1057 lines
35 KiB
Rust
1057 lines
35 KiB
Rust
use dsp::{
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atan2, cossin,
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iir_int::{IIRState, IIR},
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reciprocal_pll::TimestampHandler,
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Complex,
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};
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use std::f64::consts::PI;
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use std::vec::Vec;
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// TODO: -> dsp/src/testing.rs
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/// Maximum acceptable error between a computed and actual value given fixed and relative
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/// tolerances.
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///
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/// # Args
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/// * `a` - First input.
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/// * `b` - Second input. The relative tolerance is computed with respect to the maximum of the
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/// absolute values of the first and second inputs.
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/// * `rtol` - Relative tolerance.
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/// * `atol` - Fixed tolerance.
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///
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/// # Returns
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/// Maximum acceptable error.
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pub fn max_error(a: f64, b: f64, rtol: f64, atol: f64) -> f64 {
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rtol * a.abs().max(b.abs()) + atol
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}
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pub fn isclose(a: f64, b: f64, rtol: f64, atol: f64) -> bool {
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(a - b).abs() <= a.abs().max(b.abs()) * rtol + atol
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}
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/// ADC full scale in machine units (16 bit signed).
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const ADC_SCALE: f64 = ((1 << 15) - 1) as f64;
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struct Lockin {
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harmonic: u32,
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phase: u32,
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iir: IIR,
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iir_state: [IIRState; 2],
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}
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impl Lockin {
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pub fn new(harmonic: u32, phase: u32, iir: IIR) -> Self {
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Lockin {
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harmonic,
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phase,
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iir,
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iir_state: [IIRState::default(); 2],
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}
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}
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pub fn update(
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&mut self,
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input: Vec<i16>,
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phase: u32,
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frequency: u32,
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) -> Complex<i32> {
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let frequency = frequency.wrapping_mul(self.harmonic);
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let mut phase =
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self.phase.wrapping_add(phase.wrapping_mul(self.harmonic));
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input
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.iter()
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.map(|&s| {
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let m = cossin((phase as i32).wrapping_neg());
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phase = phase.wrapping_add(frequency);
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let signal = (s as i32) << 16;
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Complex(
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self.iir.update(
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&mut self.iir_state[0],
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((signal as i64 * m.0 as i64) >> 32) as _,
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),
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self.iir.update(
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&mut self.iir_state[1],
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((signal as i64 * m.1 as i64) >> 32) as _,
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),
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)
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})
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.last()
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.unwrap_or(Complex::default())
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}
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}
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/// Single-frequency sinusoid.
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#[derive(Copy, Clone)]
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struct Tone {
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// Frequency (in Hz).
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frequency: f64,
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// Phase offset (in radians).
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phase: f64,
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// Amplitude in dBFS (decibels relative to full-scale).
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// A 16-bit ADC has a minimum dBFS for each sample of -90.
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amplitude_dbfs: f64,
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}
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/// Convert dBFS to a linear ratio.
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fn linear(dbfs: f64) -> f64 {
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10f64.powf(dbfs / 20.)
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}
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impl Tone {
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fn eval(&self, time: f64) -> f64 {
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linear(self.amplitude_dbfs) * (self.phase + self.frequency * time).cos()
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}
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}
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/// Generate a full batch of samples with size `sample_buffer_size` starting at `time_offset`.
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fn sample_tones(
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tones: &Vec<Tone>,
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time_offset: f64,
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sample_buffer_size: u32,
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) -> Vec<i16> {
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(0..sample_buffer_size)
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.map(|i| {
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let time = 2. * PI * (time_offset + i as f64);
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let x: f64 = tones.iter().map(|t| t.eval(time)).sum();
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assert!(-1. < x && x < 1.);
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(x * ADC_SCALE) as i16
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})
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.collect()
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}
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/// Total maximum noise amplitude of the input signal after 2nd order lowpass filter.
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/// Constructive interference is assumed.
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///
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/// # Args
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/// * `tones` - Noise sources at the ADC input.
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/// * `frequency` - Frequency of the signal of interest.
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/// * `corner` - Low-pass filter 3dB corner cutoff frequency.
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///
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/// # Returns
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/// Upper bound of the total amplitude of all noise sources in linear units full scale.
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fn sampled_noise_amplitude(
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tones: &Vec<Tone>,
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frequency: f64,
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corner: f64,
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) -> f64 {
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tones
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.iter()
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.map(|t| {
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let decades =
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((t.frequency - frequency) / corner).abs().max(1.).log10();
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// 2nd-order filter: 40dB/decade rolloff.
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linear(t.amplitude_dbfs - 40. * decades)
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})
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.sum::<f64>()
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.max(1. / ADC_SCALE / 2.) // 1/2 LSB from quantization
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}
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/// Reference clock timestamp values in one ADC batch period starting at `timestamp_start`. The
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/// number of timestamps in a batch can be 0 or 1, so this returns an Option containing a timestamp
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/// only if one occurred during the batch.
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///
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/// # Args
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/// * `reference_frequency` - External reference signal frequency (in Hz).
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/// * `timestamp_start` - Start time in terms of the internal clock count. This is the start time of
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/// the current processing sequence.
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/// * `timestamp_stop` - Stop time in terms of the internal clock count.
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/// * `internal_frequency` - Internal clock frequency (in Hz).
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///
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/// # Returns
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/// An Option, containing a timestamp if one occurred during the current batch period.
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fn adc_batch_timestamps(
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reference_frequency: f64,
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timestamp_start: u64,
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timestamp_stop: u64,
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internal_frequency: f64,
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) -> Option<u32> {
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let reference_period = internal_frequency / reference_frequency;
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let start_count = timestamp_start as f64 % reference_period;
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let timestamp = (reference_period - start_count) % reference_period;
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if timestamp < (timestamp_stop - timestamp_start) as f64 {
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return Some(
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((timestamp_start + timestamp.round() as u64) % (1u64 << 32))
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as u32,
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);
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}
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None
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}
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/// Lowpass biquad filter using cutoff and sampling frequencies. Taken from:
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/// https://webaudio.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html
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///
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/// # Args
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/// * `fc` - Corner frequency, or 3dB cutoff frequency (in units of sample rate).
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/// * `q` - Quality factor (1/sqrt(2) for critical).
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/// * `k` - DC gain.
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///
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/// # Returns
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/// 2nd-order IIR filter coefficients in the form [b0,b1,b2,a1,a2]. a0 is set to -1.
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fn lowpass_iir_coefficients(fc: f64, q: f64, k: f64) -> IIRState {
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let f = 2. * PI * fc;
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let a = f.sin() / (2. * q);
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// IIR uses Q2.30 fixed point
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let a0 = (1. + a) / (1 << IIR::SHIFT) as f64;
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let b0 = (k / 2. * (1. - f.cos()) / a0).round() as _;
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let a1 = (2. * f.cos() / a0).round() as _;
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let a2 = ((a - 1.) / a0).round() as _;
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IIRState([b0, 2 * b0, b0, a1, a2])
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}
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/// Compute the maximum effect of input noise on the lock-in magnitude computation.
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///
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/// The maximum effect of noise on the magnitude computation is given by:
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///
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/// | sqrt((I+n*sin(x))**2 + (Q+n*cos(x))**2) - sqrt(I**2 + Q**2) |
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///
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/// * I is the in-phase component of the portion of the input signal with the same frequency as the
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/// demodulation signal.
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/// * Q is the quadrature component.
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/// * n is the total noise amplitude (from all contributions, after attenuation from filtering).
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/// * x is the phase of the demodulation signal.
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///
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/// We need to find the demodulation phase (x) that maximizes this expression. We can ignore the
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/// absolute value operation by also considering the expression minimum. The locations of the
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/// minimum and maximum can be computed analytically by finding the value of x when the derivative
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/// of this expression with respect to x is 0. When we solve this equation, we find:
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///
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/// x = atan(I/Q)
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///
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/// It's worth noting that this solution is technically only valid when cos(x)!=0 (i.e.,
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/// x!=pi/2,-pi/2). However, this is not a problem because we only get these values when Q=0. Rust
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/// correctly computes atan(inf)=pi/2, which is precisely what we want because x=pi/2 maximizes
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/// sin(x) and therefore also the noise effect.
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///
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/// The other maximum or minimum is pi radians away from this value.
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///
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/// # Args
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/// * `total_noise_amplitude` - Combined amplitude of all noise sources sampled by the ADC.
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/// * `in_phase_actual` - Value of the in-phase component if no noise were present at the ADC input.
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/// * `quadrature_actual` - Value of the quadrature component if no noise were present at the ADC
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/// input.
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/// * `desired_input_amplitude` - Amplitude of the desired input signal. That is, the input signal
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/// component with the same frequency as the demodulation signal.
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///
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/// # Returns
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/// Approximation of the maximum effect on the magnitude computation due to noise sources at the ADC
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/// input.
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fn magnitude_noise(
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total_noise_amplitude: f64,
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in_phase_actual: f64,
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quadrature_actual: f64,
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desired_input_amplitude: f64,
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) -> f64 {
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// See function documentation for explanation.
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let noise = |in_phase_delta: f64, quadrature_delta: f64| -> f64 {
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(((in_phase_actual + in_phase_delta).powf(2.)
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+ (quadrature_actual + quadrature_delta).powf(2.))
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.sqrt()
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- desired_input_amplitude)
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.abs()
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};
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let phase = (in_phase_actual / quadrature_actual).atan();
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let max_noise_1 = noise(
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total_noise_amplitude * phase.sin(),
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total_noise_amplitude * phase.cos(),
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);
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let max_noise_2 = noise(
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total_noise_amplitude * (phase + PI).sin(),
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total_noise_amplitude * (phase + PI).cos(),
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);
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max_noise_1.max(max_noise_2)
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}
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/// Compute the maximum phase deviation from the correct value due to the input noise sources.
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///
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/// The maximum effect of noise on the phase computation is given by:
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///
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/// | atan2(Q+n*cos(x), I+n*sin(x)) - atan2(Q, I) |
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///
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/// See `magnitude_noise` for an explanation of the terms in this mathematical expression.
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///
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/// This expression is harder to compute analytically than the expression in `magnitude_noise`. We
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/// could compute it numerically, but that's expensive. However, we can use heuristics to try to
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/// guess the values of x that will maximize the noise effect. Intuitively, the difference will be
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/// largest when the Y-argument of the atan2 function (Q+n*cos(x)) is pushed in the opposite
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/// direction of the noise effect on the X-argument (i.e., cos(x) and sin(x) have different
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/// signs). We can use:
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///
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/// * sin(x)=+-1 (+- denotes plus or minus), cos(x)=0,
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/// * sin(x)=0, cos(x)=+-1, and
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/// * the value of x that maximizes |sin(x)-cos(x)| (when sin(x)=1/sqrt(2) and cos(x)=-1/sqrt(2), or
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/// when the signs are flipped)
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///
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/// The first choice addresses cases in which |I|>>|Q|, the second choice addresses cases in which
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/// |Q|>>|I|, and the third choice addresses cases in which |I|~|Q|. We can test all of these cases
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/// as an approximation for the real maximum.
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///
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/// # Args
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/// * `total_noise_amplitude` - Total amplitude of all input noise sources.
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/// * `in_phase_actual` - Value of the in-phase component if no noise were present at the input.
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/// * `quadrature_actual` - Value of the quadrature component if no noise were present at the input.
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///
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/// # Returns
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/// Approximation of the maximum effect on the phase computation due to noise sources at the ADC
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/// input.
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fn phase_noise(
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total_noise_amplitude: f64,
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in_phase_actual: f64,
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quadrature_actual: f64,
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) -> f64 {
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// See function documentation for explanation.
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let noise = |in_phase_delta: f64, quadrature_delta: f64| -> f64 {
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((quadrature_actual + quadrature_delta)
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.atan2(in_phase_actual + in_phase_delta)
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- quadrature_actual.atan2(in_phase_actual))
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.abs()
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};
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let mut max_noise: f64 = 0.;
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for (in_phase_delta, quadrature_delta) in [
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(
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total_noise_amplitude / 2_f64.sqrt(),
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total_noise_amplitude / -2_f64.sqrt(),
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),
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(
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total_noise_amplitude / -2_f64.sqrt(),
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total_noise_amplitude / 2_f64.sqrt(),
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),
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(total_noise_amplitude, 0.),
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(-total_noise_amplitude, 0.),
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(0., total_noise_amplitude),
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(0., -total_noise_amplitude),
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]
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.iter()
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{
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max_noise = max_noise.max(noise(*in_phase_delta, *quadrature_delta));
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}
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max_noise
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}
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/// Lowpass filter test for in-phase/quadrature and magnitude/phase computations.
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///
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/// This attempts to "intelligently" model acceptable tolerance ranges for the measured in-phase,
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/// quadrature, magnitude and phase results of lock-in processing for a typical low-pass filter
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/// application. So, instead of testing whether the lock-in processing extracts the true magnitude
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/// and phase (or in-phase and quadrature components) of the input signal, it attempts to calculate
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/// what the lock-in processing should compute given any set of input noise sources. For example, if
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/// a noise source of sufficient strength differs in frequency by 1kHz from the reference frequency
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/// and the filter cutoff frequency is also 1kHz, testing if the lock-in amplifier extracts the
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/// amplitude and phase of the input signal whose frequency is equal to the demodulation frequency
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/// is doomed to failure. Instead, this function tests whether the lock-in correctly adheres to its
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/// actual transfer function, whether or not it was given reasonable inputs. The logic for computing
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/// acceptable tolerance ranges is performed in `sampled_noise_amplitude`, `magnitude_noise`, and
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/// `phase_noise`.
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///
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/// # Args
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/// * `internal_frequency` - Internal clock frequency (Hz). The internal clock increments timestamp
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/// counter values used to record the edges of the external reference.
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/// * `reference_frequency` - External reference frequency (in Hz).
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/// * `demodulation_phase_offset` - Phase offset applied to the in-phase and quadrature demodulation
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/// signals.
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/// * `harmonic` - Scaling factor for the demodulation frequency. E.g., 2 would demodulate with the
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/// first harmonic of the reference frequency.
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/// * `sample_buffer_size_log2` - The base-2 logarithm of the number of samples in a processing
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/// batch.
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/// * `pll_shift_frequency` - See `pll::update()`.
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/// * `pll_shift_phase` - See `pll::update()`.
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/// * `corner_frequency` - Lowpass filter 3dB cutoff frequency.
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/// * `desired_input` - `Tone` giving the frequency, amplitude and phase of the desired result.
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/// * `noise_inputs` - Vector of `Tone` for any noise inputs on top of `desired_input`.
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/// * `time_constant_factor` - Number of time constants after which the output is considered valid.
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/// * `tolerance` - Acceptable relative tolerance for the magnitude and angle outputs. This is added
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/// to fixed tolerance values computed inside this function. The outputs must remain within this
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/// tolerance between `time_constant_factor` and `time_constant_factor+1` time constants.
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fn lowpass_test(
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internal_frequency: f64,
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reference_frequency: f64,
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demodulation_phase_offset: f64,
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harmonic: u32,
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sample_buffer_size_log2: usize,
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pll_shift_frequency: u8,
|
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pll_shift_phase: u8,
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corner_frequency: f64,
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desired_input: Tone,
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tones: &mut Vec<Tone>,
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time_constant_factor: f64,
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tolerance: f64,
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) {
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assert!(
|
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isclose((internal_frequency).log2(), (internal_frequency).log2().round(), 0., 1e-5),
|
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"The number of internal clock cycles in one ADC sampling period must be a power-of-two."
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);
|
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|
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assert!(
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internal_frequency / reference_frequency
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>= internal_frequency
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* (1 << sample_buffer_size_log2) as f64,
|
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"Too many timestamps per batch. Each batch can have at most 1 timestamp."
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);
|
|
|
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let adc_sample_ticks_log2 = (internal_frequency).log2().round() as usize;
|
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assert!(
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adc_sample_ticks_log2 + sample_buffer_size_log2 <= 32,
|
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"The base-2 log of the number of ADC ticks in a sampling period plus the base-2 log of the sample buffer size must be less than 32."
|
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);
|
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|
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let mut lockin = Lockin::new(
|
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harmonic,
|
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(demodulation_phase_offset / (2. * PI) * (1_u64 << 32) as f64).round()
|
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as u32,
|
|
IIR {
|
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ba: lowpass_iir_coefficients(
|
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corner_frequency,
|
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1. / 2f64.sqrt(),
|
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2.,
|
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),
|
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},
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);
|
|
let mut timestamp_handler = TimestampHandler::new(
|
|
pll_shift_frequency,
|
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pll_shift_phase,
|
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adc_sample_ticks_log2,
|
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sample_buffer_size_log2,
|
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);
|
|
|
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let mut timestamp_start: u64 = 0;
|
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let time_constant: f64 = 1. / (2. * PI * corner_frequency);
|
|
// Account for the pll settling time (see its documentation).
|
|
let pll_time_constant_samples =
|
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(1 << pll_shift_phase.max(pll_shift_frequency)) as usize;
|
|
let low_pass_time_constant_samples = (time_constant_factor * time_constant
|
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/ (1 << sample_buffer_size_log2) as f64)
|
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as usize;
|
|
let samples = pll_time_constant_samples + low_pass_time_constant_samples;
|
|
// Ensure the result remains within tolerance for 1 time constant after `time_constant_factor`
|
|
// time constants.
|
|
let extra_samples = time_constant as usize;
|
|
let batch_sample_count =
|
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1_u64 << (adc_sample_ticks_log2 + sample_buffer_size_log2);
|
|
|
|
let effective_phase_offset =
|
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desired_input.phase - demodulation_phase_offset;
|
|
let in_phase_actual =
|
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linear(desired_input.amplitude_dbfs) * effective_phase_offset.cos();
|
|
let quadrature_actual =
|
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linear(desired_input.amplitude_dbfs) * effective_phase_offset.sin();
|
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|
|
let total_noise_amplitude = sampled_noise_amplitude(
|
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tones,
|
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reference_frequency * harmonic as f64,
|
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corner_frequency,
|
|
);
|
|
// Add some fixed error to account for errors introduced by the PLL, our custom trig functions
|
|
// and integer division. It's a bit difficult to be precise about this. I've added a 1%
|
|
// (relative to full scale) error.
|
|
let total_magnitude_noise = magnitude_noise(
|
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total_noise_amplitude,
|
|
in_phase_actual,
|
|
quadrature_actual,
|
|
linear(desired_input.amplitude_dbfs),
|
|
) + 1e-2;
|
|
let total_phase_noise =
|
|
phase_noise(total_noise_amplitude, in_phase_actual, quadrature_actual)
|
|
+ 1e-2 * 2. * PI;
|
|
|
|
tones.push(desired_input);
|
|
|
|
for n in 0..(samples + extra_samples) {
|
|
let adc_signal = sample_tones(
|
|
&tones,
|
|
timestamp_start as f64 / internal_frequency,
|
|
1 << sample_buffer_size_log2,
|
|
);
|
|
let timestamp = adc_batch_timestamps(
|
|
reference_frequency,
|
|
timestamp_start,
|
|
timestamp_start + batch_sample_count - 1,
|
|
internal_frequency,
|
|
);
|
|
timestamp_start += batch_sample_count;
|
|
|
|
let (demodulation_initial_phase, demodulation_frequency) =
|
|
timestamp_handler.update(timestamp);
|
|
let output = lockin.update(
|
|
adc_signal,
|
|
demodulation_initial_phase,
|
|
demodulation_frequency,
|
|
);
|
|
let magnitude = (((output.0 as i64) * (output.0 as i64)
|
|
+ (output.1 as i64) * (output.1 as i64))
|
|
>> 32) as i32;
|
|
let phase = atan2(output.1, output.0);
|
|
|
|
// Ensure stable within tolerance for 1 time constant after `time_constant_factor`.
|
|
if n >= samples {
|
|
// We want our full-scale magnitude to be 1. Our fixed-point numbers treated as integers
|
|
// set the full-scale magnitude to 1<<60. So, we must divide by this number. However,
|
|
// we've already divided by 1<<32 in the magnitude computation to keep our values within
|
|
// the i32 limits, so we just need to divide by an additional 1<<28.
|
|
let amplitude_normalized =
|
|
(magnitude as f64 / (1_u64 << 28) as f64).sqrt();
|
|
assert!(
|
|
isclose(linear(desired_input.amplitude_dbfs), amplitude_normalized, tolerance, total_magnitude_noise),
|
|
"magnitude actual: {:.4} ({:.2} dBFS), magnitude computed: {:.4} ({:.2} dBFS), tolerance: {:.4}",
|
|
linear(desired_input.amplitude_dbfs),
|
|
desired_input.amplitude_dbfs,
|
|
amplitude_normalized,
|
|
20.*amplitude_normalized.log10(),
|
|
max_error(linear(desired_input.amplitude_dbfs), amplitude_normalized, tolerance, total_magnitude_noise),
|
|
);
|
|
let phase_normalized =
|
|
phase as f64 / (1_u64 << 32) as f64 * (2. * PI);
|
|
assert!(
|
|
isclose(
|
|
effective_phase_offset,
|
|
phase_normalized,
|
|
tolerance,
|
|
total_phase_noise
|
|
),
|
|
"phase actual: {:.4}, phase computed: {:.4}, tolerance: {:.4}",
|
|
effective_phase_offset,
|
|
phase_normalized,
|
|
max_error(
|
|
effective_phase_offset,
|
|
phase_normalized,
|
|
tolerance,
|
|
total_phase_noise
|
|
),
|
|
);
|
|
|
|
let in_phase_normalized = output.0 as f64 / (1 << 30) as f64;
|
|
let quadrature_normalized = output.1 as f64 / (1 << 30) as f64;
|
|
|
|
assert!(
|
|
isclose(
|
|
in_phase_actual,
|
|
in_phase_normalized,
|
|
total_noise_amplitude,
|
|
tolerance
|
|
),
|
|
"in-phase actual: {:.4}, in-phase computed: {:.3}, tolerance: {:.4}",
|
|
in_phase_actual,
|
|
in_phase_normalized,
|
|
max_error(
|
|
in_phase_actual,
|
|
in_phase_normalized,
|
|
total_noise_amplitude,
|
|
tolerance
|
|
),
|
|
);
|
|
assert!(
|
|
isclose(
|
|
quadrature_actual,
|
|
quadrature_normalized,
|
|
total_noise_amplitude,
|
|
tolerance
|
|
),
|
|
"quadrature actual: {:.4}, quadrature computed: {:.4}, tolerance: {:.4}",
|
|
quadrature_actual,
|
|
quadrature_normalized,
|
|
max_error(
|
|
quadrature_actual,
|
|
quadrature_normalized,
|
|
total_noise_amplitude,
|
|
tolerance
|
|
),
|
|
);
|
|
}
|
|
}
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 64e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 3;
|
|
let pll_shift_phase: u8 = 2;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 6.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.1 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.9 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_demodulation_phase_offset_pi_2() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 64e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 3;
|
|
let pll_shift_phase: u8 = 2;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = PI / 2.;
|
|
let time_constant_factor: f64 = 6.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.1 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.9 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_phase_offset_pi_2() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 64e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 3;
|
|
let pll_shift_phase: u8 = 2;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 6.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: PI / 2.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.1 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.9 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_fundamental_71e_3_phase_offset_pi_4() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 71e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 3;
|
|
let pll_shift_phase: u8 = 2;
|
|
let corner_frequency: f64 = 0.6e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: PI / 4.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.1 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.9 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_first_harmonic() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 50e-3;
|
|
let harmonic: u32 = 2;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_second_harmonic() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 50e-3;
|
|
let harmonic: u32 = 3;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_third_harmonic() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 50e-3;
|
|
let harmonic: u32 = 4;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_first_harmonic_phase_shift() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 50e-3;
|
|
let harmonic: u32 = 2;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: PI / 4.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_adc_frequency_1e6() {
|
|
let internal_frequency: f64 = 32.;
|
|
let signal_frequency: f64 = 100e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_internal_frequency_125e6() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 100e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-2;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![
|
|
Tone {
|
|
frequency: 1.2 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
Tone {
|
|
frequency: 0.8 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
},
|
|
],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|
|
|
|
#[test]
|
|
fn lowpass_low_signal_frequency() {
|
|
let internal_frequency: f64 = 64.;
|
|
let signal_frequency: f64 = 10e-3;
|
|
let harmonic: u32 = 1;
|
|
let sample_buffer_size_log2: usize = 2;
|
|
let pll_shift_frequency: u8 = 2;
|
|
let pll_shift_phase: u8 = 1;
|
|
let corner_frequency: f64 = 1e-3;
|
|
let demodulation_frequency: f64 = harmonic as f64 * signal_frequency;
|
|
let demodulation_phase_offset: f64 = 0.;
|
|
let time_constant_factor: f64 = 5.;
|
|
let tolerance: f64 = 1e-1;
|
|
|
|
lowpass_test(
|
|
internal_frequency,
|
|
signal_frequency,
|
|
demodulation_phase_offset,
|
|
harmonic,
|
|
sample_buffer_size_log2,
|
|
pll_shift_frequency,
|
|
pll_shift_phase,
|
|
corner_frequency,
|
|
Tone {
|
|
frequency: demodulation_frequency,
|
|
amplitude_dbfs: -30.,
|
|
phase: 0.,
|
|
},
|
|
&mut vec![Tone {
|
|
frequency: 1.1 * demodulation_frequency,
|
|
amplitude_dbfs: -20.,
|
|
phase: 0.,
|
|
}],
|
|
time_constant_factor,
|
|
tolerance,
|
|
);
|
|
}
|