Fix pre-existing compilation failures in four test call sites that were
missing the wfm_denoise: bool argument added to ChannelDsp::new() and
SdrPipeline::start().
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
- Raise audio LPF cutoff from 15 kHz to 17 kHz to pass full FM stereo
audio bandwidth without excessive HF rolloff
- Replace 2-point linear interpolation resampler with 4-point Hermite
cubic spline for a much flatter passband up to 17 kHz
- Add FM discriminator gain normalization (fm_gain = fs / 150000) so
±75 kHz deviation maps to ±1.0 regardless of composite sample rate,
stabilizing stereo carrier amplitude reconstruction
- Double pilot PLL proportional (0.0015→0.003) and integral
(0.00002→0.00005) gains for faster lock and better tracking
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
The 19 kHz pilot notch was applied only to the L+R sum path, introducing
~22° of phase shift at 15 kHz relative to the L-R diff path. This phase
mismatch caused interchannel crosstalk (≈ −14 dB separation at 15 kHz).
Fix: remove the notch from the sum processing chain so both sum and diff
pass through identical 4th-order Butterworth LPFs, giving phase-coherent
demodulation across the full audio band. The notch is relocated to the
mono output branch where phase alignment with the diff channel is not
required. Pilot rejection on the stereo L/R outputs is still adequate
(~28 dB) from the combined LPF + deemphasis response at 19 kHz.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add a server-side toggle for the multiband stereo denoiser so it can be
enabled or disabled at runtime without restarting the server.
Backend (trx-backend-soapysdr):
- Add `denoise_enabled: bool` to `WfmStereoDecoder`; gate multiband
blend behind it (falls back to uniform single-band blend when off)
- Add `set_denoise_enabled()` method on `WfmStereoDecoder`
- Propagate `wfm_denoise: bool` through `ChannelDsp`, `SdrPipeline`,
and `SoapySdrRig`; add `set_wfm_denoise()` at each layer
- Include `wfm_denoise` in `filter_state()` so it flows into snapshots
Protocol / core (trx-core, trx-protocol, trx-server):
- Add `SetWfmDenoise(bool)` to `RigCommand` and `ClientCommand`
- Add default `set_wfm_denoise()` trait method to `RigCat`
- Handle `SetWfmDenoise` in `rig_task.rs` and update `RigFilterState`
- Add `wfm_denoise: bool` (default `true`) to `RigFilterState`
Frontend (trx-frontend-http):
- Add `POST /toggle_wfm_denoise` endpoint
- Add "Denoise On/Off" button next to the stereo/mono audio picker
- Sync button state from SSE filter snapshot (`update.filter.wfm_denoise`)
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Split the L-R diff channel into three frequency bands at audio rate and
apply SNR-weighted blending per band driven by pilot magnitude:
- 0–2 kHz: blend¹ (most stereo — low frequencies have best SNR)
- 2–8 kHz: blend² (moderate noise reduction)
- 8–15 kHz: blend⁴ (aggressive noise reduction — hiss-prone range)
Move blend from composite rate to audio rate so the crossover filters
(2nd-order Butterworth at 2 kHz and 8 kHz) operate at 48 kHz and the
pilot blend is linearly interpolated per audio sample for smooth
transitions. Unblended diff is now stored in prev_diff; prev_blend
tracks the blend value for the same interpolation.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Three bugs made the AM path sound wrong:
1. AGC attack too fast (5 ms). The slowest audio a broadcast AM station
can transmit is ~50 Hz (20 ms period). A 5 ms attack lets the AGC track
individual audio cycles, which causes severe pumping and amplitude
distortion. Change to 500 ms attack / 5 s release so the AGC only
responds to slow carrier-amplitude fading, not the audio modulation itself.
2. Bandwidth too narrow. The IQ filter cutoff is audio_bandwidth_hz / 2,
so the previous 6 000 Hz setting gave only 3 kHz audio bandwidth.
AM broadcast sidebands extend to ±4.5–5 kHz; raise the default to
12 000 (cutoff 6 kHz) to cover the full audio band.
3. DC blocker rate inconsistent. For AM the demodulated magnitude is
always ≥ 0 and the DC component equals the carrier amplitude; only true
DC needs removing. Unify all non-WFM modes to r = 0.9999 (corner
≈ 0.76 Hz @ 48 kHz), which strips carrier DC without touching any
audible bass content.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
CW signals in SDR are centred at an audio offset (e.g. 700 Hz) by the
upstream FIR filter, so demodulating as USB (taking the real part) produces
the correct side-tone. The previous magnitude/envelope approach produced a
DC pulse per key press with no audible tone.
Re-enable the DC blocker for CW/CWR (r = 0.9999): the output is now audio
that can carry a DC offset from BFO frequency error, identical to USB.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add SoftAgc — a fast-attack/slow-release envelope AGC with a max-gain cap
— to all demodulated audio paths so that switching between modes (WFM, AM,
SSB, CW, FM) no longer produces large volume jumps. AGC is applied after
every demodulator, including WFM, with a shared target level of 0.5.
Add per-mode DC blocking (DcBlocker) for USB/LSB/AM/FM/DIG paths to remove
carrier frequency-offset DC from the FM discriminator and LO bleedthrough in
SSB. CW is excluded because high-passing a non-negative envelope creates
negative-going artifacts on each key release; WFM already has internal DC
blockers on each output channel.
AGC time constants are tuned per mode:
CW/CWR – 1 ms attack / 50 ms release (follows individual dots/dashes)
AM – 5 ms attack / 200 ms release (tracks fading carriers)
all else– 5 ms attack / 500 ms release (suits voice and data)
Simplify demod_am and demod_cw: remove per-block peak normalisation and DC
removal that caused block-boundary level discontinuities ("pumping"). Both
now return raw magnitudes and rely on the downstream DC blocker and AGC for
normalisation.
DIG is already wired as Passthrough (identical to USB); no change needed.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Replace one-pole sum/diff filters with a proper 4th-order Butterworth
cascade (Q = 0.5412 / 1.3066) at 15 kHz. This reduces pilot tone
leakage from −4 dB to −12 dB at 19 kHz and suppresses the 38 kHz DSB
carrier from −9 dB to −32 dB, significantly improving stereo crosstalk.
Add a biquad notch at 19 kHz on the L+R channel to eliminate the residual
pilot tone that would otherwise be audible after downsampling to 48 kHz.
Replace nearest-neighbor (sample-hold) resampling with linear interpolation
inside WfmStereoDecoder. The output sample is now placed at the exact
fractional position between the two adjacent composite samples using the
phase accumulator state, removing timing jitter and harmonic distortion on
sustained tones.
Add DC blockers (pole at 0.9999, corner ≈ 0.75 Hz at 48 kHz) to all audio
outputs to remove carrier frequency-offset DC from the FM discriminator
without any audible bass roll-off.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Make the primary SoapySDR DSP channel follow the tuned\nfrequency so RDS decoding stays aligned with the active\nfrequency rather than the hardware center.\n\nMove the default WFM deemphasis setting to server SDR\nconfig and default it to 50 us, then pass that value into\nthe SoapySDR backend.\n\nCo-authored-by: Codex <codex@openai.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add server-side debug log when RDS data is decoded (PI, PS, PTY).
Extend the RDS panel with active mode, frame counter, and a raw JSON
dump of the last spectrum frame (bins excluded) to help diagnose why
RDS remains absent from the SSE stream.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
The FIR LPF cutoff was audio_bandwidth_hz/2; with wfm_bandwidth_hz=75000
this gave 37.5 kHz, stripping the 57 kHz RDS subcarrier before FM
demodulation. Clamp the IQ filter bandwidth to at least 130 kHz (cutoff
≥ 65 kHz) for WFM so the RDS subcarrier always reaches the decoder,
regardless of the configured audio bandwidth.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
JS: fix stereo AudioData channel copy — frame.copyTo with planeIndex:0
only fills the left plane; reading it as interleaved data caused every
other sample to be zero, making WFM stereo play at half speed. Now
calls copyTo per channel with the correct planeIndex.
Rust WFM: replace integer output_decim/output_counter in WfmStereoDecoder
with a fractional phase accumulator (output_phase_inc = audio_rate /
composite_rate). Integer division caused the effective output rate to
drift from audio_sample_rate when the SDR rate is not an exact multiple
(e.g. 2 MHz SDR → 250 kHz composite → ~50 kHz output instead of 48 kHz,
making audio play 4% slow).
Rust non-WFM: add resample_phase/resample_phase_inc to ChannelDsp and
use a fractional-phase resampler in process_block for non-WFM paths,
ensuring exactly audio_sample_rate samples/sec regardless of SDR rate.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Fix waterfall overview freezing at steady state by tracking a monotonic
push counter instead of row array length — the array size stays constant
once the waterfall is full, so the previous row-count comparison never
triggered the incremental draw path.
Fix WFM RDS not decoding when switching to WFM from a narrowband mode:
set_mode now resets audio_bandwidth_hz to the mode-appropriate default
(180 kHz for WFM) before rebuilding the FIR, preventing the 57 kHz RDS
subcarrier from being filtered out.
Add 1×/10×/100× multiplier button group next to the jog unit selector.
jogUnit × jogMult gives the effective jog step; both are persisted to
localStorage independently.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
- Compute hardware center as dial - center_offset_hz (was ignoring offset)
- Pass configured bandwidth_hz to device instead of hardcoded 1.5 MHz
- Add retune_cmd channel so set_freq repoints SDR hardware in real time
- Auto-add default channel with mode-appropriate bandwidth when [[sdr.channels]]
is empty, preventing silent audio with minimal config
- Add ChannelDsp::set_filter to rebuild FIR LPFs at runtime; wire
set_bandwidth and set_fir_taps to call it so UI filter changes take effect
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add SpectrumData struct (bins, center_hz, sample_rate) to RigState and
RigSnapshot. Add GetSpectrum RigCommand and ClientCommand plumbed through
the protocol layer. SoapySDR DSP pipeline now computes a 1024-bin FFT
(Hann window, FFT-shifted, dBFS) every 4 IQ blocks (~10 Hz update rate)
and exposes it via RigCat::get_spectrum(). The rig_task handles
GetSpectrum without persisting spectrum data in ongoing state.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Three root causes prevented APRS decoding at 144.800 MHz with PKT/FM mode:
1. `RealIqSource::read_into` returned zeros — the SoapySDR streaming API
was never wired up. `RxStream<Complex<f32>>` is `Send` and
`StreamSample` is implemented for `num_complex::Complex<f32>` in the
soapysdr 0.3 crate, so the stream can read directly into the IQ buffer.
Now creates and activates an `RxStream` in `new()` and calls
`stream.read` in `read_into`.
2. PKT mode used `Passthrough` (take `.re`) demodulation. VHF/UHF packet
radio (APRS, AX.25) is FM-encoded AFSK — it must be FM-demodulated
before the APRS decoder sees the audio tones. Changed PKT to `Fm`.
3. `iq_read_loop` always slept `block_duration_ms` after each read. Real
hardware already blocks inside `read_into`; the extra sleep doubled
latency. Added `IqSource::is_blocking()` (default `false`; `true` for
`RealIqSource`) and skip the throttle sleep for blocking sources.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Replace the per-sample ring-buffer FIR convolution with block-level
overlap-save convolution using rustfft. For a block of M samples and
N taps the old approach costs O(N·M); the new one costs O(M log M),
with rustfft using SIMD (AVX2/SSE4) internally.
Key changes:
- Add rustfft = "6" dependency
- Add BlockFirFilter: overlap-save filter with pre-computed H(f) and
a single forward+inverse FFT pair per block (no per-sample multiply)
- ChannelDsp.process_block() now:
1. Batch-mixes entire block to baseband in one vectorisable loop
2. Applies BlockFirFilter to I and Q (one FFT pair each)
3. Decimates and demodulates as before
- Keep the old FirFilter for unit tests (sample-by-sample interface)
- Add BlockFirFilter unit tests (DC passthrough, length preservation)
- IQ_BLOCK_SIZE promoted to pub const for use in filter sizing
For the default config (4096-sample blocks, 64 taps, decim=40):
Old: ~262144 multiply-adds per FIR × 2 components = ~524k per block
New: ~2 × (3 × 8192 × log2(8192)) ops, all SIMD-vectorised by rustfft
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Enhance error handling and messaging for SoapySDR device initialization:
- Add fallback logic to try empty args if device-specific args fail
- Provide detailed multi-line error messages with troubleshooting guidance
- Include suggestions to check SoapySDR plugins and run SoapySDRUtil --probe
- Clarify device lifetime management in struct
- Document why actual streaming not yet implemented (soapysdr 0.3 limitations)
- Note that sdr 0.3 requires FFI or upgrade for streaming support
This helps users diagnose device initialization failures and understand
the current architectural limitations.
Co-Authored-By: Claude Haiku 4.5 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Drop optional feature gating - SoapySDR hardware support is now required.
Make soapysdr a required dependency in Cargo.toml instead of optional.
Update server to always enable soapysdr backend and its dependencies.
Simplify initialization to always use RealIqSource instead of conditional
fallback to MockIqSource.
This assumes SoapySDR library is installed on the system.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Add RealIqSource that connects to actual SoapySDR devices when soapysdr-sys
feature is enabled. Implements proper device initialization, frequency/bandwidth
configuration, gain control, and RX stream management.
When soapysdr-sys feature is disabled (default), falls back to MockIqSource
for testing. Update feature flags in Cargo.toml dependencies to support both
real hardware and mock operation.
- Device initialization with proper error handling
- RX frequency, bandwidth, and gain configuration
- IQ sample streaming via broadcast channel
- Proper resource cleanup via Drop trait
- Throttled MockIqSource to prevent 100% CPU
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Add sleep proportional to block duration in iq_read_loop to simulate
real hardware timing. MockIqSource immediately returns samples without
any delay, causing busy-looping. Throttling prevents excessive CPU usage
when using the mock source.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Fix useless vec! allocation in demod tests, improve loop iterator usage,
and reformat code for better readability across dummy and soapysdr
backends.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Set tx/tx_limit/vfo_switch/filter_controls/signal_meter on all backends:
- FT-817, FT-450D, dummy: tx=true, tx_limit=true, vfo_switch=true,
filter_controls=false, signal_meter=true
- SoapySDR: tx=false, tx_limit=false, vfo_switch=false,
filter_controls=true, signal_meter=true
SoapySDR backend now stores bandwidth_hz and fir_taps fields; overrides
set_bandwidth, set_fir_taps, and filter_state on RigCat to expose live
DSP state in the snapshot.
Add UC-08 unit tests on dummy backend asserting tx capabilities present
and filter_controls absent, and that filter_state returns None.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Replace the stub SoapySdrRig with a full implementation: wire up SdrPipeline
from dsp.rs, implement AudioSource::subscribe_pcm on the primary channel,
add gain control (manual/auto with fallback warning), and track primary
channel freq/mode so set_freq/set_mode update the live DSP pipeline.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add 9 unit tests covering all demodulators in demod.rs: USB/LSB/Passthrough
real-part extraction, AM DC removal and varying-envelope, FM tone frequency
and silence, CW peak normalisation, mode mapping, and empty-input safety.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
Add dsp.rs with IqSource trait abstraction, MockIqSource, windowed-sinc
FIR low-pass filter, ChannelDsp (mixer/decimate/demod/frame-accumulator),
and SdrPipeline which spawns a dedicated IQ read thread. No soapysdr crate
dependency; the real device will be wired in SDR-07.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
sjg
