@swf2es/player 0.0.0-stage → 0.1.0

package/dist/media/dynamic-sound.js

@@ -0,0 +1,91 @@
+// SampleDataEvent's bytes belong to the script; only detached float buffers reach the audio host.
+import { avm2 } from "@swf2es/runtime";
+import { dispatchEvent } from "../scripting/events.js";
+import { mixerMix, outputMix } from "./sounds.js";
+export function startDynamicSound(s, owner, state, onFailure) {
+    const data = s.rt.construct(s.rt.classNamed("flash.utils::ByteArray"));
+    const bytes = avm2.bytesOf(s.rt, data);
+    let generated = 0;
+    let requestOffset = 0;
+    let finished = false;
+    let requesting = false;
+    let ready = false;
+    let starting = true;
+    let stream;
+    const position = () => stream?.positionMs ?? Math.min((generated * 1000) / 44100, s.timers.now - state.started);
+    const request = () => {
+        // AIR preserves the ByteArray's contents and endian, resetting only its write cursor.
+        bytes.position = 0;
+        const event = s.rt.construct(s.rt.classNamed("flash.events::SampleDataEvent"), "sampleData", false, false, generated + requestOffset, data);
+        dispatchEvent(s, owner, event);
+        if (state.stopped) {
+            return;
+        }
+        const frames = Math.floor(bytes.position / 8);
+        if (frames > 8192 || bytes.position > bytes.length) {
+            finished = true;
+            stream?.finish();
+            s.reportUncaught(frames > 8192 ? s.rt.error("RangeError", 2004) : s.rt.error("flash.errors::EOFError", 2030));
+            return;
+        }
+        if (frames > 0) {
+            const channels = [new Float32Array(frames), new Float32Array(frames)];
+            for (let i = 0; i < frames; i++) {
+                for (let c = 0; c < 2; c++) {
+                    const sample = bytes.view.getFloat32(i * 8 + c * 4, bytes.littleEndian);
+                    channels[c][i] = Number.isFinite(sample) ? sample : 0;
+                }
+            }
+            generated += frames;
+            stream?.append(channels);
+        }
+        if (frames < 2048) {
+            finished = true;
+            stream?.finish();
+        }
+    };
+    const pump = () => {
+        if (!ready || requesting || finished || state.stopped) {
+            return;
+        }
+        requesting = true;
+        try {
+            if (generated - (position() * 44100) / 1000 < 8192) {
+                request();
+            }
+        }
+        catch (error) {
+            if (starting) {
+                throw error;
+            }
+            finished = true;
+            onFailure(error);
+        }
+        finally {
+            requesting = false;
+        }
+    };
+    state.dynamic = {
+        position: () => state.start + position(),
+        pump,
+        complete: () => finished && (stream ? stream.ended === true : position() >= generated / 44.1),
+    };
+    stream = s.audio?.stream?.(outputMix(state.mix, mixerMix(s)), pump);
+    state.playing = stream ?? null;
+    ready = true;
+    pump();
+    // AIR fills another buffer synchronously when the first supplied the full 8192 frames.
+    if (!finished && !state.stopped && generated === 8192) {
+        requesting = true;
+        try {
+            request();
+        }
+        finally {
+            requesting = false;
+        }
+    }
+    // AIR applies startTime to later request positions, after its synchronous initial fill.
+    requestOffset = Math.floor(state.start * 44.1);
+    starting = false;
+}
+//# sourceMappingURL=dynamic-sound.js.map

package/dist/media/dynamic-sound.js.map

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package/dist/media/extract.d.ts

@@ -0,0 +1,41 @@
+import type { Sound } from "@swf2es/format";
+export declare const EXTRACT_RATE = 44100;
+/** A sound's samples to extract from. */
+export interface ExtractSource {
+    rate: number;
+    /** One channel, which extract writes to both, or two. */
+    channels: Float32Array[];
+    /** Samples at the start that extract never reaches: an MP3 DefineSound's seekSamples. */
+    skip: number;
+    /**
+     * Whether it writes whole samples of its own only, as adl does for a
+     * sound uncompressed or ADPCM in the SWF: a length short of one writes
+     * none.
+     */
+    whole: boolean;
+    /** Where a script may start it, short of its end: an ADPCM sound's samples, not the silence after them. */
+    starts?: number;
+    /** Where a start lands, for a sound whose seeking adl gets wrong. */
+    seek?: (start: number) => number;
+}
+/**
+ * Whole MP3 frames, including a header frame's silence, without gapless
+ * trimming. None where the engine could not compile the decoder: extract
+ * then writes nothing, as Flash does for a sound it cannot decode.
+ */
+export declare function mp3Source(bytes: Uint8Array): ExtractSource | null;
+/** An uncompressed, ADPCM or Nellymoser DefineSound's samples to extract, or null for another format. */
+export declare function embeddedSource(sound: Sound): ExtractSource | null;
+/** Samples at `rate` brought to 44.1 kHz as loadPCMFromByteArray does: each output sample the one it falls in. */
+export declare function toExtractRate(channels: Float32Array[], rate: number): Float32Array[];
+/**
+ * Up to `length` samples from `position`, interleaved, and the position
+ * after them. What extract returns is how many it wrote, but once the
+ * sound runs out, how many of its own samples it read, as adl counts them.
+ */
+export declare function extractSamples(source: ExtractSource, position: number, length: number): {
+    samples: Float32Array;
+    position: number;
+    count: number;
+};
+//# sourceMappingURL=extract.d.ts.map

package/dist/media/extract.d.ts.map

@@ -0,0 +1 @@
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package/dist/media/extract.js

@@ -0,0 +1,136 @@
+import { decodeAdpcm, NellymoserDecoder } from "@swf2es/native/audio";
+import { Mp3Decoder, mp3Available } from "@swf2es/native/mp3";
+import { mp3Frames } from "./mp3.js";
+export const EXTRACT_RATE = 44100;
+/**
+ * Whole MP3 frames, including a header frame's silence, without gapless
+ * trimming. None where the engine could not compile the decoder: extract
+ * then writes nothing, as Flash does for a sound it cannot decode.
+ */
+export function mp3Source(bytes) {
+    const mp3 = mp3Frames(bytes);
+    if (!mp3?.whole.length || !mp3Available()) {
+        return null;
+    }
+    const frames = (mp3.whole.length / 2) * mp3.samplesPerFrame;
+    const channels = Array.from({ length: mp3.channels }, () => new Float32Array(frames));
+    const decoder = new Mp3Decoder();
+    let at = 0;
+    for (let k = 0; k < mp3.whole.length; k += 2) {
+        const count = decoder.read(bytes.subarray(mp3.whole[k], mp3.whole[k + 1]));
+        if (decoder.channels === mp3.channels && decoder.rate === mp3.rate) {
+            const take = Math.min(count, mp3.samplesPerFrame);
+            for (let i = 0; i < take; i++) {
+                for (let c = 0; c < mp3.channels; c++) {
+                    channels[c][at + i] = decoder.pcm[i * mp3.channels + c];
+                }
+            }
+        }
+        at += mp3.samplesPerFrame;
+    }
+    return { rate: mp3.rate, channels, skip: 0, whole: false };
+}
+/** An uncompressed, ADPCM or Nellymoser DefineSound's samples to extract, or null for another format. */
+export function embeddedSource(sound) {
+    const n = sound.channels;
+    const source = { rate: sound.sampleRate, skip: 0, whole: true };
+    if (sound.format === 1) {
+        return { ...source, ...adpcmSource(sound.data, n) };
+    }
+    // Mono, at the rate its flags give, as adl extracts it, formats 4 and 5
+    // too, which play at 16 and 8 kHz (audio.ts). adl's own extract of it is
+    // erratic past 4096 samples a call and near the sound's end
+    // (scripts/SoundExtractNellymoser.as): those are not copied.
+    if (sound.format === 4 || sound.format === 5 || sound.format === 6) {
+        const samples = new NellymoserDecoder().decode(sound.data);
+        return {
+            ...source,
+            channels: [samples.subarray(0, Math.min(samples.length, sound.sampleCount))],
+        };
+    }
+    if (sound.format !== 0 && sound.format !== 3) {
+        return null;
+    }
+    const view = new DataView(sound.data.buffer, sound.data.byteOffset, sound.data.byteLength);
+    const size = sound.sampleSize / 8;
+    const frames = Math.min(sound.sampleCount, Math.floor(sound.data.byteLength / (size * n)));
+    return {
+        ...source,
+        channels: deinterleave(frames, n, (i) => size === 1 ? (view.getUint8(i) - 128) / 128 : view.getInt16(i * 2, true) / 32768),
+    };
+}
+/** Samples adl decodes in blocks of 2048, the last run on past the data as if its bits went on as zeros. */
+const ADPCM_BLOCK = 2048;
+/** Samples a packet: its header's and 4095 codes'. */
+const ADPCM_PACKET = 4096;
+/**
+ * An ADPCM sound's samples as adl extracts them: blocks of samples, the
+ * last running on as if the data's bits went on as zeros (where adl reads
+ * on into whatever follows the data), though a script may start only
+ * within the data. A start's packet's offset in bits is reckoned in 32
+ * bits: where that wraps negative, adl starts from the first packet, as
+ * far into it as the start is into its own.
+ */
+function adpcmSource(data, n) {
+    const frames = decodeAdpcm(data, n).length / n;
+    const padded = (Math.floor(frames / ADPCM_BLOCK) + 1) * ADPCM_BLOCK;
+    const bits = data.length ? (data[0] >> 6) + 2 : 4;
+    const more = new Uint8Array(data.length + Math.ceil(((padded - frames) * bits * n) / 8) + 8 * n);
+    more.set(data);
+    const samples = decodeAdpcm(more, n);
+    const packetBits = 22 * n + (ADPCM_PACKET - 1) * bits * n;
+    return {
+        channels: deinterleave(padded, n, (i) => samples[i] / 32768),
+        starts: frames,
+        seek: (start) => Math.imul(Math.floor(start / ADPCM_PACKET), packetBits) < 0 ? start % ADPCM_PACKET : start,
+    };
+}
+function deinterleave(frames, n, at) {
+    const channels = Array.from({ length: n }, () => new Float32Array(frames));
+    for (let i = 0; i < frames; i++) {
+        for (let c = 0; c < n; c++) {
+            channels[c][i] = at(i * n + c);
+        }
+    }
+    return channels;
+}
+/** Samples at `rate` brought to 44.1 kHz as loadPCMFromByteArray does: each output sample the one it falls in. */
+export function toExtractRate(channels, rate) {
+    if (rate === EXTRACT_RATE) {
+        return channels;
+    }
+    const frames = Math.floor((channels[0].length * EXTRACT_RATE) / rate);
+    return channels.map((channel) => {
+        const out = new Float32Array(frames);
+        for (let i = 0; i < frames; i++) {
+            out[i] = channel[Math.floor((i * rate) / EXTRACT_RATE)];
+        }
+        return out;
+    });
+}
+/**
+ * Up to `length` samples from `position`, interleaved, and the position
+ * after them. What extract returns is how many it wrote, but once the
+ * sound runs out, how many of its own samples it read, as adl counts them.
+ */
+export function extractSamples(source, position, length) {
+    const total = source.channels[0].length - source.skip;
+    const ratio = EXTRACT_RATE / source.rate;
+    const room = Math.max(0, total - position);
+    const wanted = Math.floor(length);
+    let written = wanted > 0 ? Math.min(wanted, Math.floor(room * ratio)) : 0;
+    if (source.whole) {
+        written = Math.floor(Math.floor(written / ratio) * ratio);
+    }
+    const read = Math.min(room, Math.ceil(written / ratio));
+    const [left, right = left] = source.channels;
+    const samples = new Float32Array(written * 2);
+    for (let i = 0; i < written; i++) {
+        const at = source.skip + position + Math.floor(i / ratio);
+        samples[i * 2] = left[at];
+        samples[i * 2 + 1] = right[at];
+    }
+    const end = position + read >= total;
+    return { samples, position: position + read, count: end ? read : written };
+}
+//# sourceMappingURL=extract.js.map

package/dist/media/extract.js.map

@@ -0,0 +1 @@
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package/dist/media/mp3.d.ts

@@ -0,0 +1,19 @@
+export interface Mp3Frames {
+    rate: number;
+    channels: 1 | 2;
+    samplesPerFrame: number;
+    /** Frames whole and cut short, as Sound.length counts them. */
+    frames: number;
+    /** Where each whole frame starts and ends, in pairs. */
+    whole: number[];
+    /** Where reading goes on from with more bytes: after the last whole frame. */
+    end: number;
+}
+/**
+ * The frames of MP3 bytes, or null for bytes with none. Given what it read
+ * of the bytes before more came after them, it reads on from there.
+ */
+export declare function mp3Frames(bytes: Uint8Array, before?: Mp3Frames | null): Mp3Frames | null;
+/** The first frame's gapless metadata, without scanning and retaining every frame of a long sound. */
+export declare function mp3HeaderFrame(bytes: Uint8Array): [number, number] | null;
+//# sourceMappingURL=mp3.d.ts.map

package/dist/media/mp3.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"mp3.d.ts","sourceRoot":"","sources":["../../src/media/mp3.ts"],"names":[],"mappings":"AAuBA,MAAM,WAAW,SAAS;IACxB,IAAI,EAAE,MAAM,CAAC;IACb,QAAQ,EAAE,CAAC,GAAG,CAAC,CAAC;IAChB,eAAe,EAAE,MAAM,CAAC;IACxB,+DAA+D;IAC/D,MAAM,EAAE,MAAM,CAAC;IACf,wDAAwD;IACxD,KAAK,EAAE,MAAM,EAAE,CAAC;IAChB,8EAA8E;IAC9E,GAAG,EAAE,MAAM,CAAC;CACb;AAwGD;;;GAGG;AACH,wBAAgB,SAAS,CAAC,KAAK,EAAE,UAAU,EAAE,MAAM,GAAE,SAAS,GAAG,IAAW,GAAG,SAAS,GAAG,IAAI,CAQ9F;AA+CD,sGAAsG;AACtG,wBAAgB,cAAc,CAAC,KAAK,EAAE,UAAU,GAAG,CAAC,MAAM,EAAE,MAAM,CAAC,GAAG,IAAI,CAOzE"}

package/dist/media/mp3.js

@@ -0,0 +1,153 @@
+// What Flash reads of MP3 bytes before it decodes them: its frames, by
+// their headers, past ID3v2 tags and other bytes between them. Flash
+// counts a frame the bytes cut short in a sound's length but decodes only
+// whole frames, and decodes a Xing, Info or VBRI frame as a frame of
+// silence like any other (adl, the sound-extract case), where a browser's
+// decoder reads it to trim the encoder's delay and padding. A header is
+// trusted as the start of a run only with the next two frames' headers
+// where it says they are, as decoders find their way into a stream, so
+// that bytes that happen to look like one, in a tag or another format,
+// are passed over.
+/** Kilobits a second by bitrate index: MPEG-1 Layer III, then MPEG-2 and 2.5. */
+const BITRATES = [
+    [0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320],
+    [0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160],
+];
+/** Rates by index for MPEG-2.5, (reserved), MPEG-2 and MPEG-1, as the version bits number them. */
+const RATES = [[11025, 12000, 8000], [], [22050, 24000, 16000], [44100, 48000, 32000]];
+/** How many headers in a row, the first included, make a run. */
+const RUN = 3;
+function header(bytes, at) {
+    if (at + 4 > bytes.length || bytes[at] !== 0xff || (bytes[at + 1] & 0xe0) !== 0xe0) {
+        return null;
+    }
+    const version = (bytes[at + 1] >> 3) & 3;
+    const layer = (bytes[at + 1] >> 1) & 3;
+    const bitrate = bytes[at + 2] >> 4;
+    const rateIndex = (bytes[at + 2] >> 2) & 3;
+    if (version === 1 || layer !== 1 || bitrate === 0 || bitrate === 15 || rateIndex === 3) {
+        return null;
+    }
+    const mpeg1 = version === 3;
+    const rate = RATES[version][rateIndex];
+    const padding = (bytes[at + 2] >> 1) & 1;
+    const kbps = BITRATES[mpeg1 ? 0 : 1][bitrate];
+    return {
+        version,
+        rate,
+        channels: bytes[at + 3] >> 6 === 3 ? 1 : 2,
+        samples: mpeg1 ? 1152 : 576,
+        length: Math.floor(((mpeg1 ? 144 : 72) * kbps * 1000) / rate) + padding,
+        mpeg1,
+    };
+}
+/** A header of the stream's kind at `at`: its version, rate and channels those of `like`. */
+function frameAt(bytes, at, like) {
+    const h = header(bytes, at);
+    if (!h ||
+        (like && (h.version !== like.version || h.rate !== like.rate || h.channels !== like.channels))) {
+        return null;
+    }
+    return h;
+}
+/** A header at `at` the next frames' headers follow, as many as the bytes hold up to a run. */
+function runAt(bytes, at, like) {
+    const first = frameAt(bytes, at, like);
+    let next = first ? at + first.length : 0;
+    for (let k = 1; first && k < RUN && next + 4 <= bytes.length; k++) {
+        const h = frameAt(bytes, next, first);
+        if (!h) {
+            return null;
+        }
+        next += h.length;
+    }
+    return first;
+}
+/** The length of an ID3v2 tag at `at`, header, body and footer, or 0 for none. */
+function id3Length(bytes, at) {
+    if (at + 10 > bytes.length ||
+        bytes[at] !== 0x49 ||
+        bytes[at + 1] !== 0x44 ||
+        bytes[at + 2] !== 0x33 ||
+        (bytes[at + 6] | bytes[at + 7] | bytes[at + 8] | bytes[at + 9]) & 0x80) {
+        return 0;
+    }
+    // Its size in seven-bit bytes.
+    const size = (bytes[at + 6] << 21) | (bytes[at + 7] << 14) | (bytes[at + 8] << 7) | bytes[at + 9];
+    return 10 + size + (bytes[at + 5] & 0x10 ? 10 : 0);
+}
+/** The next run of frames from `at`, past tags and anything else, and where it starts; null for none. */
+function nextRun(bytes, at, like) {
+    for (let p = at; p + 4 <= bytes.length;) {
+        const tag = id3Length(bytes, p);
+        if (tag) {
+            p += tag;
+            continue;
+        }
+        const h = runAt(bytes, p, like);
+        if (h) {
+            return [p, h];
+        }
+        p++;
+    }
+    return null;
+}
+/**
+ * The frames of MP3 bytes, or null for bytes with none. Given what it read
+ * of the bytes before more came after them, it reads on from there.
+ */
+export function mp3Frames(bytes, before = null) {
+    const like = before?.whole.length ? header(bytes, before.whole[0]) : null;
+    if (!like) {
+        const run = nextRun(bytes, 0, null);
+        return run ? readFrames(bytes, run[0], run[1], []) : null;
+    }
+    return readFrames(bytes, before?.end ?? 0, like, before?.whole ?? []);
+}
+function readFrames(bytes, from, like, before) {
+    const whole = before.slice();
+    let frames = whole.length / 2;
+    let at = from;
+    let end = from;
+    while (at + 4 <= bytes.length) {
+        let h = frameAt(bytes, at, like);
+        if (!h) {
+            const run = nextRun(bytes, at, like);
+            if (!run) {
+                break;
+            }
+            [at, h] = run;
+        }
+        frames++;
+        if (at + h.length > bytes.length) {
+            break;
+        }
+        whole.push(at, at + h.length);
+        at += h.length;
+        end = at;
+    }
+    return {
+        rate: like.rate,
+        channels: like.channels,
+        samplesPerFrame: like.samples,
+        frames,
+        whole,
+        end,
+    };
+}
+/** Whether the frame at `at` is a Xing, Info or VBRI header rather than sound. */
+function isHeaderFrame(bytes, at, h) {
+    const tag = (offset) => String.fromCharCode(...bytes.subarray(at + offset, at + offset + 4));
+    const side = h.mpeg1 ? (h.channels === 1 ? 17 : 32) : h.channels === 1 ? 9 : 17;
+    const xing = tag(4 + side);
+    return xing === "Xing" || xing === "Info" || tag(36) === "VBRI";
+}
+/** The first frame's gapless metadata, without scanning and retaining every frame of a long sound. */
+export function mp3HeaderFrame(bytes) {
+    const first = nextRun(bytes, 0, null);
+    if (!first || !isHeaderFrame(bytes, first[0], first[1])) {
+        return null;
+    }
+    return [first[0], Math.min(bytes.length, first[0] + first[1].length)];
+}
+//# sourceMappingURL=mp3.js.map

package/dist/media/mp3.js.map

@@ -0,0 +1 @@
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package/dist/media/sounds.d.ts

@@ -0,0 +1,157 @@
+import type { avm2 } from "@swf2es/runtime";
+import type { DisplayObject, MovieClip } from "../display/display.js";
+import type { SoundCharacter, TimelineSounds } from "../display/timeline.js";
+import type { Scripting } from "../scripting.js";
+import { type AudioHost, type DecodedSound, type PlayingSound, type SoundMix } from "./audio.js";
+import type { DynamicChannel } from "./dynamic-sound.js";
+import type { Mp3Frames } from "./mp3.js";
+type AsObject = avm2.AsObject;
+/** What the timeline's sounds play by: an AS3 movie's Scripting, or AVM1's movie. */
+export interface SoundHost {
+    readonly audio: AudioHost | null;
+    /** The player's clock, in ms. */
+    readonly now: number;
+    /** The SWF frames played: what the timeline took off this frame plays its streams no more. */
+    readonly frames: number;
+    readonly frameRate: number;
+    /** A sound decoded for the device, once; null where there is no device. */
+    soundClip(character: SoundCharacter): Promise<DecodedSound> | null;
+    /** A display object's own transform, where a script gave it one. */
+    ownMix(o: DisplayObject): SoundMix | undefined;
+    /** The transform over every sound. */
+    readonly mixer: SoundMix;
+    /** The channels a script opened that hold a sound, outside the timeline's. */
+    scriptSounds(): number;
+}
+export interface SoundState {
+    character: SoundCharacter | null;
+    bytes: Uint8Array | null;
+    url: string | null;
+    loaded: number;
+    total: number;
+    length: number;
+    used: boolean;
+    generation: number;
+    abort: AbortController | null;
+    clip: Promise<DecodedSound> | null;
+    /** The MP3 frames of loadCompressedDataFromByteArray's bytes, read so far. */
+    frames: Mp3Frames | null;
+    /** Whether `bytes` came from loadCompressedDataFromByteArray, which adds to them. */
+    compressed: boolean;
+    /** loadPCMFromByteArray's samples, at 44.1 kHz. */
+    pcm: Float32Array[] | null;
+    /** Where extract goes on from, in the sound's own samples. */
+    extracted: number;
+}
+export interface ChannelState {
+    dynamic?: DynamicChannel;
+    sound: SoundState;
+    started: number;
+    start: number;
+    loops: number;
+    mix: SoundMix;
+    playing: PlayingSound | null;
+    stopped: boolean;
+    /** Where a stopped channel's position stays. */
+    stoppedAt: number;
+}
+export declare const channels: WeakMap<Scripting, Set<any>>;
+/** SoundMixer's transform, applied over every channel's own. */
+export declare const mixerMixes: WeakMap<Scripting, SoundMix>;
+export declare const mixOf: (o: AsObject | null) => SoundMix;
+/**
+ * What a channel sends to the device: its own transform and the mixer's,
+ * combined as Ruffle's SoundTransform::concat computes them. Only
+ * transforms that both cross channels depend on the order, which no trace
+ * can show; Ruffle's is kept.
+ */
+export declare function outputMix(local: SoundMix, global: SoundMix): SoundMix;
+export declare const mixerMix: (s: Scripting) => SoundMix;
+/** An AS3 movie's SoundHost: its device, clock and symbols, its sprites' and SoundMixer's transforms. */
+export declare function soundHostOf(s: Scripting): SoundHost;
+export declare const positionOf: (s: Scripting, state: ChannelState) => number;
+export declare function stopChannel(s: Scripting, state: ChannelState): void;
+/**
+ * SoundMixer.stopAll: every channel stays where it was, and none completes;
+ * the timeline's sounds stop too, though a stream starts again at the next
+ * block its clip plays on to, as Ruffle has it.
+ */
+export declare function stopAllSounds(s: Scripting): void;
+/** Every sound the timeline, or AVM1's Sound, started stops, none completing. */
+export declare function stopTimelineSounds(h: SoundHost): void;
+/** A sound the timeline started: an event sound, a button's, a clip's stream, or AVM1 Sound's. */
+export interface TimelineSound {
+    /** The display object it plays under, whose transforms it takes; null for AVM1's global Sound. */
+    owner: DisplayObject | null;
+    character: SoundCharacter;
+    /** The clip whose stream it is; null for an event sound. */
+    clip: MovieClip | null;
+    started: number;
+    /** How long it plays, all loops, by the player's clock. */
+    duration: number;
+    playing: PlayingSound | null;
+    stopped: boolean;
+    /** Whether its decode is still to come. */
+    decoding: boolean;
+    /**
+     * When the player's clock ran its length, or null while it plays: over,
+     * it no longer counts for SyncNoMultiple, but the
+     * device may still play it, later than the clock has it where the page
+     * held the device, and a stop still reaches it until the device is done
+     * with it (`PlayingSound.ended`), or, for a host that cannot tell, until
+     * TAIL has passed and it is stopped for good.
+     */
+    endedAt: number | null;
+    /** What hears that its time has run, with the frame: AVM1's onSoundComplete. */
+    onEnd: (() => void) | null;
+    /** When it was stopped, by the clock, for AVM1 Sound's position; null while it plays. */
+    stoppedAt: number | null;
+    /** A stream's elapsed timeline time and the frame after its final block; null for event sounds. */
+    stream: {
+        elapsedMs: number;
+        endFrame: number;
+    } | null;
+}
+/**
+ * The sounds that may play at once, timeline and script alike: Flash's 32
+ * channels, Ruffle's AudioManager::MAX_SOUNDS. A timeline sound past them
+ * does not start, nor queue on a device that is not yet running.
+ */
+export declare const MAX_SOUNDS = 32;
+/**
+ * The sounds that hold a channel: a script's whose sound is there to play
+ * (one still loading holds none until it can start), and the timeline's
+ * the device has or will have.
+ */
+export declare function liveSounds(s: Scripting): number;
+/** A sprite's transform or the mixer's changed: every timeline sound's mix follows. */
+export declare function updateTimelineMixes(s: Scripting): void;
+export declare function updateMixes(h: SoundHost): void;
+export declare function stopTimelineSound(h: SoundHost, sound: TimelineSound): void;
+/**
+ * AVM1's Sound.start: an event sound of `character` from `offset` ms, `loops`
+ * times, under `owner`'s transforms (none for the global Sound), whose end
+ * `onEnd` hears with the frame; null where it cannot play.
+ */
+export declare function startScriptSound(h: SoundHost, owner: DisplayObject | null, character: SoundCharacter, offset: number, loops: number, onEnd: () => void): TimelineSound | null;
+/** The sounds playing under exactly `owner`, as AVM1's Sound.stop() of a clip stops them; null for the global Sound's. */
+export declare function soundsOf(h: SoundHost, owner: DisplayObject | null): TimelineSound[];
+/** Every sound of `character` stops, whoever started it, as AVM1's Sound.stop(name) does. */
+export declare function stopSoundsOf(h: SoundHost, character: SoundCharacter): void;
+/** Whole frames by which a playing stream's device is ahead of its timeline. */
+export declare function streamFrames(h: SoundHost): number;
+/** unloadAndStop: the sounds of the timelines under `display` stop, its event sounds too. */
+export declare function stopTimelineSoundsUnder(s: Scripting, display: DisplayObject): void;
+/** What plays the timelines' sounds of every library a host loads. */
+export declare function timelineSoundsOf(h: SoundHost): TimelineSounds;
+/**
+ * The timeline's sounds that have played their time are over, with the
+ * frame: a NoMultiple may start one again, and AVM1 Sound's hears it.
+ * Streams remain attached until the timeline passes their blocks: the
+ * device finishing first must not restart one from a lagging frame.
+ */
+export declare function finishTimelineSounds(h: SoundHost): void;
+/** Sound completions run with the next SWF frame, even if the device ended between frames. */
+export declare function finishSounds(s: Scripting): void;
+export {};
+//# sourceMappingURL=sounds.d.ts.map

package/dist/media/sounds.d.ts.map

@@ -0,0 +1 @@
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