bandsaunter/README.md
The Dust Council 2aa8739f25 Tune SSB to the carrier, and read the sideband from the signal
An SSB capture was tuned to the centroid of the detected energy, which is
what every other mode wants and the one thing SSB cannot use. Its
demodulator is a filter that opens at the suppressed carrier, so centring
on the middle of the voice filtered off its lower half and shifted the
rest down by the error. Measured against known transmitters that error was
+2445 Hz on 20 m and -2486 Hz on 80 m, against 35 Hz for NFM and 9 Hz for
AM, which do not care either way. On real speech the difference is a clean
transcript versus nothing recognisable at all.

Speech puts most of its power just above the carrier, so the occupied band
leans towards it. ssb_alignment() reads that lean: it locates the carrier
to within about 100 Hz and names the sideband at the same time, without
recourse to any convention. The offset is applied inside the demodulator
at the IF rate, where it costs a fraction of what shifting the full-rate
stream would, and the reported frequency becomes the carrier -- the
frequency an operator would dial in.

The sideband was also decided by "LSB below 10 MHz, USB above", which
overrode a band plan that already knew better and demodulated 60 m as LSB.
The measurement decides now, the band plan when the signal has no lean to
read, and the convention only when neither has anything to say.

The simulator was transmitting SSB unfiltered, several times wider than
anything on the air, because it applied no transmit audio filter -- and
that filter is what makes a signal single-sideband. Its absence hid the
whole problem from the tests.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
2026-08-21 23:09:44 -07:00

39 KiB

bandsaunter

A signal scanner, recorder and identifier for RTL-SDR receivers.

Give it any number of frequency ranges — typed in by hand or picked from a built-in US band plan — and it sweeps them, stops on anything above the noise floor, records it, and works out what kind of signal it was. CW/Morse is decoded to text.

╭──────────────────────────────── receiver ────────────────────────────────╮
│ Rafael Micro R820T/R820T2   2.048 MS/s   gain auto   +0 ppm              │
╰──────────────────────────────────────────────────────────────────────────╯
╭───────────────────────────────── sweep ──────────────────────────────────╮
│ ━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━  step 4/6   146 MHz - 146.666667 MHz  │
│               ▆                      █       ▄▄▄        peak -18.0 dBFS  │
│ recording   cycle 2   hits 3   dropped 0   detections 7   up 0:04        │
╰──────────────────────────────────────────────────────────────────────────╯
╭──────────────────────────────────────────────────────────────────────────╮
│ REC 146.52 MHz  [nfm]  ███████████░░░░░░░░░  11.4/30s  SIGNAL  SNR 27 dB │
╰──────────────────────────────────────────────────────────────────────────╯
╭───────────────────────────── recorded signals ───────────────────────────╮
│ 19:38:43      460.025 MHz   8.2s   27.0  P25 Phase 1 C4FM digital voice  │
│ 19:38:43        144.1 MHz   8.2s   27.0  CW / Morse at 18 WPM  "VVV DE…" │
│ 19:38:43       146.52 MHz   8.2s   27.0  Narrowband FM voice (CTCSS 100) │
╰──────────────────────────────────────────────────────────────────────────╯

Install

From a package (Debian, Ubuntu, Mint)

./packaging/build-deb.sh              # writes dist/bandsaunter_<version>_all.deb
sudo apt install ./dist/bandsaunter_*.deb

apt pulls in every dependency itself, and the package blacklists the DVB-T driver that would otherwise claim the receiver. Nothing else to do.

Speech transcription is the one part that cannot come from Debian, because no speech recogniser is packaged there. Installed this way, everything else works and the install prints a short note saying how to add one. To have that arrive by apt as well, build the repository below instead.

From your own apt repository

For installing on several machines, or on a fresh one, without hunting for the recogniser afterwards. build-repo.sh builds three packages and an apt index:

Package Arch Contents
bandsaunter all the application
bandsaunter-transcribe amd64 faster-whisper and its dependencies, in /usr/lib/bandsaunter/vendor
bandsaunter-model-base-en all the base.en model, in /usr/share/bandsaunter/models
./packaging/build-repo.sh                 # writes dist/repo/
rsync -a dist/repo/ server:/var/www/html/bandsaunter/

Serve that directory over HTTP from anywhere on the LAN, then on each machine:

echo 'deb [trusted=yes] http://server/bandsaunter ./' \
    | sudo tee /etc/apt/sources.list.d/bandsaunter.list
sudo apt update
sudo apt install bandsaunter

That single command brings the recogniser and its model too — they are Recommends, which apt installs by default. --no-install-recommends gets just the application. Nothing reaches the network afterwards: the model is on disk, so the first transcription works offline.

[trusted=yes] skips signing, which is the sensible trade on a private LAN. To sign it instead, run gpg --clearsign over dist/repo/Release to produce InRelease and drop the [trusted=yes].

The vendored packages are appended to sys.path, never prepended, so anything apt provides — numpy, PyYAML — still wins; the vendor directory only fills the gap Debian leaves. BANDSAUNTER_VENDOR_DIR and BANDSAUNTER_MODEL_DIR override both locations.

Rebuilding for a new version is the same command; apt upgrade picks it up.

From source

sudo apt install rtl-sdr librtlsdr0 espeak-ng    # Debian, Ubuntu, Mint
pip install -e .

Dependencies

Everything required is packaged in Debian, Fedora and Arch, so nothing has to be built.

Package Debian/Ubuntu Fedora Arch Needed for
required librtlsdr librtlsdr0 rtl-sdr rtl-sdr talking to the receiver at all
required NumPy python3-numpy python3-numpy python-numpy all signal processing
required SciPy python3-scipy python3-scipy python-scipy filters, resampling, spectra
required Rich python3-rich python3-rich python-rich menus and the live display
required PyYAML python3-yaml python3-pyyaml python-yaml settings file and profiles
recommended eSpeak NG espeak-ng espeak-ng espeak-ng clearer spoken timestamps
optional rtl-sdr tools rtl-sdr rtl-sdr rtl-sdr rtl_test and friends for diagnosis
optional a speech recogniser pip only pip only AUR transcribing speech to text
optional Matplotlib python3-matplotlib python3-matplotlib python-matplotlib nothing yet; reserved for plots

Two notes on the optional ones:

eSpeak NG is a recommendation, not a requirement. Without it the spoken timestamps come from a built-in formant synthesiser, so that feature works on a machine with nothing else installed. With it they are clearer and render about three times faster.

No speech recogniser is packaged for Debian. faster-whisper, vosk and the pocketsphinx Python bindings are all absent from the archive, so transcription can only be installed with pip:

pip install faster-whisper     # best on radio audio, ~120 MB
pip install vosk               # ~10 MB plus a 40 MB model, weaker on noise
bandsaunter transcribe --engines

That is why the plain .deb cannot depend on one. Debian Policy forbids anything in the archive from requiring software outside it, and a postinst that reaches out to PyPI would break offline and reproducible installs — so a package in the archive simply cannot pull these in. Transcription is therefore off by default and reports plainly when no recogniser is present, rather than the install failing or the feature appearing broken.

A repository of your own is not bound by that rule, which is what build-repo.sh exploits: it packages faster-whisper and its model itself, into a private directory rather than into dist-packages, and lets apt install them alongside. Nothing is downloaded at install time, and nothing collides with an apt-managed module.

Mixing the two is nonetheless fine here. Modern Debian marks the system Python as externally managed (PEP 668), so a pip install lands in your user site directory:

pip install --user faster-whisper      # ~/.local/lib/python3.x/site-packages

which is on sys.path for the system interpreter. A bandsaunter installed from the .deb into /usr/lib/python3/dist-packages picks it up with no further configuration — verified, not assumed. A virtual environment works too, as long as bandsaunter runs inside it.

Getting these into Debian proper would be a different matter: it would mean packaging ctranslate2, tokenizers, onnxruntime and their dependencies, several of which are large C++ or Rust projects, each to archive standards. That is why none of them are there, and why the local repository vendors the wheels instead of trying to do it properly.

Other distributions

sudo dnf install rtl-sdr python3-numpy python3-scipy python3-rich \
                 python3-pyyaml espeak-ng                    # Fedora
sudo pacman -S rtl-sdr python-numpy python-scipy python-rich \
               python-yaml espeak-ng                          # Arch
brew install librtlsdr espeak-ng && pip install -e .          # macOS

Letting your user reach the receiver

The DVB-T television driver claims RTL dongles on sight and has to be kept away from them. The .deb does this for you; from source:

echo 'blacklist dvb_usb_rtl28xxu' | sudo tee /etc/modprobe.d/blacklist-rtlsdr.conf
sudo rmmod dvb_usb_rtl28xxu        # or just unplug and replug the receiver

If the device is found but cannot be opened, your user needs permission for it. Most distributions ship a udev rule with rtl-sdr; failing that:

echo 'SUBSYSTEM=="usb", ATTRS{idVendor}=="0bda", ATTRS{idProduct}=="2838", MODE="0666"' \
  | sudo tee /etc/udev/rules.d/20-rtlsdr.rules
sudo udevadm control --reload-rules && sudo udevadm trigger

Check it worked

bandsaunter devices --test      # opens the receiver and captures a test block
bandsaunter scan -b 2m --simulate   # exercises everything without hardware

Versioning

Releases are named for the day they were made and a revision within that day:

2026-08-21_01      first build on the 21st
2026-08-21_02      second build the same day
2026-09-01_01

The revision is padded to two digits so that versions sort correctly as text — without it, revision 10 would sort before revision 2.

Packaging tools cannot use that form directly, so it is converted at the edge rather than kept as a second version string that could drift:

Where Form Why
the program, --version 2026-08-21_01 what you asked for
pip, pyproject.toml 2026.8.21.1 PEP 440 forbids dashes and underscores in a release
dpkg, the .deb 2026.08.21.01-1 Debian versions may not contain underscores

bandsaunter/__init__.py holds the date and revision; the other two forms are derived from it, and the test suite checks that all three describe the same release and that both pip and dpkg --compare-versions order them correctly.

Quick start

bandsaunter                                     # the menus: set up and scan
bandsaunter scan -b 2m -b marine-vhf            # band-plan presets
bandsaunter scan -r 144M-148M -r 420M-450M      # your own ranges
bandsaunter scan -b 2m --simulate               # try it without hardware

Two ways to drive it

Everything is available both ways. Run bandsaunter with no arguments for the menus, or pass flags for scripting — the two are generated from the same table of settings, so neither can offer something the other cannot.

  1  Frequency ranges       3 configured
  2  Band plan              107 US presets
  3  Settings               record no limit, hang 6s, squelch +12 dB, keep voice, cw
  4  Saved settings and profiles
  h  Help
  s  Start scanning
  q  Quit

Settings are grouped, show their current value against the built-in default, and carry their own help:

   #  setting                 value        what it does
   1  Record for *            no limit     longest one signal may hold the receiver
   2  Wait for quiet *        6 s          quiet time before the sweep resumes
   3  Absolute limit          900 s        ceiling on one capture, even when 'Record for' is 0
* differs from the built-in default

Number to change it, ?N for help on one, d to reset the group, blank to go back.

?2 explains a setting in full, including the command-line flag that does the same thing. Typing a search term instead of a number finds settings by any word in their name or description — voice score finds the speech threshold.

Values may be typed with their units: 5 s, 2.048 MHz, 12 dB, 48k, or no limit for the settings that accept 0.

Settings that persist

Settings are saved to ~/.config/bandsaunter/config.yaml and picked up by every later run. Save them from the menus (4 → s) or from the command line:

bandsaunter config                          # open the settings menu
bandsaunter config hang_seconds=6 record_seconds=0   # set and save directly
bandsaunter config --show                   # every setting, with defaults
bandsaunter config --describe hang_seconds  # explain one in full
bandsaunter config --path                   # where the file lives
bandsaunter config --reset                  # back to defaults

Three layers apply in order, each overriding the last:

  1. the saved settings file
  2. a named profile, if --profile is given
  3. any flags on the command line

So a saved squelch of 12 dB stays in force while --hang 1.5 overrides just the hang for one run. --no-config ignores the file entirely; --save stores the resulting settings as the new default.

Named profiles live beside it in the same directory:

bandsaunter scan -b 2m -b 70cm --record 0 --hang 6 --save-profile local
bandsaunter scan -p local
bandsaunter profiles

Entering frequencies

By hand — repeat -r as many times as you like; there is no limit on the number of start/end pairs.

bandsaunter scan -r 144M-148M -r 462.5M-467.8M -r 929M-932M

A range is start-end, with optional /step and @mode:

Form Meaning
144M-148M explicit start and end
144-148M the unit carries over to the left end
146.52M a single frequency
144M-148M/25k with a channel step
144M-148M/25k@nfm and a forced demodulator

Units may be written 144M, 144 MHz, 144000k, or plain Hz. A bare number below 10000 is read as MHz, so -r 162.4-162.55 does what you expect.

From the US band plan — 107 presets across 18 categories:

bandsaunter bands --categories          # list categories
bandsaunter bands --category Aviation   # everything in one category
bandsaunter bands pager                 # search
bandsaunter scan -b gmrs -b railroad -b noaa-weather

Each preset carries its own channel spacing, demodulator and bandwidth, so -b marine-vhf scans 25 kHz channels in NFM while -b fm-broadcast uses 200 kHz WFM, without being told.

Each amateur band also has a complete entry that covers the whole band and picks the demodulator per segment, because a band is not one mode:

bandsaunter scan -b 2m-complete

scans 144-148 MHz as CW below 144.1, SSB to 144.3 and FM above it, so a CW beacon at the bottom is decoded to text while a repeater at the top is demodulated as FM — in one sweep. There is one for every band from 160 m to 33 cm: 160m-complete, 80m-complete, ... 70cm-complete, 33cm-complete.

Where an amateur band overlaps another service the amateur reading wins inside a complete-band sweep — 433 MHz is treated as 70 cm rather than as the ISM band it shares — while scanning -r 433.9M on its own still treats it as ISM.

A few presets stand for a set of others, so scattered segments can be picked in one go:

bandsaunter scan -b all-cw --record 0 --hang 6

all-cw covers every CW allocation in the plan — 160, 80, 40, 30, 20, 17, 15, 12, 10, 6 and 2 metres — as eleven separate ranges rather than one span from 1.8 to 144 MHz. That is 1.25 MHz of spectrum in total, so a full pass takes under a second and CW gets decoded to text as it turns up. Direct sampling switches itself on for the HF segments and off again from 12 m upward; the HF part needs an HF antenna to be worth anything.

The menus do both: browse the band plan by category, or type in start/end pairs one after another. Ranges can be listed, removed, toggled on and off, and have their demodulator changed from the ranges menu.

The two dwell settings

These are the settings that decide how the scanner behaves when it finds something:

Setting Flag What it does
Record for X seconds before continuing --record 30 The longest a single signal may hold the receiver. 0 means stay as long as it keeps transmitting.
Wait for X seconds of no signal before continuing --hang 3 How long the channel must stay quiet before the sweep resumes. Gaps shorter than this are recorded straight through.
bandsaunter scan -b 2m --record 30 --hang 3

Whichever comes first wins, and that is worth being clear about: --record overrides --hang. A transmission still in progress at the record limit is cut off there, however long the hang time is. If a recording keeps ending at exactly 30 seconds, that is the default record limit doing it, not the hang — set --record 0. The scan reports it when this happens.

Both are measured in samples, not wall-clock time, so a 30 second setting produces a 30.0 second recording.

Supporting settings:

  • --min-record 0.5 — discard anything shorter, so brief noise spikes leave nothing behind on disk.
  • --revisit 8 — ignore a frequency for this long after recording it, so a busy repeater does not monopolise the sweep.
  • --max-record 900 — absolute ceiling on one capture, applied even when --record is 0.
  • --threshold 8 — squelch, in dB above the measured noise floor.

Capturing both sides of a conversation

--hang is what holds a recording open across the natural pauses in two-way traffic. Set it longer than the gap between overs and the whole exchange lands in one file:

bandsaunter scan -b 2m --record 0 --hang 6

--record 0 is the important half. Without it the per-signal cap cuts the exchange off mid-sentence no matter what the hang time is — the default 30 seconds is a common surprise. --max-record (default 900 s) still bounds an unlimited capture so nothing runs away.

To make it the permanent default:

bandsaunter config record_seconds=0 hang_seconds=6

"Quiet" means no real signal, not merely a closed squelch. Silence, static and interference all count towards the timer, so a burst of noise during a pause does not reset it and park the receiver on a finished conversation. Recognising that a signal carries nothing takes a couple of seconds of evidence, so expect the tail to run a little past --hang in that case.

Once a capture has produced real content it is never abandoned as noise, since a quiet spell between overs would otherwise throw the conversation away.

Only real signals get recorded

A power threshold cannot tell a transmission from a hump of interference, so every capture is checked for content before it is kept. Recording happens only for:

Category What it means
voice speech structure in the demodulated audio: a pitch track in the 70-400 Hz range that drifts the way intonation does, pauses between phrases, syllable-rate envelope modulation, and formants that move
cw a keyed carrier whose timing resolves as Morse
digital an identified keying scheme: discrete FSK levels, an M-PSK phase line, or on-off keying -- corroborated by a symbol rate

Everything else -- static, hum, switch-mode power supply harmonics, clock spurs, bare carriers -- is discarded, and the files it wrote are deleted.

The check runs while the capture is still going, so interference is dropped after a second or two instead of holding the receiver for the whole record time.

bandsaunter scan -b 2m                          # voice, CW and digital (default)
bandsaunter scan -b 2m --accept voice           # voice only
bandsaunter scan -b ism-433 --accept digital    # data bursts only
bandsaunter scan -b 2m --keep-carriers          # also keep unmodulated carriers
bandsaunter scan -b 2m --keep-everything        # no content check at all

Tuning knobs: --min-voice-score (0-1, default 0.45) sets how speech-like audio must be; --min-signal-score sets the confidence needed to keep anything; --verify-max caps how long a contentless capture is given to prove itself.

What makes this hold up against interference

Static is good at imitating most of the things that look like structure, so each test is built so that noise cannot pass it:

  • A pitch track that moves gates the voice score, rather than contributing a share of it. Dynamics, syllable-rate modulation and energy in the voice band are all things hiss does too; weighted alongside voicing they were enough to carry noise over the line on their own. Speech is the only thing here that produces a pitch period that drifts, so nothing is called voice without one. A steady tone or mains hum has a perfectly stable "pitch" and is rejected for exactly that reason. Measuring drift needs several voiced frames, so the requirement eases for a short over that cannot supply them; the steady-tone guard still applies.
  • A symbol rate has to hold still across the capture. The estimator always returns its best peak, so on noise it reports one rate for the first half of a capture and a different one for the second. Real data keeps one.
  • Keying has to land on a grid. On/off contrast alone is not evidence: a signal fading across the squelch produces plenty of it, with run lengths that fit no symbol period at all.
  • A phase line has to be created by the exponentiation. An unmodulated carrier -- including the gaps between phrases on an FM channel -- already has a line at every power, and would otherwise look like textbook PSK.
  • The demodulator is chosen from the signal, not from the band plan, by probing briefly before recording. An AM signal inside a band listed as FM would otherwise be recorded through the wrong detector, giving audio that is useless to listen to and impossible to judge. Three ratios settle it — how much the envelope varies, how far the tone swings, and how much power sits in a carrier — because those hold steady over a fraction of a second whatever is being said. Running the full classifier on so short a probe was tried and is not reliable: speech makes any modulation look bursty over half a second, and AM came back as on-off keying while FM came back as AM. The probe is played into the recording rather than discarded, so a short over does not lose its opening.
  • Content is judged only on the audio that was actually recorded, and only on the stretches the squelch called signal. Demodulating two ways and keeping whichever scored higher is cherry-picking, and on noise one of the two always flatters it.

The gate needs roughly two seconds of audio to judge speech reliably, so pair it with --record 5 or more rather than very short capture limits.

Keeping up with the radio

An RTL-SDR only delivers samples while the host is actively reading. Anything that arrives while the program is busy demodulating is discarded by the driver, and a recording then holds less than really went by -- which plays back too fast.

Two things keep that from happening:

  • Captures stream asynchronously. A ring of USB transfers stays queued in a background thread, so the dongle is never waiting for the host. Sweeping still uses plain reads, because each dwell is an independent snapshot and a gap between them costs nothing.
  • The signal path is fast enough to keep up. Decimation computes only the samples that survive, rather than filtering at the input rate and throwing away seven of every eight outputs; the quarter-rate local oscillator is the four-step cycle 1, -j, -1, +j and needs no trigonometry; and pitch tracking runs through the FFT instead of a direct autocorrelation per frame. Together those took the capture loop from 65% of the real-time budget to under 10%.

If the host does fall behind anyway, the scan reports how many samples were lost rather than silently producing a fast recording.

Why the threshold is what it is

The sweep uses peak-hold, which keeps the largest value each FFT bin reached during the dwell. That finds bursty traffic that averaging would bury -- but it also means noise alone rides several dB above the measured floor. On this hardware, empty spectrum reaches 5-9 dB above a percentile floor with nothing transmitting.

So --threshold is a margin over noise, not over the floor: the offset that noise alone clears is computed from the detector (segment count and bin count) and added automatically. A threshold of 8 means 8 dB of real headroom. It is deliberately not measured from the spectrum -- a spread estimated from the data reads five times higher across the packed broadcast FM band than on empty spectrum, which would suppress exactly the stations you are looking for.

Signal identification

Every recording is classified from its own IQ. The classifier measures occupied bandwidth, envelope statistics, discriminator levels, phase behaviour, spectral flatness and symbol rate, then combines those with the frequency to name the signal:

Family Recognised as
Analogue voice Narrowband FM (with CTCSS tone or DCS), wideband FM (stereo pilot detected), AM, SSB (USB/LSB)
CW Keyed carrier, decoded to text with the speed in WPM
Digital voice P25 C4FM, DMR (TDMA burst structure), NXDN, D-STAR
Data POCSAG and FLEX paging, ACARS, AIS, APRS/AFSK1200, 2-FSK and 4-FSK, BPSK/QPSK/8-PSK
Other Unmodulated carriers, OOK/ISM devices, ADS-B and UAT, DME/TACAN pulses, wideband OFDM/cellular

Each result carries a confidence and the reasoning behind it:

146.520038 MHz   3.0s  SNR 27.6 dB  Narrowband FM voice (CTCSS 100.0 Hz) (88%)
                       4.9 kHz wide, 0.9 kHz rms deviation

Low SNR reduces confidence rather than producing a confident wrong answer.

CW / Morse

Keyed carriers are decoded to text. The speed is measured from the signal, so nothing has to be configured, and anything from about 8 to 40 WPM reads reliably:

144.1 MHz  12.0s  SNR 50.8 dB  CW / Morse at 18 WPM  CW "VVV DE W1AW FN31"

The decoder runs its own CW detector over the captured IQ, so Morse is found even when the recording itself was made in FM or SSB.

Single sideband

SSB needs no --mode usb. Nothing else does either, but SSB is the mode where it would matter: FM and AM detectors do not care where in their passband a signal sits, while an SSB demodulator is a filter that opens at the suppressed carrier. Tune to the middle of the voice — which is where a detector naturally lands, a couple of kilohertz up — and its lower half is filtered off while the rest comes out shifted down by the error. That is the mistuned sound, and it makes the recording useless rather than merely imperfect.

So the carrier is measured rather than assumed. Speech puts most of its power in the first few hundred hertz above the carrier, so an SSB signal's occupied band is lopsided: the loud end is the carrier end. That locates the carrier to within about a hundred hertz and names the sideband at the same time — energy bunched at the low edge is upper sideband, at the high edge lower.

The frequency in the filename is therefore the carrier, the one you would dial into a radio, not the middle of the voice.

Where the signal has no lean to read — a data mode inside an SSB segment, or a steady tone — the band plan decides, and it is right where the folklore is wrong: 60 m and the HF utility bands are upper sideband well below the 10 MHz that "LSB below, USB above" splits on.

Output

Everything lands in one directory, named frequency--yyyy-mm-dd_hour_minute_second-modulation.wav:

0014.058000MHz--2026-08-21_20_35_41-cw.wav
0098.299255MHz--2026-08-21_20_24_01-wfm.wav          demodulated audio
0098.299255MHz--2026-08-21_20_24_01-wfm.json         identification, features, timing
0098.299255MHz--2026-08-21_20_24_01-wfm.cf32         raw IQ         (with --iq)
0098.299255MHz--2026-08-21_20_24_01-wfm.sigmf-meta   SigMF sidecar  (with --iq)
0098.299255MHz--2026-08-21_20_24_01-wfm_transcription.txt  (with --transcribe)
0098.361991MHz--2026-08-21_20_24_18-wfm.wav
0146.520000MHz--2026-08-21_20_31_02-nfm.wav
1090.000000MHz--2026-08-21_20_38_12-raw.wav
scan_log.jsonl                                       one line per hit
scan_log.csv                                         the same, as a spreadsheet

Frequency leads and is padded to four digits, so a plain directory listing sorts by frequency across the whole tuning range — unpadded, 1090 MHz would sort before 146 MHz. Each channel's captures group together with the timestamp ordering them. Every artefact of one capture shares a stem, and the modulation suffix is what the signal was identified as, so the file is renamed once the capture has been analysed.

Where files go

The directory is asked for the first time bandsaunter is run and remembered afterwards:

Recordings, transcripts and the scan log are all written to one directory.
Where would you like them?

  recordings directory (~/bandsaunter):

It is an ordinary setting, so it can be changed at any time:

bandsaunter config output_dir=~/somewhere-else
bandsaunter scan -b 2m -o /tmp/just-this-once

The question is only asked when there is someone to answer it: a scan run from a script or with output redirected uses the default rather than blocking, and --no-config skips it entirely.

One file per frequency

With --combine, each frequency gets a single file that every later reception is appended to, so a whole watch on a channel plays back as one recording:

bandsaunter scan -b 2m --combine --record 0 --hang 6
recordings/
  146.520000MHz.wav      every transmission heard on 146.520, in order
  147.100000MHz.wav
  146.520692MHz--2026-08-21_13_21_20-nfm.json   what each capture was
  scan_log.csv

Each transmission is preceded by its date and time read aloud, so the file says when everything was heard without needing the log open. Receptions within --combine-tolerance (6.25 kHz by default) of each other count as the same channel, which covers the few hundred hertz a detection wanders by.

The file is rewritten to stay valid after every append, so it can be opened and played while the scan is still running, and a scan stopped part-way still leaves a working recording. A later run continues the same file rather than starting a new one.

By default the per-transmission WAVs are removed once they have been added, so each recording exists in one place; --keep-individual keeps both. The .json describing each capture is written either way.

--no-announce leaves the timestamps out, and --announce-frequency reads the frequency out as well.

Where the speech comes from

The announcements use an installed text-to-speech program if there is one (espeak-ng, espeak, pico2wave, flite, say) and fall back to a built-in formant synthesiser otherwise, so the feature works with nothing else installed. --announce-engine builtin forces the built-in one; the scan plan reports which is in use.

The two are given different wording. An installed engine gets ordinary text, where punctuation is what produces the phrasing — and the obvious spellings are traps: espeak-ng reads 14:38:05 as "fourteen thirty, eight zero five", and an ISO date as "two thousand and twenty six dash zero eight dash twenty one". It is given August 21, twenty twenty six, at 14 38 and 05 seconds instead. The built-in synthesiser gets the word list it has pronunciations for. Both are level-matched, so switching between them does not change how loud the announcements sit against the recordings.

It only has to say numbers, month names and a handful of words, which makes direct synthesis practical: a glottal source through a cascade of three formant resonators, with the formant tracks interpolated between phonemes. The test suite checks by LPC analysis that all ten vowels come out within 130 Hz of their intended first formant and 250 Hz of their second — summing the resonators in parallel rather than cascading them loses the first formant entirely and makes every vowel sound the same.

Transcribing speech to text

With --transcribe, anything the content check identified as voice is passed to a speech recogniser and the words are written beside the recording:

bandsaunter scan -b 2m --transcribe
146.520000MHz--2026-08-21_12_18_38-nfm.wav
146.520000MHz--2026-08-21_12_18_38-nfm_transcription.txt
146.520000MHz--2026-08-21_12_18_38-nfm.json

Only voice is transcribed — running a recogniser over Morse or a data burst costs seconds and produces nothing. CW is decoded separately and appears in the metadata as text already.

A capture with nothing recognisable in it produces no file. Music, a carrier with an open mic, a fragment too short to make out: nothing is written, rather than a directory of placeholders. The transcript is also copied into the capture's .json, which names it only once it exists — so the metadata never points at a file that was never created. The scan reports the tally at the end:

  2 transcript(s) written, 1 with no recognisable speech

Recognition takes seconds per capture, far longer than a capture itself, so it runs on its own thread and the scan never waits for it; anything still queued is finished when the scan stops. When recordings are being combined by frequency there is one transcript per frequency too, each line stamped with the time:

[2026-08-21 12:18:38] this is what the first transmission said
[2026-08-21 12:24:02] and this is the second

Getting a recogniser

Unlike the spoken announcements, this needs an installed engine — recognition depends on a trained model, so there is no built-in fallback.

bandsaunter transcribe --engines      # what is installed
sudo apt install bandsaunter-transcribe   # from your own repository
pip install faster-whisper            # or straight from PyPI
Engine Notes
faster-whisper best on radio audio; ~120 MB of dependencies, model downloads on first use
whisper the original; heavier
whisper-cli whisper.cpp, no Python dependencies
vosk ~10 MB plus a 40 MB model, fully offline, but weaker on noisy audio
pocketsphinx tiny; poor on radio audio

The difference is easy to measure. Both engines on the same 26-second off-air recording:

faster-whisper   3.4s   "August 21, 2026, at 13.42 and 28 seconds,
                          96.108 megahertz, 13.42 and 47 seconds, ..."
vosk            11.2s   "august twenty one twenty twenty six at thirteen
                          forty two i'm twenty eight seconds ... forty
                          family factories ..."

Whisper is both more accurate and three times faster, and it punctuates. Vosk's advantage is size and that it needs nothing after its model is downloaded once.

--transcribe-model selects the size (tiny.en, base.en, small.en, medium.en) and --transcribe-language fixes the language — worth setting, since on a short noisy clip automatic detection often guesses wrong and returns nonsense in another language.

Existing recordings can be transcribed after the fact:

bandsaunter transcribe recordings/                 # every WAV in a directory
bandsaunter transcribe one.wav --stdout

Re-examine anything later:

bandsaunter analyze recordings/2026-08-19/.../iq.cf32     # identify
bandsaunter analyze recordings/2026-08-19/.../audio.wav   # decode CW

Live controls

The display sizes itself to the terminal, giving up the spectrum row, then the hit list, then the key hints as space runs short. A frame taller than the terminal cannot be redrawn in place, so an oversized one would leave a copy of itself behind on every refresh.

For the same reason the driver's own messages are suppressed while a scan runs: librtlsdr writes them straight to file descriptor 2 from C — including Allocating 15 zero-copy buffers on every capture — and they draw over the display and break its cursor tracking. bandsaunter devices still shows them, since that is the command to run when something is wrong, and BANDSAUNTER_DRIVER_MESSAGES=1 restores them everywhere.

Key Action
q stop
p pause / resume
s skip this signal, resume sweeping
l lock out this frequency for the rest of the run
+ / - adjust the squelch threshold

Built-in help

Press h in the menus for topics covering setup, how the sweep works, why nothing (or too much) is being recorded, capturing conversations, where files go, HF reception and the keys available during a scan. Typing a setting name there explains that setting instead.

From the command line, bandsaunter config --describe <setting> does the same, and bandsaunter scan --help lists every flag grouped the same way as the menus.

HF

Frequencies below 24 MHz need direct sampling, which most RTL-SDR dongles support on the Q branch. It is selected automatically:

bandsaunter scan -b 40m-cw --record 60      # 40 m CW, decoded to text
bandsaunter scan -b am-broadcast

You will need an HF antenna; the tuner is bypassed in this mode, so there is no front-end filtering or gain.

How the sweep works

  • The band is covered in steps of sample_rate x usable_fraction / 2. The local oscillator is parked below the span each step covers, so the RTL2832's DC spike never lands inside the frequencies being searched.
  • The noise floor is measured per FFT bin as a sliding low percentile, which follows the receiver's passband shape and steps over signals. There is no warm-up period, and a station that transmits constantly does not learn itself into the floor.
  • The sweep uses peak-hold rather than averaging across each dwell, so bursty traffic — CW, packet, a short over — is not averaged into the noise.
  • On a hit, the receiver retunes with a quarter-rate LO offset (moving the DC spike off the signal), probes once at ~60 Hz resolution to measure the real occupied bandwidth, and picks the demodulator from that plus the band plan.

Without hardware

--simulate swaps in a synthetic receiver carrying one of each interesting signal type, which is also what the test suite runs against:

bandsaunter scan -r 144M-148M --simulate

Testing

python -m pytest

Covers DSP invariants, frequency parsing, the classifier against synthetic signals at several SNRs and random seeds, Morse decoding from 8 to 40 WPM, the voice detector against synthetic speech and against noise, tones and hum, and full scan runs through the simulator checking that --record and --hang are obeyed, that static and bare carriers are never written to disk, and that audio, IQ and metadata are correct.

The simulator's voice transmitters carry synthesised speech -- glottal pulses through moving formants, compressed the way a real transmitter compresses, then gated into syllables and phrases -- because sine tones would not exercise the speech detector at all.

Receiving is not the same as being allowed to use or divulge what you hear. In the US, the ECPA prohibits intercepting cellular and other private communications, and rebroadcasting or acting on what you receive is separately restricted. Check your local rules.