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[feat](trx-backend-soapysdr): use fir resampler for wfm audio

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Co-authored-by: OpenAI Codex <codex@openai.com>
Signed-off-by: Stan Grams <sjg@haxx.space>
This commit is contained in:
Stan Grams
2026-02-28 23:17:15 +01:00
parent 96da7ca471
commit dbeec9fb39
1 changed files with 87 additions and 43 deletions
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src/trx-server/trx-backend/trx-backend-soapysdr/src/demod.rs
+87 -43
View File
@@ -28,18 +28,65 @@ const PILOT_BPF_Q: f32 = 20.0;
const STEREO_SEPARATION_PHASE_TRIM: f32 = 0.012; const STEREO_SEPARATION_PHASE_TRIM: f32 = 0.012;
/// Fixed gain trim on the recovered L-R channel. /// Fixed gain trim on the recovered L-R channel.
const STEREO_SEPARATION_GAIN: f32 = 1.006; const STEREO_SEPARATION_GAIN: f32 = 1.006;
/// Fractional-resampler FIR taps for WFM audio reconstruction.
const WFM_RESAMP_TAPS: usize = 6;
/// Polyphase slots for the WFM fractional FIR resampler.
const WFM_RESAMP_PHASES: usize = 32;
/// Slightly sub-Nyquist sinc cutoff to tame top-end imaging.
const WFM_RESAMP_CUTOFF: f32 = 0.94;
fn build_wfm_resample_bank() -> [[f32; WFM_RESAMP_TAPS]; WFM_RESAMP_PHASES] {
let mut bank = [[0.0; WFM_RESAMP_TAPS]; WFM_RESAMP_PHASES];
let anchor = (WFM_RESAMP_TAPS / 2 - 1) as f32;
for (phase_idx, phase) in bank.iter_mut().enumerate() {
let frac = phase_idx as f32 / WFM_RESAMP_PHASES as f32;
let center = anchor + frac;
let mut sum = 0.0_f32;
for (tap_idx, coeff) in phase.iter_mut().enumerate() {
let x = tap_idx as f32 - center;
let sinc = if x.abs() < 1e-6 {
WFM_RESAMP_CUTOFF
} else {
let arg = std::f32::consts::PI * x * WFM_RESAMP_CUTOFF;
arg.sin() / (std::f32::consts::PI * x)
};
let window = if WFM_RESAMP_TAPS == 1 {
1.0
} else {
let pos = tap_idx as f32 / (WFM_RESAMP_TAPS - 1) as f32;
0.5 - 0.5 * (2.0 * std::f32::consts::PI * pos).cos()
};
*coeff = sinc * window;
sum += *coeff;
}
if sum.abs() > 1e-9 {
let inv = 1.0 / sum;
for coeff in phase.iter_mut() {
*coeff *= inv;
}
}
}
bank
}
/// 4-point Hermite cubic interpolation.
///
/// `y` contains `[y_{n-1}, y_n, y_{n+1}, y_{n+2}]`; `frac` in `[0, 1)`
/// positions the output within the `y_n`–`y_{n+1}` interval.
#[inline] #[inline]
fn hermite4(y: &[f32; 4], frac: f32) -> f32 { fn shift_append<const N: usize>(hist: &mut [f32; N], sample: f32) {
let c0 = y[1]; hist.rotate_left(1);
let c1 = 0.5 * (y[2] - y[0]); hist[N - 1] = sample;
let c2 = y[0] - 2.5 * y[1] + 2.0 * y[2] - 0.5 * y[3]; }
let c3 = 0.5 * (y[3] - y[0]) + 1.5 * (y[1] - y[2]);
((c3 * frac + c2) * frac + c1) * frac + c0 #[inline]
fn polyphase_resample(
hist: &[f32; WFM_RESAMP_TAPS],
bank: &[[f32; WFM_RESAMP_TAPS]; WFM_RESAMP_PHASES],
frac: f32,
) -> f32 {
let phase = (frac.clamp(0.0, 0.999_999) * WFM_RESAMP_PHASES as f32).round() as usize;
let phase = phase.min(WFM_RESAMP_PHASES - 1);
hist.iter()
.zip(bank[phase].iter())
.map(|(x, c)| x * c)
.sum()
} }
#[derive(Debug, Clone)] #[derive(Debug, Clone)]
@@ -382,15 +429,16 @@ pub struct WfmStereoDecoder {
/// For standard WFM ±75 kHz deviation we want ±1.0 at full deviation, /// For standard WFM ±75 kHz deviation we want ±1.0 at full deviation,
/// so `fm_gain = fs / (2 · 75_000)`. /// so `fm_gain = fs / (2 · 75_000)`.
fm_gain: f32, fm_gain: f32,
/// History ring for 4-point Hermite cubic interpolation of sum channel. /// Shared coefficient bank for the polyphase fractional audio resampler.
/// Layout: [n-3, n-2, n-1, n] — oldest first. resample_bank: [[f32; WFM_RESAMP_TAPS]; WFM_RESAMP_PHASES],
sum_hist: [f32; 4], /// History ring for polyphase FIR resampling of the sum channel.
/// History ring for 4-point Hermite cubic interpolation of diff channel. sum_hist: [f32; WFM_RESAMP_TAPS],
diff_hist: [f32; 4], /// History ring for polyphase FIR resampling of the diff channel.
/// History ring for 4-point Hermite cubic interpolation of quadrature diff channel. diff_hist: [f32; WFM_RESAMP_TAPS],
diff_q_hist: [f32; 4], /// History ring for polyphase FIR resampling of the quadrature diff channel.
/// History ring for 4-point Hermite cubic interpolation of pilot blend. diff_q_hist: [f32; WFM_RESAMP_TAPS],
blend_hist: [f32; 4], /// Previous pilot blend sample for simple linear interpolation.
prev_blend: f32,
/// Fractional phase increment per composite sample = audio_rate / composite_rate. /// Fractional phase increment per composite sample = audio_rate / composite_rate.
/// Avoids integer-division rate error when composite_rate is not an exact /// Avoids integer-division rate error when composite_rate is not an exact
/// multiple of audio_rate (e.g. 250 kHz composite → 48 kHz audio). /// multiple of audio_rate (e.g. 250 kHz composite → 48 kHz audio).
@@ -441,10 +489,11 @@ impl WfmStereoDecoder {
stereo_detect_level: 0.0, stereo_detect_level: 0.0,
stereo_detected: false, stereo_detected: false,
fm_gain: composite_rate_f / (2.0 * 75_000.0), fm_gain: composite_rate_f / (2.0 * 75_000.0),
sum_hist: [0.0; 4], resample_bank: build_wfm_resample_bank(),
diff_hist: [0.0; 4], sum_hist: [0.0; WFM_RESAMP_TAPS],
diff_q_hist: [0.0; 4], diff_hist: [0.0; WFM_RESAMP_TAPS],
blend_hist: [0.0; 4], diff_q_hist: [0.0; WFM_RESAMP_TAPS],
prev_blend: 0.0,
output_phase_inc, output_phase_inc,
output_phase: 0.0, output_phase: 0.0,
} }
@@ -516,35 +565,30 @@ impl WfmStereoDecoder {
.diff_q_lpf2 .diff_q_lpf2
.process(self.diff_q_lpf1.process(x * (-sin_2p * 2.0))); .process(self.diff_q_lpf1.process(x * (-sin_2p * 2.0)));
// --- 4-point Hermite cubic interpolation resampling --- // --- Polyphase FIR fractional resampling ---
// Shift history rings: drop oldest, append current sample. // This uses a short windowed-sinc bank instead of cubic interpolation
// Layout after shift: [n-3, n-2, n-1, n]. // to reduce top-end overshoot/ringing near the audio cutoff.
// Hermite interpolates between hist[1] and hist[2] using hist[0] shift_append(&mut self.sum_hist, sum);
// and hist[3] as outer control points — one composite sample of shift_append(&mut self.diff_hist, diff_i);
// latency relative to linear interpolation (<5 µs at 240 kHz). shift_append(&mut self.diff_q_hist, diff_q);
self.sum_hist.rotate_left(1);
self.sum_hist[3] = sum;
self.diff_hist.rotate_left(1);
self.diff_hist[3] = diff_i;
self.diff_q_hist.rotate_left(1);
self.diff_q_hist[3] = diff_q;
self.blend_hist.rotate_left(1);
self.blend_hist[3] = stereo_blend;
let prev_phase = self.output_phase; let prev_phase = self.output_phase;
self.output_phase += self.output_phase_inc; self.output_phase += self.output_phase_inc;
if self.output_phase < 1.0 { if self.output_phase < 1.0 {
self.prev_blend = stereo_blend;
continue; continue;
} }
self.output_phase -= 1.0; self.output_phase -= 1.0;
// `frac` positions the output sample within the hist[1]–hist[2] // `frac` positions the output sample within the current fractional
// interval of the 4-point Hermite window. // interval. The FIR bank reconstructs a band-limited sample using
// a fixed two-sample lookahead in the decoder.
let frac = ((1.0 - prev_phase) / self.output_phase_inc) as f32; let frac = ((1.0 - prev_phase) / self.output_phase_inc) as f32;
let sum_i = hermite4(&self.sum_hist, frac); let sum_i = polyphase_resample(&self.sum_hist, &self.resample_bank, frac);
let diff_i = hermite4(&self.diff_hist, frac); let diff_i = polyphase_resample(&self.diff_hist, &self.resample_bank, frac);
let diff_q = hermite4(&self.diff_q_hist, frac); let diff_q = polyphase_resample(&self.diff_q_hist, &self.resample_bank, frac);
let blend_i = hermite4(&self.blend_hist, frac).clamp(0.0, 1.0); let blend_i = (self.prev_blend + frac * (stereo_blend - self.prev_blend)).clamp(0.0, 1.0);
self.prev_blend = stereo_blend;
let (trim_sin, trim_cos) = STEREO_SEPARATION_PHASE_TRIM.sin_cos(); let (trim_sin, trim_cos) = STEREO_SEPARATION_PHASE_TRIM.sin_cos();
let diff_i = (diff_i * trim_cos + diff_q * trim_sin) * STEREO_SEPARATION_GAIN; let diff_i = (diff_i * trim_cos + diff_q * trim_sin) * STEREO_SEPARATION_GAIN;