Logan Cusano
ceb2836371
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>
100 lines
4.4 KiB
Python
100 lines
4.4 KiB
Python
from pydantic_settings import BaseSettings
|
|
from typing import Optional
|
|
|
|
|
|
class Settings(BaseSettings):
|
|
# Node identity
|
|
node_id: str
|
|
node_name: str = "Unnamed Node"
|
|
node_lat: float = 0.0
|
|
node_lon: float = 0.0
|
|
|
|
# MQTT
|
|
mqtt_broker: str
|
|
mqtt_port: int = 1883
|
|
mqtt_user: Optional[str] = None
|
|
mqtt_pass: Optional[str] = None
|
|
|
|
# C2 server (audio upload destination); None disables upload
|
|
c2_url: Optional[str] = None
|
|
|
|
# Local Icecast — live listening only (frontend / mobile).
|
|
# NOT used for call recording or Discord voice: it lags 1s and drifts to 100s+.
|
|
icecast_host: str = "localhost"
|
|
icecast_port: int = 8000
|
|
icecast_mount: str = "/radio"
|
|
icecast_source_password: str = "hackme"
|
|
|
|
# PulseAudio — the low-latency path used for call recording and Discord voice.
|
|
# Liquidsoap (op25 container) writes into the `drb_sink` null sink; we capture
|
|
# its monitor. Addressed explicitly rather than via "default" because the op25
|
|
# entrypoint starts pulseaudio with -n and never applies system.pa's
|
|
# `set-default-source` line.
|
|
pulse_source: str = "drb_sink.monitor"
|
|
# Bounded wait for the shared PulseAudio socket before launching FFmpeg.
|
|
pulse_wait_timeout: float = 30.0
|
|
|
|
# Call segmentation — seconds with no active transmission before the current
|
|
# recording is closed out. Consecutive grants on the SAME talkgroup inside this
|
|
# window are kept in one recording so back-and-forth traffic stays together.
|
|
#
|
|
# Do NOT tune this against measured *audio* silence: audio gaps also contain
|
|
# the ~1.9 s P25 grant→speech delay, so they are always longer than the
|
|
# control-channel idle this timer measures. metadata_watcher logs the real
|
|
# measured idle on every idle-timeout close — tune from that.
|
|
call_idle_timeout: float = 3.0
|
|
|
|
# Audio kept after the observed end of the last transmission. The srcaddr
|
|
# 1→0 edge can be up to one poll (0.5 s) late and the encoder adds its own
|
|
# latency, so this is the only headroom protecting the last word of a
|
|
# transmission — which is usually the disposition or the address.
|
|
#
|
|
# Raised 1.0 -> 3.0 after field measurement showed the recording WINDOW
|
|
# (anchored to OP25 control-channel timestamps) closing well before the
|
|
# actual voice audio arrives: grant->speech offset measured 0.84-1.62s
|
|
# across 7 calls (~1.5s typical). At the old 1.0s pad, a short
|
|
# transmission (e.g. a 0.97s control-channel call) had its window close
|
|
# at T+1.97 while voice didn't start until ~T+1.5 — leaving ~0.4s of
|
|
# captured speech, clipped mid-word. Confirmed by a 0.57s output file
|
|
# whose final 0.10s measured -12.2dB, louder than its own -18.2dB
|
|
# average (i.e. clipped speech, not trailing silence), and by two short
|
|
# calls that produced no "Trimmed" log line at all because there was no
|
|
# trailing silence left to trim.
|
|
#
|
|
# Safe to be generous here: trim_silence already strips trailing silence
|
|
# back to trim_silence_guard_seconds before upload, so a larger pad costs
|
|
# long calls nothing (the extra is trimmed away) while giving short
|
|
# transmissions enough window to actually capture the voice. Over-capture
|
|
# is free; under-capture loses words permanently. Do not tune this back
|
|
# down without new field data showing the grant->speech offset has
|
|
# shrunk — see DEFERRED.md for the call_idle_timeout coupling this value
|
|
# now sits at.
|
|
call_tail_pad_seconds: float = 3.0
|
|
|
|
# Strip leading/trailing dead air before upload. ~63% of a typical recording
|
|
# is silence (the grant→speech delay plus the tail pad), which inflates
|
|
# Whisper cost and is a well-documented trigger for hallucinated transcript
|
|
# text. Trimming is conservative — see internal/audio_trim.py.
|
|
trim_silence: bool = True
|
|
# Anything quieter than this counts as silence for detection purposes.
|
|
trim_silence_threshold_db: float = -40.0
|
|
# Guard margin kept around detected speech so no syllable is clipped.
|
|
trim_silence_guard_seconds: float = 0.25
|
|
|
|
# OP25 container
|
|
op25_api_url: str = "http://localhost:8001"
|
|
op25_terminal_url: str = "http://localhost:8081"
|
|
|
|
# Paths (volume mounts)
|
|
config_path: str = "/configs"
|
|
recordings_path: str = "/recordings"
|
|
|
|
# Offline call buffer — how many call_end events to keep while disconnected
|
|
offline_call_buffer_size: int = 35
|
|
|
|
class Config:
|
|
env_file = ".env"
|
|
|
|
|
|
settings = Settings()
|