use dsp::{
    iir_int::{IIRState, IIR},
    reciprocal_pll::TimestampHandler,
    trig::{atan2, cossin},
    Complex,
};

use std::f64::consts::PI;
use std::vec::Vec;

// TODO: -> dsp/src/testing.rs
/// Maximum acceptable error between a computed and actual value given fixed and relative
/// tolerances.
///
/// # Args
/// * `a` - First input.
/// * `b` - Second input. The relative tolerance is computed with respect to the maximum of the
/// absolute values of the first and second inputs.
/// * `rtol` - Relative tolerance.
/// * `atol` - Fixed tolerance.
///
/// # Returns
/// Maximum acceptable error.
pub fn max_error(a: f64, b: f64, rtol: f64, atol: f64) -> f64 {
    rtol * a.abs().max(b.abs()) + atol
}

pub fn isclose(a: f64, b: f64, rtol: f64, atol: f64) -> bool {
    (a - b).abs() <= a.abs().max(b.abs()) * rtol + atol
}

const ADC_MAX_COUNT: f64 = (1 << 15) as f64;

struct Lockin {
    harmonic: u32,
    phase: u32,
    iir: IIR,
    iir_state: [IIRState; 2],
}

impl Lockin {
    pub fn new(harmonic: u32, phase: u32, iir: IIR) -> Self {
        Lockin {
            harmonic,
            phase,
            iir,
            iir_state: [IIRState::default(); 2],
        }
    }

    pub fn update(
        &mut self,
        input: Vec<i16>,
        phase: u32,
        frequency: u32,
    ) -> Complex<i32> {
        let frequency = frequency.wrapping_mul(self.harmonic);
        let mut phase =
            self.phase.wrapping_add(phase.wrapping_mul(self.harmonic));
        let mut last = Complex::default();

        for &s in input.iter() {
            let m = cossin((phase as i32).wrapping_neg());
            phase = phase.wrapping_add(frequency);

            let signal = (s as i32) << 16;
            last = Complex(
                self.iir.update(
                    &mut self.iir_state[0],
                    ((signal as i64 * m.0 as i64) >> 32) as i32,
                ),
                self.iir.update(
                    &mut self.iir_state[1],
                    ((signal as i64 * m.1 as i64) >> 32) as i32,
                ),
            );
        }
        last
    }
}

/// Single-frequency sinusoid.
#[derive(Copy, Clone)]
struct Tone {
    // Frequency (in Hz).
    frequency: f64,
    // Phase offset (in radians).
    phase: f64,
    // Amplitude in dBFS (decibels relative to full-scale).
    // A 16-bit ADC has a minimum dBFS for each sample of -90.
    amplitude_dbfs: f64,
}

/// Convert dBFS to a linear ratio.
fn linear(dbfs: f64) -> f64 {
    10f64.powf(dbfs / 20.)
}

/// Generate a full batch of samples starting at `time_offset`.
fn adc_sampled_signal(
    tones: &Vec<Tone>,
    time_offset: f64,
    sampling_frequency: f64,
    sample_buffer_size: u32,
) -> Vec<i16> {
    let mut signal = Vec::<i16>::new();

    for i in 0..sample_buffer_size {
        let time = 2. * PI * (time_offset + i as f64 / sampling_frequency);
        let x: f64 = tones
            .iter()
            .map(|&t| {
                linear(t.amplitude_dbfs) * (t.phase + t.frequency * time).cos()
            })
            .sum();
        assert!(-1. < x && x < 1.);
        signal.push((x * ADC_MAX_COUNT) as i16);
    }
    signal
}

/// Reference clock timestamp values in one ADC batch period starting at `timestamp_start`. The
/// number of timestamps in a batch can be 0 or 1, so this returns an Option containing a timestamp
/// only if one occurred during the batch.
///
/// # Args
/// * `reference_frequency` - External reference signal frequency (in Hz).
/// * `timestamp_start` - Start time in terms of the internal clock count. This is the start time of
/// the current processing sequence.
/// * `timestamp_stop` - Stop time in terms of the internal clock count.
/// * `internal_frequency` - Internal clock frequency (in Hz).
///
/// # Returns
/// An Option, containing a timestamp if one occurred during the current batch period.
fn adc_batch_timestamps(
    reference_frequency: f64,
    timestamp_start: u64,
    timestamp_stop: u64,
    internal_frequency: f64,
) -> Option<u32> {
    let reference_period = internal_frequency / reference_frequency;
    let start_count = timestamp_start as f64 % reference_period;

    let timestamp = (reference_period - start_count) % reference_period;

    if timestamp < (timestamp_stop - timestamp_start) as f64 {
        return Some(
            ((timestamp_start + timestamp.round() as u64) % (1u64 << 32))
                as u32,
        );
    }

    None
}

/// Lowpass biquad filter using cutoff and sampling frequencies.  Taken from:
/// https://webaudio.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html
///
/// # Args
/// * `fc` - Corner frequency, or 3dB cutoff frequency (in Hz).
/// * `q` - Quality factor (1/sqrt(2) for critical).
/// * `k` - DC gain.
///
/// # Returns
/// 2nd-order IIR filter coefficients in the form [b0,b1,b2,a1,a2]. a0 is set to -1.
fn lowpass_iir_coefficients(fc: f64, q: f64, k: f64) -> IIRState {
    let f = 2. * PI * fc;
    let a = f.sin() / (2. * q);
    // IIR uses Q2.30 fixed point
    let a0 = (1. + a) / (1 << IIR::SHIFT) as f64;
    let b0 = (k / 2. * (1. - f.cos()) / a0).round() as _;
    let a1 = (2. * f.cos() / a0).round() as _;
    let a2 = ((a - 1.) / a0).round() as _;

    IIRState([b0, 2 * b0, b0, a1, a2])
}

/// Total noise amplitude of the input signal after sampling by the ADC. This computes an upper
/// bound of the total noise amplitude, rather than its actual value.
///
/// # Args
/// * `tones` - Noise sources at the ADC input.
/// * `demodulation_frequency` - Frequency of the demodulation signal (in Hz).
/// * `corner_frequency` - Low-pass filter 3dB corner (cutoff) frequency.
///
/// # Returns
/// Upper bound of the total amplitude of all noise sources.
fn sampled_noise_amplitude(
    tones: &Vec<Tone>,
    demodulation_frequency: f64,
    corner_frequency: f64,
) -> f64 {
    // There is not a simple way to compute the amplitude of a superpostition of sinusoids with
    // different frequencies and phases. Although we can compute the amplitude in special cases
    // (e.g., two signals whose periods have a common multiple), these do not help us in the general
    // case. However, we can say that the total amplitude will not be greater than the sum of the
    // amplitudes of the individual noise sources. We treat this as an upper bound, and use it as an
    // approximation of the actual amplitude.

    let mut noise: f64 = tones
        .iter()
        .map(|n| {
            // Noise inputs create an oscillation at the output, where the oscillation magnitude is
            // determined by the strength of the noise and its attenuation (attenuation is
            // determined by its proximity to the demodulation frequency and filter rolloff).
            let octaves = ((n.frequency - demodulation_frequency).abs()
                / corner_frequency)
                .log2();
            // 2nd-order filter. Approximately 12dB/octave rolloff.
            let attenuation = -2. * 20. * 2f64.log10() * octaves;
            linear(n.amplitude_dbfs + attenuation)
        })
        .sum();

    // Add in 1/2 LSB for the maximum amplitude deviation resulting from quantization.
    noise += 1. / ADC_MAX_COUNT / 2.;

    noise
}

/// Compute the maximum effect of input noise on the lock-in magnitude computation.
///
/// The maximum effect of noise on the magnitude computation is given by:
///
/// | sqrt((I+n*sin(x))**2 + (Q+n*cos(x))**2) - sqrt(I**2 + Q**2) |
///
/// * I is the in-phase component of the portion of the input signal with the same frequency as the
/// demodulation signal.
/// * Q is the quadrature component.
/// * n is the total noise amplitude (from all contributions, after attenuation from filtering).
/// * x is the phase of the demodulation signal.
///
/// We need to find the demodulation phase (x) that maximizes this expression. We can ignore the
/// absolute value operation by also considering the expression minimum. The locations of the
/// minimum and maximum can be computed analytically by finding the value of x when the derivative
/// of this expression with respect to x is 0. When we solve this equation, we find:
///
/// x = atan(I/Q)
///
/// It's worth noting that this solution is technically only valid when cos(x)!=0 (i.e.,
/// x!=pi/2,-pi/2). However, this is not a problem because we only get these values when Q=0. Rust
/// correctly computes atan(inf)=pi/2, which is precisely what we want because x=pi/2 maximizes
/// sin(x) and therefore also the noise effect.
///
/// The other maximum or minimum is pi radians away from this value.
///
/// # Args
/// * `total_noise_amplitude` - Combined amplitude of all noise sources sampled by the ADC.
/// * `in_phase_actual` - Value of the in-phase component if no noise were present at the ADC input.
/// * `quadrature_actual` - Value of the quadrature component if no noise were present at the ADC
/// input.
/// * `desired_input_amplitude` - Amplitude of the desired input signal. That is, the input signal
/// component with the same frequency as the demodulation signal.
///
/// # Returns
/// Approximation of the maximum effect on the magnitude computation due to noise sources at the ADC
/// input.
fn magnitude_noise(
    total_noise_amplitude: f64,
    in_phase_actual: f64,
    quadrature_actual: f64,
    desired_input_amplitude: f64,
) -> f64 {
    // See function documentation for explanation.
    let noise = |in_phase_delta: f64, quadrature_delta: f64| -> f64 {
        (((in_phase_actual + in_phase_delta).powf(2.)
            + (quadrature_actual + quadrature_delta).powf(2.))
        .sqrt()
            - desired_input_amplitude)
            .abs()
    };

    let phase = (in_phase_actual / quadrature_actual).atan();
    let max_noise_1 = noise(
        total_noise_amplitude * phase.sin(),
        total_noise_amplitude * phase.cos(),
    );
    let max_noise_2 = noise(
        total_noise_amplitude * (phase + PI).sin(),
        total_noise_amplitude * (phase + PI).cos(),
    );

    max_noise_1.max(max_noise_2)
}

/// Compute the maximum phase deviation from the correct value due to the input noise sources.
///
/// The maximum effect of noise on the phase computation is given by:
///
/// | atan2(Q+n*cos(x), I+n*sin(x)) - atan2(Q, I) |
///
/// See `magnitude_noise` for an explanation of the terms in this mathematical expression.
///
/// This expression is harder to compute analytically than the expression in `magnitude_noise`. We
/// could compute it numerically, but that's expensive. However, we can use heuristics to try to
/// guess the values of x that will maximize the noise effect. Intuitively, the difference will be
/// largest when the Y-argument of the atan2 function (Q+n*cos(x)) is pushed in the opposite
/// direction of the noise effect on the X-argument (i.e., cos(x) and sin(x) have different
/// signs). We can use:
///
/// * sin(x)=+-1 (+- denotes plus or minus), cos(x)=0,
/// * sin(x)=0, cos(x)=+-1, and
/// * the value of x that maximizes |sin(x)-cos(x)| (when sin(x)=1/sqrt(2) and cos(x)=-1/sqrt(2), or
/// when the signs are flipped)
///
/// The first choice addresses cases in which |I|>>|Q|, the second choice addresses cases in which
/// |Q|>>|I|, and the third choice addresses cases in which |I|~|Q|. We can test all of these cases
/// as an approximation for the real maximum.
///
/// # Args
/// * `total_noise_amplitude` - Total amplitude of all input noise sources.
/// * `in_phase_actual` - Value of the in-phase component if no noise were present at the input.
/// * `quadrature_actual` - Value of the quadrature component if no noise were present at the input.
///
/// # Returns
/// Approximation of the maximum effect on the phase computation due to noise sources at the ADC
/// input.
fn phase_noise(
    total_noise_amplitude: f64,
    in_phase_actual: f64,
    quadrature_actual: f64,
) -> f64 {
    // See function documentation for explanation.
    let noise = |in_phase_delta: f64, quadrature_delta: f64| -> f64 {
        ((quadrature_actual + quadrature_delta)
            .atan2(in_phase_actual + in_phase_delta)
            - quadrature_actual.atan2(in_phase_actual))
        .abs()
    };

    let mut max_noise: f64 = 0.;
    for (in_phase_delta, quadrature_delta) in [
        (
            total_noise_amplitude / 2_f64.sqrt(),
            total_noise_amplitude / -2_f64.sqrt(),
        ),
        (
            total_noise_amplitude / -2_f64.sqrt(),
            total_noise_amplitude / 2_f64.sqrt(),
        ),
        (total_noise_amplitude, 0.),
        (-total_noise_amplitude, 0.),
        (0., total_noise_amplitude),
        (0., -total_noise_amplitude),
    ]
    .iter()
    {
        max_noise = max_noise.max(noise(*in_phase_delta, *quadrature_delta));
    }

    max_noise
}

/// Lowpass filter test for in-phase/quadrature and magnitude/phase computations.
///
/// This attempts to "intelligently" model acceptable tolerance ranges for the measured in-phase,
/// quadrature, magnitude and phase results of lock-in processing for a typical low-pass filter
/// application. So, instead of testing whether the lock-in processing extracts the true magnitude
/// and phase (or in-phase and quadrature components) of the input signal, it attempts to calculate
/// what the lock-in processing should compute given any set of input noise sources. For example, if
/// a noise source of sufficient strength differs in frequency by 1kHz from the reference frequency
/// and the filter cutoff frequency is also 1kHz, testing if the lock-in amplifier extracts the
/// amplitude and phase of the input signal whose frequency is equal to the demodulation frequency
/// is doomed to failure. Instead, this function tests whether the lock-in correctly adheres to its
/// actual transfer function, whether or not it was given reasonable inputs. The logic for computing
/// acceptable tolerance ranges is performed in `sampled_noise_amplitude`, `magnitude_noise`, and
/// `phase_noise`.
///
/// # Args
/// * `internal_frequency` - Internal clock frequency (Hz). The internal clock increments timestamp
/// counter values used to record the edges of the external reference.
/// * `adc_frequency` - ADC sampling frequency (in Hz).
/// * `reference_frequency` - External reference frequency (in Hz).
/// * `demodulation_phase_offset` - Phase offset applied to the in-phase and quadrature demodulation
/// signals.
/// * `harmonic` - Scaling factor for the demodulation frequency. E.g., 2 would demodulate with the
/// first harmonic of the reference frequency.
/// * `sample_buffer_size_log2` - The base-2 logarithm of the number of samples in a processing
/// batch.
/// * `pll_shift_frequency` - See `pll::update()`.
/// * `pll_shift_phase` - See `pll::update()`.
/// * `corner_frequency` - Lowpass filter 3dB cutoff frequency.
/// * `desired_input` - `Tone` giving the frequency, amplitude and phase of the desired result.
/// * `noise_inputs` - Vector of `Tone` for any noise inputs on top of `desired_input`.
/// * `time_constant_factor` - Number of time constants after which the output is considered valid.
/// * `tolerance` - Acceptable relative tolerance for the magnitude and angle outputs. This is added
/// to fixed tolerance values computed inside this function. The outputs must remain within this
/// tolerance between `time_constant_factor` and `time_constant_factor+1` time constants.
fn lowpass_test(
    internal_frequency: f64,
    adc_frequency: f64,
    reference_frequency: f64,
    demodulation_phase_offset: f64,
    harmonic: u32,
    sample_buffer_size_log2: usize,
    pll_shift_frequency: u8,
    pll_shift_phase: u8,
    corner_frequency: f64,
    desired_input: Tone,
    tones: &mut Vec<Tone>,
    time_constant_factor: f64,
    tolerance: f64,
) {
    assert!(
        isclose((internal_frequency / adc_frequency).log2(), (internal_frequency / adc_frequency).log2().round(), 0., 1e-5),
        "The number of internal clock cycles in one ADC sampling period must be a power-of-two."
    );

    assert!(
        internal_frequency / reference_frequency
            >= internal_frequency / adc_frequency
            * (1 << sample_buffer_size_log2) as f64,
        "Too many timestamps per batch. Each batch can have at most 1 timestamp."
    );

    let adc_sample_ticks_log2 =
        (internal_frequency / adc_frequency).log2().round() as usize;
    assert!(
        adc_sample_ticks_log2 + sample_buffer_size_log2 <= 32,
        "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."
    );

    let mut lockin = Lockin::new(
        harmonic,
        (demodulation_phase_offset / (2. * PI) * (1_u64 << 32) as f64).round()
            as u32,
        IIR {
            ba: lowpass_iir_coefficients(
                corner_frequency / adc_frequency,
                1. / 2f64.sqrt(),
                2.,
            ),
        },
    );
    let mut timestamp_handler = TimestampHandler::new(
        pll_shift_frequency,
        pll_shift_phase,
        adc_sample_ticks_log2,
        sample_buffer_size_log2,
    );

    let mut timestamp_start: u64 = 0;
    let time_constant: f64 = 1. / (2. * PI * corner_frequency);
    // Account for the pll settling time (see its documentation).
    let pll_time_constant_samples =
        (1 << pll_shift_phase.max(pll_shift_frequency)) as usize;
    let low_pass_time_constant_samples =
        (time_constant_factor * time_constant * adc_frequency
            / (1 << sample_buffer_size_log2) as f64) 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 * adc_frequency) as usize;
    let batch_sample_count =
        1_u64 << (adc_sample_ticks_log2 + sample_buffer_size_log2);

    let effective_phase_offset =
        desired_input.phase - demodulation_phase_offset;
    let in_phase_actual =
        linear(desired_input.amplitude_dbfs) * effective_phase_offset.cos();
    let quadrature_actual =
        linear(desired_input.amplitude_dbfs) * effective_phase_offset.sin();

    let total_noise_amplitude = sampled_noise_amplitude(
        tones,
        reference_frequency * harmonic as f64,
        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(
        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 = adc_sampled_signal(
            &tones,
            timestamp_start as f64 / internal_frequency,
            adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 100e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 100e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 100e3;
    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 = 1e3;
    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,
        adc_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_111e3_phase_offset_pi_4() {
    let internal_frequency: f64 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 111e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 50e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 50e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 50e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 50e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 32.;
    let signal_frequency: f64 = 100e3;
    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 = 1e3;
    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,
        adc_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 = 125e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 100e3;
    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 = 1e3;
    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,
        adc_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 = 100e6;
    let adc_frequency: f64 = internal_frequency / 64.;
    let signal_frequency: f64 = 10e3;
    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 = 1e3;
    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,
        adc_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,
    );
}