Logan Cusano
d6dfe5a293
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>
131 lines
6.1 KiB
Python
131 lines
6.1 KiB
Python
from pydantic_settings import BaseSettings
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from typing import Optional
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class Settings(BaseSettings):
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# Node identity
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node_id: str
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node_name: str = "Unnamed Node"
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node_lat: float = 0.0
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node_lon: float = 0.0
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# MQTT
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mqtt_broker: str
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mqtt_port: int = 1883
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mqtt_user: Optional[str] = None
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mqtt_pass: Optional[str] = None
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# C2 server (audio upload destination); None disables upload
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c2_url: Optional[str] = None
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# Local Icecast — live listening only (frontend / mobile).
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# NOT used for call recording or Discord voice: it lags 1s and drifts to 100s+.
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icecast_host: str = "localhost"
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icecast_port: int = 8000
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icecast_mount: str = "/radio"
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icecast_source_password: str = "hackme"
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# PulseAudio — the low-latency path used for call recording and Discord voice.
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# Liquidsoap (op25 container) writes into the `drb_sink` null sink; we capture
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# its monitor. Addressed explicitly rather than via "default" because the op25
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# entrypoint starts pulseaudio with -n and never applies system.pa's
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# `set-default-source` line.
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pulse_source: str = "drb_sink.monitor"
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# Bounded wait for the shared PulseAudio socket before launching FFmpeg.
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pulse_wait_timeout: float = 30.0
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# ------------------------------------------------------------------
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# Call segmentation
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#
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# Boundaries come from the AUDIO, not the control channel. A recording
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# starts at voice onset and ends after call_silence_timeout seconds of
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# silence actually heard in the stream. See internal/metadata_watcher.py
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# for why the control channel is no longer trusted for either edge.
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# ------------------------------------------------------------------
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# Seconds of continuous silence IN THE AUDIO before the current recording is
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# closed. This is the primary segmentation control. Consecutive
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# transmissions on the SAME talkgroup separated by less than this stay in
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# one recording, so back-and-forth traffic is one file.
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#
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# Defaults to 3.0 to match the behaviour of the control-channel idle timer
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# it replaces, but it is NOT the same clock: this one measures real silence
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# in the audio, with no grant->speech delay mixed in. metadata_watcher logs
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# the measured trailing silence on every close — tune from that number.
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call_silence_timeout: float = 3.0
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# dBFS (RMS, measured over one ~46ms capture chunk) below which audio counts
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# as silence for the purpose of ending a recording.
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#
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# This does NOT need field calibration against radio noise. Between
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# transmissions the capture is the monitor of a PulseAudio *null sink*,
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# which emits digital silence, not an analog noise floor: measured on a live
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# node the gap sits at about -91 dBFS, i.e. one least-significant bit of a
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# 16-bit sample. Speech on the same node averages about -18 dBFS. Anything
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# between roughly -70 and -40 therefore behaves identically; -50 is chosen
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# to sit far below even quiet speech while staying far above the floor.
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call_silence_threshold_db: float = -50.0
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# DEPRECATED as a primary control — used ONLY in console fallback mode, i.e.
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# when PulseAudio capture is not producing audio and there is nothing to
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# segment on. Then, and only then, the old control-channel state machine
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# runs and closes a segment this many seconds after the last observed
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# transmission. Those segments carry no audio; they exist so the node keeps
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# reporting real radio activity to C2 while its audio path is broken.
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#
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# Do NOT tune this against measured *audio* silence — use
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# call_silence_timeout for that.
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call_idle_timeout: float = 3.0
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# Audio kept past a CONSOLE-DERIVED segment boundary, covering the fact that
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# buffered audio lags control-channel timestamps by ~1.5s (grant->speech
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# offset measured 0.84-1.62s across 7 field calls).
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#
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# Still needed, with a narrower job than before. It no longer pads the
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# normal end of a call — that boundary now comes from the audio itself and
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# needs no pad at all. It applies to the three boundaries that are still
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# control-channel timestamps:
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#
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# tgid_change close the outgoing call at the new grant + pad
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# tgid_change_unlogged close at the observing poll + pad
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# idle_timeout console fallback mode only
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#
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# Safe to be generous: trim_silence strips trailing silence back to
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# trim_silence_guard_seconds before upload, so a larger pad costs long calls
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# nothing. Over-capture is free; under-capture loses words permanently. If
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# the outgoing and incoming recordings overlap in the underlying audio
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# because of this pad, that is correct — the audio contains both.
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call_tail_pad_seconds: float = 3.0
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# Strip leading/trailing dead air before upload. A recording deliberately
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# over-captures at both ends (pre-roll at the head, the whole measured
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# silence run at the tail), which inflates Whisper cost and is a
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# well-documented trigger for hallucinated transcript text. Trimming is a
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# sample-offset slice of the buffered PCM — no re-encode — and only ever
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# touches the head and tail. See internal/audio_trim.py.
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trim_silence: bool = True
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# dBFS (RMS) below which audio counts as silence when trimming the ends.
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# Kept above call_silence_threshold_db on purpose: the closer must not miss
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# speech (permissive), the trimmer must not leave dead air (stricter), and
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# trim_silence_guard_seconds protects the syllable either way.
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trim_silence_threshold_db: float = -40.0
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# Guard margin kept around detected speech so no syllable is clipped.
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trim_silence_guard_seconds: float = 0.25
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# OP25 container
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op25_api_url: str = "http://localhost:8001"
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op25_terminal_url: str = "http://localhost:8081"
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# Paths (volume mounts)
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config_path: str = "/configs"
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recordings_path: str = "/recordings"
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# Offline call buffer — how many call_end events to keep while disconnected
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offline_call_buffer_size: int = 35
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class Config:
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env_file = ".env"
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settings = Settings()
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