@gjsify/webrtc 0.4.30 → 0.4.31

package/lib/esm/rtc-peer-connection/data-channel.js

@@ -1 +1 @@
-import"../_virtual/_rolldown/runtime.js";import{Gst as e}from"../gst-init.js";import{RTCDataChannel as t}from"../rtc-data-channel.js";import{DOMException as n}from"@gjsify/dom-exception";function coerceUnsignedShort(e,t){let n=Number(t);if(!Number.isFinite(n))throw TypeError(`createDataChannel: ${e} must be a finite number, got ${String(t)}`);let r=Math.trunc(n);if(r<0||r>65535)throw TypeError(`createDataChannel: ${e}=${r} is outside the [0, 65535] range`);return r}const r={createDataChannel(r,i={}){if(this._closed)throw new n(`Cannot create a data channel on a closed RTCPeerConnection`,`InvalidStateError`);if(typeof r!=`string`)throw TypeError(`createDataChannel: label must be a string`);if(new TextEncoder().encode(r).byteLength>65535)throw TypeError(`createDataChannel: label too long (> 65535 bytes)`);let a=i.maxPacketLifeTime==null?void 0:coerceUnsignedShort(`maxPacketLifeTime`,i.maxPacketLifeTime),o=i.maxRetransmits==null?void 0:coerceUnsignedShort(`maxRetransmits`,i.maxRetransmits),s=i.id==null?void 0:coerceUnsignedShort(`id`,i.id);if(a!==void 0&&o!==void 0)throw TypeError(`createDataChannel: maxPacketLifeTime and maxRetransmits are mutually exclusive`);if(i.negotiated===!0&&s===void 0)throw TypeError(`createDataChannel: negotiated=true requires an id`);if(s===65535)throw TypeError(`createDataChannel: id 65535 is reserved`);let c=e.Structure.new_empty(`data-channel-opts`);this._setStructureField(c,`ordered`,`boolean`,i.ordered),this._setStructureField(c,`max-packet-lifetime`,`int`,a),this._setStructureField(c,`max-retransmits`,`int`,o),this._setStructureField(c,`protocol`,`string`,i.protocol),this._setStructureField(c,`negotiated`,`boolean`,i.negotiated),this._setStructureField(c,`id`,`int`,s);let l=null;try{l=this._webrtcbin.emit(`create-data-channel`,r,c)}catch(e){throw Error(`create-data-channel failed: ${e?.message??e}`)}if(!l)throw Error(`webrtcbin returned null data channel (check id/label/options)`);this._ensureSctpTransport();let u=new t(l);return this._dataChannels.set(l,u),u.addEventListener(`close`,()=>{this._dataChannels.delete(l)}),u}};function installDataChannelMethods(e){Object.assign(e,r)}export{installDataChannelMethods};
+import"../_virtual/_rolldown/runtime.js";import{Gst as e}from"../gst-init.js";import{RTCDataChannel as t}from"../rtc-data-channel.js";import{DOMException as n}from"@gjsify/dom-exception";function coerceUnsignedShort(e,t){let n=Number(t);if(!Number.isFinite(n))throw TypeError(`createDataChannel: ${e} must be a finite number, got ${String(t)}`);let r=Math.trunc(n);if(r<0||r>65535)throw TypeError(`createDataChannel: ${e}=${r} is outside the [0, 65535] range`);return r}const r={createDataChannel(r,i={}){if(this._closed)throw new n(`Cannot create a data channel on a closed RTCPeerConnection`,`InvalidStateError`);if(typeof r!=`string`)throw TypeError(`createDataChannel: label must be a string`);if(new TextEncoder().encode(r).byteLength>65535)throw TypeError(`createDataChannel: label too long (> 65535 bytes)`);let a=i.maxPacketLifeTime==null?void 0:coerceUnsignedShort(`maxPacketLifeTime`,i.maxPacketLifeTime),o=i.maxRetransmits==null?void 0:coerceUnsignedShort(`maxRetransmits`,i.maxRetransmits),s=i.id==null?void 0:coerceUnsignedShort(`id`,i.id);if(a!==void 0&&o!==void 0)throw TypeError(`createDataChannel: maxPacketLifeTime and maxRetransmits are mutually exclusive`);if(i.negotiated===!0&&s===void 0)throw TypeError(`createDataChannel: negotiated=true requires an id`);if(s===65535)throw TypeError(`createDataChannel: id 65535 is reserved`);let c=e.Structure.new_empty(`data-channel-opts`);this._setStructureField(c,`ordered`,`boolean`,i.ordered),this._setStructureField(c,`max-packet-lifetime`,`int`,a),this._setStructureField(c,`max-retransmits`,`int`,o),this._setStructureField(c,`protocol`,`string`,i.protocol),this._setStructureField(c,`negotiated`,`boolean`,i.negotiated),this._setStructureField(c,`id`,`int`,s);let l=null;try{l=this._webrtcbin.emit(`create-data-channel`,r,c)}catch(e){let t=e instanceof Error?e.message:String(e);throw Error(`create-data-channel failed: ${t}`)}if(!l)throw Error(`webrtcbin returned null data channel (check id/label/options)`);this._ensureSctpTransport();let u=new t(l);return this._dataChannels.set(l,u),u.addEventListener(`close`,()=>{this._dataChannels.delete(l)}),u}};function installDataChannelMethods(e){Object.assign(e,r)}export{installDataChannelMethods};

package/lib/types/media-stream-track.d.ts

@@ -1,8 +1,10 @@
 import '@gjsify/dom-events/register/event-target';
+import type GstNs from 'gi://Gst?version=1.0';
+import type { TeeMultiplexer } from './tee-multiplexer.js';
 /** @internal GStreamer backing for tracks created by getUserMedia */
 export interface MediaStreamTrackGstInit {
-    source: any;
-    pipeline: any;
+    source: GstNs.Element;
+    pipeline: GstNs.Pipeline;
 }
 export interface MediaStreamTrackInit {
     kind: 'audio' | 'video';
@@ -24,13 +26,13 @@ export declare class MediaStreamTrack extends EventTarget {
     private _onmute;
     private _onunmute;
     /** @internal GStreamer source element (e.g. pulsesrc, audiotestsrc) */
-    _gstSource: any;
+    _gstSource: GstNs.Element | null;
     /** @internal Pipeline the source currently lives in (updated by VideoBridge) */
-    _gstPipeline: any;
+    _gstPipeline: GstNs.Pipeline | null;
     /** @internal Tee element inserted by VideoBridge for preview fan-out */
-    _gstTee: any;
+    _gstTee: GstNs.Element | null;
     /** @internal TeeMultiplexer for multi-PC fan-out (created on second addTrack) */
-    _teeMultiplexer: any;
+    _teeMultiplexer: TeeMultiplexer | null;
     /** @internal Callback set by RTCRtpSender to control valve drop property */
     private _enableCallback;
     constructor(init: MediaStreamTrackInit);

package/lib/types/rtc-rtp-receiver.d.ts

@@ -26,7 +26,7 @@ export declare class RTCRtpReceiver {
     private _receiverBridge;
     /** @internal — stats callback set by RTCPeerConnection */
     _getStatsForTrack: ((track: MediaStreamTrack) => Promise<RTCStatsReport>) | null;
-    constructor(kind: 'audio' | 'video', gstReceiver: GstWebRTC.WebRTCRTPReceiver | null, pipeline?: any);
+    constructor(kind: 'audio' | 'video', gstReceiver: GstWebRTC.WebRTCRTPReceiver | null, pipeline?: Gst.Pipeline);
     /** @internal — called from RTCPeerConnection._handlePadAdded */
     _connectToPad(pad: Gst.Pad): void;
     /** @internal — called from RTCPeerConnection.close() */

package/lib/types/tee-multiplexer.d.ts

@@ -1,3 +1,4 @@
+import type GstNs from 'gi://Gst?version=1.0';
 /**
  * Manages a GStreamer `tee` element that fans out one source to multiple
  * consumer branches. Each branch gets its own src pad from the tee.
@@ -10,16 +11,16 @@ export declare class TeeMultiplexer {
      * 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);
+    constructor(pipeline: GstNs.Pipeline, source: GstNs.Element);
     /** Request a new src pad from the tee for a consumer branch. */
-    requestSrcPad(): any;
+    requestSrcPad(): GstNs.Pad | null;
     /**
      * Release a branch's src pad from the tee.
      * Adds a DROP probe before unlinking to prevent errors.
      */
-    releaseSrcPad(srcPad: any): void;
+    releaseSrcPad(srcPad: GstNs.Pad | null): void;
     /** Number of active branches. */
     get branchCount(): number;
     /** The tee element (for pipeline queries). */
-    get element(): any;
+    get element(): GstNs.Element;
 }

package/package.json

@@ -1,6 +1,6 @@
 {
     "name": "@gjsify/webrtc",
-    "version": "0.4.30",
+    "version": "0.4.31",
     "description": "W3C WebRTC API for GJS using GStreamer webrtcbin as the peer-connection backend",
     "type": "module",
     "module": "lib/esm/index.js",
@@ -58,10 +58,10 @@
         "peer-connection"
     ],
     "dependencies": {
-        "@gjsify/buffer": "^0.4.30",
-        "@gjsify/dom-events": "^0.4.30",
-        "@gjsify/dom-exception": "^0.4.30",
-        "@gjsify/webrtc-native": "^0.4.30"
+        "@gjsify/buffer": "^0.4.31",
+        "@gjsify/dom-events": "^0.4.31",
+        "@gjsify/dom-exception": "^0.4.31",
+        "@gjsify/webrtc-native": "^0.4.31"
     },
     "devDependencies": {
         "@girs/gjs": "4.0.1",
@@ -70,8 +70,8 @@
         "@girs/gst-1.0": "1.28.1-4.0.1",
         "@girs/gstsdp-1.0": "1.0.0-4.0.1",
         "@girs/gstwebrtc-1.0": "1.0.0-4.0.1",
-        "@gjsify/cli": "^0.4.30",
-        "@gjsify/unit": "^0.4.30",
+        "@gjsify/cli": "^0.4.31",
+        "@gjsify/unit": "^0.4.31",
         "@types/node": "^25.9.1",
         "typescript": "^6.0.3"
     }