@gjsify/webrtc 0.4.0 → 0.4.4

package/src/rtc-rtp-transceiver.ts

@@ -1,131 +0,0 @@
-// W3C RTCRtpTransceiver for GJS.
-//
-// Wraps GstWebRTC.WebRTCRTPTransceiver. Reads mid, direction, currentDirection
-// from the native object. direction setter maps back to GStreamer enum.
-//
-// Reference: refs/node-gst-webrtc/src/webrtc/RTCRtpTransceiver.ts (ISC)
-// Reference: W3C WebRTC spec § 5.4
-
-import type GstWebRTC from 'gi://GstWebRTC?version=1.0';
-
-import { gstDirectionToW3C, w3cDirectionToGst } from './gst-enum-maps.js';
-import { RTCRtpSender, type RTCRtpTransceiverDirection, type RTCRtpCodecCapability } from './rtc-rtp-sender.js';
-import { RTCRtpReceiver } from './rtc-rtp-receiver.js';
-
-export class RTCRtpTransceiver {
-    private _gstTrans: GstWebRTC.WebRTCRTPTransceiver;
-    readonly sender: RTCRtpSender;
-    readonly receiver: RTCRtpReceiver;
-    private _stopped = false;
-    private _codecPreferences: RTCRtpCodecCapability[] = [];
-
-    constructor(
-        gstTrans: GstWebRTC.WebRTCRTPTransceiver,
-        sender: RTCRtpSender,
-        receiver: RTCRtpReceiver,
-    ) {
-        this._gstTrans = gstTrans;
-        this.sender = sender;
-        this.receiver = receiver;
-    }
-
-    get mid(): string | null {
-        if (this._stopped) return null;
-        const m = this._gstTrans.mid;
-        return (m === '' || m == null) ? null : String(m);
-    }
-
-    get direction(): RTCRtpTransceiverDirection {
-        if (this._stopped) return 'stopped';
-        return gstDirectionToW3C(this._gstTrans.direction);
-    }
-
-    set direction(d: RTCRtpTransceiverDirection) {
-        if (this._stopped) {
-            throw new DOMException(
-                "Cannot set direction on a stopped transceiver",
-                'InvalidStateError',
-            );
-        }
-        if (d === 'stopped') {
-            throw new TypeError("The provided value 'stopped' is not a valid enum value of type RTCRtpTransceiverDirection.");
-        }
-        const valid: RTCRtpTransceiverDirection[] = ['sendrecv', 'sendonly', 'recvonly', 'inactive'];
-        if (!valid.includes(d)) {
-            throw new TypeError(`The provided value '${d}' is not a valid enum value of type RTCRtpTransceiverDirection.`);
-        }
-        this._gstTrans.direction = w3cDirectionToGst(d);
-    }
-
-    get currentDirection(): RTCRtpTransceiverDirection | null {
-        if (this._stopped) return null;
-        // GIR exposes both snake_case `current_direction` and camelCase
-        // `currentDirection` getters — they refer to the same GObject
-        // property. Read snake_case first; fall back to camelCase for
-        // older GstWebRTC bindings that omitted the snake_case alias.
-        const cd = this._gstTrans.current_direction ?? this._gstTrans.currentDirection;
-        if (cd == null) return null;
-        const w3c = gstDirectionToW3C(cd);
-        return w3c === 'inactive' ? null : w3c;
-    }
-
-    get stopped(): boolean {
-        return this._stopped;
-    }
-
-    stop(): void {
-        if (this._stopped) return;
-        this._stopped = true;
-    }
-
-    setCodecPreferences(codecs: RTCRtpCodecCapability[]): void {
-        if (!Array.isArray(codecs)) {
-            throw new TypeError('codecs must be an array');
-        }
-        if (codecs.length === 0) {
-            this._codecPreferences = [];
-            return;
-        }
-
-        const kind = this.receiver.track.kind;
-        const recvCaps = RTCRtpReceiver.getCapabilities(kind);
-        const sendCaps = RTCRtpSender.getCapabilities(kind);
-        if (!recvCaps || !sendCaps) {
-            throw new DOMException('No capabilities available', 'InvalidModificationError');
-        }
-
-        const allCaps = [...recvCaps.codecs, ...sendCaps.codecs];
-
-        for (const codec of codecs) {
-            if (!codec || typeof codec !== 'object') {
-                throw new TypeError('Each codec must be an object');
-            }
-            if (typeof codec.mimeType !== 'string' || typeof codec.clockRate !== 'number') {
-                throw new TypeError('codec must have mimeType (string) and clockRate (number)');
-            }
-
-            const isResiliency = /\/(rtx|red|ulpfec)$/i.test(codec.mimeType);
-            if (isResiliency) continue;
-
-            const match = allCaps.find((c) =>
-                c.mimeType.toLowerCase() === codec.mimeType.toLowerCase() &&
-                c.clockRate === codec.clockRate &&
-                (codec.channels === undefined || c.channels === codec.channels) &&
-                (codec.sdpFmtpLine === undefined || c.sdpFmtpLine === codec.sdpFmtpLine)
-            );
-            if (!match) {
-                throw new DOMException(
-                    `Codec ${codec.mimeType} ${codec.clockRate} is not in capabilities`,
-                    'InvalidModificationError',
-                );
-            }
-        }
-
-        this._codecPreferences = [...codecs];
-    }
-
-    /** @internal */
-    get _nativeTransceiver(): GstWebRTC.WebRTCRTPTransceiver {
-        return this._gstTrans;
-    }
-}

package/src/rtc-sctp-transport.ts

@@ -1,48 +0,0 @@
-// W3C RTCSctpTransport for GJS.
-//
-// Created when a data channel is negotiated. Wraps the SCTP association
-// over the DTLS transport. State transitions from connecting → connected
-// when the first data channel opens.
-//
-// Reference: W3C WebRTC spec § 6.1
-// Reference: refs/wpt/webrtc/RTCSctpTransport-constructor.html
-
-import '@gjsify/dom-events/register/event-target';
-
-import { RTCDtlsTransport } from './rtc-dtls-transport.js';
-
-export type RTCSctpTransportState = 'connecting' | 'connected' | 'closed';
-
-type EventHandler = ((ev: Event) => void) | null;
-
-export class RTCSctpTransport extends EventTarget {
-    readonly transport: RTCDtlsTransport;
-    private _state: RTCSctpTransportState = 'connecting';
-    private _maxMessageSize: number = 262144; // 256 KB default per SCTP
-    private _maxChannels: number | null = 65535;
-
-    private _onstatechange: EventHandler = null;
-
-    constructor(dtlsTransport: RTCDtlsTransport) {
-        super();
-        this.transport = dtlsTransport;
-    }
-
-    get state(): RTCSctpTransportState { return this._state; }
-    get maxMessageSize(): number { return this._maxMessageSize; }
-    get maxChannels(): number | null { return this._maxChannels; }
-
-    get onstatechange(): EventHandler { return this._onstatechange; }
-    set onstatechange(v: EventHandler) { this._onstatechange = v; }
-
-    // ---- Internal setters (called by RTCPeerConnection) ---------------------
-
-    /** @internal */
-    _setState(state: RTCSctpTransportState): void {
-        if (this._state === state) return;
-        this._state = state;
-        const ev = new Event('statechange');
-        this._onstatechange?.call(this, ev);
-        this.dispatchEvent(ev);
-    }
-}

package/src/rtc-session-description.ts

@@ -1,64 +0,0 @@
-// RTCSessionDescription — W3C WebRTC session description (offer/answer/rollback).
-//
-// Reference: refs/node-gst-webrtc/src/webrtc/RTCSessionDescription.ts (ISC)
-// Adapted for GJS using GstSdp + GstWebRTC.
-
-import GstSdp from 'gi://GstSdp?version=1.0';
-import GstWebRTC from 'gi://GstWebRTC?version=1.0';
-
-export type RTCSdpType = 'offer' | 'pranswer' | 'answer' | 'rollback';
-
-export interface RTCSessionDescriptionInit {
-    type?: RTCSdpType;
-    sdp?: string;
-}
-
-function sdpTypeToGst(type: RTCSdpType): GstWebRTC.WebRTCSDPType {
-    switch (type) {
-        case 'offer':    return GstWebRTC.WebRTCSDPType.OFFER;
-        case 'pranswer': return GstWebRTC.WebRTCSDPType.PRANSWER;
-        case 'answer':   return GstWebRTC.WebRTCSDPType.ANSWER;
-        case 'rollback': return GstWebRTC.WebRTCSDPType.ROLLBACK;
-    }
-}
-
-function sdpTypeFromGst(type: GstWebRTC.WebRTCSDPType): RTCSdpType {
-    switch (type) {
-        case GstWebRTC.WebRTCSDPType.OFFER:    return 'offer';
-        case GstWebRTC.WebRTCSDPType.PRANSWER: return 'pranswer';
-        case GstWebRTC.WebRTCSDPType.ANSWER:   return 'answer';
-        case GstWebRTC.WebRTCSDPType.ROLLBACK: return 'rollback';
-        default: return 'offer';
-    }
-}
-
-export class RTCSessionDescription {
-    readonly type: RTCSdpType;
-    readonly sdp: string;
-
-    constructor(init?: RTCSessionDescriptionInit) {
-        this.type = (init?.type ?? 'offer') as RTCSdpType;
-        this.sdp = init?.sdp ?? '';
-    }
-
-    toJSON(): { type: RTCSdpType; sdp: string } {
-        return { type: this.type, sdp: this.sdp };
-    }
-
-    /** Build a GstWebRTC.WebRTCSessionDescription for use with webrtcbin signals. */
-    toGstDesc(): GstWebRTC.WebRTCSessionDescription {
-        const [ret, sdp] = GstSdp.SDPMessage.new_from_text(this.sdp);
-        if (ret !== GstSdp.SDPResult.OK) {
-            throw new Error(`Failed to parse SDP text (GstSDPResult=${ret})`);
-        }
-        return GstWebRTC.WebRTCSessionDescription.new(sdpTypeToGst(this.type), sdp);
-    }
-
-    static fromGstDesc(desc: GstWebRTC.WebRTCSessionDescription): RTCSessionDescription {
-        const sdpText = desc.sdp?.as_text?.() ?? '';
-        return new RTCSessionDescription({
-            type: sdpTypeFromGst(desc.type),
-            sdp: sdpText,
-        });
-    }
-}

package/src/rtc-stats-report.ts

@@ -1,39 +0,0 @@
-// W3C RTCStatsReport — read-only Map<string, object> of WebRTC statistics.
-//
-// Reference: W3C WebRTC spec § 8.5
-// The report is a frozen snapshot; .set()/.delete()/.clear() are no-ops.
-
-export interface RTCStats {
-    timestamp: number;
-    type: string;
-    id: string;
-    [key: string]: unknown;
-}
-
-/**
- * Read-only Map-like collection of stats entries keyed by id.
- * W3C requires iteration support but not mutation.
- */
-export class RTCStatsReport {
-    private readonly _map: Map<string, RTCStats>;
-
-    constructor(entries?: Iterable<[string, RTCStats]>) {
-        this._map = new Map(entries);
-    }
-
-    get size(): number { return this._map.size; }
-
-    get(key: string): RTCStats | undefined { return this._map.get(key); }
-    has(key: string): boolean { return this._map.has(key); }
-
-    forEach(callbackfn: (value: RTCStats, key: string, map: RTCStatsReport) => void, thisArg?: unknown): void {
-        this._map.forEach((value, key) => {
-            callbackfn.call(thisArg, value, key, this);
-        });
-    }
-
-    entries(): IterableIterator<[string, RTCStats]> { return this._map.entries(); }
-    keys(): IterableIterator<string> { return this._map.keys(); }
-    values(): IterableIterator<RTCStats> { return this._map.values(); }
-    [Symbol.iterator](): IterableIterator<[string, RTCStats]> { return this._map.entries(); }
-}

package/src/rtc-track-event.ts

@@ -1,45 +0,0 @@
-// W3C RTCTrackEvent for GJS.
-//
-// Ported from refs/wpt/webrtc/RTCTrackEvent-constructor.html (W3C, BSD-3-Clause)
-// Reference: W3C WebRTC spec § 5.7
-
-import '@gjsify/dom-events/register/event-target';
-
-import type { RTCRtpReceiver } from './rtc-rtp-receiver.js';
-import type { RTCRtpTransceiver } from './rtc-rtp-transceiver.js';
-import type { MediaStreamTrack } from './media-stream-track.js';
-import type { MediaStream } from './media-stream.js';
-
-export interface RTCTrackEventInit extends EventInit {
-    receiver: RTCRtpReceiver;
-    track: MediaStreamTrack;
-    streams?: MediaStream[];
-    transceiver: RTCRtpTransceiver;
-}
-
-export class RTCTrackEvent extends Event {
-    readonly receiver: RTCRtpReceiver;
-    readonly track: MediaStreamTrack;
-    readonly streams: ReadonlyArray<MediaStream>;
-    readonly transceiver: RTCRtpTransceiver;
-
-    constructor(type: string, init: RTCTrackEventInit) {
-        super(type, init);
-        if (!init || typeof init !== 'object') {
-            throw new TypeError('RTCTrackEventInit is required');
-        }
-        if (!('receiver' in init)) {
-            throw new TypeError("Failed to construct 'RTCTrackEvent': required member receiver is not provided.");
-        }
-        if (!('track' in init)) {
-            throw new TypeError("Failed to construct 'RTCTrackEvent': required member track is not provided.");
-        }
-        if (!('transceiver' in init)) {
-            throw new TypeError("Failed to construct 'RTCTrackEvent': required member transceiver is not provided.");
-        }
-        this.receiver = init.receiver;
-        this.track = init.track;
-        this.streams = Object.freeze(init.streams ? [...init.streams] : []);
-        this.transceiver = init.transceiver;
-    }
-}

package/src/rtp-capabilities.ts

@@ -1,48 +0,0 @@
-// Shared RTP codec capabilities for RTCRtpSender and RTCRtpReceiver.
-//
-// Both classes expose identical getCapabilities(kind) static methods per
-// W3C WebRTC spec §§ 5.2 and 5.3. This module deduplicates the data.
-
-import type { RTCRtpCapabilities } from './rtc-rtp-sender.js';
-
-const AUDIO_CAPABILITIES: RTCRtpCapabilities = {
-    codecs: [
-        { mimeType: 'audio/opus', clockRate: 48000, channels: 2, sdpFmtpLine: 'minptime=10;useinbandfec=1' },
-        { mimeType: 'audio/G722', clockRate: 8000, channels: 1 },
-        { mimeType: 'audio/PCMU', clockRate: 8000, channels: 1 },
-        { mimeType: 'audio/PCMA', clockRate: 8000, channels: 1 },
-        { mimeType: 'audio/telephone-event', clockRate: 8000, channels: 1 },
-        { mimeType: 'audio/red', clockRate: 48000, channels: 2 },
-    ],
-    headerExtensions: [
-        { uri: 'urn:ietf:params:rtp-hdrext:ssrc-audio-level' },
-        { uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time' },
-        { uri: 'urn:ietf:params:rtp-hdrext:sdes:mid' },
-    ],
-};
-
-const VIDEO_CAPABILITIES: RTCRtpCapabilities = {
-    codecs: [
-        { mimeType: 'video/VP8', clockRate: 90000 },
-        { mimeType: 'video/rtx', clockRate: 90000 },
-        { mimeType: 'video/H264', clockRate: 90000, sdpFmtpLine: 'level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f' },
-        { mimeType: 'video/VP9', clockRate: 90000 },
-        { mimeType: 'video/red', clockRate: 90000 },
-        { mimeType: 'video/ulpfec', clockRate: 90000 },
-    ],
-    headerExtensions: [
-        { uri: 'urn:ietf:params:rtp-hdrext:toffset' },
-        { uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time' },
-        { uri: 'urn:3gpp:video-orientation' },
-        { uri: 'urn:ietf:params:rtp-hdrext:sdes:mid' },
-        { uri: 'urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id' },
-        { uri: 'urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id' },
-    ],
-};
-
-/** Shared implementation for RTCRtpSender.getCapabilities / RTCRtpReceiver.getCapabilities. */
-export function getRtpCapabilities(kind: string): RTCRtpCapabilities | null {
-    if (kind === 'audio') return AUDIO_CAPABILITIES;
-    if (kind === 'video') return VIDEO_CAPABILITIES;
-    return null;
-}

package/src/tee-multiplexer.ts

@@ -1,75 +0,0 @@
-// GStreamer tee element manager for multi-PC fan-out.
-//
-// When a single MediaStreamTrack (backed by a GStreamer source) needs to
-// feed multiple RTCPeerConnections, a tee element is inserted after the
-// source. Each PC gets its own branch (encoder chain → webrtcbin sink).
-//
-// Reference: refs/node-gst-webrtc/src/media/TeeMultiplexer.ts (ISC)
-// Reference: packages/web/webrtc-native/src/vala/receiver-bridge.vala (tee pattern)
-
-import { Gst } from './gst-init.js';
-
-/**
- * Manages a GStreamer `tee` element that fans out one source to multiple
- * consumer branches. Each branch gets its own src pad from the tee.
- */
-export class TeeMultiplexer {
-    private _tee: any; // Gst.Element
-    private _pipeline: any; // Gst.Pipeline
-    private _branchCount = 0;
-
-    /**
-     * Create a tee in the given pipeline and link it to the source's output.
-     * The source must already be in the pipeline.
-     */
-    constructor(pipeline: any, source: any) {
-        this._pipeline = pipeline;
-        this._tee = Gst.ElementFactory.make('tee', null)!;
-        (this._tee as any).allow_not_linked = true;
-
-        pipeline.add(this._tee);
-        this._tee.sync_state_with_parent();
-
-        // Link source → tee
-        source.link(this._tee);
-    }
-
-    /** Request a new src pad from the tee for a consumer branch. */
-    requestSrcPad(): any /* Gst.Pad */ {
-        const padName = 'src_%u';
-        const srcPad = this._tee.request_pad_simple
-            ? this._tee.request_pad_simple(padName)
-            : this._tee.get_request_pad(padName);
-        if (srcPad) this._branchCount++;
-        return srcPad;
-    }
-
-    /**
-     * Release a branch's src pad from the tee.
-     * Adds a DROP probe before unlinking to prevent errors.
-     */
-    releaseSrcPad(srcPad: any): void {
-        if (!srcPad) return;
-        try {
-            // Block data on the pad before unlinking
-            srcPad.add_probe(
-                Gst.PadProbeType.BLOCK_DOWNSTREAM,
-                () => Gst.PadProbeReturn.DROP,
-            );
-        } catch { /* ignore if probe fails */ }
-        try {
-            const peer = srcPad.get_peer();
-            if (peer) srcPad.unlink(peer);
-        } catch { /* ignore */ }
-        try {
-            this._tee.release_request_pad(srcPad);
-        } catch { /* ignore */ }
-        this._branchCount--;
-    }
-
-    /** Number of active branches. */
-    get branchCount(): number { return this._branchCount; }
-
-    /** The tee element (for pipeline queries). */
-    get element(): any { return this._tee; }
-}

package/src/test.mts

@@ -1,11 +0,0 @@
-import webrtcSpec from './webrtc.spec.js';
-import wptSpec from './wpt.spec.js';
-import wptMediaSpec from './wpt-media.spec.js';
-import registerSpec from './register.spec.js';
-
-const results = {
-    webrtc: await webrtcSpec(),
-    wpt: await wptSpec(),
-    wptMedia: await wptMediaSpec(),
-    register: await registerSpec(),
-};