Commit Graph
12 Commits
Author SHA1 Message Date
Logan CusanoandClaude Opus 5.5 6330609a75 Node: report every radio heard on a call, not just the last (server-26#304)
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P1 of the radio-ID plan. metadata_watcher records each transmission as
[radio, first seen, last seen] from channel rows and call_log grants in both
audio and console modes. Sightings of the same radio under 2 s apart are
merged, because the P0 probe found the ID in only 44% of polls during a
burst. call_end gains `radios: [{radio, start, end}]`.

main.py rebases the offsets into the trimmed audio file's frame, so
wall clock = audio_start_epoch + offset, the mapping C2 already uses. It
also sends the list on /upload (UploadMeta.radios), because C2 starts the
pipeline from /upload before call_end arrives.

srcaddr keeps its meaning (latest talker). Metadata only: no boundary,
timestamp or attribution change (drb-audio-review).

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
2026-11-08 23:03:18 -05:00
Logan CusanoandClaude Opus 5.5 d07d312e92 P25 conventional decode mode + conventional-channel attribution (node-26#24, #20)
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op25-container
- generate-config takes p25_mode "conventional" + conv_channels [{frequency,
  name, nac}]. It builds fixed fsk4 channels (up to 2 on one non-tunable SDR,
  DC spike kept off the channels) with no control channel to hunt. Two
  channels send audio to UDP 23456/23457, and audio.py -2 + amplify(2) keeps
  levels.
- tk_drb_conv.py (a tk_p25.rx_ctl subclass, copied into OP25 apps) reports
  voice / NAC / encryption (LC service options + LDU2 ESS algid/keyid) for
  conventional channels and applies the per-channel NAC filter, which stock
  tk_p25 does not. Analog uses it too.
- Trunked P25 config is unchanged when p25_mode is absent.

drb-edge-node
- radio_config: system kind from type + config.mode (missing = trunked).
- conventional.py normalizes the console: a distinctive LC TGID, else the
  channel's last TGID (30 s), else a synthetic per-channel id (freq/100 Hz).
  Encrypted rows are never talkers. With exactly one conventional channel,
  orphan audio is attributed to it at close, which fixes analog recording
  (#20). The synthetic id and the real TGID on one channel never split.
- Health/dashboard: P25 conventional listen tab (/api/listen p25conv), and
  conventional decode + encryption checks. The checkin carries radio_channels.
- Tests: edge 296 pass; op25-container 18 (new CI job). Reviewed by
  drb-audio-review (findings fixed; simultaneous two-channel mixdown filed
  as #26) and drb-qa-review.

Not verified on SDR hardware.

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
2026-10-04 15:57:15 -04:00
Logan CusanoandClaude Opus 5.5 10f7db0a2c Local dashboard: plain-language status checklist and C2-free listening
- /api/health: ordered checklist (password, C2 approval, MQTT, SDR, source,
  OP25, decoding, audio, Icecast) with what's wrong and what to do; pure
  build_checks() so every state (incl. Toughny's unapproved node) is tested
- /api/listen: pick P25 control channels / one analog freq / a cached C2
  system, validated in MHz, stored as a local override; OP25 restarted with
  the start failure reason surfaced
- /api/config/system no longer writes a raw blob OP25 can't read
- OP25 resumes after reboot on a locally chosen system, not only C2-assigned
- op25-container: analog (NBFM) config generation fixed (AttributeError +
  wrong field names), /op25/log tail, start refuses without a config,
  status detects a dead multi_rx ("crashed") -- node-26#12
- index.html rebuilt on UI_REDESIGN tokens; scanner drops DMR, shows System key

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
2026-10-04 01:20:11 -04:00
Logan CusanoandClaude Opus 5 d6dfe5a293 Drive call boundaries from audio, use the console only for the label
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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>
2026-08-06 18:19:45 -04:00
Logan CusanoandClaude Opus 5 f1de157d69 Pad the tgid_change closes so a talkgroup switch stops clipping the tail
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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>
2026-08-04 23:53:27 -04:00
Logan CusanoandClaude Opus 5 ceb2836371 Raise tail pad to 3s so short transmissions are not clipped
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>
2026-08-04 22:34:17 -04:00
Logan CusanoandClaude Opus 5 b0a8ed2a5a Fix tail truncation, decouple call length from buffer, trim silence
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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
Logan Cusano efdbe7d803 Move recording and Discord voice to PulseAudio
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2026-08-04 19:53:45 -04:00
Logan Cusano 3fbdc50536 feat: implement FastAPI router, web + local UI updates, and metadata monitoring for the edge node
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2026-07-12 21:30:30 -04:00
Logan be785a453c metadata fixes 2026-04-12 21:58:24 -04:00
Logan 7de55f9885 changes 2026-04-06 00:23:33 -04:00
Logan 1a9c92b6db Initial commit — DRB client (edge node) stack
Includes edge-node (FastAPI/MQTT/Discord voice), op25-container (SDR decoder),
and icecast (audio streaming).
2026-04-05 19:01:51 -04:00