Invalidate main-decoder windows whenever the rig's main frequency
changes, including direct SetFreq, scheduler-driven retunes, and
external CAT retunes observed during polling.
The decoder loops now resubscribe their PCM receivers on reset and drop
results that finish after the reset sequence advances, preventing false
decodes from stale pre-retune audio.
Co-Authored-By: OpenAI Codex <noreply@openai.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add NEON (aarch64) vectorized paths mirroring the existing AVX2 paths:
- demod/math_arm.rs: replace the no-op placeholder with a full NEON FM
discriminator that processes 4 samples per iteration using a 7th-order
minimax atan polynomial and branchless atan2 with argument reduction,
matching the accuracy of the AVX2 path (max error ~2.4e-7 rad).
32-bit ARM retains the scalar fallback.
- dsp/filter.rs: add mul_freq_domain_neon() that deinterleaves 4 complex
pairs via vuzpq/vzipq, performs complex multiply with vmulq/vaddq/vsubq,
then reinterleaves. On aarch64 this path is always taken (NEON is
mandatory); scalar fallback remains for other targets.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
clamp_bandwidth_for_mode now floors AMC bandwidth at 9 kHz so that both
the sum (L+R) and difference (L−R) sidebands are always captured,
regardless of what the user or a set_filter call requests.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
RigMode::AMC was implemented but omitted from the supported_modes vec,
so the HTTP frontend never received it in capabilities and the mode
selector did not show AMC-QUAM.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
- remote_client tests: add missing server_connected field and import
AtomicBool in the test module
- pskreporter: replace map_or(true, …) with is_none_or and
repeat(x).take(n) with repeat_n(x, n)
- dsp.rs, scheduler.rs: suppress intentional too_many_arguments with
#[allow(clippy::too_many_arguments)]
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add a RigMode::AMC arm to encode_mode returning an Err so that
attempting to set C-QUAM on the FT-450D returns a descriptive error.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add RigMode::AMC to the None return arm of encode_mode so that
attempting to set C-QUAM on the FT-817 returns a descriptive error
rather than a compile-time non-exhaustive match.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add RigMode::AMC arm to default_audio_bandwidth_for_mode, returning
9_000 Hz (same as AM).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add CquamDemod (demod/amcquam.rs) with first-order IIR carrier phase
tracker (τ = 50 ms) that rotates baseband IQ to align I with the sum
audio and Q with the difference audio, then DC-blocks each channel to
yield L/R stereo PCM.
Wire AmCQuam into the Demodulator enum, add ChannelDsp::cquam_decoder
field initialized for RigMode::AMC, and insert the C-QUAM audio path
between the WFM and fallback branches in process_block. Update all
mode-dispatch tables (agc_for_mode, iq_agc_for_mode, dc_for_mode,
default_bandwidth_for_mode, lib.rs, vchan_impl.rs) with AMC arms.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Drop fir_taps field from SdrChannelConfig and its default. Remove the
SetFirTaps dispatch arm from rig_task. The DSP layer now auto-calculates
tap count from audio_bandwidth_hz.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Replace the static fir_taps parameter with auto_taps(cutoff_norm) which
computes ceil(3.32 / cutoff_norm).clamp(63, 16383). This ensures the
filter transition band equals one passband width regardless of SDR sample
rate, giving correct image rejection when the user sets audio_bandwidth_hz.
At 912 ksps with 3 kHz audio bandwidth this yields ~2018 taps instead
of the previous hardcoded 64, eliminating the 114 kHz stopband gap that
caused adjacent-band signals to alias into the audio output.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add read_named_gain to IqSource (default: None) and implement it in
RealIqSource via Device::gain_element. Read the "LNA" element before
boxing the source so the initial sdr_lna_gain_db reflects the actual
hardware state, making the UI control visible and correct on first
connect. Devices without an LNA element (e.g. RTL-SDR with "TUNER")
return None and the control stays hidden.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Handle RigCommand::SetSdrLnaGain by calling set_sdr_lna_gain on the
rig and refreshing filter state, matching the pattern used by
SetSdrGain and SetSdrAgc.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add set_named_gain to the IqSource trait and implement it in
RealIqSource via soapysdr Device::set_gain_element. Wire a
lna_gain_cmd channel through SdrPipeline so the IQ read loop applies
LNA gain changes on the next iteration. Add set_sdr_lna_gain to
SoapySdrRig and expose the current value via filter_state.
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add hardware AGC on/off control for SoapySDR backend, wired through the
full stack from RigCommand to the web UI:
- RigCommand::SetSdrAgc(bool) + ClientCommand::SetSdrAgc in protocol
- set_sdr_agc() on RigCat trait (not-supported default)
- SoapySdrRig: agc_enabled field, set_sdr_agc() via pipeline agc_cmd,
sdr_agc_enabled in filter_state(); removes the "not yet implemented"
warning — gain_mode="auto" now properly enables hardware AGC via
SoapySDR set_gain_mode()
- IqSource::set_gain_mode() trait method; RealIqSource implements it
- SdrPipeline: agc_cmd channel, read loop applies it each iteration
- POST /set_sdr_agc endpoint in trx-frontend-http
- New "SDR settings" full-row in index.html with Hardware AGC checkbox
and RF Gain (moved out of WFM controls); row hidden when
show_sdr_gain_control is false
- app.js: AGC checkbox handler, disables RF gain input while AGC is on,
syncs checkbox state from filter.sdr_agc_enabled
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Three issues caused audible distortion on AM reception:
1. DC blocker shared r=0.9999 (τ≈1.25 s at 8 kHz) across all modes.
For AM the envelope detector outputs A_c+m(t) — always positive —
so the blocker needs to track the carrier bias quickly. AM now uses
r=0.999 (τ≈125 ms), 10× faster, while keeping the highpass cutoff
below 2 Hz so speech is unaffected.
2. Audio AGC time constants were inverted relative to good AM AGC design:
attack=200 ms (should be fast to prevent overload) and
release=3500 ms (unreasonably sluggish). Changed to attack=5 ms /
release=200 ms, target=0.5, max_gain=36 dB.
3. No IQ AGC before envelope detection meant carrier amplitude variation
went directly into the audio chain, forcing the slow audio AGC to
handle both RF level and audio level simultaneously. Added an AM IQ
AGC (attack=0.5 ms, release=50 ms, target=0.7, max=30 dB) that
normalizes carrier power before demod_am, so the DC blocker always
sees the same steady-state bias regardless of signal strength.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Three bugs caused USB/LSB to sound like AM:
1. The IQ low-pass filter was symmetric (passband ±BW/2), so both
sidebands were passed equally — taking .re then produced DSB-SC
rather than SSB audio.
2. cutoff_hz was computed as bandwidth_hz/2, halving the usable audio
bandwidth (1500 Hz for a 3 kHz USB channel).
3. demod_lsb claimed spectrum inversion was "handled upstream by
negating channel_if_hz", but that negation was never applied; USB
and LSB were functionally identical.
Fix: add a shift_norm parameter to build_fir_kernel / BlockFirFilterPair
that complex-modulates the time-domain FIR coefficients by
e^{j·2π·shift_norm·n}, shifting the passband in the frequency domain.
A new ssb_shift_norm() helper returns +cutoff_norm for USB/CW/DIG
([0, BW] Hz passband) and -cutoff_norm for LSB/CWR ([-BW, 0] Hz
passband); all other modes get 0.0 (symmetric LPF unchanged).
After the one-sided filter, taking .re correctly reconstructs the
selected sideband. No IF negation is needed for LSB.
Also fix two unit tests missing the force_mono_pcm argument introduced
after they were last updated.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Use millisecond-based slot indexing for FT4 decode windows in\nforeground and background decoders to avoid premature 7s resets.\n\nCo-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Use FT2-specific analysis settings and a wider receive span.
Switch server-side FT2 decoding to a rolling async window.
Widen FT2 candidate timing search in the vendored decoder.
Co-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Implement a distinct FT2 protocol path in the decoder stack and align\nits timing with the confirmed FT2 framing used by Decodium.\n\nCo-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stanislaw Grams <stanislawgrams@gmail.com>
Add periodic IGate position beacon support to the APRS-IS uplink.
New AprsFiConfig fields: beacon (bool), beacon_interval_secs (default
1200), beacon_symbol (default "/-"), latitude/longitude overrides.
A beacon is sent immediately on connect then every beacon_interval_secs.
Coordinates fall back from [aprsfi] to [general].latitude/longitude.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Allow specifying the IGate callsign directly in [aprsfi] instead of
relying on [general].callsign. The aprsfi-specific callsign takes
precedence; [general].callsign is used as fallback.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Move aprsfi and pskreporter modules from trx-server into a new
standalone trx-reporting library crate. Config types (AprsFiConfig,
PskReporterConfig) move to trx-reporting and are re-exported from
trx-server::config for backwards compatibility.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
- Append mandatory q-construct (,qAR,<callsign>) to all forwarded
TNC2 packets via updated format_tnc2(pkt, igate_call)
- Add TCPIP/TCPXX loop-prevention check before forwarding
- Drain server-sent data in select! loop to prevent TCP backpressure
- Enable TCP_NODELAY for low-latency packet forwarding
- Guard against history replays: skip packets older than 2 minutes
- Use "trx-rs" in login string and keepalive comment
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add [workspace.package] version = "0.1.0" to the root Cargo.toml and
switch all 21 member crates to version.workspace = true so the entire
workspace is versioned from a single place.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Two bugs introduced by 60697bb:
1. dsp.rs passed `channel_idx == 0` as force_mono_pcm, which forced the
primary pipeline channel to output mono samples. The Opus encoder was
configured for stereo, so it received half the expected frame data,
causing distortion for all connected audio clients.
Fixed by passing `false` — hidden virtual channels already set
force_mono_pcm=true via set_force_mono_pcm() in vchan_impl.rs.
2. main.rs short-circuited channel conversion when no audio clients were
connected, sending raw frames to pcm_tx (decoders). When clients then
connected, decoders switched to receiving stereo-interleaved frames,
making decoder input format dependent on client presence.
Fixed by always performing the channel conversion before sending to
pcm_tx; the no-client skip now only bypasses Opus encode + rx_audio_tx.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Skip APRS packets whose ts_ms is older than 120 seconds. Live RF-decoded
packets arrive within milliseconds; history replay items can be up to 24
hours old and must not be re-uploaded to APRS-IS as live traffic.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Rewrite the PSKReporter uplink to match the protocol spec exactly:
- Fix template FlowSetIDs: receiver uses 0x0003 (Options Template Set),
sender uses 0x0002 (Template Set); previously both used 0x9992/0x9993
- Add missing enterprise numbers (0x0000768F = 30351) to all enterprise
field specifiers in both template blocks
- Fix sender template field IDs: use correct attributes (senderCallsign
30351.1, frequency 30351.5, sNR 30351.6, iMD 30351.7, mode 30351.10,
informationSource 30351.11, senderLocator 30351.3, flowStartSeconds 150)
- Fix sender data field order to match the template declaration
- Add iMD byte (0) required by the 8-field template
- Add 4-byte null padding on receiver and sender data records
- Batch spots into one UDP packet per 5-minute window (spec requirement)
- Deduplicate by callsign within each window (keep most-recent spot)
- Send template descriptors only in first 3 packets then once per hour
- Increment sequence number by report count, not packet count
- Guard against history replays: drop any spot older than the flush
window (live FT8/WSPR is seconds old; history can be 24 h old)
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Remove hidden background decode channels when the owning audio client disconnects to avoid stale DSP and decoder buildup.\n\nCo-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Run FT8 and WSPR decode steps in blocking sections so the server listener stays responsive under decode load.\n\nCo-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Handle AUDIO_MSG_VCHAN_BW in the audio server path and apply per-channel filter bandwidth through the SoapySDR virtual channel manager.
Co-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
sjg
