@gjsify/webrtc 0.26.1 → 0.27.0
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/lib/esm/gst-init.js +8 -3
- package/package.json +8 -8
package/lib/esm/gst-init.js
CHANGED
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@@ -1,6 +1,11 @@
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1
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-
import"./_virtual/_rolldown/runtime.js";import e from"gi://Gst?version=1.0";import{DOMException as t}from"@gjsify/dom-exception";let n=!1;function ensureGstInit(){n||=(e.init(null),!0)}function ensureWebrtcbinAvailable(){ensureGstInit(),e.ElementFactory.find(`webrtcbin`)||throwNotSupported(`GStreamer element "webrtcbin" not available
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-
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-
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1
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import"./_virtual/_rolldown/runtime.js";import e from"gi://Gst?version=1.0";import{DOMException as t}from"@gjsify/dom-exception";let n=!1;function ensureGstInit(){n||=(e.init(null),!0)}function ensureWebrtcbinAvailable(){ensureGstInit(),e.ElementFactory.find(`webrtcbin`)||throwNotSupported(e.ElementFactory.find(`dtlsenc`)!==null||e.ElementFactory.find(`sctpenc`)!==null?`GStreamer element "webrtcbin" not available, but gst-plugins-bad IS installed
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(its dtls/sctp elements are registered). This distro does not ship the
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plugin — verify with: ls /usr/lib/gstreamer-1.0/libgstwebrtc.so
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Alpine/postmarketOS: absent since Alpine v3.24; no package provides it.
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Otherwise: install the plugin from source, or use a distro that ships it.`:`GStreamer element "webrtcbin" not available. Install gst-plugins-bad:
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Fedora: dnf install gstreamer1-plugins-bad-free gstreamer1-plugins-bad-free-extras
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Ubuntu/Debian: apt install gstreamer1.0-plugins-bad
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8
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Alpine/postmarketOS: apk add gst-plugins-bad`),e.ElementFactory.find(`nicesrc`)||throwNotSupported(`GStreamer "nice" plugin (libnice-gstreamer) not available — required by webrtcbin.
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Fedora: dnf install libnice-gstreamer1
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Ubuntu/Debian: apt install gstreamer1.0-nice
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Verify with: gst-inspect-1.0 nicesrc`)}function throwNotSupported(e){throw new t(e,`NotSupportedError`)}export{e as Gst,ensureGstInit,ensureWebrtcbinAvailable};
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package/package.json
CHANGED
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@@ -1,6 +1,6 @@
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1
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{
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"name": "@gjsify/webrtc",
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"version": "0.
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"version": "0.27.0",
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"description": "W3C WebRTC API for GJS using GStreamer webrtcbin as the peer-connection backend",
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"type": "module",
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"module": "lib/esm/index.js",
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@@ -41,7 +41,7 @@
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"./globals.mjs"
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],
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"scripts": {
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"clear": "
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"clear": "gjsify clear lib tsconfig.tsbuildinfo tsconfig.types.tsbuildinfo test.gjs.mjs",
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"check": "gjsify tsc --noEmit",
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"build": "gjsify run build:gjsify && gjsify run build:types",
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"build:gjsify": "gjsify build --library 'src/**/*.{ts,js}' --exclude 'src/**/*.spec.{mts,ts}' 'src/test.{mts,ts}'",
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@@ -61,10 +61,10 @@
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"peer-connection"
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],
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"dependencies": {
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64
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"@gjsify/buffer": "^0.
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"@gjsify/dom-events": "^0.
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"@gjsify/dom-exception": "^0.
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"@gjsify/webrtc-native": "^0.
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"@gjsify/buffer": "^0.27.0",
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"@gjsify/dom-events": "^0.27.0",
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"@gjsify/dom-exception": "^0.27.0",
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"@gjsify/webrtc-native": "^0.27.0"
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},
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"devDependencies": {
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"@girs/gjs": "^4.1.0",
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@@ -73,8 +73,8 @@
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"@girs/gst-1.0": "^4.1.0",
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"@girs/gstsdp-1.0": "^4.1.0",
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"@girs/gstwebrtc-1.0": "^4.1.0",
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"@gjsify/cli": "^0.
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"@gjsify/unit": "^0.
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"@gjsify/cli": "^0.27.0",
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"@gjsify/unit": "^0.27.0",
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"@types/node": "^25.9.2",
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"typescript": "^6.0.3"
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},
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