@datamoc/mw_games 0.44.2 → 0.45.0

package/dist/audio/Positional.js

@@ -1,138 +0,0 @@
-/**
- * The gain a listener at `distance` hears a source at, clamped to [0, 1]: 1 at or inside
- * `refDistance`, 0 at or beyond `maxDistance`, an inverse-distance curve between them.
- *
- * @example
- * ```ts
- * import { audioGain } from '@datamoc/mw_games/audio';
- *
- * audioGain(0, { refDistance: 2 }); // 1 - inside the full-volume radius
- * audioGain(2, { refDistance: 2 }); // 1 - the reference distance itself
- * audioGain(4, { refDistance: 2 }); // 0.5 - one doubling away
- * audioGain(4, { refDistance: 2, maxDistance: 4 }); // 0 - at the edge of hearing
- * ```
- */
-export function audioGain(distance, falloff = {}) {
-    const refDistance = falloff.refDistance ?? 1;
-    const maxDistance = falloff.maxDistance ?? Infinity;
-    const rolloff = falloff.rolloff ?? 1;
-    if (!(distance > refDistance))
-        return 1;
-    if (!(distance < maxDistance))
-        return 0;
-    const gain = Math.pow(refDistance / distance, rolloff);
-    return Math.max(0, Math.min(1, gain));
-}
-/**
- * Stereo pan in [-1, 1] for a source: -1 is hard left of the listener, 1 hard right, 0
- * directly ahead or behind. The listener's `facing` is its heading in radians (0 faces +x,
- * increasing clockwise on a screen where y grows downward), so turning the listener turns
- * what counts as "right" with it - a sound to the world's east is centred when the listener
- * faces east and hard right when the listener faces north.
- *
- * A game with a stereo backend applies this itself; the framework has no panner node of its
- * own, so it reports the number rather than pretending to pan.
- *
- * @example
- * ```ts
- * import { audioPan } from '@datamoc/mw_games/audio';
- *
- * const listener = { x: 0, y: 0, facing: 0 }; // facing east
- * audioPan(listener, { x: 10, y: 0 }); // 0 - straight ahead
- * audioPan(listener, { x: 0, y: 10 }); // 1 - to the listener's right
- * audioPan(listener, { x: 0, y: -10 }); // -1 - to its left
- * ```
- */
-export function audioPan(listener, source) {
-    const dx = source.x - listener.x;
-    const dy = source.y - listener.y;
-    const distance = Math.hypot(dx, dy);
-    if (distance === 0)
-        return 0;
-    //the listener's right-hand direction at heading `facing`, in screen coordinates
-    const rightX = Math.sin(listener.facing);
-    const rightY = Math.cos(listener.facing);
-    const pan = (dx * rightX + dy * rightY) / distance;
-    return Math.max(-1, Math.min(1, pan));
-}
-/**
- * Where sounds are heard from: a game moves one listener to its camera or its party leader
- * each frame, and every `SoundSource` is mixed against it.
- *
- * @example
- * ```ts
- * import { AudioListener } from '@datamoc/mw_games/audio';
- *
- * const listener = new AudioListener({ x: 40, y: 12 });
- * listener.moveTo(41, 12); // following the camera
- * listener.face(Math.PI / 2); // now facing the world's south
- * ```
- */
-export class AudioListener {
-    constructor(options = {}) {
-        this.x = options.x ?? 0;
-        this.y = options.y ?? 0;
-        this.facing = options.facing ?? 0;
-    }
-    moveTo(x, y) {
-        this.x = x;
-        this.y = y;
-    }
-    face(radians) {
-        this.facing = radians;
-    }
-}
-/**
- * One positional one-shot: a `Sound` attached to a point in the world, heard from a
- * listener with the distance gain `audioGain` computes. `playFor` applies that gain through
- * `Sound.play(gain)`, and stays silent past `maxDistance` rather than playing at volume 0.
- *
- * Long-running position (a waterfall, an engine) is the game's own loop calling `playFor` on
- * a cadence it chooses; this is the placement and the mix, not a scheduler.
- *
- * @example
- * ```ts
- * import { AudioListener, Sound, SoundSource } from '@datamoc/mw_games/audio';
- *
- * const listener = new AudioListener({ x: 0, y: 0 });
- * const arrow = new Sound('sounds/arrow.mp3');
- * const source = new SoundSource(arrow, { x: 6, y: 8, refDistance: 4, maxDistance: 20 });
- *
- * const gain = source.gainTo(listener); // 0.4 at distance 10
- * source.playFor(listener); // plays at that gain, or not at all when silent
- * ```
- */
-export class SoundSource {
-    constructor(sound, options = {}) {
-        this.sound = sound;
-        this.x = options.x ?? 0;
-        this.y = options.y ?? 0;
-        this.falloff = {
-            refDistance: options.refDistance,
-            maxDistance: options.maxDistance,
-            rolloff: options.rolloff,
-        };
-    }
-    moveTo(x, y) {
-        this.x = x;
-        this.y = y;
-    }
-    gainTo(listener) {
-        return audioGain(Math.hypot(this.x - listener.x, this.y - listener.y), this.falloff);
-    }
-    panTo(listener) {
-        return audioPan(listener, this);
-    }
-    /**
-     * Plays the source at the gain the listener hears it; returns the gain actually used. A listener
-     * with a `facing` also pans the sound, for a sound that takes a pan (a `Sound` ignores it).
-     */
-    playFor(listener) {
-        const gain = this.gainTo(listener);
-        if (gain <= 0)
-            return 0;
-        this.sound.play(gain, 1, 'facing' in listener ? this.panTo(listener) : 0);
-        return gain;
-    }
-}
-//# sourceMappingURL=Positional.js.map

package/dist/audio/Positional.js.map

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package/dist/audio/Resample.d.ts

@@ -1,61 +0,0 @@
-/**
- * Plain-array helpers for fitting rendered audio to a budget: cut to a length with a fade,
- * fold to mono, halve the rate, convert between rates. They work on the `Float32Array`
- * pairs a render produces, so nothing here needs an `AudioContext`.
- */
-/** stereo frames at a sample rate: the shape every renderer in this module hands back */
-export interface StereoFrames {
-    sampleRate: number;
-    left: Float32Array;
-    right: Float32Array;
-}
-/**
- * Converts stereo frames to another sample rate by linear interpolation. Linear is audibly
- * soft above the new Nyquist, which is the right trade for ambience and background music where
- * this is used to save memory; a game that needs a clean conversion renders at the target
- * rate instead.
- *
- * @example
- * ```ts
- * import { resampleStereo } from '@datamoc/mw_games/audio';
- *
- * declare const frames: { sampleRate: number; left: Float32Array; right: Float32Array };
- *
- * const half = resampleStereo(frames, frames.sampleRate / 2);
- * console.log(half.left.length);
- * ```
- */
-export declare function resampleStereo<T extends StereoFrames>(frames: T, sampleRate: number): T;
-/** what a track may cost: every field optional, an omitted one leaves that dimension alone */
-export interface RenderBudget {
-    /** cut the track to this many seconds, fading the cut so it does not click */
-    maxSeconds?: number;
-    /** seconds of fade at a cut; 0.5 when omitted */
-    fadeSeconds?: number;
-    /** fold both channels into one, stored twice-referenced so a stereo consumer still works */
-    mono?: boolean;
-    /** halve the sample rate, for ambience that carries no high frequencies */
-    halfRate?: boolean;
-}
-/**
- * Fits a rendered track to a budget: cuts it to `maxSeconds` with a fade, folds it to mono
- * and halves its rate, in that order. A mono track shares one array between both channels, so
- * it costs half the memory in the renderer's hands even though a stereo `AudioBuffer` made
- * from it later still has two. Loop points past the cut are pulled back to the new end.
- *
- * @example
- * ```ts
- * import { fitToBudget, renderMidiToBuffer, parseMidi } from '@datamoc/mw_games/audio';
- *
- * declare const midiBytes: ArrayBuffer;
- *
- * const ambience = fitToBudget(renderMidiToBuffer(parseMidi(midiBytes)), { maxSeconds: 60, mono: true, halfRate: true });
- * console.log(ambience.duration);
- * ```
- */
-export declare function fitToBudget<T extends StereoFrames & {
-    loopStart?: number;
-    loopEnd?: number;
-    duration?: number;
-}>(track: T, budget: RenderBudget): T;
-//# sourceMappingURL=Resample.d.ts.map

package/dist/audio/Resample.d.ts.map

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package/dist/audio/Resample.js

@@ -1,97 +0,0 @@
-/**
- * Plain-array helpers for fitting rendered audio to a budget: cut to a length with a fade,
- * fold to mono, halve the rate, convert between rates. They work on the `Float32Array`
- * pairs a render produces, so nothing here needs an `AudioContext`.
- */
-/**
- * Converts stereo frames to another sample rate by linear interpolation. Linear is audibly
- * soft above the new Nyquist, which is the right trade for ambience and background music where
- * this is used to save memory; a game that needs a clean conversion renders at the target
- * rate instead.
- *
- * @example
- * ```ts
- * import { resampleStereo } from '@datamoc/mw_games/audio';
- *
- * declare const frames: { sampleRate: number; left: Float32Array; right: Float32Array };
- *
- * const half = resampleStereo(frames, frames.sampleRate / 2);
- * console.log(half.left.length);
- * ```
- */
-export function resampleStereo(frames, sampleRate) {
-    if (!(sampleRate > 0))
-        throw new Error('target sample rate must be positive');
-    if (sampleRate === frames.sampleRate)
-        return frames;
-    const length = Math.max(1, Math.round((frames.left.length * sampleRate) / frames.sampleRate));
-    const step = frames.sampleRate / sampleRate;
-    const left = new Float32Array(length);
-    const right = new Float32Array(length);
-    const last = frames.left.length - 1;
-    for (let i = 0; i < length; i++) {
-        const position = i * step;
-        const floor = Math.min(last, Math.floor(position));
-        const next = Math.min(last, floor + 1);
-        const fraction = position - floor;
-        left[i] = frames.left[floor] + (frames.left[next] - frames.left[floor]) * fraction;
-        right[i] = frames.right[floor] + (frames.right[next] - frames.right[floor]) * fraction;
-    }
-    return { ...frames, sampleRate, left, right };
-}
-/**
- * Fits a rendered track to a budget: cuts it to `maxSeconds` with a fade, folds it to mono
- * and halves its rate, in that order. A mono track shares one array between both channels, so
- * it costs half the memory in the renderer's hands even though a stereo `AudioBuffer` made
- * from it later still has two. Loop points past the cut are pulled back to the new end.
- *
- * @example
- * ```ts
- * import { fitToBudget, renderMidiToBuffer, parseMidi } from '@datamoc/mw_games/audio';
- *
- * declare const midiBytes: ArrayBuffer;
- *
- * const ambience = fitToBudget(renderMidiToBuffer(parseMidi(midiBytes)), { maxSeconds: 60, mono: true, halfRate: true });
- * console.log(ambience.duration);
- * ```
- */
-export function fitToBudget(track, budget) {
-    let result = track;
-    if (budget.maxSeconds !== undefined) {
-        if (!(budget.maxSeconds > 0))
-            throw new Error('budget maxSeconds must be positive');
-        const limit = Math.max(1, Math.floor(budget.maxSeconds * result.sampleRate));
-        if (result.left.length > limit) {
-            const left = result.left.slice(0, limit);
-            const right = result.right.slice(0, limit);
-            const fade = Math.min(limit, Math.floor((budget.fadeSeconds ?? 0.5) * result.sampleRate));
-            for (let i = 0; i < fade; i++) {
-                const gain = i / fade;
-                left[limit - 1 - i] *= gain;
-                right[limit - 1 - i] *= gain;
-            }
-            const seconds = limit / result.sampleRate;
-            result = withEnd({ ...result, left, right }, seconds);
-        }
-    }
-    if (budget.mono) {
-        const mixed = new Float32Array(result.left.length);
-        for (let i = 0; i < mixed.length; i++)
-            mixed[i] = (result.left[i] + result.right[i]) / 2;
-        result = { ...result, left: mixed, right: mixed };
-    }
-    if (budget.halfRate)
-        result = resampleStereo(result, result.sampleRate / 2);
-    return result;
-}
-function withEnd(track, seconds) {
-    const next = { ...track };
-    if (next.duration !== undefined)
-        next.duration = seconds;
-    if (next.loopEnd !== undefined)
-        next.loopEnd = Math.min(next.loopEnd, seconds);
-    if (next.loopStart !== undefined)
-        next.loopStart = Math.min(next.loopStart, Math.max(0, seconds - 0.1));
-    return next;
-}
-//# sourceMappingURL=Resample.js.map

package/dist/audio/Resample.js.map

@@ -1 +0,0 @@
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package/dist/audio/SampleDecoder.d.ts

@@ -1,55 +0,0 @@
-/**
- * Turns the Ogg Vorbis bytes of one SoundFont 3 sample into mono 16-bit frames at the
- * sample's own rate, which is the rate its header states and the frame count its loop
- * points were written against. Synchronous or asynchronous: a browser decodes through
- * WebAudio (`webAudioSampleDecoder`), a Node tool through whatever it has installed.
- */
-export type SoundFontSampleDecoder = (ogg: Uint8Array, rate: number) => Int16Array | Promise<Int16Array>;
-/** the slice of `OfflineAudioContext` the decoder uses, so a test can stand in for it */
-export interface SampleDecodingContext {
-    decodeAudioData(data: ArrayBuffer): Promise<{
-        sampleRate: number;
-        length: number;
-        getChannelData(channel: number): Float32Array;
-    }>;
-}
-/**
- * What a sample header may claim. The rate and the Ogg bytes come from a font file that may have been
- * downloaded, so what they ask the decoder to allocate is bounded: the SoundFont 2 format itself
- * stops at 50 kHz, and sixteen million frames is six minutes at 44.1 kHz for a single instrument sample.
- */
-export declare const MIN_SAMPLE_RATE = 400;
-export declare const MAX_SAMPLE_RATE = 200000;
-export declare const MAX_SAMPLE_FRAMES: number;
-/**
- * How much a SoundFont 3 may decode to, in frames. The defaults (`MAX_SAMPLE_FRAMES` for one sample, 128
- * million, 256 MB of 16-bit audio, for the whole font) suit a game font; raise them for a very large one
- * you trust, since each frame is two bytes of memory and a damaged font can ask for any number.
- */
-export interface SoundFontLimits {
-    /** frames one decoded sample may have */
-    sampleFrames?: number;
-    /** frames all the decoded samples of one font may have */
-    totalFrames?: number;
-}
-/** the whole-font default of {@link SoundFontLimits} */
-export declare const MAX_TOTAL_FRAMES: number;
-/**
- * A decoder over WebAudio's `decodeAudioData`, for a browser: each sample decodes in a context
- * created at the sample's own rate, so the browser does not resample it and the loop points keep
- * meaning what the font wrote. A rate outside what a context accepts (8 to 96 kHz) decodes at
- * the nearest accepted rate and is resampled back. `create` replaces the context factory and `limits`
- * the longest sample it will return.
- *
- * @example
- * ```ts
- * import { loadSoundFont, webAudioSampleDecoder } from '@datamoc/mw_games/audio';
- *
- * declare const sf3Bytes: ArrayBuffer;
- *
- * const font = await loadSoundFont(sf3Bytes, { decode: webAudioSampleDecoder() });
- * console.log(font.presetCount);
- * ```
- */
-export declare function webAudioSampleDecoder(create?: (rate: number) => SampleDecodingContext, limits?: SoundFontLimits): SoundFontSampleDecoder;
-//# sourceMappingURL=SampleDecoder.d.ts.map

package/dist/audio/SampleDecoder.d.ts.map

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package/dist/audio/SampleDecoder.js

@@ -1,74 +0,0 @@
-/** the narrowest rate range every engine's `OfflineAudioContext` accepts */
-const MIN_RATE = 8000;
-const MAX_RATE = 96000;
-/**
- * What a sample header may claim. The rate and the Ogg bytes come from a font file that may have been
- * downloaded, so what they ask the decoder to allocate is bounded: the SoundFont 2 format itself
- * stops at 50 kHz, and sixteen million frames is six minutes at 44.1 kHz for a single instrument sample.
- */
-export const MIN_SAMPLE_RATE = 400;
-export const MAX_SAMPLE_RATE = 200000;
-export const MAX_SAMPLE_FRAMES = 1 << 24;
-/** the whole-font default of {@link SoundFontLimits} */
-export const MAX_TOTAL_FRAMES = 1 << 27;
-function toInt16(frames) {
-    const out = new Int16Array(frames.length);
-    for (let index = 0; index < frames.length; index++) {
-        out[index] = Math.round(Math.max(-1, Math.min(1, frames[index])) * 32767);
-    }
-    return out;
-}
-/** linear resampling, only for the rare sample whose rate a context cannot be created at */
-function resample(frames, from, to, maxFrames) {
-    const length = Math.max(1, Math.round((frames.length * to) / from));
-    if (length > maxFrames)
-        throw new Error('a decoded sample is longer than a SoundFont sample may be');
-    const out = new Float32Array(length);
-    for (let index = 0; index < out.length; index++) {
-        const at = (index * from) / to;
-        const base = Math.floor(at);
-        const next = Math.min(base + 1, frames.length - 1);
-        out[index] = frames[base] + (frames[next] - frames[base]) * (at - base);
-    }
-    return out;
-}
-function offlineContext(rate) {
-    if (typeof OfflineAudioContext === 'undefined') {
-        throw new Error('no OfflineAudioContext here: pass a decoder of your own to decode a SoundFont 3');
-    }
-    return new OfflineAudioContext(1, 1, rate);
-}
-/**
- * A decoder over WebAudio's `decodeAudioData`, for a browser: each sample decodes in a context
- * created at the sample's own rate, so the browser does not resample it and the loop points keep
- * meaning what the font wrote. A rate outside what a context accepts (8 to 96 kHz) decodes at
- * the nearest accepted rate and is resampled back. `create` replaces the context factory and `limits`
- * the longest sample it will return.
- *
- * @example
- * ```ts
- * import { loadSoundFont, webAudioSampleDecoder } from '@datamoc/mw_games/audio';
- *
- * declare const sf3Bytes: ArrayBuffer;
- *
- * const font = await loadSoundFont(sf3Bytes, { decode: webAudioSampleDecoder() });
- * console.log(font.presetCount);
- * ```
- */
-export function webAudioSampleDecoder(create = offlineContext, limits = {}) {
-    const maxFrames = limits.sampleFrames ?? MAX_SAMPLE_FRAMES;
-    return async (ogg, rate) => {
-        if (!(rate >= MIN_SAMPLE_RATE && rate <= MAX_SAMPLE_RATE)) {
-            throw new Error(`a sample rate of ${rate} Hz is not one a SoundFont can state`);
-        }
-        const contextRate = Math.max(MIN_RATE, Math.min(MAX_RATE, rate));
-        //decodeAudioData detaches what it is given, and `ogg` is a view into the caller's font
-        const copy = ogg.slice().buffer;
-        const decoded = await create(contextRate).decodeAudioData(copy);
-        if (decoded.length > maxFrames)
-            throw new Error('a decoded sample is longer than a SoundFont sample may be');
-        const channel = decoded.getChannelData(0);
-        return toInt16(decoded.sampleRate === rate ? channel : resample(channel, decoded.sampleRate, rate, maxFrames));
-    };
-}
-//# sourceMappingURL=SampleDecoder.js.map

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package/dist/audio/Settings.d.ts

@@ -1,99 +0,0 @@
-import type { AudioBus, BusKind } from './Channels.ts';
-/** the television standard a retro machine ran at: it sets the clock its music was timed by */
-export type AudioRegion = 'pal' | 'ntsc';
-/**
- * How a machine profile's output stage is treated: `faithful` keeps what the hardware did
- * (DC offset, aliasing and all), `clean` removes the DC and softens the top end.
- */
-export type OutputStage = 'faithful' | 'clean';
-/** player-facing audio settings as plain data, the shape a save or a settings file holds */
-export interface AudioSettingsData {
-    /** volume per bus kind, 0 to 100 like every other volume in the bus API */
-    busVolumes: Record<BusKind, number>;
-    /** the MIDI engine name content is voiced by first, or null for the internal path */
-    preferredEngine: string | null;
-    region: AudioRegion;
-    outputStage: OutputStage;
-    /** whether caption text is shown for captioned sounds */
-    captions: boolean;
-    /** dynamics control, 0 (untouched) to 1 (a soft limiter at full), for loudness comfort */
-    dynamics: number;
-    /** softening of bright or broadband sounds, 0 (untouched) to 1 (most softened) */
-    soften: number;
-}
-/**
- * The playback rate a tune authored for one region runs at on the other: a PAL machine's
- * 50 Hz clock against NTSC's 60 Hz makes the same file slower and lower, which is the
- * difference a player who grew up with the other standard hears first.
- *
- * @example
- * ```ts
- * import { regionPlaybackRate } from '@datamoc/mw_games/audio';
- *
- * console.log(regionPlaybackRate('pal', 'ntsc')); // 0.8333..., slower on PAL
- * console.log(regionPlaybackRate('ntsc', 'ntsc')); // 1, its own region
- * ```
- */
-export declare function regionPlaybackRate(region: AudioRegion, authored?: AudioRegion): number;
-/**
- * The low-pass cutoff, in hertz, that a `soften` amount asks an output stage for: nothing
- * at 0, down to a muffled 3.5 kHz at 1, never above what the sample rate can represent.
- *
- * @example
- * ```ts
- * import { softenCutoff } from '@datamoc/mw_games/audio';
- *
- * console.log(softenCutoff(0, 44100)); // 18000, effectively open
- * console.log(softenCutoff(1, 44100) < softenCutoff(0.5, 44100)); // true, more soften, lower cutoff
- * ```
- */
-export declare function softenCutoff(soften: number, sampleRate: number): number;
-/**
- * One sample through the dynamics control: at 0 the sample is untouched, and as the amount
- * rises a soft knee pulls peaks toward full scale without ever passing it, so a loud game
- * gets quieter peaks rather than clipped ones.
- *
- * @example
- * ```ts
- * import { dynamicsLimit } from '@datamoc/mw_games/audio';
- *
- * console.log(dynamicsLimit(0.5, 0)); // 0.5, untouched
- * console.log(dynamicsLimit(4, 1) < 1); // true, a hot sample is pulled under full scale
- * ```
- */
-export declare function dynamicsLimit(sample: number, dynamics: number): number;
-/**
- * The player's audio settings in one place, over the pieces that already exist: bus
- * volumes apply to an `AudioBus` (`applyToBus`), the preferred engine names what
- * `resolveMidiSynth` is asked for first, the region and output stage describe the machine
- * profile a retro package renders for, and captions, dynamics and soften are the
- * accessibility dials. Everything clamps on the way in, and `snapshot`/`restore` round-trip
- * the plain-data shape a settings screen persists.
- *
- * @example
- * ```ts
- * import { AudioSettings } from '@datamoc/mw_games/audio';
- *
- * const settings = new AudioSettings({ preferredEngine: 'retro:mt32', soften: 0.4 });
- * settings.setBusVolume('se', 130);
- * console.log(settings.snapshot().busVolumes.se); // 100, clamped
- * ```
- */
-export declare class AudioSettings {
-    private data;
-    constructor(initial?: Partial<AudioSettingsData>);
-    /** a copy of the current settings, safe to store */
-    snapshot(): AudioSettingsData;
-    /** takes settings in, clamping every dial; a partial update keeps the rest */
-    restore(data: Partial<AudioSettingsData>): void;
-    setBusVolume(kind: BusKind, volume: number): void;
-    /** pushes the per-bus volumes onto a bus's mixer, the settings' one live connection */
-    applyToBus(bus: AudioBus): void;
-    /** the rate content authored for `authored` should play at under this region */
-    playbackRate(authored?: AudioRegion): number;
-    /** the cutoff the current soften amount (and a clean stage's extra softening) asks for */
-    cutoff(sampleRate: number): number;
-    /** one sample through the current dynamics amount */
-    limit(sample: number): number;
-}
-//# sourceMappingURL=Settings.d.ts.map

package/dist/audio/Settings.d.ts.map

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