@gjsify/webaudio 0.3.21 → 0.4.3

package/src/gst-decoder.ts

@@ -1,109 +0,0 @@
-// GStreamer decode pipeline: ArrayBuffer (MP3/WAV/OGG) → AudioBuffer (PCM Float32)
-//
-// Pipeline: appsrc → decodebin → audioconvert → audioresample → capsfilter(F32LE) → appsink
-// Uses try_pull_sample() for synchronous decoding (avoids GJS thread-safety issues).
-//
-// Reference: GStreamer 1.0 via gi://Gst, GstApp via gi://GstApp
-
-import { ensureGstInit, Gst } from './gst-init.js';
-import { AudioBuffer } from './audio-buffer.js';
-
-// Force GstApp typelib load so get_by_name() resolves AppSrc/AppSink types
-import GstApp from 'gi://GstApp?version=1.0';
-void GstApp;
-
-const PIPELINE_DESC =
-    'appsrc name=src ! decodebin ! audioconvert ! audioresample ! ' +
-    'capsfilter caps=audio/x-raw,format=F32LE,layout=interleaved ! ' +
-    'appsink name=sink sync=false';
-
-/**
- * Decode encoded audio data (MP3, WAV, OGG, FLAC, etc.) into an AudioBuffer
- * containing PCM Float32 channel data.
- *
- * This is a synchronous operation that blocks until decoding completes.
- * It must be called from the main thread (GJS requirement).
- */
-export function decodeAudioDataSync(arrayBuffer: ArrayBuffer): AudioBuffer {
-    ensureGstInit();
-
-    // Reject non-ArrayBuffer / empty input before touching GStreamer —
-    // gst_memory_new_wrapped() asserts data != NULL and empty TypedArrays
-    // marshal to a NULL pointer through GI.
-    if (!(arrayBuffer instanceof ArrayBuffer) || arrayBuffer.byteLength === 0) {
-        throw new DOMException('Unable to decode audio data', 'EncodingError');
-    }
-
-    const pipeline = Gst.parse_launch(PIPELINE_DESC) as Gst.Bin;
-    const appsrc = pipeline.get_by_name('src')!;
-    const appsink = pipeline.get_by_name('sink')!;
-
-    pipeline.set_state(Gst.State.PLAYING);
-
-    // Push encoded data into the pipeline
-    const data = new Uint8Array(arrayBuffer);
-    (appsrc as any).push_buffer(Gst.Buffer.new_wrapped(data));
-    (appsrc as any).end_of_stream();
-
-    // Pull decoded PCM samples
-    const chunks: Uint8Array[] = [];
-    let sampleRate = 0;
-    let channels = 0;
-
-    while (true) {
-        const sample = (appsink as any).try_pull_sample(2 * Number(Gst.SECOND));
-        if (!sample) break;
-
-        // Read format from the first sample's negotiated caps
-        if (sampleRate === 0) {
-            const caps = sample.get_caps();
-            if (caps) {
-                const struct = caps.get_structure(0);
-                [, sampleRate] = struct.get_int('rate');
-                [, channels] = struct.get_int('channels');
-            }
-        }
-
-        const buffer = sample.get_buffer();
-        if (!buffer) continue;
-
-        const [ok, mapInfo] = buffer.map(Gst.MapFlags.READ);
-        if (ok) {
-            // Copy data — mapInfo.data is only valid until unmap
-            chunks.push(new Uint8Array(mapInfo.data));
-            buffer.unmap(mapInfo);
-        }
-    }
-
-    pipeline.set_state(Gst.State.NULL);
-
-    if (sampleRate === 0 || channels === 0) {
-        throw new DOMException('Unable to decode audio data', 'EncodingError');
-    }
-
-    // Concatenate chunks into a single interleaved Float32 buffer
-    let totalBytes = 0;
-    for (const c of chunks) totalBytes += c.length;
-    const totalFrames = totalBytes / (4 * channels);
-
-    const audioBuffer = new AudioBuffer({
-        numberOfChannels: channels,
-        length: totalFrames,
-        sampleRate,
-    });
-
-    // De-interleave into per-channel Float32Arrays
-    let offset = 0;
-    for (const chunk of chunks) {
-        const f32 = new Float32Array(chunk.buffer, chunk.byteOffset, chunk.length / 4);
-        const framesInChunk = f32.length / channels;
-        for (let frame = 0; frame < framesInChunk; frame++) {
-            for (let ch = 0; ch < channels; ch++) {
-                audioBuffer._channelData[ch][offset + frame] = f32[frame * channels + ch];
-            }
-        }
-        offset += framesInChunk;
-    }
-
-    return audioBuffer;
-}

package/src/gst-init.ts

@@ -1,15 +0,0 @@
-// Lazy GStreamer initialization — call ensureGstInit() before any Gst API usage.
-// Reference: GStreamer 1.0 via gi://Gst
-
-import Gst from 'gi://Gst?version=1.0';
-
-let initialized = false;
-
-export function ensureGstInit(): void {
-    if (!initialized) {
-        Gst.init(null);
-        initialized = true;
-    }
-}
-
-export { Gst };

package/src/gst-player.ts

@@ -1,185 +0,0 @@
-// GStreamer playback pipeline: AudioBuffer PCM → audio output
-//
-// Pipeline: appsrc(F32LE) → audioconvert → volume → autoaudiosink
-// Each GstPlayer instance is single-use (matches W3C AudioBufferSourceNode).
-//
-// Reference: GStreamer 1.0 via gi://Gst
-
-import GLib from 'gi://GLib?version=2.0';
-import { ensureGstInit, Gst } from './gst-init.js';
-import type { AudioBuffer } from './audio-buffer.js';
-
-// Force GstApp typelib load
-import GstApp from 'gi://GstApp?version=1.0';
-void GstApp;
-
-export interface GstPlayerOptions {
-    audioBuffer: AudioBuffer;
-    volume: number;
-    loop: boolean;
-    offset: number;      // start offset in seconds
-    duration?: number;    // play duration in seconds (undefined = full)
-    playbackRate: number;
-    onEnded: () => void;
-}
-
-/**
- * Manages a single GStreamer playback pipeline for one AudioBufferSourceNode.start() call.
- */
-export class GstPlayer {
-    private _pipeline: any = null;
-    private _volumeElement: any = null;
-    private _busWatchId: number | null = null;
-    private _ended = false;
-    private _loop: boolean;
-    private _onEnded: () => void;
-    private _audioBuffer: AudioBuffer;
-
-    constructor(options: GstPlayerOptions) {
-        ensureGstInit();
-        this._loop = options.loop;
-        this._onEnded = options.onEnded;
-        this._audioBuffer = options.audioBuffer;
-
-        const { audioBuffer, volume, offset, duration, playbackRate } = options;
-        const sr = audioBuffer.sampleRate;
-        const ch = audioBuffer.numberOfChannels;
-
-        // Build interleaved PCM data
-        const pcmData = this._interleave(audioBuffer, offset, duration);
-        if (pcmData.length === 0) {
-            // Empty buffer — fire ended immediately
-            this._fireEnded();
-            return;
-        }
-
-        // Build pipeline — format=3 (TIME) ensures downstream gets TIME-based
-        // segments, preventing gst_segment_to_stream_time assertion failures.
-        const capsStr = `audio/x-raw,format=F32LE,rate=${sr},channels=${ch},layout=interleaved`;
-        const desc = `appsrc name=src caps="${capsStr}" format=3 ! audioconvert ! volume name=vol ! autoaudiosink`;
-        this._pipeline = Gst.parse_launch(desc);
-        this._volumeElement = this._pipeline.get_by_name('vol');
-        const appsrc = this._pipeline.get_by_name('src')!;
-
-        // Set volume
-        this._volumeElement.set_property('volume', Math.max(0, Math.min(volume, 10)));
-
-        // Set up bus watch for EOS/ERROR
-        const bus = this._pipeline.get_bus();
-        this._busWatchId = bus.add_watch(0, (_bus: any, msg: any) => {
-            if (msg.type === Gst.MessageType.EOS) {
-                if (this._loop && !this._ended) {
-                    // Restart: push data again
-                    this._restartPlayback(appsrc, pcmData);
-                } else {
-                    this._fireEnded();
-                }
-            } else if (msg.type === Gst.MessageType.ERROR) {
-                this._fireEnded();
-            }
-            return true; // keep watching
-        });
-
-        // Push PCM data with proper timestamps for TIME-format segments
-        const gstBuf = Gst.Buffer.new_wrapped(pcmData);
-        const totalFrames = pcmData.length / (4 * ch);
-        gstBuf.pts = 0;
-        gstBuf.duration = Math.floor((totalFrames / sr) * Number(Gst.SECOND));
-        appsrc.push_buffer(gstBuf);
-        appsrc.end_of_stream();
-
-        // Apply playback rate if not 1.0
-        if (playbackRate !== 1.0) {
-            this._pipeline.set_state(Gst.State.PAUSED);
-            // Seek with rate change
-            this._pipeline.seek(
-                playbackRate,
-                Gst.Format.TIME,
-                Gst.SeekFlags.FLUSH | Gst.SeekFlags.ACCURATE,
-                Gst.SeekType.SET, 0,
-                Gst.SeekType.NONE, -1
-            );
-        }
-
-        this._pipeline.set_state(Gst.State.PLAYING);
-    }
-
-    /** Update volume on a running pipeline */
-    setVolume(value: number): void {
-        if (this._volumeElement && !this._ended) {
-            this._volumeElement.set_property('volume', Math.max(0, Math.min(value, 10)));
-        }
-    }
-
-    /** Update loop flag */
-    setLoop(value: boolean): void {
-        this._loop = value;
-    }
-
-    /** Stop playback and clean up */
-    stop(): void {
-        if (this._ended) return;
-        this._fireEnded();
-    }
-
-    /** Whether playback has ended */
-    get ended(): boolean {
-        return this._ended;
-    }
-
-    private _restartPlayback(appsrc: any, pcmData: Uint8Array): void {
-        // For looping: seek pipeline to start
-        if (this._pipeline) {
-            this._pipeline.seek_simple(Gst.Format.TIME, Gst.SeekFlags.FLUSH, 0);
-        }
-    }
-
-    private _fireEnded(): void {
-        if (this._ended) return;
-        this._ended = true;
-        this._cleanup();
-        this._onEnded();
-    }
-
-    private _cleanup(): void {
-        const pipeline = this._pipeline;
-        this._pipeline = null;
-        this._volumeElement = null;
-        this._busWatchId = null;
-
-        if (pipeline) {
-            // Defer NULL-state transition to a low-priority idle so this method does not
-            // block the GLib main loop when called from within the bus-watch callback (EOS/ERROR).
-            // autoaudiosink teardown flushes the audio device, which can take several ms and
-            // causes frame drops when SFX fire frequently during gameplay.
-            GLib.idle_add(GLib.PRIORITY_LOW, () => {
-                pipeline.set_state(Gst.State.NULL);
-                return GLib.SOURCE_REMOVE;
-            });
-        }
-    }
-
-    /**
-     * Interleave AudioBuffer's per-channel Float32Arrays into a single Uint8Array.
-     * Applies offset (seconds) and optional duration (seconds).
-     */
-    private _interleave(buf: AudioBuffer, offsetSec: number, durationSec?: number): Uint8Array {
-        const ch = buf.numberOfChannels;
-        const startFrame = Math.min(Math.floor(offsetSec * buf.sampleRate), buf.length);
-        const maxFrames = buf.length - startFrame;
-        const frames = durationSec !== undefined
-            ? Math.min(Math.floor(durationSec * buf.sampleRate), maxFrames)
-            : maxFrames;
-
-        if (frames <= 0) return new Uint8Array(0);
-
-        const interleaved = new Float32Array(frames * ch);
-        for (let frame = 0; frame < frames; frame++) {
-            for (let c = 0; c < ch; c++) {
-                interleaved[frame * ch + c] = buf._channelData[c][startFrame + frame];
-            }
-        }
-
-        return new Uint8Array(interleaved.buffer);
-    }
-}

package/src/html-audio-element.ts

@@ -1,76 +0,0 @@
-// HTMLAudioElement — format detection + basic playback via GStreamer playbin.
-// Used by Excalibur.js for canPlayType() format sniffing.
-//
-// Reference: https://developer.mozilla.org/en-US/docs/Web/API/HTMLAudioElement
-
-import { ensureGstInit, Gst } from './gst-init.js';
-
-// GStreamer-supported MIME types (common on GNOME systems)
-const SUPPORTED_TYPES = new Set([
-    'audio/mpeg',
-    'audio/mp3',
-    'audio/wav',
-    'audio/x-wav',
-    'audio/ogg',
-    'audio/webm',
-    'audio/flac',
-    'audio/x-flac',
-    'audio/aac',
-    'audio/mp4',
-]);
-
-export class HTMLAudioElement {
-    src = '';
-    volume = 1;
-    loop = false;
-    paused = true;
-    currentTime = 0;
-    duration = 0;
-    readyState = 0;
-
-    private _pipeline: any = null;
-
-    canPlayType(type: string): CanPlayTypeResult {
-        // Strip codecs parameter: "audio/ogg; codecs=vorbis" → "audio/ogg"
-        const mime = type.split(';')[0].trim().toLowerCase();
-        return SUPPORTED_TYPES.has(mime) ? 'maybe' : '';
-    }
-
-    play(): Promise<void> {
-        if (!this.src) return Promise.resolve();
-
-        ensureGstInit();
-        this._cleanup();
-
-        this._pipeline = Gst.ElementFactory.make('playbin', 'player');
-        if (!this._pipeline) return Promise.resolve();
-
-        this._pipeline.set_property('uri', this.src);
-        this._pipeline.set_property('volume', this.volume);
-        this._pipeline.set_state(Gst.State.PLAYING);
-        this.paused = false;
-
-        return Promise.resolve();
-    }
-
-    pause(): void {
-        if (this._pipeline) {
-            this._pipeline.set_state(Gst.State.PAUSED);
-            this.paused = true;
-        }
-    }
-
-    load(): void {
-        this._cleanup();
-    }
-
-    addEventListener(_type: string, _listener: any): void {}
-    removeEventListener(_type: string, _listener: any): void {}
-
-    private _cleanup(): void {
-        if (this._pipeline) {
-            this._pipeline.set_state(Gst.State.NULL);
-            this._pipeline = null;
-        }
-    }
-}

package/src/index.ts

@@ -1,14 +0,0 @@
-// Web Audio API for GJS — backed by GStreamer 1.0.
-//
-// This module has no side effects. Importing @gjsify/webaudio gives
-// named access to Web Audio classes but does NOT register globals.
-// Use @gjsify/webaudio/register to set globalThis.AudioContext, etc.
-
-export { AudioContext } from './audio-context.js';
-export { AudioBuffer } from './audio-buffer.js';
-export { AudioNode } from './audio-node.js';
-export { AudioDestinationNode } from './audio-destination-node.js';
-export { AudioBufferSourceNode } from './audio-buffer-source-node.js';
-export { GainNode } from './gain-node.js';
-export { AudioParam } from './audio-param.js';
-export { HTMLAudioElement } from './html-audio-element.js';

package/src/register.ts

@@ -1,17 +0,0 @@
-// Side-effect module: registers Web Audio API globals on GJS.
-// On Node.js the alias layer routes this to @gjsify/empty.
-
-import { AudioContext, HTMLAudioElement } from './index.js';
-
-if (typeof (globalThis as any).AudioContext === 'undefined') {
-    (globalThis as any).AudioContext = AudioContext;
-}
-if (typeof (globalThis as any).webkitAudioContext === 'undefined') {
-    (globalThis as any).webkitAudioContext = AudioContext;
-}
-if (typeof (globalThis as any).Audio === 'undefined') {
-    (globalThis as any).Audio = HTMLAudioElement;
-}
-if (typeof (globalThis as any).HTMLAudioElement === 'undefined') {
-    (globalThis as any).HTMLAudioElement = HTMLAudioElement;
-}

package/src/test.mts

@@ -1,5 +0,0 @@
-import webaudioSpec from './webaudio.spec.js';
-
-const results = {
-    webaudio: await webaudioSpec(),
-};