The control channel was wrong in both directions. Grants fire 0.84-1.62s
before anyone speaks, and srcaddr can drop to 0 while someone is still
talking - one recording came back "-1.61s lead, -0.00s tail", the trim
finding nothing to remove because the window had closed on live speech.
Confirmed by ear: the cut lands at a word boundary on an unfinished word.
Audio is ground truth for WHEN. The console remains the only source of
WHO, so it still supplies talkgroup, alias and rid.
START voice onset in the captured audio, with a 0.25s pre-roll that
now covers only chunk quantisation and threshold ramp-up rather
than a variable control-channel offset.
STOP call_silence_timeout seconds of silence heard in the audio.
LABEL resolved AT CLOSE from a bounded rolling history of console
observations overlapping the window, +4s/-2s, because there is
no guaranteed ordering between a grant and its audio.
SPLIT a console talkgroup change still forces a cut, since two calls
with no silence between them would otherwise merge into one.
Capture now emits raw PCM instead of MP3. Silence detection becomes
integer arithmetic per chunk with no decode, trimming becomes a byte
offset slice rather than a second ffmpeg pass, and MP3 encoding happens
exactly once at save - uploads are no longer double-encoded.
Audio with no talkgroup anywhere in its window is discarded rather than
uploaded: an untagged call silently poisons incident correlation, which
is worse than losing the audio. Logged at ERROR and counted on
/api/status.
When capture produces no audio at all the old console state machine
still runs, so a node with a broken audio path keeps reporting radio
activity. That is now the only consumer of call_idle_timeout.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The tgid_change and tgid_change_unlogged paths closed the outgoing
segment with no tail pad, on the reasoning that the new grant's
timestamp is an exact, already-known boundary. That is exact only in
control-channel time. The buffered audio lags control timestamps by
~1.5s (measured 0.84-1.62s across seven field calls), so slicing there
cut roughly the outgoing call's last 1.5s of speech - recordings ending
mid-word with ~0s trailing silence.
Both paths now pad, and the recorder's bounded tail wait blocks until
that audio has actually been captured. TAIL_WAIT_TIMEOUT_SECONDS goes
2.0 -> 4.0 so it can satisfy the 3.0s pad instead of giving up and
warning on every talkgroup switch.
The incoming call's pre-roll is served from the ring buffer, so the
delay costs it nothing. The two slices overlapping in the underlying
audio is correct: the stream genuinely contains one call's tail and
then the next call's start.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The pulse_socket named volume survives container recreation, so after a
compose recreate the previous container's /run/pulse/pid and native
socket were still present. PulseAudio read the stale pid file, decided a
daemon was already running, and refused to start:
E: [pulseaudio] pid.c: Daemon already running.
The entrypoint still reported "PulseAudio socket ready" because it only
checked that the socket file existed - and a stale one did. Capture then
failed in a restart loop against a dead daemon.
Readiness in both the op25 entrypoint and drb-edge-node now means a
pactl probe actually succeeds. Stale pid/socket are removed only when
that probe fails, so a live daemon's socket is never deleted.
pulseaudio-utils was missing from the edge-node image (only libpulse0
was installed), so no pactl binary existed there at all - added.
Capture exits are now classified: a missing source logs at ERROR and
names the configured PULSE_SOURCE, rather than looking identical to
"daemon not up yet". Retrying forever against a wrong source name is how
the April PulseAudio failure stayed hidden.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Measured six recordings off a live P25 node and found two independent
defects causing lost audio at the end of calls:
- stop_recording() sliced the ring buffer immediately, so if the MP3
muxer had not yet delivered the tail the file was silently short.
Now waits (bounded, 2s) until buffered audio covers the end epoch.
- TGID-change closes used the new grant's epoch as the end with no pad
at all, guaranteeing truncation on every split. Tail pad is now a
setting, default raised 0.5s -> 1.0s.
The ring buffer also capped maximum call length: a call longer than the
buffer had its front silently clamped. The ring now serves the pre-roll
only, with a per-call accumulator for the rest, bounded at 4.8MB.
Clamping is loudly warned rather than silent.
Uploads averaged 63% silence, which inflates STT cost and is a known
Whisper hallucination trigger. Leading/trailing silence is now trimmed
conservatively (-40dB, 0.25s guard, internal pauses untouched).
started_at/ended_at still describe the call; new audio_* fields carry
the trimmed audio bounds so playback can map back to wall clock.
All-silence recordings are skipped and logged instead of uploaded.
Also: log measured control-channel idle on idle-timeout closes so
CALL_IDLE_TIMEOUT can be tuned from data, and quiet the httpx logger
which emitted ~170k lines/day of poll noise.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
op25-container/system.pa New PulseAudio config: auth-anonymous=1 so the edge-node can connect without group membership
op25-container/Dockerfile Copies system.pa into /etc/pulse/system.pa
drb-edge-node/Dockerfile Added libpulse0 — ffmpeg needs this runtime library to use -f pulse
drb-edge-node/app/internal/call_recorder.py Switched recording from Icecast (burst-buffer overlap) to PulseAudio monitor (zero-delay, exact call boundaries)
Logan Cusano