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>
Three unrelated failures from `make test` on the node:
test_metadata_watcher: two tests asserted the old zero-pad tgid_change
contract. Both were off by exactly call_tail_pad_seconds, i.e. the code
was doing what f1de157 intended and the tests encoded the behaviour we
deliberately changed. Updated to expect the pad.
test_pulse: four async tests errored with "async def functions are not
natively supported". Root cause is not the tests - pytest.ini sets
asyncio_mode = auto but the Dockerfile only copies app/ and tests/, so
the container had no pytest config at all and fell back to strict mode.
That is also why these passed in a local venv and failed on the node.
Copy pytest.ini into the image, and add explicit @pytest.mark.asyncio so
the tests hold up regardless of how pytest is configured.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
The recording window is anchored to OP25 control-channel timestamps, but
the buffered audio lags those by roughly 1.5s. Trim logs across seven
calls measured the offset at 0.84-1.62s, consistently present.
At a 1.0s pad a short call closed its window before the voice arrived:
a 0.97s control-channel call closed at T+1.97 while voice started around
T+1.5, capturing ~0.4s of speech and cutting mid-word. Confirmed by a
0.57s file whose final 0.10s measured -12.2dB against its own -18.2dB
average - clipped speech, not a tail - and by two short calls that
logged no trim at all because no trailing silence remained.
Being generous is free here: trim_silence already strips trailing
silence back to the guard margin before upload, so long calls are
unaffected while short ones gain the window they need. Over-capture
costs nothing; under-capture loses words permanently.
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>
Logan Cusano