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Fix tail truncation, decouple call length from buffer, trim silence
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>
2026-08-04 21:39:36 -04:00

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# Node Identity
NODE_ID=node-001
NODE_NAME="My Radio Node"
NODE_LAT=0.0
NODE_LON=0.0
# MQTT — point to your C2 server
MQTT_BROKER=localhost
MQTT_PORT=1883
# Must match MQTT_NODE_USER/MQTT_NODE_PASS in the server's top-level .env
MQTT_USER=drb-node
MQTT_PASS=change-me-node
# C2 server for audio upload (leave blank to disable upload)
C2_URL=http://localhost:8888
# API key is provisioned automatically via MQTT after admin approves the node
# Icecast (local container — usually no need to change)
# Live listening only. Call recording and Discord voice use PulseAudio instead.
ICECAST_SOURCE_PASSWORD=hackme
ICECAST_ADMIN_PASSWORD=admin
ICECAST_HOST=localhost
ICECAST_PORT=8000
ICECAST_MOUNT=/radio
# PulseAudio capture (usually no need to change)
# Monitor of the drb_sink null sink that Liquidsoap writes into.
PULSE_SOURCE=drb_sink.monitor
# Seconds to wait for the shared PulseAudio socket before giving up and retrying.
PULSE_WAIT_TIMEOUT=30
# Call segmentation: seconds of radio silence before the current recording is
# closed. Grants on the same talkgroup within this window stay in ONE recording.
# Tune ONLY from the "measured control-channel idle" line the edge node logs on
# every idle-timeout close — silence measured in the audio is a different clock
# (it also contains the ~1.9s P25 grant-to-speech delay).
CALL_IDLE_TIMEOUT=3
# Seconds of audio kept after the last transmission ends. This is the only
# headroom protecting the final word of a transmission — usually the disposition
# or the address. Measured at 0.5s it left ~0.3s of real margin and one recording
# ended mid-word, hence 1.0.
CALL_TAIL_PAD_SECONDS=1.0
# Strip leading/trailing dead air before upload. ~63% of an untrimmed recording
# is silence, which costs Whisper spend and makes it hallucinate text that was
# never spoken. Only the head and tail are touched, with a guard margin so no
# syllable is clipped. Set to false to upload raw audio.
TRIM_SILENCE=true
# dBFS below which audio counts as silence for detection.
TRIM_SILENCE_THRESHOLD_DB=-40
# Seconds of audio kept either side of detected speech.
TRIM_SILENCE_GUARD_SECONDS=0.25
# OP25 container (usually no need to change)
OP25_API_URL=http://localhost:8001
OP25_TERMINAL_URL=http://localhost:8081
# Container registry — set these to pull pre-built images instead of building locally.
# Must match the DOCKER_ORG variable and repo name configured in Gitea.
# Leave blank to always build locally.
IMAGE_REGISTRY=git.vpn.cusano.net
DOCKER_ORG=
DOCKER_REPO=