EAwDA,SAAgBC,GAAqC,CACjD,OAAO,OAAO,cAAkB,KAAeC,EAAkB,IAAM,IAC3E,CAgBA,SAAgBA,EACZ,EAA+BF,EAClB,CACb,OAAOH,EAAAA,sBAAsBM,CAAS,CAC1C,CAsBA,SAAgBC,EACZ,EACA,EAAgC,CAAC,EACd,CACnB,OAAON,EAAAA,oBAAoBO,EAAQ,CAC/B,WAAYL,EACZ,KAAM,QACN,SAAUM,EAAQ,SAClB,mBAAoBA,EAAQ,mBAC5B,YAAaA,EAAQ,YACrB,QAASA,EAAQ,QACjB,QAASA,EAAQ,OACrB,CAAC,CACL,C"}

package/dist/audio/audio-recorder.js.map

@@ -1 +1 @@
-{"version":3,"file":"audio-recorder.js","names":[],"sources":["../../src/audio/audio-recorder.ts"],"sourcesContent":["import {\n    createMediaRecorder,\n    pickRecordingMimeType,\n    type MediaRecorderStatus,\n} from \"@/capture/media-recorder\";\n\n/** Lifecycle of a recording. Shared with the video recorder — the states are the same. */\nexport type AudioRecorderStatus = MediaRecorderStatus;\n\n/**\n * Container/codec candidates, best first.\n *\n * No browser supports all of these, and none supports MP3 or WAV from\n * `MediaRecorder` — see the note on {@link pickAudioMimeType}. Opus in WebM is the\n * first choice because it is the smallest at speech bitrates and is what Chromium\n * and Firefox produce natively; `audio/mp4` is here for Safari, which produces AAC\n * and nothing else.\n *\n * Ported from the intersection of what `MediaRecorder.isTypeSupported` accepts in\n * Chrome 120+, Firefox 115+ and Safari 17. Re-check against\n * <https://developer.mozilla.org/docs/Web/API/MediaRecorder/isTypeSupported> before\n * adding to the list.\n */\nexport const AUDIO_MIME_CANDIDATES: readonly string[] = [\n    \"audio/webm;codecs=opus\",\n    \"audio/webm\",\n    \"audio/ogg;codecs=opus\",\n    \"audio/mp4;codecs=mp4a.40.2\",\n    \"audio/mp4\",\n];\n\n/** A finished recording. */\nexport interface AudioRecording {\n    /** The audio. Wrap with `useObjectUrl` to play it, or POST it as-is. */\n    blob: Blob;\n    /** What the browser actually produced — not necessarily what you asked for. */\n    mimeType: string;\n    /** Recorded length, excluding time spent paused. */\n    durationMs: number;\n}\n\n/** Options for {@link createAudioRecorder}. */\nexport interface AudioRecorderOptions {\n    /**\n     * Force a container. Throws when the browser cannot produce it.\n     *\n     * Leave it out. The default negotiates from {@link AUDIO_MIME_CANDIDATES}, which\n     * is the only way one call site works on both Chromium and Safari.\n     */\n    mimeType?: string;\n    /** Target bitrate. 32000–64000 is plenty for speech in Opus. */\n    audioBitsPerSecond?: number;\n    /**\n     * Emit a chunk every N ms through `onChunk`, for streaming upload.\n     *\n     * Without it the whole recording is buffered in memory until `stop()` — fine for\n     * a voice note, not fine for an hour-long meeting.\n     */\n    timesliceMs?: number;\n    /** Receives each chunk when `timesliceMs` is set. Chunks are **not** independently playable. */\n    onChunk?: (chunk: Blob) => void;\n    /** Recorder-level failure (device unplugged mid-recording, encoder error). */\n    onError?: (error: unknown) => void;\n}\n\n/** Imperative recorder. */\nexport interface AudioRecorderHandle {\n    /** Begin recording. No-op when already recording or paused. */\n    start: () => void;\n    /** Pause. The clock stops; `durationMs` freezes. */\n    pause: () => void;\n    /** Resume after `pause()`. */\n    resume: () => void;\n    /** Stop and resolve with the assembled recording. */\n    stop: () => Promise<AudioRecording>;\n    /** Stop and throw the audio away. */\n    cancel: () => void;\n    status: () => AudioRecorderStatus;\n    /** Recorded length so far, excluding paused time. */\n    durationMs: () => number;\n    /** The negotiated container. */\n    mimeType: string;\n}\n\n/** Whether `MediaRecorder` exists and can produce at least one audio container. */\nexport function isAudioRecordingSupported(): boolean {\n    return typeof MediaRecorder !== \"undefined\" && pickAudioMimeType() !== null;\n}\n\n/**\n * First container in `preferred` the browser can actually produce, or `null`.\n *\n * **There is no MP3 or WAV here, and that is not an omission.** `MediaRecorder`\n * emits Opus (in WebM or Ogg) on Chromium and Firefox and AAC (in MP4) on Safari —\n * no engine implements an MP3 or WAV encoder for it. If a backend needs WAV,\n * {@link blobToWav} converts one client-side with no dependency; if it needs MP3,\n * transcode on the server. Shipping an MP3 encoder would mean a WASM build of the\n * order of 150 KB in every consumer's bundle to serve one format, which is the\n * trade this SDK does not make.\n *\n * @param preferred - Candidates, best first. Defaults to {@link AUDIO_MIME_CANDIDATES}.\n * @returns A supported MIME type, or `null` when none is.\n */\nexport function pickAudioMimeType(\n    preferred: readonly string[] = AUDIO_MIME_CANDIDATES,\n): string | null {\n    return pickRecordingMimeType(preferred);\n}\n\n/**\n * Wrap a `MediaStream` in a recorder with a real state machine and an honest clock.\n *\n * The engine is shared with the video recorder (`createMediaRecorder`), because the\n * subtle parts are not audio-specific: `MediaRecorder` reports **no duration** (WebM\n * it writes carries none in its header, which is why `<audio>` shows `Infinity` for a\n * fresh recording) so the clock is kept by hand and subtracts paused time; and\n * `stop()` returns *before* the last `dataavailable` event, so the blob can only be\n * assembled in `onstop`. What stays here is the part that genuinely differs — the\n * container list.\n *\n * The stream is **not** owned here: `stop()` leaves the microphone open so a retake\n * does not need a second permission round-trip. Release it with the owning\n * `useMicrophone().stop()`.\n *\n * @param stream - A live audio stream, usually from {@link useMicrophone}.\n * @param options - See {@link AudioRecorderOptions}.\n * @returns The imperative recorder.\n * @throws When `MediaRecorder` is unavailable, or an explicit `mimeType` is unsupported.\n */\nexport function createAudioRecorder(\n    stream: MediaStream,\n    options: AudioRecorderOptions = {},\n): AudioRecorderHandle {\n    return createMediaRecorder(stream, {\n        candidates: AUDIO_MIME_CANDIDATES,\n        kind: \"audio\",\n        mimeType: options.mimeType,\n        audioBitsPerSecond: options.audioBitsPerSecond,\n        timesliceMs: options.timesliceMs,\n        onChunk: options.onChunk,\n        onError: options.onError,\n    });\n}\n"],"mappings":";;AAuBA,IAAa,IAA2C;CACpD;CACA;CACA;CACA;CACA;AACJ;AAwDA,SAAgB,IAAqC;CACjD,OAAO,OAAO,gBAAkB,OAAe,EAAkB,MAAM;AAC3E;AAgBA,SAAgB,EACZ,IAA+B,GAClB;CACb,OAAO,EAAsB,CAAS;AAC1C;AAsBA,SAAgB,EACZ,GACA,IAAgC,CAAC,GACd;CACnB,OAAO,EAAoB,GAAQ;EAC/B,YAAY;EACZ,MAAM;EACN,UAAU,EAAQ;EAClB,oBAAoB,EAAQ;EAC5B,aAAa,EAAQ;EACrB,SAAS,EAAQ;EACjB,SAAS,EAAQ;CACrB,CAAC;AACL"}
+{"version":3,"file":"audio-recorder.js","names":["AUDIO_MIME_CANDIDATES","isAudioRecordingSupported","pickAudioMimeType","pickRecordingMimeType","preferred","createAudioRecorder","createMediaRecorder","stream","options"],"sources":["../../src/audio/audio-recorder.ts"],"sourcesContent":["import {\n    createMediaRecorder,\n    pickRecordingMimeType,\n    type MediaRecorderStatus,\n} from \"@/capture/media-recorder\";\n\n/** Lifecycle of a recording. Shared with the video recorder — the states are the same. */\nexport type AudioRecorderStatus = MediaRecorderStatus;\n\n/**\n * Container/codec candidates, best first.\n *\n * No browser supports all of these, and none supports MP3 or WAV from\n * `MediaRecorder` — see the note on {@link pickAudioMimeType}. Opus in WebM is the\n * first choice because it is the smallest at speech bitrates and is what Chromium\n * and Firefox produce natively; `audio/mp4` is here for Safari, which produces AAC\n * and nothing else.\n *\n * Ported from the intersection of what `MediaRecorder.isTypeSupported` accepts in\n * Chrome 120+, Firefox 115+ and Safari 17. Re-check against\n * <https://developer.mozilla.org/docs/Web/API/MediaRecorder/isTypeSupported> before\n * adding to the list.\n */\nexport const AUDIO_MIME_CANDIDATES: readonly string[] = [\n    \"audio/webm;codecs=opus\",\n    \"audio/webm\",\n    \"audio/ogg;codecs=opus\",\n    \"audio/mp4;codecs=mp4a.40.2\",\n    \"audio/mp4\",\n];\n\n/** A finished recording. */\nexport interface AudioRecording {\n    /** The audio. Wrap with `useObjectUrl` to play it, or POST it as-is. */\n    blob: Blob;\n    /** What the browser actually produced — not necessarily what you asked for. */\n    mimeType: string;\n    /** Recorded length, excluding time spent paused. */\n    durationMs: number;\n}\n\n/** Options for {@link createAudioRecorder}. */\nexport interface AudioRecorderOptions {\n    /**\n     * Force a container. Throws when the browser cannot produce it.\n     *\n     * Leave it out. The default negotiates from {@link AUDIO_MIME_CANDIDATES}, which\n     * is the only way one call site works on both Chromium and Safari.\n     */\n    mimeType?: string;\n    /** Target bitrate. 32000–64000 is plenty for speech in Opus. */\n    audioBitsPerSecond?: number;\n    /**\n     * Emit a chunk every N ms through `onChunk`, for streaming upload.\n     *\n     * Without it the whole recording is buffered in memory until `stop()` — fine for\n     * a voice note, not fine for an hour-long meeting.\n     */\n    timesliceMs?: number;\n    /** Receives each chunk when `timesliceMs` is set. Chunks are **not** independently playable. */\n    onChunk?: (chunk: Blob) => void;\n    /** Recorder-level failure (device unplugged mid-recording, encoder error). */\n    onError?: (error: unknown) => void;\n}\n\n/** Imperative recorder. */\nexport interface AudioRecorderHandle {\n    /** Begin recording. No-op when already recording or paused. */\n    start: () => void;\n    /** Pause. The clock stops; `durationMs` freezes. */\n    pause: () => void;\n    /** Resume after `pause()`. */\n    resume: () => void;\n    /** Stop and resolve with the assembled recording. */\n    stop: () => Promise<AudioRecording>;\n    /** Stop and throw the audio away. */\n    cancel: () => void;\n    status: () => AudioRecorderStatus;\n    /** Recorded length so far, excluding paused time. */\n    durationMs: () => number;\n    /** The negotiated container. */\n    mimeType: string;\n}\n\n/** Whether `MediaRecorder` exists and can produce at least one audio container. */\nexport function isAudioRecordingSupported(): boolean {\n    return typeof MediaRecorder !== \"undefined\" && pickAudioMimeType() !== null;\n}\n\n/**\n * First container in `preferred` the browser can actually produce, or `null`.\n *\n * **There is no MP3 or WAV here, and that is not an omission.** `MediaRecorder`\n * emits Opus (in WebM or Ogg) on Chromium and Firefox and AAC (in MP4) on Safari —\n * no engine implements an MP3 or WAV encoder for it. If a backend needs WAV,\n * {@link blobToWav} converts one client-side with no dependency; if it needs MP3,\n * transcode on the server. Shipping an MP3 encoder would mean a WASM build of the\n * order of 150 KB in every consumer's bundle to serve one format, which is the\n * trade this SDK does not make.\n *\n * @param preferred - Candidates, best first. Defaults to {@link AUDIO_MIME_CANDIDATES}.\n * @returns A supported MIME type, or `null` when none is.\n */\nexport function pickAudioMimeType(\n    preferred: readonly string[] = AUDIO_MIME_CANDIDATES,\n): string | null {\n    return pickRecordingMimeType(preferred);\n}\n\n/**\n * Wrap a `MediaStream` in a recorder with a real state machine and an honest clock.\n *\n * The engine is shared with the video recorder (`createMediaRecorder`), because the\n * subtle parts are not audio-specific: `MediaRecorder` reports **no duration** (WebM\n * it writes carries none in its header, which is why `<audio>` shows `Infinity` for a\n * fresh recording) so the clock is kept by hand and subtracts paused time; and\n * `stop()` returns *before* the last `dataavailable` event, so the blob can only be\n * assembled in `onstop`. What stays here is the part that genuinely differs — the\n * container list.\n *\n * The stream is **not** owned here: `stop()` leaves the microphone open so a retake\n * does not need a second permission round-trip. Release it with the owning\n * `useMicrophone().stop()`.\n *\n * @param stream - A live audio stream, usually from {@link useMicrophone}.\n * @param options - See {@link AudioRecorderOptions}.\n * @returns The imperative recorder.\n * @throws When `MediaRecorder` is unavailable, or an explicit `mimeType` is unsupported.\n */\nexport function createAudioRecorder(\n    stream: MediaStream,\n    options: AudioRecorderOptions = {},\n): AudioRecorderHandle {\n    return createMediaRecorder(stream, {\n        candidates: AUDIO_MIME_CANDIDATES,\n        kind: \"audio\",\n        mimeType: options.mimeType,\n        audioBitsPerSecond: options.audioBitsPerSecond,\n        timesliceMs: options.timesliceMs,\n        onChunk: options.onChunk,\n        onError: options.onError,\n    });\n}\n"],"mappings":";;AAuBA,IAAaA,IAA2C;CACpD;CACA;CACA;CACA;CACA;AACJ;AAwDA,SAAgBC,IAAqC;CACjD,OAAO,OAAO,gBAAkB,OAAeC,EAAkB,MAAM;AAC3E;AAgBA,SAAgBA,EACZ,IAA+BF,GAClB;CACb,OAAOG,EAAsBC,CAAS;AAC1C;AAsBA,SAAgBC,EACZ,GACA,IAAgC,CAAC,GACd;CACnB,OAAOC,EAAoBC,GAAQ;EAC/B,YAAYP;EACZ,MAAM;EACN,UAAUQ,EAAQ;EAClB,oBAAoBA,EAAQ;EAC5B,aAAaA,EAAQ;EACrB,SAASA,EAAQ;EACjB,SAASA,EAAQ;CACrB,CAAC;AACL"}

package/dist/audio/duration.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"duration.cjs","names":[],"sources":["../../src/audio/duration.ts"],"sourcesContent":["/**\n * Format a length in milliseconds as a clock: `0:07`, `2:41`, `1:03:12`.\n *\n * Returns `\"--:--\"` for anything not finite, which is the case that matters: a fresh\n * `MediaRecorder` WebM carries no duration in its header, so `<audio>.duration` reads\n * `Infinity` until the browser has been coaxed into resolving it. Rendering `NaN:aN`\n * there is the bug every home-made audio player ships with.\n *\n * The hour field appears only when there is one, so a voice note reads `0:07` rather\n * than `0:00:07`.\n *\n * @param ms - Length in milliseconds, or a non-finite value for \"unknown\".\n * @returns A clock string.\n */\nexport function formatDuration(ms: number): string {\n    if (!Number.isFinite(ms) || ms < 0) return \"--:--\";\n    const total = Math.floor(ms / 1000);\n    const seconds = total % 60;\n    const minutes = Math.floor(total / 60) % 60;\n    const hours = Math.floor(total / 3600);\n    const pad = (value: number): string => String(value).padStart(2, \"0\");\n    return hours > 0 ? `${hours}:${pad(minutes)}:${pad(seconds)}` : `${minutes}:${pad(seconds)}`;\n}\n"],"mappings":"AAcA,SAAgB,EAAe,EAAoB,CAC/C,GAAI,CAAC,OAAO,SAAS,CAAE,GAAK,EAAK,EAAG,MAAO,QAC3C,IAAM,EAAQ,KAAK,MAAM,EAAK,GAAI,EAC5B,EAAU,EAAQ,GAClB,EAAU,KAAK,MAAM,EAAQ,EAAE,EAAI,GACnC,EAAQ,KAAK,MAAM,EAAQ,IAAI,EAC/B,EAAO,GAA0B,OAAO,CAAK,CAAC,CAAC,SAAS,EAAG,GAAG,EACpE,OAAO,EAAQ,EAAI,GAAG,EAAM,GAAG,EAAI,CAAO,EAAE,GAAG,EAAI,CAAO,IAAM,GAAG,EAAQ,GAAG,EAAI,CAAO,GAC7F"}
+{"version":3,"file":"duration.cjs","names":["formatDuration","ms","total","seconds","minutes","hours","pad","value"],"sources":["../../src/audio/duration.ts"],"sourcesContent":["/**\n * Format a length in milliseconds as a clock: `0:07`, `2:41`, `1:03:12`.\n *\n * Returns `\"--:--\"` for anything not finite, which is the case that matters: a fresh\n * `MediaRecorder` WebM carries no duration in its header, so `<audio>.duration` reads\n * `Infinity` until the browser has been coaxed into resolving it. Rendering `NaN:aN`\n * there is the bug every home-made audio player ships with.\n *\n * The hour field appears only when there is one, so a voice note reads `0:07` rather\n * than `0:00:07`.\n *\n * @param ms - Length in milliseconds, or a non-finite value for \"unknown\".\n * @returns A clock string.\n */\nexport function formatDuration(ms: number): string {\n    if (!Number.isFinite(ms) || ms < 0) return \"--:--\";\n    const total = Math.floor(ms / 1000);\n    const seconds = total % 60;\n    const minutes = Math.floor(total / 60) % 60;\n    const hours = Math.floor(total / 3600);\n    const pad = (value: number): string => String(value).padStart(2, \"0\");\n    return hours > 0 ? `${hours}:${pad(minutes)}:${pad(seconds)}` : `${minutes}:${pad(seconds)}`;\n}\n"],"mappings":"AAcA,SAAgBA,EAAe,EAAoB,CAC/C,GAAI,CAAC,OAAO,SAASC,CAAE,GAAKA,EAAK,EAAG,MAAO,QAC3C,IAAMC,EAAQ,KAAK,MAAMD,EAAK,GAAI,EAC5BE,EAAUD,EAAQ,GAClBE,EAAU,KAAK,MAAMF,EAAQ,EAAE,EAAI,GACnCG,EAAQ,KAAK,MAAMH,EAAQ,IAAI,EAC/BI,EAAO,GAA0B,OAAOC,CAAK,CAAC,CAAC,SAAS,EAAG,GAAG,EACpE,OAAOF,EAAQ,EAAI,GAAGA,EAAM,GAAGC,EAAIF,CAAO,EAAE,GAAGE,EAAIH,CAAO,IAAM,GAAGC,EAAQ,GAAGE,EAAIH,CAAO,GAC7F,C"}

package/dist/audio/duration.js.map

@@ -1 +1 @@
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+{"version":3,"file":"duration.js","names":["formatDuration","ms","total","seconds","minutes","hours","pad","value"],"sources":["../../src/audio/duration.ts"],"sourcesContent":["/**\n * Format a length in milliseconds as a clock: `0:07`, `2:41`, `1:03:12`.\n *\n * Returns `\"--:--\"` for anything not finite, which is the case that matters: a fresh\n * `MediaRecorder` WebM carries no duration in its header, so `<audio>.duration` reads\n * `Infinity` until the browser has been coaxed into resolving it. Rendering `NaN:aN`\n * there is the bug every home-made audio player ships with.\n *\n * The hour field appears only when there is one, so a voice note reads `0:07` rather\n * than `0:00:07`.\n *\n * @param ms - Length in milliseconds, or a non-finite value for \"unknown\".\n * @returns A clock string.\n */\nexport function formatDuration(ms: number): string {\n    if (!Number.isFinite(ms) || ms < 0) return \"--:--\";\n    const total = Math.floor(ms / 1000);\n    const seconds = total % 60;\n    const minutes = Math.floor(total / 60) % 60;\n    const hours = Math.floor(total / 3600);\n    const pad = (value: number): string => String(value).padStart(2, \"0\");\n    return hours > 0 ? `${hours}:${pad(minutes)}:${pad(seconds)}` : `${minutes}:${pad(seconds)}`;\n}\n"],"mappings":";AAcA,SAAgBA,EAAe,GAAoB;CAC/C,IAAI,CAAC,OAAO,SAASC,CAAE,KAAKA,IAAK,GAAG,OAAO;CAC3C,IAAMC,IAAQ,KAAK,MAAMD,IAAK,GAAI,GAC5BE,IAAUD,IAAQ,IAClBE,IAAU,KAAK,MAAMF,IAAQ,EAAE,IAAI,IACnCG,IAAQ,KAAK,MAAMH,IAAQ,IAAI,GAC/BI,KAAO,MAA0B,OAAOC,CAAK,CAAC,CAAC,SAAS,GAAG,GAAG;CACpE,OAAOF,IAAQ,IAAI,GAAGA,EAAM,GAAGC,EAAIF,CAAO,EAAE,GAAGE,EAAIH,CAAO,MAAM,GAAGC,EAAQ,GAAGE,EAAIH,CAAO;AAC7F"}

package/dist/audio/level-meter.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"level-meter.cjs","names":[],"sources":["../../src/audio/level-meter.ts"],"sourcesContent":["/** Options for {@link createLevelMeter}. */\nexport interface LevelMeterOptions {\n    /**\n     * FFT window. 1024 is a good default: ~21 ms at 48 kHz, long enough to be stable\n     * and short enough to react to a syllable.\n     */\n    fftSize?: number;\n    /**\n     * Extra smoothing on the way down, 0–1. Default `0.7`.\n     *\n     * A raw RMS reading is jittery enough to make a meter look broken. Attack is left\n     * instant — a meter that lags the voice going *up* reads as \"not recording\" —\n     * while the decay is eased, which is how every hardware meter behaves.\n     *\n     * `0` turns the smoothing off and reports the window's RMS as measured, which is\n     * what a detector wants: {@link monitorVoiceActivity} applies its own release,\n     * and two of them stacked would hold the indicator open long past the speech.\n     */\n    decay?: number;\n    /**\n     * Reuse an existing `AudioContext` instead of creating one.\n     *\n     * Browsers cap live contexts (Chrome allows around six), and a meter per remote\n     * participant reaches that in a five-person call. A caller measuring several\n     * streams should open one context and hand it to every meter. `stop()` closes\n     * only a context this meter created — an injected one belongs to the caller.\n     */\n    context?: AudioContext;\n}\n\n/** A running level meter. */\nexport interface LevelMeter {\n    /** Latest level, 0–1. Read it on your own frame loop. */\n    level: () => number;\n    /** Stop sampling and release the Web Audio graph. */\n    stop: () => void;\n}\n\n/**\n * Sample the loudness of a live stream, 0–1.\n *\n * This exists because a recorder with no visible level is indistinguishable from a\n * broken one. A muted OS input, a headset whose mic arm is switched off, the wrong\n * device selected — all three produce a perfectly successful recording of silence,\n * and the user only finds out after they finish talking.\n *\n * The value is RMS, not peak: peak reacts to a single sample and flickers, RMS tracks\n * perceived loudness. It is deliberately **not** React state — updating state per\n * frame would re-render the tree 60 times a second. Poll `level()` from a\n * `requestAnimationFrame` loop and write to the DOM, or read it inside an existing\n * animation.\n *\n * The `AudioContext` is created here and closed by `stop()`. Browsers cap the number\n * of live contexts (Chrome allows around six), so a meter left running on unmount\n * eventually breaks every later one on the page.\n *\n * `level()` applies instant attack and eased release: a value above the current one\n * is taken as is, a lower one decays by `decay`. A meter that fell as fast as it rose\n * reads as noise instead of as loudness.\n *\n * @tempest-limits empty-catch — `stop()` disconnects nodes whose context may already\n * be closed (the tab was backgrounded, the stream's track ended, `stop()` raced an\n * unmount), and a disconnect on a dead graph throws. There is nothing to report and\n * nothing to retry: the resource this call would have released is already gone, and\n * the `context.close()` right after it is what actually matters.\n *\n * @param stream - A live audio stream.\n * @param options - See {@link LevelMeterOptions}.\n * @returns A `level()` reader and a `stop()`.\n *\n * @example\n * const meter = createLevelMeter(stream);\n * const tick = () => { bar.style.transform = `scaleX(${meter.level()})`; raf = requestAnimationFrame(tick); };\n */\nexport function createLevelMeter(\n    stream: MediaStream,\n    { fftSize = 1024, decay = 0.7, context: injectedContext }: LevelMeterOptions = {},\n): LevelMeter {\n    const Ctor: typeof AudioContext | undefined =\n        typeof AudioContext !== \"undefined\"\n            ? 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AudioContext\n            : (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n\n    if (!injectedContext && !Ctor) {\n        return { level: () => 0, stop: () => undefined };\n    }\n\n    const context = injectedContext ?? new (Ctor as typeof AudioContext)();\n    const source = context.createMediaStreamSource(stream);\n    const analyser = context.createAnalyser();\n    analyser.fftSize = fftSize;\n    source.connect(analyser);\n\n    const samples = new Float32Array(analyser.fftSize);\n    let smoothed = 0;\n    let closed = false;\n\n    const read = (): number => {\n        if (closed) return 0;\n        analyser.getFloatTimeDomainData(samples);\n        let sum = 0;\n        for (let index = 0; index < samples.length; index += 1) {\n            sum += samples[index] * samples[index];\n        }\n        const rms = Math.sqrt(sum / samples.length);\n        smoothed = rms > smoothed ? rms : smoothed * decay + rms * (1 - decay);\n        return Math.min(1, smoothed);\n    };\n\n    return {\n        level: read,\n        stop: (): void => {\n            if (closed) return;\n            closed = true;\n            try {\n                source.disconnect();\n                analyser.disconnect();\n            } catch {\n                /* empty */\n            }\n            if (!injectedContext) void context.close().catch(() => undefined);\n        },\n    };\n}\n"],"mappings":"AA0EA,SAAgBA,EACZ,EACA,CAAE,UAAU,KAAM,QAAQ,GAAK,QAASC,GAAuC,CAAC,EACtE,CACV,IAAMC,EACF,OAAO,aAAiB,IAClB,aACC,WAA4D,mBAEvE,GAAI,CAACD,GAAmB,CAACC,EACrB,MAAO,CAAE,UAAa,EAAG,SAAY,IAAA,EAAU,EAGnD,IAAMC,EAAUF,GAAmB,IAAKC,EAClCE,EAASD,EAAQ,wBAAwBE,CAAM,EAC/CC,EAAWH,EAAQ,eAAe,EACxC,EAAS,QAAUI,EACnBH,EAAO,QAAQE,CAAQ,EAEvB,IAAME,EAAU,IAAI,aAAaF,EAAS,OAAO,EAC7CG,EAAW,EACXC,EAAS,GAcb,MAAO,CACH,UAbuB,CACvB,GAAIA,EAAQ,MAAO,GACnBJ,EAAS,uBAAuBE,CAAO,EACvC,IAAIG,EAAM,EACV,IAAK,IAAIC,EAAQ,EAAGA,EAAQJ,EAAQ,OAAQ,GAAS,EACjD,GAAOA,EAAQI,GAASJ,EAAQI,GAEpC,IAAMC,EAAM,KAAK,KAAKF,EAAMH,EAAQ,MAAM,EAE1C,MADA,GAAWK,EAAMJ,EAAWI,EAAMJ,EAAWK,EAAQD,GAAO,EAAIC,GACzD,KAAK,IAAI,EAAGL,CAAQ,CAC/B,EAII,SAAkB,CACd,GAAI,GACJ,GAAS,GACT,GAAI,CACAL,EAAO,WAAW,EAClBE,EAAS,WAAW,CACxB,MAAQ,CAER,CACKL,GAAiBE,EAAa,MAAM,CAAC,CAAC,UAAY,IAAA,EAAS,CAPvD,CAQb,CACJ,CACJ,C"}

package/dist/audio/level-meter.js.map

@@ -1 +1 @@
-{"version":3,"file":"level-meter.js","names":[],"sources":["../../src/audio/level-meter.ts"],"sourcesContent":["/** Options for {@link createLevelMeter}. */\nexport interface LevelMeterOptions {\n    /**\n     * FFT window. 1024 is a good default: ~21 ms at 48 kHz, long enough to be stable\n     * and short enough to react to a syllable.\n     */\n    fftSize?: number;\n    /**\n     * Extra smoothing on the way down, 0–1. Default `0.7`.\n     *\n     * A raw RMS reading is jittery enough to make a meter look broken. Attack is left\n     * instant — a meter that lags the voice going *up* reads as \"not recording\" —\n     * while the decay is eased, which is how every hardware meter behaves.\n     *\n     * `0` turns the smoothing off and reports the window's RMS as measured, which is\n     * what a detector wants: {@link monitorVoiceActivity} applies its own release,\n     * and two of them stacked would hold the indicator open long past the speech.\n     */\n    decay?: number;\n    /**\n     * Reuse an existing `AudioContext` instead of creating one.\n     *\n     * Browsers cap live contexts (Chrome allows around six), and a meter per remote\n     * participant reaches that in a five-person call. A caller measuring several\n     * streams should open one context and hand it to every meter. `stop()` closes\n     * only a context this meter created — an injected one belongs to the caller.\n     */\n    context?: AudioContext;\n}\n\n/** A running level meter. */\nexport interface LevelMeter {\n    /** Latest level, 0–1. Read it on your own frame loop. */\n    level: () => number;\n    /** Stop sampling and release the Web Audio graph. */\n    stop: () => void;\n}\n\n/**\n * Sample the loudness of a live stream, 0–1.\n *\n * This exists because a recorder with no visible level is indistinguishable from a\n * broken one. A muted OS input, a headset whose mic arm is switched off, the wrong\n * device selected — all three produce a perfectly successful recording of silence,\n * and the user only finds out after they finish talking.\n *\n * The value is RMS, not peak: peak reacts to a single sample and flickers, RMS tracks\n * perceived loudness. It is deliberately **not** React state — updating state per\n * frame would re-render the tree 60 times a second. Poll `level()` from a\n * `requestAnimationFrame` loop and write to the DOM, or read it inside an existing\n * animation.\n *\n * The `AudioContext` is created here and closed by `stop()`. Browsers cap the number\n * of live contexts (Chrome allows around six), so a meter left running on unmount\n * eventually breaks every later one on the page.\n *\n * `level()` applies instant attack and eased release: a value above the current one\n * is taken as is, a lower one decays by `decay`. A meter that fell as fast as it rose\n * reads as noise instead of as loudness.\n *\n * @tempest-limits empty-catch — `stop()` disconnects nodes whose context may already\n * be closed (the tab was backgrounded, the stream's track ended, `stop()` raced an\n * unmount), and a disconnect on a dead graph throws. There is nothing to report and\n * nothing to retry: the resource this call would have released is already gone, and\n * the `context.close()` right after it is what actually matters.\n *\n * @param stream - A live audio stream.\n * @param options - See {@link LevelMeterOptions}.\n * @returns A `level()` reader and a `stop()`.\n *\n * @example\n * const meter = createLevelMeter(stream);\n * const tick = () => { bar.style.transform = `scaleX(${meter.level()})`; raf = requestAnimationFrame(tick); };\n */\nexport function createLevelMeter(\n    stream: MediaStream,\n    { fftSize = 1024, decay = 0.7, context: injectedContext }: LevelMeterOptions = {},\n): LevelMeter {\n    const Ctor: typeof AudioContext | undefined =\n        typeof AudioContext !== \"undefined\"\n            ? AudioContext\n            : (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n\n    if (!injectedContext && !Ctor) {\n        return { level: () => 0, stop: () => undefined };\n    }\n\n    const context = injectedContext ?? new (Ctor as typeof AudioContext)();\n    const source = context.createMediaStreamSource(stream);\n    const analyser = context.createAnalyser();\n    analyser.fftSize = fftSize;\n    source.connect(analyser);\n\n    const samples = new Float32Array(analyser.fftSize);\n    let smoothed = 0;\n    let closed = false;\n\n    const read = (): number => {\n        if (closed) return 0;\n        analyser.getFloatTimeDomainData(samples);\n        let sum = 0;\n        for (let index = 0; index < samples.length; index += 1) {\n            sum += samples[index] * samples[index];\n        }\n        const rms = Math.sqrt(sum / samples.length);\n        smoothed = rms > smoothed ? rms : smoothed * decay + rms * (1 - decay);\n        return Math.min(1, smoothed);\n    };\n\n    return {\n        level: read,\n        stop: (): void => {\n            if (closed) return;\n            closed = true;\n            try {\n                source.disconnect();\n                analyser.disconnect();\n            } catch {\n                /* empty */\n            }\n            if (!injectedContext) void context.close().catch(() => undefined);\n        },\n    };\n}\n"],"mappings":";AA0EA,SAAgB,EACZ,GACA,EAAE,aAAU,MAAM,WAAQ,IAAK,SAAS,MAAuC,CAAC,GACtE;CACV,IAAM,IACF,OAAO,eAAiB,MAClB,eACC,WAA4D;CAEvE,IAAI,CAAC,KAAmB,CAAC,GACrB,OAAO;EAAE,aAAa;EAAG,YAAY,KAAA;CAAU;CAGnD,IAAM,IAAU,KAAmB,IAAK,EAA6B,GAC/D,IAAS,EAAQ,wBAAwB,CAAM,GAC/C,IAAW,EAAQ,eAAe;CAExC,AADA,EAAS,UAAU,GACnB,EAAO,QAAQ,CAAQ;CAEvB,IAAM,IAAU,IAAI,aAAa,EAAS,OAAO,GAC7C,IAAW,GACX,IAAS;CAcb,OAAO;EACH,aAbuB;GACvB,IAAI,GAAQ,OAAO;GACnB,EAAS,uBAAuB,CAAO;GACvC,IAAI,IAAM;GACV,KAAK,IAAI,IAAQ,GAAG,IAAQ,EAAQ,QAAQ,KAAS,GACjD,KAAO,EAAQ,KAAS,EAAQ;GAEpC,IAAM,IAAM,KAAK,KAAK,IAAM,EAAQ,MAAM;GAE1C,OADA,IAAW,IAAM,IAAW,IAAM,IAAW,IAAQ,KAAO,IAAI,IACzD,KAAK,IAAI,GAAG,CAAQ;EAC/B;EAII,YAAkB;GACV,QACJ;QAAS;IACT,IAAI;KAEA,AADA,EAAO,WAAW,GAClB,EAAS,WAAW;IACxB,QAAQ,CAER;IACA,AAAK,KAAiB,EAAa,MAAM,CAAC,CAAC,YAAY,KAAA,CAAS;GAPvD;EAQb;CACJ;AACJ"}
+{"version":3,"file":"level-meter.js","names":["createLevelMeter","injectedContext","Ctor","context","source","stream","analyser","fftSize","samples","smoothed","closed","sum","index","rms","decay"],"sources":["../../src/audio/level-meter.ts"],"sourcesContent":["/** Options for {@link createLevelMeter}. */\nexport interface LevelMeterOptions {\n    /**\n     * FFT window. 1024 is a good default: ~21 ms at 48 kHz, long enough to be stable\n     * and short enough to react to a syllable.\n     */\n    fftSize?: number;\n    /**\n     * Extra smoothing on the way down, 0–1. Default `0.7`.\n     *\n     * A raw RMS reading is jittery enough to make a meter look broken. Attack is left\n     * instant — a meter that lags the voice going *up* reads as \"not recording\" —\n     * while the decay is eased, which is how every hardware meter behaves.\n     *\n     * `0` turns the smoothing off and reports the window's RMS as measured, which is\n     * what a detector wants: {@link monitorVoiceActivity} applies its own release,\n     * and two of them stacked would hold the indicator open long past the speech.\n     */\n    decay?: number;\n    /**\n     * Reuse an existing `AudioContext` instead of creating one.\n     *\n     * Browsers cap live contexts (Chrome allows around six), and a meter per remote\n     * participant reaches that in a five-person call. A caller measuring several\n     * streams should open one context and hand it to every meter. `stop()` closes\n     * only a context this meter created — an injected one belongs to the caller.\n     */\n    context?: AudioContext;\n}\n\n/** A running level meter. */\nexport interface LevelMeter {\n    /** Latest level, 0–1. Read it on your own frame loop. */\n    level: () => number;\n    /** Stop sampling and release the Web Audio graph. */\n    stop: () => void;\n}\n\n/**\n * Sample the loudness of a live stream, 0–1.\n *\n * This exists because a recorder with no visible level is indistinguishable from a\n * broken one. A muted OS input, a headset whose mic arm is switched off, the wrong\n * device selected — all three produce a perfectly successful recording of silence,\n * and the user only finds out after they finish talking.\n *\n * The value is RMS, not peak: peak reacts to a single sample and flickers, RMS tracks\n * perceived loudness. It is deliberately **not** React state — updating state per\n * frame would re-render the tree 60 times a second. Poll `level()` from a\n * `requestAnimationFrame` loop and write to the DOM, or read it inside an existing\n * animation.\n *\n * The `AudioContext` is created here and closed by `stop()`. Browsers cap the number\n * of live contexts (Chrome allows around six), so a meter left running on unmount\n * eventually breaks every later one on the page.\n *\n * `level()` applies instant attack and eased release: a value above the current one\n * is taken as is, a lower one decays by `decay`. A meter that fell as fast as it rose\n * reads as noise instead of as loudness.\n *\n * @tempest-limits empty-catch — `stop()` disconnects nodes whose context may already\n * be closed (the tab was backgrounded, the stream's track ended, `stop()` raced an\n * unmount), and a disconnect on a dead graph throws. There is nothing to report and\n * nothing to retry: the resource this call would have released is already gone, and\n * the `context.close()` right after it is what actually matters.\n *\n * @param stream - A live audio stream.\n * @param options - See {@link LevelMeterOptions}.\n * @returns A `level()` reader and a `stop()`.\n *\n * @example\n * const meter = createLevelMeter(stream);\n * const tick = () => { bar.style.transform = `scaleX(${meter.level()})`; raf = requestAnimationFrame(tick); };\n */\nexport function createLevelMeter(\n    stream: MediaStream,\n    { fftSize = 1024, decay = 0.7, context: injectedContext }: LevelMeterOptions = {},\n): LevelMeter {\n    const Ctor: typeof AudioContext | undefined =\n        typeof AudioContext !== \"undefined\"\n            ? AudioContext\n            : (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n\n    if (!injectedContext && !Ctor) {\n        return { level: () => 0, stop: () => undefined };\n    }\n\n    const context = injectedContext ?? new (Ctor as typeof AudioContext)();\n    const source = context.createMediaStreamSource(stream);\n    const analyser = context.createAnalyser();\n    analyser.fftSize = fftSize;\n    source.connect(analyser);\n\n    const samples = new Float32Array(analyser.fftSize);\n    let smoothed = 0;\n    let closed = false;\n\n    const read = (): number => {\n        if (closed) return 0;\n        analyser.getFloatTimeDomainData(samples);\n        let sum = 0;\n        for (let index = 0; index < samples.length; index += 1) {\n            sum += samples[index] * samples[index];\n        }\n        const rms = Math.sqrt(sum / samples.length);\n        smoothed = rms > smoothed ? rms : smoothed * decay + rms * (1 - decay);\n        return Math.min(1, smoothed);\n    };\n\n    return {\n        level: read,\n        stop: (): void => {\n            if (closed) return;\n            closed = true;\n            try {\n                source.disconnect();\n                analyser.disconnect();\n            } catch {\n                /* empty */\n            }\n            if (!injectedContext) void context.close().catch(() => undefined);\n        },\n    };\n}\n"],"mappings":";AA0EA,SAAgBA,EACZ,GACA,EAAE,aAAU,MAAM,WAAQ,IAAK,SAASC,MAAuC,CAAC,GACtE;CACV,IAAMC,IACF,OAAO,eAAiB,MAClB,eACC,WAA4D;CAEvE,IAAI,CAACD,KAAmB,CAACC,GACrB,OAAO;EAAE,aAAa;EAAG,YAAY,KAAA;CAAU;CAGnD,IAAMC,IAAUF,KAAmB,IAAKC,EAA6B,GAC/DE,IAASD,EAAQ,wBAAwBE,CAAM,GAC/CC,IAAWH,EAAQ,eAAe;CAExC,AADA,EAAS,UAAUI,GACnBH,EAAO,QAAQE,CAAQ;CAEvB,IAAME,IAAU,IAAI,aAAaF,EAAS,OAAO,GAC7CG,IAAW,GACXC,IAAS;CAcb,OAAO;EACH,aAbuB;GACvB,IAAIA,GAAQ,OAAO;GACnB,EAAS,uBAAuBF,CAAO;GACvC,IAAIG,IAAM;GACV,KAAK,IAAIC,IAAQ,GAAGA,IAAQJ,EAAQ,QAAQ,KAAS,GACjD,KAAOA,EAAQI,KAASJ,EAAQI;GAEpC,IAAMC,IAAM,KAAK,KAAKF,IAAMH,EAAQ,MAAM;GAE1C,OADA,IAAWK,IAAMJ,IAAWI,IAAMJ,IAAWK,IAAQD,KAAO,IAAIC,IACzD,KAAK,IAAI,GAAGL,CAAQ;EAC/B;EAII,YAAkB;GACd,IAAI,IACJ;QAAS;IACT,IAAI;KAEA,AADAL,EAAO,WAAW,GAClBE,EAAS,WAAW;IACxB,QAAQ,CAER;IACA,AAAKL,KAAiBE,EAAa,MAAM,CAAC,CAAC,YAAY,KAAA,CAAS;GAPvD;EAQb;CACJ;AACJ"}

package/dist/audio/media-access.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"media-access.cjs","names":[],"sources":["../../src/audio/media-access.ts"],"sourcesContent":["/**\n * What a media error is about.\n *\n * `\"screen\"` is not a device in the `getUserMedia` sense — it is what\n * `getDisplayMedia` captures — but it fails through the same `DOMException` names, so\n * it is classified by the same function rather than by a second copy of the table.\n */\nexport type MediaDeviceKindLabel = \"microphone\" | \"camera\" | \"screen\";\n\n/** Classified reason a capture device could not be acquired. */\nexport type MediaAccessErrorKind =\n    \"unsupported\" | \"insecure\" | \"permission-denied\" | \"not-found\" | \"in-use\" | \"unknown\";\n\n/** A classified capture error with a human-readable, English message. */\nexport interface MediaAccessError {\n    kind: MediaAccessErrorKind;\n    message: string;\n}\n\n/** Sentence fragment naming the device, so one classifier serves both. */\nconst NOUN: Record<MediaDeviceKindLabel, string> = {\n    microphone: \"Microphone\",\n    camera: \"Camera\",\n    screen: \"Screen\",\n};\n\n/**\n * Map an unknown `getUserMedia` failure into a {@link MediaAccessError}.\n *\n * The order matters. A missing `navigator.mediaDevices` is almost never \"this\n * browser cannot do audio\" — it is a page served over plain HTTP, where the whole\n * API is simply absent. Reporting `unsupported` there sends the developer looking\n * for a polyfill for a problem an `https://` URL fixes, so the secure-context check\n * runs first and wins.\n *\n * The `DOMException.name` values are then collapsed into kinds an app can actually\n * branch on, because the raw names do not group the way a UI needs them:\n * `NotFoundError` and `OverconstrainedError` both mean \"you will not get a device\n * with these constraints\", and `NotReadableError` and `AbortError` both mean \"the\n * hardware is busy\" — most often another tab of the same app.\n *\n * @param err - Whatever `getUserMedia` (or a device query) rejected with.\n * @param device - Which device was being opened, for the message.\n * @returns A stable kind plus a message safe to show a user.\n */\nexport function classifyMediaError(\n    err: unknown,\n    device: MediaDeviceKindLabel = \"microphone\",\n): MediaAccessError {\n    const noun = NOUN[device];\n\n    if (typeof window === \"undefined\") {\n        return { kind: \"unsupported\", message: `${noun} is unavailable in this environment.` };\n    }\n    if (!window.isSecureContext) {\n        return {\n            kind: \"insecure\",\n            message: `${noun} access requires a secure (HTTPS) connection.`,\n        };\n    }\n    if (err instanceof DOMException) {\n        switch (err.name) {\n            case \"NotAllowedError\":\n            case \"SecurityError\":\n                return {\n                    kind: \"permission-denied\",\n                    message: `${noun} permission denied. Enable access in your browser settings.`,\n                };\n            case \"NotFoundError\":\n            case \"OverconstrainedError\":\n                return {\n                    kind: \"not-found\",\n                    message: `No ${device} available on this device.`,\n                };\n            case \"NotReadableError\":\n            case \"AbortError\":\n                return {\n                    kind: \"in-use\",\n                    message: `The ${device} is in use by another app. Close it and try again.`,\n                };\n        }\n    }\n    return {\n        kind: \"unknown\",\n        message:\n            err instanceof Error ? err.message : `Unexpected error while accessing the ${device}.`,\n    };\n}\n\n/**\n * Whether `getUserMedia` is reachable at all.\n *\n * Separate from the classifier because a page often wants to hide a \"record\" button\n * entirely rather than render it and fail on click.\n */\nexport function isMediaCaptureSupported(): boolean {\n    return (\n        typeof navigator !== \"undefined\" &&\n        navigator.mediaDevices !== undefined &&\n        typeof navigator.mediaDevices.getUserMedia === \"function\"\n    );\n}\n\n/**\n * The error for \"the API is not here at all\", which is a different question from\n * \"the API rejected\".\n *\n * `classifyMediaError` takes a thrown value; a missing `navigator.mediaDevices` never\n * threw one, and handing it `null` would come back `unknown` — the least actionable\n * kind there is. The two real causes are an insecure page (the fix is a URL) and an\n * engine that genuinely cannot capture (the fix is another browser), and telling them\n * apart is the whole point.\n *\n * @param device - Which device was being opened, for the message.\n */\nexport function missingCaptureApiError(\n    device: MediaDeviceKindLabel = \"microphone\",\n): MediaAccessError {\n    if (typeof window !== \"undefined\" && !window.isSecureContext) {\n        return classifyMediaError(null, device);\n    }\n    return {\n        kind: \"unsupported\",\n        message: `${NOUN[device]} capture is not supported in this browser.`,\n    };\n}\n"],"mappings":"AAoBA,IAAM,EAA6C,CAC/C,WAAY,aACZ,OAAQ,SACR,OAAQ,QACZ,EAqBA,SAAgB,EACZ,EACA,EAA+B,aACf,CAChB,IAAM,EAAO,EAAK,GAElB,GAAI,OAAO,OAAW,IAClB,MAAO,CAAE,KAAM,cAAe,QAAS,GAAG,EAAK,qCAAsC,EAEzF,GAAI,CAAC,OAAO,gBACR,MAAO,CACH,KAAM,WACN,QAAS,GAAG,EAAK,8CACrB,EAEJ,GAAI,aAAe,aACf,OAAQ,EAAI,KAAZ,CACI,IAAK,kBACL,IAAK,gBACD,MAAO,CACH,KAAM,oBACN,QAAS,GAAG,EAAK,4DACrB,EACJ,IAAK,gBACL,IAAK,uBACD,MAAO,CACH,KAAM,YACN,QAAS,MAAM,EAAO,2BAC1B,EACJ,IAAK,mBACL,IAAK,aACD,MAAO,CACH,KAAM,SACN,QAAS,OAAO,EAAO,mDAC3B,CACR,CAEJ,MAAO,CACH,KAAM,UACN,QACI,aAAe,MAAQ,EAAI,QAAU,wCAAwC,EAAO,EAC5F,CACJ,CAQA,SAAgB,GAAmC,CAC/C,OACI,OAAO,UAAc,KACrB,UAAU,eAAiB,IAAA,IAC3B,OAAO,UAAU,aAAa,cAAiB,UAEvD,CAcA,SAAgB,EACZ,EAA+B,aACf,CAIhB,OAHI,OAAO,OAAW,KAAe,CAAC,OAAO,gBAClC,EAAmB,KAAM,CAAM,EAEnC,CACH,KAAM,cACN,QAAS,GAAG,EAAK,GAAQ,2CAC7B,CACJ"}
+{"version":3,"file":"media-access.cjs","names":["NOUN","classifyMediaError","noun","device","err","isMediaCaptureSupported","missingCaptureApiError"],"sources":["../../src/audio/media-access.ts"],"sourcesContent":["/**\n * What a media error is about.\n *\n * `\"screen\"` is not a device in the `getUserMedia` sense — it is what\n * `getDisplayMedia` captures — but it fails through the same `DOMException` names, so\n * it is classified by the same function rather than by a second copy of the table.\n */\nexport type MediaDeviceKindLabel = \"microphone\" | \"camera\" | \"screen\";\n\n/** Classified reason a capture device could not be acquired. */\nexport type MediaAccessErrorKind =\n    \"unsupported\" | \"insecure\" | \"permission-denied\" | \"not-found\" | \"in-use\" | \"unknown\";\n\n/** A classified capture error with a human-readable, English message. */\nexport interface MediaAccessError {\n    kind: MediaAccessErrorKind;\n    message: string;\n}\n\n/** Sentence fragment naming the device, so one classifier serves both. */\nconst NOUN: Record<MediaDeviceKindLabel, string> = {\n    microphone: \"Microphone\",\n    camera: \"Camera\",\n    screen: \"Screen\",\n};\n\n/**\n * Map an unknown `getUserMedia` failure into a {@link MediaAccessError}.\n *\n * The order matters. A missing `navigator.mediaDevices` is almost never \"this\n * browser cannot do audio\" — it is a page served over plain HTTP, where the whole\n * API is simply absent. 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Enable access in your browser settings.`,\n                };\n            case \"NotFoundError\":\n            case \"OverconstrainedError\":\n                return {\n                    kind: \"not-found\",\n                    message: `No ${device} available on this device.`,\n                };\n            case \"NotReadableError\":\n            case \"AbortError\":\n                return {\n                    kind: \"in-use\",\n                    message: `The ${device} is in use by another app. Close it and try again.`,\n                };\n        }\n    }\n    return {\n        kind: \"unknown\",\n        message:\n            err instanceof Error ? err.message : `Unexpected error while accessing the ${device}.`,\n    };\n}\n\n/**\n * Whether `getUserMedia` is reachable at all.\n *\n * Separate from the classifier because a page often wants to hide a \"record\" button\n * entirely rather than render it and fail on click.\n */\nexport function isMediaCaptureSupported(): boolean {\n    return (\n        typeof navigator !== \"undefined\" &&\n        navigator.mediaDevices !== undefined &&\n        typeof navigator.mediaDevices.getUserMedia === \"function\"\n    );\n}\n\n/**\n * The error for \"the API is not here at all\", which is a different question from\n * \"the API rejected\".\n *\n * `classifyMediaError` takes a thrown value; a missing `navigator.mediaDevices` never\n * threw one, and handing it `null` would come back `unknown` — the least actionable\n * kind there is. The two real causes are an insecure page (the fix is a URL) and an\n * engine that genuinely cannot capture (the fix is another browser), and telling them\n * apart is the whole point.\n *\n * @param device - Which device was being opened, for the message.\n */\nexport function missingCaptureApiError(\n    device: MediaDeviceKindLabel = \"microphone\",\n): MediaAccessError {\n    if (typeof window !== \"undefined\" && !window.isSecureContext) {\n        return classifyMediaError(null, device);\n    }\n    return {\n        kind: \"unsupported\",\n        message: `${NOUN[device]} capture is not supported in this browser.`,\n    };\n}\n"],"mappings":"AAoBA,IAAMA,EAA6C,CAC/C,WAAY,aACZ,OAAQ,SACR,OAAQ,QACZ,EAqBA,SAAgBC,EACZ,EACA,EAA+B,aACf,CAChB,IAAMC,EAAOF,EAAKG,GAElB,GAAI,OAAO,OAAW,IAClB,MAAO,CAAE,KAAM,cAAe,QAAS,GAAGD,EAAK,qCAAsC,EAEzF,GAAI,CAAC,OAAO,gBACR,MAAO,CACH,KAAM,WACN,QAAS,GAAGA,EAAK,8CACrB,EAEJ,GAAIE,aAAe,aACf,OAAQA,EAAI,KAAZ,CACI,IAAK,kBACL,IAAK,gBACD,MAAO,CACH,KAAM,oBACN,QAAS,GAAGF,EAAK,4DACrB,EACJ,IAAK,gBACL,IAAK,uBACD,MAAO,CACH,KAAM,YACN,QAAS,MAAMC,EAAO,2BAC1B,EACJ,IAAK,mBACL,IAAK,aACD,MAAO,CACH,KAAM,SACN,QAAS,OAAOA,EAAO,mDAC3B,CACR,CAEJ,MAAO,CACH,KAAM,UACN,QACIC,aAAe,MAAQA,EAAI,QAAU,wCAAwCD,EAAO,EAC5F,CACJ,CAQA,SAAgBE,GAAmC,CAC/C,OACI,OAAO,UAAc,KACrB,UAAU,eAAiB,IAAA,IAC3B,OAAO,UAAU,aAAa,cAAiB,UAEvD,CAcA,SAAgBC,EACZ,EAA+B,aACf,CAIhB,OAHI,OAAO,OAAW,KAAe,CAAC,OAAO,gBAClCL,EAAmB,KAAME,CAAM,EAEnC,CACH,KAAM,cACN,QAAS,GAAGH,EAAKG,GAAQ,2CAC7B,CACJ,C"}

package/dist/audio/media-access.js.map

@@ -1 +1 @@
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A missing `navigator.mediaDevices` is almost never \"this\n * browser cannot do audio\" — it is a page served over plain HTTP, where the whole\n * API is simply absent. 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package/dist/audio/noise-floor.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"noise-floor.cjs","names":[],"sources":["../../src/audio/noise-floor.ts"],"sourcesContent":["import { windowRms } from \"./voice-chain-stages\";\n\n/** How long the measurement listens by default, in milliseconds. */\nconst DEFAULT_DURATION_MS = 3000;\n\n/** How often it samples the room, in milliseconds. */\nconst SAMPLE_MS = 50;\n\n/**\n * Headroom over the measured floor, so ordinary speech clears the gate.\n *\n * A gate set exactly at the noise level chatters, opening and closing on the\n * room's own variation. Ported from `tempest-mirror-screen`'s `voiceChain.ts`,\n * where it was arrived at by listening.\n */\nconst GATE_MARGIN = 2.4;\n\n/** The narrowest and widest thresholds worth suggesting. */\nconst GATE_MIN = 0.004;\nconst GATE_MAX = 0.12;\n\n/** Options for {@link measureNoiseFloor}. */\nexport interface NoiseFloorOptions {\n    /** Called with `0..1` so a dialog can show the wait. */\n    onProgress?: (fraction: number) => void;\n    /** How long to listen. 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package/dist/audio/noise-floor.js.map

@@ -1 +1 @@
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Ported from `tempest-mirror-screen`'s `voiceChain.ts`,\n * where it was arrived at by listening.\n */\nconst GATE_MARGIN = 2.4;\n\n/** The narrowest and widest thresholds worth suggesting. */\nconst GATE_MIN = 0.004;\nconst GATE_MAX = 0.12;\n\n/** Options for {@link measureNoiseFloor}. */\nexport interface NoiseFloorOptions {\n    /** Called with `0..1` so a dialog can show the wait. */\n    onProgress?: (fraction: number) => void;\n    /** How long to listen. Default `3000`. */\n    durationMs?: number;\n    /** Reuse an existing `AudioContext` rather than opening one. */\n    context?: AudioContext;\n}\n\n/** The `AudioContext` constructor this engine exposes, prefixed or not. */\nfunction audioContextConstructor(): typeof AudioContext | undefined {\n    if (typeof AudioContext !== \"undefined\") return AudioContext;\n    return (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n}\n\n/**\n * Listen to a room at rest and report how loud its silence is.\n *\n * The number a person cannot guess. Setting a gate by ear means switching it on\n * and hearing what disappeared, which only works if you already know what should\n * have disappeared — so the usual outcome is leaving it at a default that is\n * wrong for that room. Measuring replaces the guess.\n *\n * The result is the **peak** RMS window observed, not the average: a threshold\n * under the loudest thing the quiet room did will be opened by that thing.\n *\n * @param stream - A live capture of the room while nobody speaks.\n * @param options - See {@link NoiseFloorOptions}.\n * @returns The peak RMS observed, `0..1`. `0` when this engine has no Web Audio\n *   or the stream carries no audio track — which {@link suggestGateThreshold}\n *   turns into its own floor rather than into a gate that never opens.\n *\n * @example\n * const floor = await measureNoiseFloor(stream, { onProgress: setProgress });\n * setThreshold(suggestGateThreshold(floor));\n */\nexport async function measureNoiseFloor(\n    stream: MediaStream,\n    {\n        onProgress,\n        durationMs = DEFAULT_DURATION_MS,\n        context: injectedContext,\n    }: NoiseFloorOptions = {},\n): Promise<number> {\n    const Ctor = audioContextConstructor();\n    const context = injectedContext ?? (Ctor ? new Ctor() : null);\n    const track = stream.getAudioTracks()[0];\n    if (!context || !track) return 0;\n\n    const source = context.createMediaStreamSource(stream);\n    const analyser = context.createAnalyser();\n    analyser.fftSize = 2048;\n    source.connect(analyser);\n\n    const buffer = new Float32Array(analyser.fftSize);\n    const started = performance.now();\n    let peak = 0;\n\n    await new Promise<void>((resolve) => {\n        const timer = setInterval(() => {\n            analyser.getFloatTimeDomainData(buffer);\n            peak = Math.max(peak, windowRms(buffer));\n\n            const elapsed = performance.now() - started;\n            onProgress?.(Math.min(1, elapsed / durationMs));\n            if (elapsed < durationMs) return;\n            clearInterval(timer);\n            resolve();\n        }, SAMPLE_MS);\n    });\n\n    source.disconnect();\n    analyser.disconnect();\n    return peak;\n}\n\n/**\n * Turn a measured noise floor into a threshold worth using.\n *\n * Sits above the floor by a margin rather than on it, and is clamped at both\n * ends: a room measured as perfectly silent (or not measured at all) would\n * otherwise get a threshold of zero, which is a gate that never closes, and a\n * room measured while somebody talked would get one nothing opens.\n *\n * @param noiseFloor - Peak RMS measured while nobody spoke.\n * @returns A threshold for {@link VoiceChainSettings.gateThreshold}, `0.004..0.12`.\n */\nexport function suggestGateThreshold(noiseFloor: number): number {\n    if (!Number.isFinite(noiseFloor)) return GATE_MIN;\n    return Math.min(GATE_MAX, Math.max(GATE_MIN, noiseFloor * GATE_MARGIN));\n}\n"],"mappings":";;AAGA,IAAM,IAAsB,KAGtB,IAAY,IASZ,IAAc,KAGd,IAAW,MACX,IAAW;AAajB,SAAS,IAA2D;CAEhE,OADI,OAAO,eAAiB,MAAoB,eACxC,WAA4D;AACxE;AAuBA,eAAsB,EAClB,GACA,EACI,eACA,gBAAa,GACb,SAAS,MACU,CAAC,GACT;CACf,IAAM,IAAO,EAAwB,GAC/B,IAAU,MAAoB,IAAO,IAAI,EAAK,IAAI,OAClD,IAAQ,EAAO,eAAe,CAAC,CAAC;CACtC,IAAI,CAAC,KAAW,CAAC,GAAO,OAAO;CAE/B,IAAM,IAAS,EAAQ,wBAAwB,CAAM,GAC/C,IAAW,EAAQ,eAAe;CAExC,AADA,EAAS,UAAU,MACnB,EAAO,QAAQ,CAAQ;CAEvB,IAAM,IAAS,IAAI,aAAa,EAAS,OAAO,GAC1C,IAAU,YAAY,IAAI,GAC5B,IAAO;CAiBX,OAfA,MAAM,IAAI,SAAe,MAAY;EACjC,IAAM,IAAQ,kBAAkB;GAE5B,AADA,EAAS,uBAAuB,CAAM,GACtC,IAAO,KAAK,IAAI,GAAM,EAAU,CAAM,CAAC;GAEvC,IAAM,IAAU,YAAY,IAAI,IAAI;GACpC,IAAa,KAAK,IAAI,GAAG,IAAU,CAAU,CAAC,GAC1C,MAAU,OACd,cAAc,CAAK,GACnB,EAAQ;EACZ,GAAG,CAAS;CAChB,CAAC,GAED,EAAO,WAAW,GAClB,EAAS,WAAW,GACb;AACX;AAaA,SAAgB,EAAqB,GAA4B;CAE7D,OADK,OAAO,SAAS,CAAU,IACxB,KAAK,IAAI,GAAU,KAAK,IAAI,GAAU,IAAa,CAAW,CAAC,IAD7B;AAE7C"}
+{"version":3,"file":"noise-floor.js","names":["windowRms","rms","DEFAULT_DURATION_MS","SAMPLE_MS","GATE_MARGIN","GATE_MIN","GATE_MAX","audioContextConstructor","measureNoiseFloor","injectedContext","Ctor","context","track","stream","source","analyser","buffer","started","peak","timer","elapsed","onProgress","durationMs","resolve","suggestGateThreshold","noiseFloor"],"sources":["../../src/audio/noise-floor.ts"],"sourcesContent":["import { windowRms } from \"./voice-chain-stages\";\n\n/** How long the measurement listens by default, in milliseconds. */\nconst DEFAULT_DURATION_MS = 3000;\n\n/** How often it samples the room, in milliseconds. */\nconst SAMPLE_MS = 50;\n\n/**\n * Headroom over the measured floor, so ordinary speech clears the gate.\n *\n * A gate set exactly at the noise level chatters, opening and closing on the\n * room's own variation. Ported from `tempest-mirror-screen`'s `voiceChain.ts`,\n * where it was arrived at by listening.\n */\nconst GATE_MARGIN = 2.4;\n\n/** The narrowest and widest thresholds worth suggesting. */\nconst GATE_MIN = 0.004;\nconst GATE_MAX = 0.12;\n\n/** Options for {@link measureNoiseFloor}. */\nexport interface NoiseFloorOptions {\n    /** Called with `0..1` so a dialog can show the wait. */\n    onProgress?: (fraction: number) => void;\n    /** How long to listen. Default `3000`. */\n    durationMs?: number;\n    /** Reuse an existing `AudioContext` rather than opening one. */\n    context?: AudioContext;\n}\n\n/** The `AudioContext` constructor this engine exposes, prefixed or not. */\nfunction audioContextConstructor(): typeof AudioContext | undefined {\n    if (typeof AudioContext !== \"undefined\") return AudioContext;\n    return (globalThis as { webkitAudioContext?: typeof AudioContext }).webkitAudioContext;\n}\n\n/**\n * Listen to a room at rest and report how loud its silence is.\n *\n * The number a person cannot guess. Setting a gate by ear means switching it on\n * and hearing what disappeared, which only works if you already know what should\n * have disappeared — so the usual outcome is leaving it at a default that is\n * wrong for that room. Measuring replaces the guess.\n *\n * The result is the **peak** RMS window observed, not the average: a threshold\n * under the loudest thing the quiet room did will be opened by that thing.\n *\n * @param stream - A live capture of the room while nobody speaks.\n * @param options - See {@link NoiseFloorOptions}.\n * @returns The peak RMS observed, `0..1`. `0` when this engine has no Web Audio\n *   or the stream carries no audio track — which {@link suggestGateThreshold}\n *   turns into its own floor rather than into a gate that never opens.\n *\n * @example\n * const floor = await measureNoiseFloor(stream, { onProgress: setProgress });\n * setThreshold(suggestGateThreshold(floor));\n */\nexport async function measureNoiseFloor(\n    stream: MediaStream,\n    {\n        onProgress,\n        durationMs = DEFAULT_DURATION_MS,\n        context: injectedContext,\n    }: NoiseFloorOptions = {},\n): Promise<number> {\n    const Ctor = audioContextConstructor();\n    const context = injectedContext ?? (Ctor ? new Ctor() : null);\n    const track = stream.getAudioTracks()[0];\n    if (!context || !track) return 0;\n\n    const source = context.createMediaStreamSource(stream);\n    const analyser = context.createAnalyser();\n    analyser.fftSize = 2048;\n    source.connect(analyser);\n\n    const buffer = new Float32Array(analyser.fftSize);\n    const started = performance.now();\n    let peak = 0;\n\n    await new Promise<void>((resolve) => {\n        const timer = setInterval(() => {\n            analyser.getFloatTimeDomainData(buffer);\n            peak = Math.max(peak, windowRms(buffer));\n\n            const elapsed = performance.now() - started;\n            onProgress?.(Math.min(1, elapsed / durationMs));\n            if (elapsed < durationMs) return;\n            clearInterval(timer);\n            resolve();\n        }, SAMPLE_MS);\n    });\n\n    source.disconnect();\n    analyser.disconnect();\n    return peak;\n}\n\n/**\n * Turn a measured noise floor into a threshold worth using.\n *\n * Sits above the floor by a margin rather than on it, and is clamped at both\n * ends: a room measured as perfectly silent (or not measured at all) would\n * otherwise get a threshold of zero, which is a gate that never closes, and a\n * room measured while somebody talked would get one nothing opens.\n *\n * @param noiseFloor - Peak RMS measured while nobody spoke.\n * @returns A threshold for {@link VoiceChainSettings.gateThreshold}, `0.004..0.12`.\n */\nexport function suggestGateThreshold(noiseFloor: number): number {\n    if (!Number.isFinite(noiseFloor)) return GATE_MIN;\n    return Math.min(GATE_MAX, Math.max(GATE_MIN, noiseFloor * GATE_MARGIN));\n}\n"],"mappings":";;AAGA,IAAME,IAAsB,KAGtBC,IAAY,IASZC,IAAc,KAGdC,IAAW,MACXC,IAAW;AAajB,SAASC,IAA2D;CAEhE,OADI,OAAO,eAAiB,MAAoB,eACxC,WAA4D;AACxE;AAuBA,eAAsBC,EAClB,GACA,EACI,eACA,gBAAaN,GACb,SAASO,MACU,CAAC,GACT;CACf,IAAMC,IAAOH,EAAwB,GAC/BI,IAAUF,MAAoBC,IAAO,IAAIA,EAAK,IAAI,OAClDE,IAAQC,EAAO,eAAe,CAAC,CAAC;CACtC,IAAI,CAACF,KAAW,CAACC,GAAO,OAAO;CAE/B,IAAME,IAASH,EAAQ,wBAAwBE,CAAM,GAC/CE,IAAWJ,EAAQ,eAAe;CAExC,AADA,EAAS,UAAU,MACnBG,EAAO,QAAQC,CAAQ;CAEvB,IAAMC,IAAS,IAAI,aAAaD,EAAS,OAAO,GAC1CE,IAAU,YAAY,IAAI,GAC5BC,IAAO;CAiBX,OAfA,MAAM,IAAI,SAAe,MAAY;EACjC,IAAMC,IAAQ,kBAAkB;GAE5B,AADAJ,EAAS,uBAAuBC,CAAM,GACtC,IAAO,KAAK,IAAIE,GAAMlB,EAAUgB,CAAM,CAAC;GAEvC,IAAMI,IAAU,YAAY,IAAI,IAAIH;GAEpC,AADAI,IAAa,KAAK,IAAI,GAAGD,IAAUE,CAAU,CAAC,GAC1C,MAAUA,OACd,cAAcH,CAAK,GACnBI,EAAQ;EACZ,GAAGpB,CAAS;CAChB,CAAC,GAEDW,EAAO,WAAW,GAClBC,EAAS,WAAW,GACbG;AACX;AAaA,SAAgBM,EAAqB,GAA4B;CAE7D,OADK,OAAO,SAASC,CAAU,IACxB,KAAK,IAAInB,GAAU,KAAK,IAAID,GAAUoB,IAAarB,CAAW,CAAC,IAD7BC;AAE7C"}

package/dist/audio/sfx-pool.cjs.map

@@ -1 +1 @@
-{"version":3,"file":"sfx-pool.cjs","names":[],"sources":["../../src/audio/sfx-pool.ts"],"sourcesContent":["/** Options for {@link createSfxPool}. */\nexport interface SfxPoolOptions {\n    /**\n     * Master volume, `0`–`1`, multiplied into every per-play volume. Default\n     * `1`. Wire it to the app's sound setting so one value governs the lot.\n     */\n    volume?: number;\n    /**\n     * Prefix applied to sources that are not absolute URLs — typically Vite's\n     * `import.meta.env.BASE_URL`, so a build served from a subpath resolves.\n     * Default `\"\"`.\n     */\n    baseUrl?: string;\n    /**\n     * Elements kept per source. `1` (the default) restarts the clip on every\n     * play, which is what a menu blip wants. Raise it to let a sound overlap\n     * itself — a hit landing while the previous one is still ringing.\n     */\n    voices?: number;\n    /**\n     * Maximum number of distinct sources held. When exceeded, the least\n     * recently played source is released. Default `48`.\n     */\n    maxSources?: number;\n}\n\n/** Per-play overrides. */\nexport interface PlaySfxOptions {\n    /** Volume for this play, `0`–`1`, multiplied by the pool's master. Default `1`. */\n    volume?: number;\n}\n\n/** Imperative handle over a pool of short sound effects. */\nexport interface SfxPool {\n    /** Play a clip, allocating and caching its element on first use. */\n    play: (src: string, options?: PlaySfxOptions) => void;\n    /** Fetch clips ahead of the first play, so it is not silent while the file downloads. */\n    preload: (src: string | string[]) => void;\n    /** Set the master volume, applying it to anything already sounding. */\n    setVolume: (volume: number) => void;\n    /** Stop one source, or every source when called with no argument. */\n    stop: (src?: string) => void;\n    /** Release every element. Call on unmount. */\n    dispose: () => void;\n}\n\n/** Clamp to the range `HTMLMediaElement.volume` accepts, rejecting `NaN`. */\nfunction clampVolume(value: number): number {\n    if (!Number.isFinite(value)) return 1;\n    return Math.min(1, Math.max(0, value));\n}\n\n/**\n * A pool of preallocated `<audio>` elements for short sound effects.\n *\n * `new Audio(src)` on every play allocates an element and re-enters the network\n * stack for a file the browser already has, which is the wrong shape for a\n * sound that fires dozens of times a minute — a UI blip, a hit, a pickup. The\n * pool allocates once per source and replays.\n *\n * This is deliberately not {@link createAudioPlayer}: that handle tracks a\n * single \"current\" clip with loop, sink routing and lifecycle callbacks, which\n * is what background music needs. Effects are the opposite case — many\n * sources, all short, fire-and-forget, and the only thing that matters is that\n * firing one is cheap.\n *\n * A blocked `play()` is swallowed. Browsers reject playback until the user has\n * interacted with the page, and a sound effect is by definition not worth\n * interrupting anything over; call {@link SfxPool.preload} after the first\n * interaction if you want the pool warm.\n *\n * @param options - Master volume, base URL, voices per source and pool size.\n * @returns The pool handle.\n *\n * @example\n * const sfx = createSfxPool({ volume: 0.6, baseUrl: import.meta.env.BASE_URL });\n * sfx.preload([\"sfx/select.mp3\", \"sfx/back.mp3\"]);\n *\n * <button onClick={() => sfx.play(\"sfx/select.mp3\")}>Confirmar</button>\n */\n/** One source's voices plus the round-robin cursor over them. */\ninterface SourceEntry {\n    /** The audio elements sharing this source. */\n    elements: HTMLAudioElement[];\n    /** Index of the element the next play claims. */\n    next: number;\n}\n\n/**\n * Resolve a source against the pool's base URL.\n *\n * An absolute URL, a protocol-relative one and a `data:` URI are already\n * addresses — prefixing them would produce a path that resolves to nothing.\n *\n * @param baseUrl - Pool base, possibly empty.\n * @param src - Source as the caller wrote it.\n * @returns The URL to load.\n */\nfunction resolveSource(baseUrl: string, src: string): string {\n    if (!baseUrl || /^(https?:)?\\/\\//.test(src) || src.startsWith(\"data:\")) return src;\n    return `${baseUrl.replace(/\\/$/, \"\")}/${src.replace(/^\\//, \"\")}`;\n}\n\n/**\n * Stop every voice of an entry and let the browser drop its buffer.\n *\n * Removing `src` before `load()` is what frees the decoded audio: `load()` on an\n * element that still has a source re-fetches it instead.\n *\n * @param entry - The entry being evicted.\n */\nfunction releaseEntry(entry: { elements: HTMLAudioElement[] }): void {\n    for (const element of entry.elements) {\n        element.pause();\n        element.removeAttribute(\"src\");\n        element.load();\n    }\n}\n\n/**\n * Drop the least recently played source once the pool is over its cap.\n *\n * The map is insertion-ordered by last play, so the first key is the coldest.\n * The source that was just admitted is never the one evicted, which would\n * otherwise happen with a cap of one.\n *\n * @param sources - The live pool, mutated in place.\n * @param keepUrl - The source that must survive this eviction.\n * @param maxSources - How many sources the pool may hold.\n */\nfunction evictLeastRecent(\n    sources: Map<string, SourceEntry>,\n    keepUrl: string,\n    maxSources: number,\n): void {\n    if (sources.size <= maxSources) return;\n    const oldest = sources.keys().next();\n    if (oldest.done || oldest.value === keepUrl) return;\n    const evicted = sources.get(oldest.value);\n    if (evicted) releaseEntry(evicted);\n    sources.delete(oldest.value);\n}\n\nexport function createSfxPool(options: SfxPoolOptions = {}): SfxPool {\n    const { baseUrl = \"\", voices = 1, maxSources = 48 } = options;\n\n    let master = clampVolume(options.volume ?? 1);\n    const voiceCount = Math.max(1, Math.floor(voices));\n\n    /**\n     * Insertion-ordered by last play, so evicting the first key drops the\n     * least recently used source.\n     */\n    const sources = new Map<string, SourceEntry>();\n\n    /**\n     * Per-play gain of whatever each element last played at.\n     *\n     * `setVolume` has to rescale live playback by the same factor it was\n     * started with; assigning the master directly would yank a clip that\n     * started at half volume up to full.\n     */\n    const gains = new WeakMap<HTMLAudioElement, number>();\n\n    /**\n     * The pool entry for a source, creating and admitting it when it is new.\n     *\n     * Re-inserts an existing entry so the map order stays \"least recently used\n     * first\", which is what makes evicting the first key correct.\n     *\n     * `created` is reported because assigning `src` on an element whose `preload`\n     * is `\"auto\"` already starts the fetch: calling `load()` on an entry that was\n     * merely looked up would abort that fetch and discard whatever is already\n     * buffered, per the media element load algorithm.\n     *\n     * @param src - Raw source, resolved against `baseUrl`.\n     * @returns The entry and whether this call created it, or `null` where\n     *   `Audio` does not exist.\n     */\n    function acquire(src: string): { entry: SourceEntry; created: boolean } | null {\n        if (typeof Audio === \"undefined\") return null;\n\n        const url = resolveSource(baseUrl, src);\n        const existing = sources.get(url);\n        if (existing) {\n            sources.delete(url);\n            sources.set(url, existing);\n            return { entry: existing, created: false };\n        }\n\n        const entry: SourceEntry = {\n            elements: Array.from({ length: voiceCount }, () => {\n                const element = new Audio(url);\n                element.preload = \"auto\";\n                return element;\n            }),\n            next: 0,\n        };\n        sources.set(url, entry);\n\n        evictLeastRecent(sources, url, maxSources);\n\n        return { entry, created: true };\n    }\n\n    return {\n        play(src, playOptions = {}) {\n            const acquired = acquire(src);\n            if (!acquired) return;\n\n            const entry = acquired.entry;\n            const element = entry.elements[entry.next];\n            entry.next = (entry.next + 1) % entry.elements.length;\n\n            const gain = clampVolume(playOptions.volume ?? 1);\n            gains.set(element, gain);\n\n            element.pause();\n            element.currentTime = 0;\n            element.volume = clampVolume(gain * master);\n            void element.play().catch(() => {});\n        },\n\n        preload(src) {\n            for (const one of Array.isArray(src) ? src : [src]) {\n                const acquired = acquire(one);\n                if (!acquired?.created) continue;\n                for (const element of acquired.entry.elements) element.load();\n            }\n        },\n\n        setVolume(volume) {\n            master = clampVolume(volume);\n            for (const entry of sources.values()) {\n                for (const element of entry.elements) {\n                    if (element.paused) continue;\n                    element.volume = clampVolume((gains.get(element) ?? 1) * master);\n                }\n            }\n        },\n\n        stop(src) {\n            const targets = src\n                ? [sources.get(resolveSource(baseUrl, src))]\n                : [...sources.values()];\n            for (const entry of targets) {\n                if (!entry) continue;\n                for (const element of entry.elements) {\n                    element.pause();\n                    element.currentTime = 0;\n                }\n            }\n        },\n\n        dispose() {\n            for (const entry of sources.values()) releaseEntry(entry);\n            sources.clear();\n        },\n    };\n}\n"],"mappings":"AA+CA,SAAS,EAAY,EAAuB,CAExC,OADK,OAAO,SAAS,CAAK,EACnB,KAAK,IAAI,EAAG,KAAK,IAAI,EAAG,CAAK,CAAC,EADD,CAExC,CAgDA,SAAS,EAAc,EAAiB,EAAqB,CAEzD,MADI,CAAC,GAAW,kBAAkB,KAAK,CAAG,GAAK,EAAI,WAAW,OAAO,EAAU,EACxE,GAAG,EAAQ,QAAQ,MAAO,EAAE,EAAE,GAAG,EAAI,QAAQ,MAAO,EAAE,GACjE,CAUA,SAAS,EAAa,EAA+C,CACjE,IAAK,IAAM,KAAW,EAAM,SACxB,EAAQ,MAAM,EACd,EAAQ,gBAAgB,KAAK,EAC7B,EAAQ,KAAK,CAErB,CAaA,SAAS,EACL,EACA,EACA,EACI,CACJ,GAAI,EAAQ,MAAQ,EAAY,OAChC,IAAM,EAAS,EAAQ,KAAK,CAAC,CAAC,KAAK,EACnC,GAAI,EAAO,MAAQ,EAAO,QAAU,EAAS,OAC7C,IAAM,EAAU,EAAQ,IAAI,EAAO,KAAK,EACpC,GAAS,EAAa,CAAO,EACjC,EAAQ,OAAO,EAAO,KAAK,CAC/B,CAEA,SAAgB,EAAc,EAA0B,CAAC,EAAY,CACjE,GAAM,CAAE,UAAU,GAAI,SAAS,EAAG,aAAa,IAAO,EAElD,EAAS,EAAY,EAAQ,QAAU,CAAC,EACtC,EAAa,KAAK,IAAI,EAAG,KAAK,MAAM,CAAM,CAAC,EAM3C,EAAU,IAAI,IASd,EAAQ,IAAI,QAiBlB,SAAS,EAAQ,EAA8D,CAC3E,GAAI,OAAO,MAAU,IAAa,OAAO,KAEzC,IAAM,EAAM,EAAc,EAAS,CAAG,EAChC,EAAW,EAAQ,IAAI,CAAG,EAChC,GAAI,EAGA,OAFA,EAAQ,OAAO,CAAG,EAClB,EAAQ,IAAI,EAAK,CAAQ,EAClB,CAAE,MAAO,EAAU,QAAS,EAAM,EAG7C,IAAM,EAAqB,CACvB,SAAU,MAAM,KAAK,CAAE,OAAQ,CAAW,MAAS,CAC/C,IAAM,EAAU,IAAI,MAAM,CAAG,EAE7B,MADA,GAAQ,QAAU,OACX,CACX,CAAC,EACD,KAAM,CACV,EAKA,OAJA,EAAQ,IAAI,EAAK,CAAK,EAEtB,EAAiB,EAAS,EAAK,CAAU,EAElC,CAAE,QAAO,QAAS,EAAK,CAClC,CAEA,MAAO,CACH,KAAK,EAAK,EAAc,CAAC,EAAG,CACxB,IAAM,EAAW,EAAQ,CAAG,EAC5B,GAAI,CAAC,EAAU,OAEf,IAAM,EAAQ,EAAS,MACjB,EAAU,EAAM,SAAS,EAAM,MACrC,EAAM,MAAQ,EAAM,KAAO,GAAK,EAAM,SAAS,OAE/C,IAAM,EAAO,EAAY,EAAY,QAAU,CAAC,EAChD,EAAM,IAAI,EAAS,CAAI,EAEvB,EAAQ,MAAM,EACd,EAAQ,YAAc,EACtB,EAAQ,OAAS,EAAY,EAAO,CAAM,EAC1C,EAAa,KAAK,CAAC,CAAC,UAAY,CAAC,CAAC,CACtC,EAEA,QAAQ,EAAK,CACT,IAAK,IAAM,KAAO,MAAM,QAAQ,CAAG,EAAI,EAAM,CAAC,CAAG,EAAG,CAChD,IAAM,EAAW,EAAQ,CAAG,EACvB,MAAU,QACf,IAAK,IAAM,KAAW,EAAS,MAAM,SAAU,EAAQ,KAAK,CAChE,CACJ,EAEA,UAAU,EAAQ,CACd,EAAS,EAAY,CAAM,EAC3B,IAAK,IAAM,KAAS,EAAQ,OAAO,EAC/B,IAAK,IAAM,KAAW,EAAM,SACpB,EAAQ,SACZ,EAAQ,OAAS,GAAa,EAAM,IAAI,CAAO,GAAK,GAAK,CAAM,EAG3E,EAEA,KAAK,EAAK,CACN,IAAM,EAAU,EACV,CAAC,EAAQ,IAAI,EAAc,EAAS,CAAG,CAAC,CAAC,EACzC,CAAC,GAAG,EAAQ,OAAO,CAAC,EAC1B,IAAK,IAAM,KAAS,EACX,KACL,IAAK,IAAM,KAAW,EAAM,SACxB,EAAQ,MAAM,EACd,EAAQ,YAAc,CAGlC,EAEA,SAAU,CACN,IAAK,IAAM,KAAS,EAAQ,OAAO,EAAG,EAAa,CAAK,EACxD,EAAQ,MAAM,CAClB,CACJ,CACJ"}
+{"version":3,"file":"sfx-pool.cjs","names":["clampVolume","value","resolveSource","baseUrl","src","releaseEntry","element","entry","evictLeastRecent","sources","maxSources","oldest","keepUrl","evicted","createSfxPool","options","master","voiceCount","voices","gains","acquire","url","existing","acquired","gain","playOptions","one","volume","targets"],"sources":["../../src/audio/sfx-pool.ts"],"sourcesContent":["/** Options for {@link createSfxPool}. */\nexport interface SfxPoolOptions {\n    /**\n     * Master volume, `0`–`1`, multiplied into every per-play volume. Default\n     * `1`. Wire it to the app's sound setting so one value governs the lot.\n     */\n    volume?: number;\n    /**\n     * Prefix applied to sources that are not absolute URLs — typically Vite's\n     * `import.meta.env.BASE_URL`, so a build served from a subpath resolves.\n     * Default `\"\"`.\n     */\n    baseUrl?: string;\n    /**\n     * Elements kept per source. `1` (the default) restarts the clip on every\n     * play, which is what a menu blip wants. Raise it to let a sound overlap\n     * itself — a hit landing while the previous one is still ringing.\n     */\n    voices?: number;\n    /**\n     * Maximum number of distinct sources held. When exceeded, the least\n     * recently played source is released. Default `48`.\n     */\n    maxSources?: number;\n}\n\n/** Per-play overrides. */\nexport interface PlaySfxOptions {\n    /** Volume for this play, `0`–`1`, multiplied by the pool's master. Default `1`. */\n    volume?: number;\n}\n\n/** Imperative handle over a pool of short sound effects. */\nexport interface SfxPool {\n    /** Play a clip, allocating and caching its element on first use. */\n    play: (src: string, options?: PlaySfxOptions) => void;\n    /** Fetch clips ahead of the first play, so it is not silent while the file downloads. */\n    preload: (src: string | string[]) => void;\n    /** Set the master volume, applying it to anything already sounding. */\n    setVolume: (volume: number) => void;\n    /** Stop one source, or every source when called with no argument. */\n    stop: (src?: string) => void;\n    /** Release every element. Call on unmount. */\n    dispose: () => void;\n}\n\n/** Clamp to the range `HTMLMediaElement.volume` accepts, rejecting `NaN`. */\nfunction clampVolume(value: number): number {\n    if (!Number.isFinite(value)) return 1;\n    return Math.min(1, Math.max(0, value));\n}\n\n/**\n * A pool of preallocated `<audio>` elements for short sound effects.\n *\n * `new Audio(src)` on every play allocates an element and re-enters the network\n * stack for a file the browser already has, which is the wrong shape for a\n * sound that fires dozens of times a minute — a UI blip, a hit, a pickup. The\n * pool allocates once per source and replays.\n *\n * This is deliberately not {@link createAudioPlayer}: that handle tracks a\n * single \"current\" clip with loop, sink routing and lifecycle callbacks, which\n * is what background music needs. Effects are the opposite case — many\n * sources, all short, fire-and-forget, and the only thing that matters is that\n * firing one is cheap.\n *\n * A blocked `play()` is swallowed. Browsers reject playback until the user has\n * interacted with the page, and a sound effect is by definition not worth\n * interrupting anything over; call {@link SfxPool.preload} after the first\n * interaction if you want the pool warm.\n *\n * @param options - Master volume, base URL, voices per source and pool size.\n * @returns The pool handle.\n *\n * @example\n * const sfx = createSfxPool({ volume: 0.6, baseUrl: import.meta.env.BASE_URL });\n * sfx.preload([\"sfx/select.mp3\", \"sfx/back.mp3\"]);\n *\n * <button onClick={() => sfx.play(\"sfx/select.mp3\")}>Confirmar</button>\n */\n/** One source's voices plus the round-robin cursor over them. */\ninterface SourceEntry {\n    /** The audio elements sharing this source. */\n    elements: HTMLAudioElement[];\n    /** Index of the element the next play claims. */\n    next: number;\n}\n\n/**\n * Resolve a source against the pool's base URL.\n *\n * An absolute URL, a protocol-relative one and a `data:` URI are already\n * addresses — prefixing them would produce a path that resolves to nothing.\n *\n * @param baseUrl - Pool base, possibly empty.\n * @param src - Source as the caller wrote it.\n * @returns The URL to load.\n */\nfunction resolveSource(baseUrl: string, src: string): string {\n    if (!baseUrl || /^(https?:)?\\/\\//.test(src) || src.startsWith(\"data:\")) return src;\n    return `${baseUrl.replace(/\\/$/, \"\")}/${src.replace(/^\\//, \"\")}`;\n}\n\n/**\n * Stop every voice of an entry and let the browser drop its buffer.\n *\n * Removing `src` before `load()` is what frees the decoded audio: `load()` on an\n * element that still has a source re-fetches it instead.\n *\n * @param entry - The entry being evicted.\n */\nfunction releaseEntry(entry: { elements: HTMLAudioElement[] }): void {\n    for (const element of entry.elements) {\n        element.pause();\n        element.removeAttribute(\"src\");\n        element.load();\n    }\n}\n\n/**\n * Drop the least recently played source once the pool is over its cap.\n *\n * The map is insertion-ordered by last play, so the first key is the coldest.\n * The source that was just admitted is never the one evicted, which would\n * otherwise happen with a cap of one.\n *\n * @param sources - The live pool, mutated in place.\n * @param keepUrl - The source that must survive this eviction.\n * @param maxSources - How many sources the pool may hold.\n */\nfunction evictLeastRecent(\n    sources: Map<string, SourceEntry>,\n    keepUrl: string,\n    maxSources: number,\n): void {\n    if (sources.size <= maxSources) return;\n    const oldest = sources.keys().next();\n    if (oldest.done || oldest.value === keepUrl) return;\n    const evicted = sources.get(oldest.value);\n    if (evicted) releaseEntry(evicted);\n    sources.delete(oldest.value);\n}\n\nexport function createSfxPool(options: SfxPoolOptions = {}): SfxPool {\n    const { baseUrl = \"\", voices = 1, maxSources = 48 } = options;\n\n    let master = clampVolume(options.volume ?? 1);\n    const voiceCount = Math.max(1, Math.floor(voices));\n\n    /**\n     * Insertion-ordered by last play, so evicting the first key drops the\n     * least recently used source.\n     */\n    const sources = new Map<string, SourceEntry>();\n\n    /**\n     * Per-play gain of whatever each element last played at.\n     *\n     * `setVolume` has to rescale live playback by the same factor it was\n     * started with; assigning the master directly would yank a clip that\n     * started at half volume up to full.\n     */\n    const gains = new WeakMap<HTMLAudioElement, number>();\n\n    /**\n     * The pool entry for a source, creating and admitting it when it is new.\n     *\n     * Re-inserts an existing entry so the map order stays \"least recently used\n     * first\", which is what makes evicting the first key correct.\n     *\n     * `created` is reported because assigning `src` on an element whose `preload`\n     * is `\"auto\"` already starts the fetch: calling `load()` on an entry that was\n     * merely looked up would abort that fetch and discard whatever is already\n     * buffered, per the media element load algorithm.\n     *\n     * @param src - Raw source, resolved against `baseUrl`.\n     * @returns The entry and whether this call created it, or `null` where\n     *   `Audio` does not exist.\n     */\n    function acquire(src: string): { entry: SourceEntry; created: boolean } | null {\n        if (typeof Audio === \"undefined\") return null;\n\n        const url = resolveSource(baseUrl, src);\n        const existing = sources.get(url);\n        if (existing) {\n            sources.delete(url);\n            sources.set(url, existing);\n            return { entry: existing, created: false };\n        }\n\n        const entry: SourceEntry = {\n            elements: Array.from({ length: voiceCount }, () => {\n                const element = new Audio(url);\n                element.preload = \"auto\";\n                return element;\n            }),\n            next: 0,\n        };\n        sources.set(url, entry);\n\n        evictLeastRecent(sources, url, maxSources);\n\n        return { entry, created: true };\n    }\n\n    return {\n        play(src, playOptions = {}) {\n            const acquired = acquire(src);\n            if (!acquired) return;\n\n            const entry = acquired.entry;\n            const element = entry.elements[entry.next];\n            entry.next = (entry.next + 1) % entry.elements.length;\n\n            const gain = clampVolume(playOptions.volume ?? 1);\n            gains.set(element, gain);\n\n            element.pause();\n            element.currentTime = 0;\n            element.volume = clampVolume(gain * master);\n            void element.play().catch(() => {});\n        },\n\n        preload(src) {\n            for (const one of Array.isArray(src) ? src : [src]) {\n                const acquired = acquire(one);\n                if (!acquired?.created) continue;\n                for (const element of acquired.entry.elements) element.load();\n            }\n        },\n\n        setVolume(volume) {\n            master = clampVolume(volume);\n            for (const entry of sources.values()) {\n                for (const element of entry.elements) {\n                    if (element.paused) continue;\n                    element.volume = clampVolume((gains.get(element) ?? 1) * master);\n                }\n            }\n        },\n\n        stop(src) {\n            const targets = src\n                ? [sources.get(resolveSource(baseUrl, src))]\n                : [...sources.values()];\n            for (const entry of targets) {\n                if (!entry) continue;\n                for (const element of entry.elements) {\n                    element.pause();\n                    element.currentTime = 0;\n                }\n            }\n        },\n\n        dispose() {\n            for (const entry of sources.values()) releaseEntry(entry);\n            sources.clear();\n        },\n    };\n}\n"],"mappings":"AA+CA,SAASA,EAAY,EAAuB,CAExC,OADK,OAAO,SAASC,CAAK,EACnB,KAAK,IAAI,EAAG,KAAK,IAAI,EAAGA,CAAK,CAAC,EADD,CAExC,CAgDA,SAASC,EAAc,EAAiB,EAAqB,CAEzD,MADI,CAACC,GAAW,kBAAkB,KAAKC,CAAG,GAAKA,EAAI,WAAW,OAAO,EAAUA,EACxE,GAAGD,EAAQ,QAAQ,MAAO,EAAE,EAAE,GAAGC,EAAI,QAAQ,MAAO,EAAE,GACjE,CAUA,SAASC,EAAa,EAA+C,CACjE,IAAK,IAAMC,KAAWC,EAAM,SACxBD,EAAQ,MAAM,EACdA,EAAQ,gBAAgB,KAAK,EAC7BA,EAAQ,KAAK,CAErB,CAaA,SAASE,EACL,EACA,EACA,EACI,CACJ,GAAIC,EAAQ,MAAQC,EAAY,OAChC,IAAMC,EAASF,EAAQ,KAAK,CAAC,CAAC,KAAK,EACnC,GAAIE,EAAO,MAAQA,EAAO,QAAUC,EAAS,OAC7C,IAAMC,EAAUJ,EAAQ,IAAIE,EAAO,KAAK,EACpCE,GAASR,EAAaQ,CAAO,EACjCJ,EAAQ,OAAOE,EAAO,KAAK,CAC/B,CAEA,SAAgBG,EAAc,EAA0B,CAAC,EAAY,CACjE,GAAM,CAAE,UAAU,GAAI,SAAS,EAAG,aAAa,IAAOC,EAElDC,EAAShB,EAAYe,EAAQ,QAAU,CAAC,EACtCE,EAAa,KAAK,IAAI,EAAG,KAAK,MAAMC,CAAM,CAAC,EAM3CT,EAAU,IAAI,IASdU,EAAQ,IAAI,QAiBlB,SAASC,EAAQ,EAA8D,CAC3E,GAAI,OAAO,MAAU,IAAa,OAAO,KAEzC,IAAMC,EAAMnB,EAAcC,EAASC,CAAG,EAChCkB,EAAWb,EAAQ,IAAIY,CAAG,EAChC,GAAIC,EAGA,OAFAb,EAAQ,OAAOY,CAAG,EAClBZ,EAAQ,IAAIY,EAAKC,CAAQ,EAClB,CAAE,MAAOA,EAAU,QAAS,EAAM,EAG7C,IAAMf,EAAqB,CACvB,SAAU,MAAM,KAAK,CAAE,OAAQU,CAAW,MAAS,CAC/C,IAAMX,EAAU,IAAI,MAAMe,CAAG,EAE7B,MADA,GAAQ,QAAU,OACXf,CACX,CAAC,EACD,KAAM,CACV,EAKA,OAJAG,EAAQ,IAAIY,EAAKd,CAAK,EAEtBC,EAAiBC,EAASY,EAAKX,CAAU,EAElC,CAAE,QAAO,QAAS,EAAK,CAClC,CAEA,MAAO,CACH,KAAK,EAAK,EAAc,CAAC,EAAG,CACxB,IAAMa,EAAWH,EAAQhB,CAAG,EAC5B,GAAI,CAACmB,EAAU,OAEf,IAAMhB,EAAQgB,EAAS,MACjBjB,EAAUC,EAAM,SAASA,EAAM,MACrC,EAAM,MAAQA,EAAM,KAAO,GAAKA,EAAM,SAAS,OAE/C,IAAMiB,EAAOxB,EAAYyB,EAAY,QAAU,CAAC,EAChDN,EAAM,IAAIb,EAASkB,CAAI,EAEvBlB,EAAQ,MAAM,EACd,EAAQ,YAAc,EACtB,EAAQ,OAASN,EAAYwB,EAAOR,CAAM,EAC1CV,EAAa,KAAK,CAAC,CAAC,UAAY,CAAC,CAAC,CACtC,EAEA,QAAQ,EAAK,CACT,IAAK,IAAMoB,KAAO,MAAM,QAAQtB,CAAG,EAAIA,EAAM,CAACA,CAAG,EAAG,CAChD,IAAMmB,EAAWH,EAAQM,CAAG,EAC5B,GAAKH,GAAU,QACf,IAAK,IAAMjB,KAAWiB,EAAS,MAAM,SAAUjB,EAAQ,KAAK,CAChE,CACJ,EAEA,UAAU,EAAQ,CACd,EAASN,EAAY2B,CAAM,EAC3B,IAAK,IAAMpB,KAASE,EAAQ,OAAO,EAC/B,IAAK,IAAMH,KAAWC,EAAM,SACpBD,EAAQ,SACZ,EAAQ,OAASN,GAAamB,EAAM,IAAIb,CAAO,GAAK,GAAKU,CAAM,EAG3E,EAEA,KAAK,EAAK,CACN,IAAMY,EAAUxB,EACV,CAACK,EAAQ,IAAIP,EAAcC,EAASC,CAAG,CAAC,CAAC,EACzC,CAAC,GAAGK,EAAQ,OAAO,CAAC,EAC1B,IAAK,IAAMF,KAASqB,EAChB,GAAKrB,EACL,IAAK,IAAMD,KAAWC,EAAM,SACxBD,EAAQ,MAAM,EACd,EAAQ,YAAc,CAGlC,EAEA,SAAU,CACN,IAAK,IAAMC,KAASE,EAAQ,OAAO,EAAGJ,EAAaE,CAAK,EACxDE,EAAQ,MAAM,CAClB,CACJ,CACJ,C"}