WebRTC audio mirror: AAC cannot cross WebRTC, so transcode audio to Opus into <path>.webrtc

bin/mtx-on-ready.sh (runOnReady inside the mediamtx container) posts the is-live notice and, for sources whose
audio is not Opus, runs ffmpeg: video copy, audio -> Opus, published over MediaMTX's loopback RTSP to
<user>/<Title>.webrtc (record: no). dumpd accepts .webrtc publishes only from 127.0.0.1. The portal hides
mirror paths, plays the mirror over WHEP (HLS stays on the source), holds an AAC source for 15 s while its
mirror comes up, and corrects the viewer count. Verified headless with a mocked path list.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>

live/src/Player.jsx

@@ -6,6 +6,7 @@ import { audio, effectiveMuted } from './useAudio.js'
 // WebRTC (WHEP) first, HLS fallback. Mute/volume come from the shared audio store.
 export default function Player({ params, api }) {
   const { id, user, title } = params
+  const whep = params.whep || id                       // Opus audio mirror path when the source audio is AAC
   const videoRef = useRef(null)
   const [status, setStatus] = useState('connecting…')
   const [viewers, setViewers] = useState(params.viewers ?? 0)
@@ -78,7 +79,7 @@ export default function Player({ params, api }) {
         const offer = await pc.createOffer()
         await pc.setLocalDescription(offer)
         await new Promise(res => { if (pc.iceGatheringState === 'complete') return res(); const t = setTimeout(res, 1500); pc.onicegatheringstatechange = () => { if (pc.iceGatheringState === 'complete') { clearTimeout(t); res() } } })
-        const r = await fetch(whepUrl(id), { method: 'POST', headers: { 'Content-Type': 'application/sdp' }, body: pc.localDescription.sdp, credentials: 'same-origin' })
+        const r = await fetch(whepUrl(whep), { method: 'POST', headers: { 'Content-Type': 'application/sdp' }, body: pc.localDescription.sdp, credentials: 'same-origin' })
         if (!r.ok) throw new Error('whep ' + r.status)
         await pc.setRemoteDescription({ type: 'answer', sdp: await r.text() })
         setStatus('webrtc')
@@ -90,7 +91,7 @@ export default function Player({ params, api }) {
     }
     startWhep()
     return () => { cancelled = true; try { pc?.close() } catch {}; try { hls?.destroy() } catch {}; v.srcObject = null; v.removeAttribute('src') }
-  }, [id])
+  }, [id, whep])
 
   const icon = muted ? '🔇' : volume === 0 ? '🔈' : volume < 0.5 ? '🔉' : '🔊'
   return (