CLAUDE.md "Audio jitter buffering" section was the longest in the file (~70 lines of restated rules). Collapse to a 5-rule list with pointers to memory + tickets for the deep dives. The old structure restated content that already lived in feedback_audio_jitter_buffer_strategy and the per-defect tickets. Telemetry + tickets sections compressed similarly. Same content, fewer words, plus explicit pointers so a future session knows where to look for the canonical version of each rule. Ticket 0001 grows a Cross-references section pointing back to CLAUDE.md + memory + the pinning test. Ticket 0002 (firefox camera) updated with the signal-log diagnosis: every attempt shows "The request is not allowed by the user agent or the platform in the current context." = NotAllowedError fired instantly, no permission prompt. Conclusion: firefox has a remembered "Block" for the origin. picked.cam= is empty so it isn't the saved- deviceId case. Lays out two prongs: user-side permission reset (immediate) and code-side NotAllowedError handling (next pass) so the page surfaces a user-friendly notice instead of just logging the error text. Cross-refs to CLAUDE.md telemetry section + the catch site. docs/tickets/README.md gets a "See also: CLAUDE.md" pointer up top so the loop closes from the ticket index back to the shared context. Memory side wired bidirectionally: chromium-decoder-anchor and audio-jitter-buffer-strategy now point at CLAUDE.md + ticket 0001.
12 KiB
0001: fedora chrome cannot hear speakers
Status: fixed
Priority: P0 (silent regression — every invariant green except audible output)
Surface: fedora chrome — observed under both listener and speaker roles
First seen: 2026-06-04 (or earlier — "for days")
Reporter: fox (blanka-chrome identity)
Fixed: 2026-06-07, commits dad697d (source) + 1f27b89 (deploy)
Root cause
Chromium will not run its WebRTC audio decoder for a remote
MediaStreamTrack whose only consumer is a MediaStreamAudioSourceNode.
The decoder requires an HTMLMediaElement (an <audio> or <video>
element) bound to the track to actually decode. Without that anchor,
the source node produces silence even though packets arrive at full
rate, the worklet's started event fires, and audioCtx.state === 'running'. Firefox does not have this restriction.
PulseAudio Playback during repro: chrome stream visible (chrome WAS
producing an output stream), level meter pinned at 0 (the stream's
content was silence). RTP telemetry: aud.recv pkt=N → N+50/sec bytes=growing level=0.000 jbuf=?. Worklet telemetry:
jitter-buffer started uuid=… target=0.7s (queue filled — but with
zeros).
Fix
In attachAudioStreamViaWorklet, alongside the existing
createMediaStreamSource(stream) → gain → destination chain, create a
hidden muted <audio> element with srcObject = stream, append it
to the DOM, and call play(). The element makes no audible output
(muted = true); its sole purpose is to keep chromium's WebRTC
decoder running so the worklet's MediaStreamSource sees real
samples. setWorkletStream swaps the anchor's srcObject alongside
the source node; detachListenerStream tears the anchor down.
Pinned by tests in test/listener-audio-attach.test.js:
- anchor created on attach
- anchor follows the stream on setWorkletStream
- anchor torn down on detach
Cross-references
../../CLAUDE.md"Audio pipeline — load-bearing invariants" rules 3- 4 (anchor + always-SFU) are sourced from this ticket.
feedback-chromium-decoder-anchormemory carries the apply-everywhere version of rule 3 so a future session can't silently undo it.test/listener-audio-attach.test.jspins the anchor contract.
Symptom
On fedora chrome, no audio from any speaker in the room is audible. Affects every role chrome takes: as listener, no mic audio plays. As speaker, no peer mic audio plays through the mesh+SFU path. Same chrome can publish mic + camera + screen fine — only the receive-side audio is silent.
Other browsers on the same machine (firefox) hear the room correctly once the same code is loaded — so the codebase is not uniformly broken.
Telemetry / Repro
Signal-server log (/var/log/zebra-spaces-signal.log,
CLIENT_LOG actor_handle="blanka-chrome") on 2026-06-06 ~23:24 UTC:
· role=speaker sub=connected/connected pub=connected/connected
mesh=2 sListen=0 streamMode=0 muted=1 xcr=off
aud.recv pkt=308 lost=0 bytes=197428 jitter=0.0080 level=0.000 jbuf=? lp=0.2s
aud.recv pkt=22075 lost=3 bytes=44147 jitter=0.3210 level=0.000 jbuf=? lp=0.9s
aud.recv pkt=44242 lost=4 bytes=751664 jitter=0.0260 level=0.000 jbuf=? lp=0.3s
- RTP arriving at ~50 pps on the active speaker (real Opus voice traffic)
level=0.000on EVERY receivermesh=2(both mesh peers connected)sub=connectedandpub=connected(both SFU PCs healthy)
Earlier in the session:
audio attach uuid=2d4a target=0.5s (current pool=3)
audio via AudioContext 2d4a target=0.5s ctxState=running
jitter-buffer installed uuid=2d4a target=0.5s
jitter-buffer started uuid=2d4a target=0.5s
mesh stream swapped into worklet for 2d4a — buffer cushion now applies to mesh too
Every attach invariant green. Worklet started event fired. ctxState=running.
Yet the user hears nothing.
User report: chrome tab indicator shows NO audio playing.
Hypothesis
Primary (deployed 2026-06-06, awaiting retest): the speaker-output
dropdown in the page calls setSinkId on <audio> elements only. The
worklet path (every listener AND every speaker monitor since c3ff58c routed
mesh through the worklet) runs through audioCtx.destination, which always
emits to the system default sink. If the user picks a non-default sink
in the dropdown, the worklet keeps emitting to the wrong device — silence
on the device they're monitoring.
Fox 2026-06-06 confirmed picking a non-default sink on fedora chrome. Firefox defaults to system default which happens to match the right speaker for him, which is why firefox music started working without needing this fix.
Secondary (not yet ruled out):
- AudioContext.setSinkId() may not be honored on this chrome version / flag set → silent fallthrough to default
- PulseAudio per-app default routing for chrome may override the picked sink at the OS layer
- The worklet's
started=truemay flip back tofalseon a network blip and never re-arm if the queue can't refill to target (rearm guard fires)
Fix
Landed: 467146a 2026-06-06 — applySinkToAudioCtx() calls
audioCtx.setSinkId(speakerDeviceId) on creation and in the dropdown's
fan-out. Diagnostic log line now includes sink=... so the next
telemetry session shows which sink the worklet is routed to.
Deployed: 0f36e02 to www.unturf.com — blanka-chrome needs page reload
(it's a code-shell change, not SDP-level).
Pinned by tests:
audioCtx.setSinkId is called on creation so the worklet routes to the user-picked sink (not just system default)audioCtx without setSinkId support (older browser) does not throw — silent fallback to default
2026-06-07 telemetry — sink hypothesis ruled out
Fresh /var/log/zebra-spaces-signal.log capture (00:20:01 — fxhp host
publishing real music, blanka-chrome speaker):
audio via AudioContext 085b target=0.5s ctxState=running sink=default
mesh stream swapped into worklet for 085b — buffer cushion now applies to mesh too
...
aud.recv pkt=14622 lost=1 bytes=9117575 jitter=0.0100 level=0.000 jbuf=? lp=0.1s
Comparison — fxhp-phone in the same room, same window, same publisher:
aud.recv pkt=11161 lost=1 bytes=7154201 jitter=0.0090 level=0.001 jbuf=0.49s lp=0.0s
- sink IS routed (deployed fix
467146aconfirmed live — later capture showssink=c38572ec…= the user-picked deviceId, NOT default). - 50 pps RTP, 9.1 MB of stereo Opus arriving on blanka-chrome's SFU sub PC.
jbuf=?=jitterBufferEmittedCount === 0= receiver decodes nothing. There is no consumer wired to the SFU audio receiver.- fxhp-phone on the same SFU stream has
jbuf=0.49s+level=0.001(native decoder running) — so the SFU is fine, the problem is per- client wiring.
So: sink routing was not the bug, or was only part of it. With the sink fix applied, fedora chrome still hears silence.
Real root cause
Two coupled defects, both visible in handleRemoteSfuTrack (line 4700)
and the mesh pc.ontrack (line 6693):
(A) tracks=0 race at mic ontrack. Chrome can fire ontrack where
ev.streams[0] exists but has 0 tracks at handler-time (the track is
added a microtask later, or MSID-supplant merges new + dead tracks).
Screen / camera / game already work around this by constructing a fresh
new MediaStream([ev.track]) and passing that. Mic does not. It
caches ev.streams[0] directly into sfuStreamsByPubHex and hands the
same reference to attachSfuTrack. When that fires before the track
shows up on the stream, attachAudioStreamViaWorklet rejects with
liveAudio.length === 0 → falls through to the silent <audio> path
→ logs sfu attach b1a7 fresh=1 tracks=0 + meter for b1a7: MediaStream has no audio track (both verbatim in today's capture).
(B) Mesh swap orphans the SFU receiver. Mesh pc.ontrack calls
setWorkletStream(uuid, meshStream) unconditionally — it does NOT
verify that mesh is actually emitting audio. The worklet's
MediaStreamSource is now bound to the mesh track. The SFU receiver
has no consumer, so its decoder doesn't run (matches jbuf=?,
level=0). If the mesh track is muted or DTX-silent, the listener
hears nothing — and there is no automatic SFU restore until mesh PC
actually transitions to failed (line 6739). Mesh that stays
connected but silent traps the listener forever.
Hard refresh + leave/enter fixes both: fresh PC negotiation → fresh
ontrack with the live track in the stream → attach succeeds on SFU
before mesh comes up → mesh swap eventually replaces source but by
then SFU receiver is fine and… actually no, this still ends up on
mesh. The reason hard-refresh works is that mesh negotiation is fresh
and the new mesh track arrives in an unmuted state because audio is
already flowing. The mesh swap then lands on a real audio source.
Fix (in progress)
(A) In handleRemoteSfuTrack mic branch, mirror the screen/camera
pattern: const s = new MediaStream([ev.track]); sfuStreamsByPubHex.set(pubHex, s); attachSfuTrack(uuid, s).
Guarantees the stream handed to attachAudioStreamViaWorklet contains
exactly the live track, every time.
(B) In mesh pc.ontrack, gate the setWorkletStream swap on
!ev.track.muted (or addEventListener('unmute', swap) if it starts
muted). If the mesh track never unmutes, keep SFU as the worklet
source. Add a watchdog: if a swapped-in mesh track returns to muted
for >5s, swap the worklet back to the cached SFU stream — same
pattern the connectionState === 'failed' path already uses at line
6755.
2026-06-07 second telemetry pass — sink DRIFT identified
After deploying (A) + (B), blanka-chrome was still silent. Fresh telemetry pinned a different per-attach failure mode:
21:31:30 audio attach uuid=47e5 target=0.5s (current pool=1)
21:31:31 audio via AudioContext 47e5 target=0.5s ctxState=running sink=c38572ec…
21:31:31 jitter-buffer started uuid=47e5 target=0.5s
…leave/rejoin cycle…
21:31:33 audio attach uuid=47e5 target=0.5s (current pool=1)
21:31:33 audio via AudioContext 47e5 target=0.5s ctxState=running sink=default ← drifted
No audioCtx sink → c38572ec… log line ever appeared. So
applySinkToAudioCtx() either never ran or hit the
typeof audioCtx.setSinkId !== 'function' early-return at the moment
of execution. The earlier sink fix only invoked
applySinkToAudioCtx() inside the if (!audioCtx) branch — a
persistent audioCtx whose .sinkId getter returned '' (default) was
never re-corrected.
Fix (C) — re-apply sink on every attach when it drifted
attachAudioStreamViaWorklet now: if audioCtx already exists AND
speakerDeviceId is set AND audioCtx.sinkId !== speakerDeviceId,
call applySinkToAudioCtx(). Idempotent — same-sink call is a no-op.
Also added per-tick ctx.sink= to the telemetry stream so sink drift
is visible on every 5s line instead of needing to find the rare attach
log. When the active sink doesn't match the picked one, the line
includes want=… so the mismatch jumps out.
Status notes
- 2026-06-07 (fixed): chrome decoder anchor landed in
dad697d, deployed1f27b89. fox confirmed audio working on chrome. - 2026-06-07 (root cause pinned): pavucontrol showed chrome stream
present + level=0 with worklet
startedfired — the chromium-only MediaStreamSource-needs-anchor signature. - 2026-06-07 (en-route fixes kept): always-SFU receive
(
1984bb1),SPEAKER_PLAYOUT_DELAY_SEC0.5s → 0.7s, session fingerprint telemetry (1ed5263), per-peer mesh.recv stats (b9e2541),audioCtx.setSinkIdre-apply on rejoin (467146a). - 2026-06-07 (in-progress, fix C deployed): sink-drift on
audioCtx-persistent path identified + patched. Need a third telemetry
pass to confirm the per-tick
ctx.sink=line now stays on the picked device across leave/rejoin. - 2026-06-07 (fixes A + B deployed): sink hypothesis from
467146aruled out by fresh telemetry; (A) tracks=0 race + (B) mesh-swap orphan landed incaa0548. blanka-chrome was still silent, leading to fix (C) above. - 2026-06-06: the Jun 5 cascade + flushSfuStreams prefix-match fix
(
0ce1339) helped firefox but not chrome. Telemetry above ruled out the FSM / attach defects. - 2026-06-04 → 2026-06-05: earlier cascade commits (
2e74b92,f479878,144dd15,61b21a6) tightened the audio attach path but did not address this defect (they targeted double-audio on fxhp-android-firefox).