zebra-report: per-speaker "stream" toggle (DJ mode, Ogg/Opus over HTTP)

Each speaker row carries a stream toggle that swaps their audio from
WebRTC subscribe (~700ms latency, glitchy under loss) to a plain
HTTP <audio> pulling Ogg/Opus from the SFU's new /stream endpoint
(~2s latency, browser-managed jitter buffer, bulletproof for DJ
playback). Requires matching proxy.unturf.com SFU build.

zebra-report/zebra-spaces.html

@@ -3853,6 +3853,10 @@ function tearPeer(uuid){
   if (pc){ try { pc.close(); } catch(_){} peers.delete(uuid); }
   const a = remoteAudio.get(uuid);
   if (a){ try { a.srcObject = null; a.remove(); } catch(_){} remoteAudio.delete(uuid); }
+  /* stream-mode <audio> for this uuid (DJ mode) — close the HTTP pull
+   * so the SFU stops fanning Ogg pages to a dead client */
+  const s = streamAudio.get(uuid);
+  if (s){ try { s.pause(); s.removeAttribute('src'); s.load(); s.remove(); } catch(_){} streamAudio.delete(uuid); }
 }
 function waitForIceGathering(p, timeoutMs=6000){
   return new Promise(res=>{
@@ -3930,6 +3934,56 @@ const MIC_OFF = '<svg viewBox="0 0 24 24" fill="none" stroke="#b00" stroke-width
 
 function rankOf(role){ return {host:4, cohost:3, speaker:2, listener:1}[role] || 0; }
 
+/* ----- HTTP broadcast stream (DJ mode) — parallel to SFU mic subscribe.
+ * For a given speaker's pubHex, the user can flip into "stream" mode:
+ * we mute the WebRTC remote audio for that uuid and attach a separate
+ * <audio> element pulling /zebra-spaces-sfu/stream?room=R&pub=PUBHEX
+ * as Ogg/Opus. ~1.5–3s extra latency, but the browser's jitter buffer
+ * smooths out network glitches the WebRTC path can't. State is keyed
+ * by pubHex (not uuid) so it survives session-uuid churn on rejoin. */
+const streamMode = new Set();        // pubHex strings currently streaming
+const streamAudio = new Map();       // uuid -> <audio> pulling stream
+function streamUrlFor(pubHex){
+  return SFU_BASE + '/stream?room=' + encodeURIComponent(roomID) + '&pub=' + pubHex;
+}
+function startStream(uuid, pubHex){
+  /* mute the WebRTC mic for this uuid so they don't double-play */
+  const w = remoteAudio.get(uuid);
+  if (w) try { w.muted = true; } catch(_){}
+  let a = streamAudio.get(uuid);
+  if (!a){
+    a = document.createElement('audio');
+    a.autoplay = true; a.controls = false;
+    document.body.appendChild(a);
+    streamAudio.set(uuid, a);
+    applySinkTo(a);
+  }
+  a.muted = false;
+  a.src = streamUrlFor(pubHex);
+  try { const p = a.play(); if (p && p.catch) p.catch(()=>{}); } catch(_){}
+  logLine('', 'stream on for '+pubHex.slice(0,12)+' — DJ mode (~2s delay, glitch-free)');
+}
+function stopStream(uuid){
+  const a = streamAudio.get(uuid);
+  if (a){
+    try { a.pause(); } catch(_){}
+    try { a.removeAttribute('src'); a.load(); } catch(_){}
+  }
+  /* unmute the WebRTC mic so conversation resumes */
+  const w = remoteAudio.get(uuid);
+  if (w) try { w.muted = false; } catch(_){}
+}
+function toggleStreamFor(uuid, pubHex){
+  if (streamMode.has(pubHex)){
+    streamMode.delete(pubHex);
+    stopStream(uuid);
+  } else {
+    streamMode.add(pubHex);
+    startStream(uuid, pubHex);
+  }
+  renderRoom();
+}
+
 function renderRoom(){
   const wrap = $('members'); wrap.innerHTML = '';
   /* sort host → cohosts → speakers → listeners, then by joined_at */
@@ -3956,6 +4010,25 @@ function renderRoom(){
     m._meterEl = fill;
     /* mod controls — only render when we can act on this row */
     const acts = document.createElement('span'); acts.className = 'mod-actions';
+    /* per-speaker "stream" toggle: anyone can flip another speaker's
+     * audio from WebRTC subscribe → HTTP Ogg/Opus tap. Higher latency,
+     * cleaner playback under loss. Not a mod action — every listener
+     * gets to pick their own path per speaker. */
+    if (canSpeak(m.role) && m.uuid !== myUUID && m.pubkey){
+      let mPubHex = '';
+      try { mPubHex = hex(unb64(m.pubkey)); } catch(_){}
+      if (mPubHex){
+        const sb = document.createElement('button'); sb.className = 'small';
+        const on = streamMode.has(mPubHex);
+        sb.textContent = on ? '◉ stream' : '○ stream';
+        sb.title = on
+          ? 'streaming — click to switch back to live (low-latency WebRTC)'
+          : 'switch to HTTP Ogg/Opus stream (~2s delay, glitch-free for DJ sets)';
+        if (on) sb.classList.add('invert');
+        sb.onclick = () => toggleStreamFor(m.uuid, mPubHex);
+        acts.appendChild(sb);
+      }
+    }
     if (isMod(myRole) && m.uuid !== myUUID){
       if (m.role === 'listener'){
         const b = document.createElement('button'); b.className='small invert';
@@ -4466,8 +4539,8 @@ logLine('', 'ready — pick a handle, type a rendezvous code, enter the space');
 
 <footer style="margin:2.2rem auto 0;font-size:0.65rem;color:#999;line-height:1.7;word-break:break-all;font-family:monospace">
   <span id="pi-seal" style="color:#777;cursor:default;user-select:none" title="">page integrity</span> &nbsp;·&nbsp; built <span class="stamp-date">2026-06-03</span><br>
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+  md5 <span class="stamp-md5">beebb8a6a5d5675bfcf12f83230fe283</span><br>
+  sha256 <span class="stamp-sha">898175e9ffac08ae9e56390a594701728bd94ec8b0d36942589df921efcfa716</span><br>
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