@gjsify/webrtc 0.4.0 → 0.4.3

package/package.json

@@ -1,75 +1,78 @@
 {
-  "name": "@gjsify/webrtc",
-  "version": "0.4.0",
-  "description": "W3C WebRTC API for GJS using GStreamer webrtcbin as the peer-connection backend",
-  "type": "module",
-  "module": "lib/esm/index.js",
-  "types": "lib/types/index.d.ts",
-  "exports": {
-    ".": {
-      "types": "./lib/types/index.d.ts",
-      "default": "./lib/esm/index.js"
+    "name": "@gjsify/webrtc",
+    "version": "0.4.3",
+    "description": "W3C WebRTC API for GJS using GStreamer webrtcbin as the peer-connection backend",
+    "type": "module",
+    "module": "lib/esm/index.js",
+    "types": "lib/types/index.d.ts",
+    "exports": {
+        ".": {
+            "types": "./lib/types/index.d.ts",
+            "default": "./lib/esm/index.js"
+        },
+        "./register": {
+            "types": "./lib/types/register.d.ts",
+            "default": "./lib/esm/register.js"
+        },
+        "./register/peer-connection": {
+            "default": "./lib/esm/register/peer-connection.js"
+        },
+        "./register/data-channel": {
+            "default": "./lib/esm/register/data-channel.js"
+        },
+        "./register/error": {
+            "default": "./lib/esm/register/error.js"
+        },
+        "./register/media": {
+            "default": "./lib/esm/register/media.js"
+        },
+        "./register/media-devices": {
+            "default": "./lib/esm/register/media-devices.js"
+        }
     },
-    "./register": {
-      "types": "./lib/types/register.d.ts",
-      "default": "./lib/esm/register.js"
+    "files": [
+        "lib"
+    ],
+    "sideEffects": [
+        "./lib/esm/register.js",
+        "./lib/esm/register/*.js"
+    ],
+    "scripts": {
+        "clear": "rm -rf lib tsconfig.tsbuildinfo tsconfig.types.tsbuildinfo test.gjs.mjs || exit 0",
+        "check": "tsc --noEmit",
+        "build": "gjsify run build:gjsify && gjsify run build:types",
+        "build:gjsify": "gjsify build --library 'src/**/*.{ts,js}' --exclude 'src/**/*.spec.{mts,ts}' 'src/test.{mts,ts}'",
+        "build:types": "tsc",
+        "build:test": "gjsify run build:test:gjs",
+        "build:test:gjs": "gjsify build src/test.mts --app gjs --outfile test.gjs.mjs",
+        "test": "gjsify run build:gjsify && gjsify run build:test && gjsify run test:gjs",
+        "test:gjs": "gjsify run test.gjs.mjs"
     },
-    "./register/peer-connection": {
-      "default": "./lib/esm/register/peer-connection.js"
+    "keywords": [
+        "gjs",
+        "webrtc",
+        "gstreamer",
+        "webrtcbin",
+        "rtc",
+        "data-channel",
+        "peer-connection"
+    ],
+    "dependencies": {
+        "@gjsify/buffer": "workspace:^",
+        "@gjsify/dom-events": "workspace:^",
+        "@gjsify/dom-exception": "workspace:^",
+        "@gjsify/webrtc-native": "workspace:^"
     },
-    "./register/data-channel": {
-      "default": "./lib/esm/register/data-channel.js"
-    },
-    "./register/error": {
-      "default": "./lib/esm/register/error.js"
-    },
-    "./register/media": {
-      "default": "./lib/esm/register/media.js"
-    },
-    "./register/media-devices": {
-      "default": "./lib/esm/register/media-devices.js"
+    "devDependencies": {
+        "@girs/gjs": "4.0.0-rc.15",
+        "@girs/glib-2.0": "2.88.0-4.0.0-rc.15",
+        "@girs/gobject-2.0": "2.88.0-4.0.0-rc.15",
+        "@girs/gst-1.0": "1.28.1-4.0.0-rc.15",
+        "@girs/gstsdp-1.0": "1.0.0-4.0.0-rc.15",
+        "@girs/gstwebrtc-1.0": "1.0.0-4.0.0-rc.15",
+        "@gjsify/cli": "workspace:^",
+        "@gjsify/unit": "workspace:^",
+        "@types/node": "^25.6.2",
+        "typescript": "^6.0.3"
     }
-  },
-  "sideEffects": [
-    "./lib/esm/register.js",
-    "./lib/esm/register/*.js"
-  ],
-  "scripts": {
-    "clear": "rm -rf lib tsconfig.tsbuildinfo tsconfig.types.tsbuildinfo test.gjs.mjs || exit 0",
-    "check": "tsc --noEmit",
-    "build": "yarn build:gjsify && yarn build:types",
-    "build:gjsify": "gjsify build --library 'src/**/*.{ts,js}' --exclude 'src/**/*.spec.{mts,ts}' 'src/test.{mts,ts}'",
-    "build:types": "tsc",
-    "build:test": "yarn build:test:gjs",
-    "build:test:gjs": "gjsify build src/test.mts --app gjs --outfile test.gjs.mjs",
-    "test": "yarn build:gjsify && yarn build:test && yarn test:gjs",
-    "test:gjs": "gjsify run test.gjs.mjs"
-  },
-  "keywords": [
-    "gjs",
-    "webrtc",
-    "gstreamer",
-    "webrtcbin",
-    "rtc",
-    "data-channel",
-    "peer-connection"
-  ],
-  "dependencies": {
-    "@gjsify/buffer": "^0.4.0",
-    "@gjsify/dom-events": "^0.4.0",
-    "@gjsify/dom-exception": "^0.4.0",
-    "@gjsify/webrtc-native": "^0.4.0"
-  },
-  "devDependencies": {
-    "@girs/gjs": "4.0.0-rc.15",
-    "@girs/glib-2.0": "2.88.0-4.0.0-rc.15",
-    "@girs/gobject-2.0": "2.88.0-4.0.0-rc.15",
-    "@girs/gst-1.0": "1.28.1-4.0.0-rc.15",
-    "@girs/gstsdp-1.0": "1.0.0-4.0.0-rc.15",
-    "@girs/gstwebrtc-1.0": "1.0.0-4.0.0-rc.15",
-    "@gjsify/cli": "^0.4.0",
-    "@gjsify/unit": "^0.4.0",
-    "@types/node": "^25.6.2",
-    "typescript": "^6.0.3"
-  }
-}
+}

package/src/get-user-media.ts

@@ -1,131 +0,0 @@
-// getUserMedia for GJS — wraps GStreamer source elements as MediaStreamTracks.
-//
-// Phase 3: basic media capture. Tries real audio/video sources first
-// (pipewiresrc, pulsesrc, v4l2src), falls back to test sources.
-// Phase 4.3: constraint support — width, height, frameRate, sampleRate,
-// channelCount mapped to GStreamer capsfilter elements.
-//
-// Reference: W3C Media Capture and Streams spec § 10.3
-
-import { Gst, ensureGstInit } from './gst-init.js';
-import { MediaStreamTrack } from './media-stream-track.js';
-import { MediaStream } from './media-stream.js';
-
-export interface MediaTrackConstraints {
-    deviceId?: string;
-    sampleRate?: number;
-    channelCount?: number;
-    width?: number;
-    height?: number;
-    frameRate?: number;
-}
-
-export interface MediaStreamConstraints {
-    audio?: boolean | MediaTrackConstraints;
-    video?: boolean | MediaTrackConstraints;
-}
-
-export async function getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream> {
-    ensureGstInit();
-
-    if (!constraints.audio && !constraints.video) {
-        throw new TypeError(
-            "Failed to execute 'getUserMedia': At least one of audio or video must be requested",
-        );
-    }
-
-    const tracks: MediaStreamTrack[] = [];
-
-    if (constraints.audio) {
-        const audioConstraints = typeof constraints.audio === 'object' ? constraints.audio : {};
-        const source = _createAudioSource();
-        const pipeline = new Gst.Pipeline() as any;
-        pipeline.add(source);
-
-        // Apply audio constraints via capsfilter
-        const capsStr = _buildAudioCaps(audioConstraints);
-        if (capsStr) {
-            const capsfilter = Gst.ElementFactory.make('capsfilter', null)!;
-            (capsfilter as any).caps = Gst.Caps.from_string(capsStr);
-            pipeline.add(capsfilter);
-            source.link(capsfilter);
-        }
-
-        tracks.push(new MediaStreamTrack({
-            kind: 'audio',
-            label: source.name ?? 'audio',
-            _gst: { source, pipeline },
-        }));
-    }
-
-    if (constraints.video) {
-        const videoConstraints = typeof constraints.video === 'object' ? constraints.video : {};
-        const source = _createVideoSource();
-        const pipeline = new Gst.Pipeline() as any;
-        pipeline.add(source);
-
-        // Apply video constraints via capsfilter
-        const capsStr = _buildVideoCaps(videoConstraints);
-        if (capsStr) {
-            const capsfilter = Gst.ElementFactory.make('capsfilter', null)!;
-            (capsfilter as any).caps = Gst.Caps.from_string(capsStr);
-            pipeline.add(capsfilter);
-            source.link(capsfilter);
-        }
-
-        tracks.push(new MediaStreamTrack({
-            kind: 'video',
-            label: source.name ?? 'video',
-            _gst: { source, pipeline },
-        }));
-    }
-
-    return new MediaStream(tracks);
-}
-
-/** Build a GStreamer caps string for audio constraints. */
-function _buildAudioCaps(c: MediaTrackConstraints): string | null {
-    const parts: string[] = [];
-    if (c.sampleRate != null) parts.push(`rate=${Math.trunc(c.sampleRate)}`);
-    if (c.channelCount != null) parts.push(`channels=${Math.trunc(c.channelCount)}`);
-    if (parts.length === 0) return null;
-    return `audio/x-raw,${parts.join(',')}`;
-}
-
-/** Build a GStreamer caps string for video constraints. */
-function _buildVideoCaps(c: MediaTrackConstraints): string | null {
-    const parts: string[] = [];
-    if (c.width != null) parts.push(`width=${Math.trunc(c.width)}`);
-    if (c.height != null) parts.push(`height=${Math.trunc(c.height)}`);
-    if (c.frameRate != null) parts.push(`framerate=${Math.trunc(c.frameRate)}/1`);
-    if (parts.length === 0) return null;
-    return `video/x-raw,${parts.join(',')}`;
-}
-
-function _createAudioSource(): any {
-    // Try real sources in priority order
-    for (const name of ['pipewiresrc', 'pulsesrc', 'autoaudiosrc']) {
-        const el = Gst.ElementFactory.make(name, null);
-        if (el) {
-            try { (el as any).is_live = true; } catch { /* not all sources have is-live */ }
-            return el;
-        }
-    }
-    // Fallback: test source (sine wave, audible for debugging)
-    const el = Gst.ElementFactory.make('audiotestsrc', null)!;
-    (el as any).is_live = true;
-    (el as any).wave = 0; // sine
-    return el;
-}
-
-function _createVideoSource(): any {
-    for (const name of ['pipewiresrc', 'v4l2src', 'autovideosrc']) {
-        const el = Gst.ElementFactory.make(name, null);
-        if (el) return el;
-    }
-    // Fallback: test pattern
-    const el = Gst.ElementFactory.make('videotestsrc', null)!;
-    (el as any).is_live = true;
-    (el as any).pattern = 0; // SMPTE bars
-    return el;
-}

package/src/gst-enum-maps.ts

@@ -1,125 +0,0 @@
-// GStreamer ↔ W3C enum conversions for WebRTC types.
-//
-// Centralises the bidirectional mapping between GstWebRTC C enums and
-// W3C string literals used across RTCPeerConnection and RTCRtpTransceiver.
-
-import GstWebRTC from 'gi://GstWebRTC?version=1.0';
-
-import type {
-    RTCSignalingState,
-    RTCPeerConnectionState,
-    RTCIceConnectionState,
-    RTCIceGatheringState,
-} from './rtc-peer-connection.js';
-import type { RTCRtpTransceiverDirection } from './rtc-rtp-sender.js';
-
-// ---- Signaling state --------------------------------------------------------
-
-const SIGNALING_STATE_MAP: Record<number, RTCSignalingState> = {
-    [GstWebRTC.WebRTCSignalingState.STABLE]: 'stable',
-    [GstWebRTC.WebRTCSignalingState.CLOSED]: 'closed',
-    [GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER]: 'have-local-offer',
-    [GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER]: 'have-remote-offer',
-    [GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER]: 'have-local-pranswer',
-    [GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER]: 'have-remote-pranswer',
-};
-
-export function gstToSignalingState(v: number): RTCSignalingState {
-    return SIGNALING_STATE_MAP[v] ?? 'stable';
-}
-
-// ---- Connection state -------------------------------------------------------
-
-const CONNECTION_STATE_MAP: Record<number, RTCPeerConnectionState> = {
-    [GstWebRTC.WebRTCPeerConnectionState.NEW]: 'new',
-    [GstWebRTC.WebRTCPeerConnectionState.CONNECTING]: 'connecting',
-    [GstWebRTC.WebRTCPeerConnectionState.CONNECTED]: 'connected',
-    [GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED]: 'disconnected',
-    [GstWebRTC.WebRTCPeerConnectionState.FAILED]: 'failed',
-    [GstWebRTC.WebRTCPeerConnectionState.CLOSED]: 'closed',
-};
-
-export function gstToConnectionState(v: number): RTCPeerConnectionState {
-    return CONNECTION_STATE_MAP[v] ?? 'new';
-}
-
-// ---- ICE connection state ---------------------------------------------------
-
-const ICE_CONNECTION_STATE_MAP: Record<number, RTCIceConnectionState> = {
-    [GstWebRTC.WebRTCICEConnectionState.NEW]: 'new',
-    [GstWebRTC.WebRTCICEConnectionState.CHECKING]: 'checking',
-    [GstWebRTC.WebRTCICEConnectionState.CONNECTED]: 'connected',
-    [GstWebRTC.WebRTCICEConnectionState.COMPLETED]: 'completed',
-    [GstWebRTC.WebRTCICEConnectionState.FAILED]: 'failed',
-    [GstWebRTC.WebRTCICEConnectionState.DISCONNECTED]: 'disconnected',
-    [GstWebRTC.WebRTCICEConnectionState.CLOSED]: 'closed',
-};
-
-export function gstToIceConnectionState(v: number): RTCIceConnectionState {
-    return ICE_CONNECTION_STATE_MAP[v] ?? 'new';
-}
-
-// ---- ICE gathering state ----------------------------------------------------
-
-const ICE_GATHERING_STATE_MAP: Record<number, RTCIceGatheringState> = {
-    [GstWebRTC.WebRTCICEGatheringState.NEW]: 'new',
-    [GstWebRTC.WebRTCICEGatheringState.GATHERING]: 'gathering',
-    [GstWebRTC.WebRTCICEGatheringState.COMPLETE]: 'complete',
-};
-
-export function gstToIceGatheringState(v: number): RTCIceGatheringState {
-    return ICE_GATHERING_STATE_MAP[v] ?? 'new';
-}
-
-// ---- Transceiver direction (bidirectional) ----------------------------------
-
-const DIRECTION_GST_TO_W3C: Record<number, RTCRtpTransceiverDirection> = {
-    [GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV]: 'sendrecv',
-    [GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY]: 'sendonly',
-    [GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY]: 'recvonly',
-};
-
-const DIRECTION_W3C_TO_GST: Record<string, number> = {
-    sendrecv: GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV,
-    sendonly: GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY,
-    recvonly: GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY,
-    inactive: GstWebRTC.WebRTCRTPTransceiverDirection.NONE,
-};
-
-export function gstDirectionToW3C(v: number): RTCRtpTransceiverDirection {
-    return DIRECTION_GST_TO_W3C[v] ?? 'inactive';
-}
-
-export function w3cDirectionToGst(d: RTCRtpTransceiverDirection): GstWebRTC.WebRTCRTPTransceiverDirection {
-    return DIRECTION_W3C_TO_GST[d] ?? GstWebRTC.WebRTCRTPTransceiverDirection.NONE;
-}
-
-// ---- Stats type (GstWebRTC → W3C RTCStatsType string) -----------------------
-
-export type RTCStatsType =
-    | 'codec' | 'inbound-rtp' | 'outbound-rtp'
-    | 'remote-inbound-rtp' | 'remote-outbound-rtp'
-    | 'csrc' | 'peer-connection' | 'data-channel'
-    | 'stream' | 'transport' | 'candidate-pair'
-    | 'local-candidate' | 'remote-candidate' | 'certificate';
-
-const STATS_TYPE_MAP: Record<number, RTCStatsType> = {
-    [GstWebRTC.WebRTCStatsType.CODEC]: 'codec',
-    [GstWebRTC.WebRTCStatsType.INBOUND_RTP]: 'inbound-rtp',
-    [GstWebRTC.WebRTCStatsType.OUTBOUND_RTP]: 'outbound-rtp',
-    [GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP]: 'remote-inbound-rtp',
-    [GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP]: 'remote-outbound-rtp',
-    [GstWebRTC.WebRTCStatsType.CSRC]: 'csrc',
-    [GstWebRTC.WebRTCStatsType.PEER_CONNECTION]: 'peer-connection',
-    [GstWebRTC.WebRTCStatsType.DATA_CHANNEL]: 'data-channel',
-    [GstWebRTC.WebRTCStatsType.STREAM]: 'stream',
-    [GstWebRTC.WebRTCStatsType.TRANSPORT]: 'transport',
-    [GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR]: 'candidate-pair',
-    [GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE]: 'local-candidate',
-    [GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE]: 'remote-candidate',
-    [GstWebRTC.WebRTCStatsType.CERTIFICATE]: 'certificate',
-};
-
-export function gstToStatsType(v: number): RTCStatsType | undefined {
-    return STATS_TYPE_MAP[v];
-}

package/src/gst-init.ts

@@ -1,49 +0,0 @@
-// Lazy GStreamer initialization for the WebRTC backend.
-//
-// `webrtcbin` ships with GStreamer's gst-plugins-bad (`libgstwebrtc.so`).
-// On Fedora: gstreamer1-plugins-bad-free + gstreamer1-plugins-bad-free-extras.
-// On Ubuntu/Debian: gstreamer1.0-plugins-bad + gstreamer1.0-nice.
-//
-// This module is GJS-only — the Node alias layer routes it to @gjsify/empty.
-
-import Gst from 'gi://Gst?version=1.0';
-import { DOMException } from '@gjsify/dom-exception';
-
-let initialized = false;
-
-export function ensureGstInit(): void {
-    if (initialized) return;
-    Gst.init(null);
-    initialized = true;
-}
-
-/** Throws if the `webrtcbin` element is not registered (gst-plugins-bad missing). */
-export function ensureWebrtcbinAvailable(): void {
-    ensureGstInit();
-    const webrtcFactory = Gst.ElementFactory.find('webrtcbin');
-    if (!webrtcFactory) {
-        throwNotSupported(
-            'GStreamer element "webrtcbin" not available. Install gst-plugins-bad:\n' +
-            '  Fedora:        dnf install gstreamer1-plugins-bad-free gstreamer1-plugins-bad-free-extras\n' +
-            '  Ubuntu/Debian: apt install gstreamer1.0-plugins-bad',
-        );
-    }
-    // webrtcbin requires libnice's GStreamer plugin for ICE transport —
-    // without it, pipeline state-change to PLAYING fails and createDataChannel
-    // hits the "webrtc->priv->is_closed" assertion.
-    const niceFactory = Gst.ElementFactory.find('nicesrc');
-    if (!niceFactory) {
-        throwNotSupported(
-            'GStreamer "nice" plugin (libnice-gstreamer) not available — required by webrtcbin.\n' +
-            '  Fedora:        dnf install libnice-gstreamer1\n' +
-            '  Ubuntu/Debian: apt install gstreamer1.0-nice\n' +
-            '  Verify with:   gst-inspect-1.0 nicesrc',
-        );
-    }
-}
-
-function throwNotSupported(message: string): never {
-    throw new DOMException(message, 'NotSupportedError');
-}
-
-export { Gst };

package/src/gst-stats-parser.ts

@@ -1,137 +0,0 @@
-// GstStructure → W3C RTCStatsReport parser.
-//
-// webrtcbin's `get-stats` action signal returns a GstStructure where each
-// field is a nested GstStructure containing one stats entry. Each entry
-// has a `type` field (GstWebRTCStatsType enum), an `id` field, a `timestamp`
-// field, plus type-specific fields in GStreamer snake_case naming.
-//
-// This module iterates the top-level fields, converts each nested structure
-// to a W3C-compliant stats dictionary with camelCase keys.
-
-import type Gst from 'gi://Gst?version=1.0';
-
-import { gstToStatsType } from './gst-enum-maps.js';
-import { RTCStatsReport, type RTCStats } from './rtc-stats-report.js';
-
-/**
- * Convert a GStreamer snake_case field name to W3C camelCase.
- * Examples: `bytes-sent` → `bytesSent`, `packets-received` → `packetsReceived`
- */
-function snakeToCamel(name: string): string {
-    return name.replace(/-([a-z])/g, (_, c) => c.toUpperCase());
-}
-
-/**
- * Extract all fields from a GstStructure into a plain object.
- * GJS unboxes `get_value` for boxed/simple types directly — no
- * GObject.Value wrapper is returned for primitive types.
- */
-function extractFields(structure: Gst.Structure): Record<string, unknown> {
-    const result: Record<string, unknown> = {};
-    const n = structure.n_fields();
-    for (let i = 0; i < n; i++) {
-        const fieldName = structure.nth_field_name(i);
-        try {
-            const value = structure.get_value(fieldName);
-            result[fieldName] = value;
-        } catch {
-            // Some fields may not be extractable — skip silently
-        }
-    }
-    return result;
-}
-
-/**
- * Parse a single stats entry (nested GstStructure) into a W3C RTCStats object.
- */
-function parseStatsEntry(structure: Gst.Structure): RTCStats | null {
-    const raw = extractFields(structure);
-
-    // The `type` field is a GstWebRTCStatsType enum value
-    const gstType = raw['type'];
-    if (gstType == null) return null;
-    const w3cType = gstToStatsType(Number(gstType));
-    if (!w3cType) return null;
-
-    const id = String(raw['id'] ?? structure.get_name() ?? '');
-    // GStreamer provides timestamp in nanoseconds (pipeline clock).
-    // Convert to milliseconds for W3C (performance.now()-relative).
-    const tsRaw = raw['timestamp'];
-    const timestamp = tsRaw != null ? Number(tsRaw) / 1_000_000 : performance.now();
-
-    const stats: RTCStats = { type: w3cType, id, timestamp };
-
-    // Copy remaining fields, converting snake_case → camelCase
-    for (const [key, value] of Object.entries(raw)) {
-        if (key === 'type' || key === 'id' || key === 'timestamp') continue;
-        const camelKey = snakeToCamel(key);
-        stats[camelKey] = value;
-    }
-
-    return stats;
-}
-
-/**
- * Parse the top-level GstStructure returned by webrtcbin's `get-stats`
- * signal into a W3C RTCStatsReport.
- *
- * The top-level structure has one field per stats entry. Each field's value
- * is a nested GstStructure (GJS unboxes boxed types automatically).
- */
-export function parseGstStats(reply: Gst.Structure | null): RTCStatsReport {
-    if (!reply) return new RTCStatsReport();
-
-    const entries: Array<[string, RTCStats]> = [];
-    const n = reply.n_fields();
-
-    for (let i = 0; i < n; i++) {
-        const fieldName = reply.nth_field_name(i);
-        try {
-            const nested = reply.get_value(fieldName);
-            // GJS unboxes the nested GstStructure directly
-            if (nested && typeof (nested as any).n_fields === 'function') {
-                const stats = parseStatsEntry(nested as unknown as Gst.Structure);
-                if (stats) {
-                    entries.push([stats.id, stats]);
-                }
-            }
-        } catch {
-            // Non-structure fields (e.g. metadata) — skip
-        }
-    }
-
-    return new RTCStatsReport(entries);
-}
-
-/**
- * Filter an RTCStatsReport to entries relevant to a specific track identifier.
- * Used by RTCRtpSender.getStats() and RTCRtpReceiver.getStats() to return
- * only the stats associated with their track.
- */
-export function filterStatsByTrackId(report: RTCStatsReport, trackId: string): RTCStatsReport {
-    const entries: Array<[string, RTCStats]> = [];
-    const relatedIds = new Set<string>();
-
-    // First pass: find stats entries that reference this track
-    for (const [id, stats] of report) {
-        if (stats.trackIdentifier === trackId) {
-            entries.push([id, stats]);
-            relatedIds.add(id);
-            // Collect referenced IDs (transportId, codecId, remoteId, etc.)
-            for (const value of Object.values(stats)) {
-                if (typeof value === 'string' && report.has(value)) {
-                    relatedIds.add(value);
-                }
-            }
-        }
-    }
-
-    // Second pass: include referenced stats (codec, transport, candidate-pair, etc.)
-    for (const [id, stats] of report) {
-        if (relatedIds.has(id) && !entries.some(([entryId]) => entryId === id)) {
-            entries.push([id, stats]);
-        }
-    }
-
-    return new RTCStatsReport(entries);
-}

package/src/gst-utils.ts

@@ -1,41 +0,0 @@
-// Async bridging between GstPromise and JavaScript Promises.
-//
-// Reference: refs/node-gst-webrtc/src/gstUtils.ts (ISC, Ratchanan Srirattanamet)
-//
-// Why a native bridge? Gst.Promise.new_with_change_func() invokes its
-// callback on GStreamer's internal streaming thread. GJS blocks any JS
-// callback invoked from a non-main thread (to prevent SpiderMonkey VM
-// corruption), so the change_func is never delivered to JS — the Promise
-// would hang forever.
-//
-// `@gjsify/webrtc-native/PromiseBridge` is a Vala helper that registers
-// the change_func on the C side, hops through `g_main_context_invoke()`
-// to the GLib main thread, and only then emits `replied` / `rejected`
-// signals which JS can safely consume.
-
-import type Gst from 'gi://Gst?version=1.0';
-import { PromiseBridge } from '@gjsify/webrtc-native';
-
-/**
- * Wrap a GstPromise-consuming emit into a JS Promise that resolves on the
- * main thread.
- *
- * Usage:
- *   const reply = await withGstPromise((promise) => {
- *     webrtcbin.emit('create-offer', options, promise);
- *   });
- */
-export function withGstPromise(
-    emit: (promise: Gst.Promise) => void,
-): Promise<Gst.Structure | null> {
-    return new Promise((resolve, reject) => {
-        const bridge = new PromiseBridge();
-        bridge.connect('replied', (_b: unknown, reply: Gst.Structure | null) => {
-            resolve(reply);
-        });
-        bridge.connect('rejected', (_b: unknown, message: string) => {
-            reject(new Error(message));
-        });
-        emit(bridge.promise);
-    });
-}