HexoDSP/src/dsp/helpers.rs

// Copyright (c) 2021 Weird Constructor <weirdconstructor@gmail.com>
// This is a part of HexoDSP. Released under (A)GPLv3 or any later.
// See README.md and COPYING for details.

/// Logarithmic table size of the table in [fast_cos] / [fast_sin].
static FAST_COS_TAB_LOG2_SIZE : usize = 9;
/// Table size of the table in [fast_cos] / [fast_sin].
static FAST_COS_TAB_SIZE : usize      = 1 << FAST_COS_TAB_LOG2_SIZE; // =512
/// The wave table of [fast_cos] / [fast_sin].
static mut FAST_COS_TAB : [f32; 513] = [0.0; 513];

/// Initializes the cosine wave table for [fast_cos] and [fast_sin].
pub fn init_cos_tab() {
    for i in 0..(FAST_COS_TAB_SIZE+1) {
        let phase : f32 =
            (i as f32)
            * ((std::f32::consts::TAU)
               / (FAST_COS_TAB_SIZE as f32));
        unsafe {
            // XXX: note: mutable statics can be mutated by multiple
            //      threads: aliasing violations or data races
            //      will cause undefined behavior
            FAST_COS_TAB[i] = phase.cos();
        }
    }
}

/// Internal phase increment/scaling for [fast_cos].
const PHASE_SCALE : f32 = 1.0_f32 / (std::f32::consts::TAU);

/// A faster implementation of cosine. It's not that much faster than
/// Rust's built in cosine function. But YMMV.
///
/// Don't forget to call [init_cos_tab] before using this!
///
///```
/// use hexodsp::dsp::helpers::*;
/// init_cos_tab(); // Once on process initialization.
///
/// // ...
/// assert!((fast_cos(std::f32::consts::PI) - -1.0).abs() < 0.001);
///```
pub fn fast_cos(mut x: f32) -> f32 {
    x = x.abs(); // cosine is symmetrical around 0, let's get rid of negative values

    // normalize range from 0..2PI to 1..2
    let phase = x * PHASE_SCALE;

    let index = FAST_COS_TAB_SIZE as f32 * phase;

    let fract = index.fract();
    let index = index.floor() as usize;

    unsafe {
        // XXX: note: mutable statics can be mutated by multiple
        //      threads: aliasing violations or data races
        //      will cause undefined behavior
        let left         = FAST_COS_TAB[index as usize];
        let right        = FAST_COS_TAB[index as usize + 1];

        return left + (right - left) * fract;
    }
}

/// A faster implementation of sine. It's not that much faster than
/// Rust's built in sine function. But YMMV.
///
/// Don't forget to call [init_cos_tab] before using this!
///
///```
/// use hexodsp::dsp::helpers::*;
/// init_cos_tab(); // Once on process initialization.
///
/// // ...
/// assert!((fast_sin(0.5 * std::f32::consts::PI) - 1.0).abs() < 0.001);
///```
pub fn fast_sin(x: f32) -> f32 {
    fast_cos(x - (std::f32::consts::PI / 2.0))
}

/// A wavetable filled entirely with white noise.
/// Don't forget to call [init_white_noise_tab] before using it.
static mut WHITE_NOISE_TAB: [f64; 1024] = [0.0; 1024];

/// Initializes [WHITE_NOISE_TAB].
pub fn init_white_noise_tab() {
    let mut rng = RandGen::new();
    unsafe {
        for i in 0..WHITE_NOISE_TAB.len() {
            WHITE_NOISE_TAB[i as usize] = rng.next_open01();
        }
    }
}

#[derive(Debug, Copy, Clone, PartialEq)]
/// Random number generator based on xoroshiro128.
/// Requires two internal state variables. You may prefer [SplitMix64] or [Rng].
pub struct RandGen {
    r: [u64; 2],
}

// Taken from xoroshiro128 crate under MIT License
// Implemented by Matthew Scharley (Copyright 2016)
// https://github.com/mscharley/rust-xoroshiro128
/// Given the mutable `state` generates the next pseudo random number.
pub fn next_xoroshiro128(state: &mut [u64; 2]) -> u64 {
    let s0: u64     = state[0];
    let mut s1: u64 = state[1];
    let result: u64 = s0.wrapping_add(s1);

    s1 ^= s0;
    state[0] = s0.rotate_left(55) ^ s1 ^ (s1 << 14); // a, b
    state[1] = s1.rotate_left(36); // c

    result
}

// Taken from rand::distributions
// Licensed under the Apache License, Version 2.0
// Copyright 2018 Developers of the Rand project.
/// Maps any `u64` to a `f64` in the open interval `[0.0, 1.0)`.
pub fn u64_to_open01(u: u64) -> f64 {
    use core::f64::EPSILON;
    let float_size         = std::mem::size_of::<f64>() as u32 * 8;
    let fraction           = u >> (float_size - 52);
    let exponent_bits: u64 = (1023 as u64) << 52;
    f64::from_bits(fraction | exponent_bits) - (1.0 - EPSILON / 2.0)
}

impl RandGen {
    pub fn new() -> Self {
        RandGen {
            r: [0x193a6754a8a7d469, 0x97830e05113ba7bb],
        }
    }

    /// Next random unsigned 64bit integer.
    pub fn next(&mut self) -> u64 {
        next_xoroshiro128(&mut self.r)
    }

    /// Next random float between `[0.0, 1.0)`.
    pub fn next_open01(&mut self) -> f64 {
        u64_to_open01(self.next())
    }
}

#[derive(Debug, Copy, Clone)]
/// Random number generator based on [SplitMix64].
/// Requires two internal state variables. You may prefer [SplitMix64] or [Rng].
pub struct Rng {
    sm: SplitMix64,
}

impl Rng {
    pub fn new() -> Self {
        Self { sm: SplitMix64::new(0x193a67f4a8a6d769) }
    }

    pub fn seed(&mut self, seed: u64) {
        println!("SEED {}", seed);
        self.sm = SplitMix64::new(seed);
    }

    #[inline]
    pub fn next(&mut self) -> f32 {
        self.sm.next_open01() as f32
    }
}

// Copyright 2018 Developers of the Rand project.
//
// Licensed under the Apache License, Version 2.0 <LICENSE-APACHE or
// https://www.apache.org/licenses/LICENSE-2.0> or the MIT license
// <LICENSE-MIT or https://opensource.org/licenses/MIT>, at your
// option. This file may not be copied, modified, or distributed
// except according to those terms.
//- splitmix64 (http://xoroshiro.di.unimi.it/splitmix64.c) 
//
/// A splitmix64 random number generator.
///
/// The splitmix algorithm is not suitable for cryptographic purposes, but is
/// very fast and has a 64 bit state.
///
/// The algorithm used here is translated from [the `splitmix64.c`
/// reference source code](http://xoshiro.di.unimi.it/splitmix64.c) by
/// Sebastiano Vigna. For `next_u32`, a more efficient mixing function taken
/// from [`dsiutils`](http://dsiutils.di.unimi.it/) is used.
#[derive(Debug, Copy, Clone)]
pub struct SplitMix64(pub u64);

/// Internal random constant for [SplitMix64].
const PHI: u64 = 0x9e3779b97f4a7c15;

impl SplitMix64 {
    pub fn new(seed: u64) -> Self { Self(seed) }
    pub fn new_from_i64(seed: i64) -> Self {
        Self::new(u64::from_be_bytes(seed.to_be_bytes()))
    }

    #[inline]
    pub fn next_u64(&mut self) -> u64 {
        self.0 = self.0.wrapping_add(PHI);
        let mut z = self.0;
        z = (z ^ (z >> 30)).wrapping_mul(0xbf58476d1ce4e5b9);
        z = (z ^ (z >> 27)).wrapping_mul(0x94d049bb133111eb);
        z ^ (z >> 31)
    }

    #[inline]
    pub fn next_i64(&mut self) -> i64 {
        i64::from_be_bytes(
            self.next_u64().to_be_bytes())
    }

    #[inline]
    pub fn next_open01(&mut self) -> f64 {
        u64_to_open01(self.next_u64())
    }
}

#[inline]
pub fn crossfade(v1: f32, v2: f32, mix: f32) -> f32 {
    v1 * (1.0 - mix) + v2 * mix
}

#[inline]
pub fn clamp(f: f32, min: f32, max: f32) -> f32 {
         if f < min { min }
    else if f > max { max }
    else            {   f }
}

pub fn square_135(phase: f32) -> f32 {
      fast_sin(phase)
    + fast_sin(phase * 3.0) / 3.0
    + fast_sin(phase * 5.0) / 5.0
}

pub fn square_35(phase: f32) -> f32 {
      fast_sin(phase * 3.0) / 3.0
    + fast_sin(phase * 5.0) / 5.0
}

// note: MIDI note value?
pub fn note_to_freq(note: f32) -> f32 {
    440.0 * (2.0_f32).powf((note - 69.0) / 12.0)
}

// Ported from LMMS under GPLv2
// * DspEffectLibrary.h - library with template-based inline-effects
// * Copyright (c) 2006-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
//
/// Signal distortion
/// ```text
/// gain:        0.1 - 5.0       default = 1.0
/// threshold:   0.0 - 100.0     default = 0.8
/// i:           signal
/// ```
pub fn f_distort(gain: f32, threshold: f32, i: f32) -> f32 {
    gain * (
        i * ( i.abs() + threshold )
        / ( i * i + (threshold - 1.0) * i.abs() + 1.0 ))
}

// Ported from LMMS under GPLv2
// * DspEffectLibrary.h - library with template-based inline-effects
// * Copyright (c) 2006-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
//
/// Foldback Signal distortion
/// ```text
/// gain:        0.1 - 5.0       default = 1.0
/// threshold:   0.0 - 100.0     default = 0.8
/// i:           signal
/// ```
pub fn f_fold_distort(gain: f32, threshold: f32, i: f32) -> f32 {
    if i >= threshold || i < -threshold {
        gain
        * ((
            ((i - threshold) % threshold * 4.0).abs()
            - threshold * 2.0).abs()
           - threshold)
    } else {
        gain * i
    }
}

pub fn lerp(x: f32, a: f32, b: f32) -> f32 {
    (a * (1.0 - x)) + (b * x)
}

pub fn lerp64(x: f64, a: f64, b: f64) -> f64 {
    (a * (1.0 - x)) + (b * x)
}

pub fn p2range(x: f32, a: f32, b: f32) -> f32 {
    lerp(x, a, b)
}

pub fn p2range_exp(x: f32, a: f32, b: f32) -> f32 {
    let x = x * x;
    (a * (1.0 - x)) + (b * x)
}

pub fn p2range_exp4(x: f32, a: f32, b: f32) -> f32 {
    let x = x * x * x * x;
    (a * (1.0 - x)) + (b * x)
}


pub fn range2p(v: f32, a: f32, b: f32) -> f32 {
    ((v - a) / (b - a)).abs()
}

pub fn range2p_exp(v: f32, a: f32, b: f32) -> f32 {
    (((v - a) / (b - a)).abs()).sqrt()
}

pub fn range2p_exp4(v: f32, a: f32, b: f32) -> f32 {
    (((v - a) / (b - a)).abs()).sqrt().sqrt()
}

/// ```text
/// gain: 24.0 - -90.0   default = 0.0
/// ```
pub fn gain2coef(gain: f32) -> f32 {
    if gain > -90.0 {
        10.0_f32.powf(gain * 0.05)
    } else {
        0.0
    }
}

// quickerTanh / quickerTanh64 credits to mopo synthesis library:
// Under GPLv3 or any later.
// Little IO <littleioaudio@gmail.com>
// Matt Tytel <matthewtytel@gmail.com>
pub fn quicker_tanh64(v: f64) -> f64 {
    let square = v * v;
    v / (1.0 + square / (3.0 + square / 5.0))
}

pub fn quicker_tanh(v: f32) -> f32 {
    let square = v * v;
    v / (1.0 + square / (3.0 + square / 5.0))
}

// quickTanh / quickTanh64 credits to mopo synthesis library:
// Under GPLv3 or any later.
// Little IO <littleioaudio@gmail.com>
// Matt Tytel <matthewtytel@gmail.com>
pub fn quick_tanh64(v: f64) -> f64 {
    let abs_v = v.abs();
    let square = v * v;
    let num =
        v * (2.45550750702956
             + 2.45550750702956 * abs_v
             + square * (0.893229853513558
                         + 0.821226666969744 * abs_v));
    let den =
        2.44506634652299
        + (2.44506634652299 + square)
          * (v + 0.814642734961073 * v * abs_v).abs();

    num / den
}

pub fn quick_tanh(v: f32) -> f32 {
    let abs_v = v.abs();
    let square = v * v;
    let num =
        v * (2.45550750702956
             + 2.45550750702956 * abs_v
             + square * (0.893229853513558
                         + 0.821226666969744 * abs_v));
    let den =
        2.44506634652299
        + (2.44506634652299 + square)
          * (v + 0.814642734961073 * v * abs_v).abs();

    num / den
}

/// A helper function for exponential envelopes.
/// It's a bit faster than calling the `pow` function of Rust.
///
/// * `x` the input value
/// * `v' the shape value.
/// Which is linear at `0.5`, the forth root of `x` at `1.0` and x to the power
/// of 4 at `0.0`. You can vary `v` as you like.
///
///```
/// use hexodsp::dsp::helpers::*;
///
/// assert!(((sqrt4_to_pow4(0.25, 0.0) - 0.25_f32 * 0.25 * 0.25 * 0.25)
///          .abs() - 1.0)
///         < 0.0001);
///
/// assert!(((sqrt4_to_pow4(0.25, 1.0) - (0.25_f32).sqrt().sqrt())
///          .abs() - 1.0)
///         < 0.0001);
///
/// assert!(((sqrt4_to_pow4(0.25, 0.5) - 0.25_f32).abs() - 1.0) < 0.0001);
///```
#[inline]
pub fn sqrt4_to_pow4(x: f32, v: f32) -> f32 {
    if v > 0.75 {
        let xsq1 = x.sqrt();
        let xsq = xsq1.sqrt();
        let v = (v - 0.75) * 4.0;
        xsq1 * (1.0 - v) + xsq * v

    } else if v > 0.5 {
        let xsq = x.sqrt();
        let v = (v - 0.5) * 4.0;
        x * (1.0 - v) + xsq * v

    } else if v > 0.25 {
        let xx = x * x;
        let v = (v - 0.25) * 4.0;
        x * v + xx * (1.0 - v)

    } else {
        let xx = x * x;
        let xxxx = xx * xx;
        let v = v * 4.0;
        xx * v + xxxx * (1.0 - v)
    }
}

/// A-100 Eurorack states, that a trigger is usually 2-10 milliseconds.
const TRIG_SIGNAL_LENGTH_MS : f32 = 2.0;

#[derive(Debug, Clone, Copy)]
pub struct TrigSignal {
    length:     u32,
    scount:     u32,
}

impl TrigSignal {
    pub fn new() -> Self {
        Self {
            length: ((44100.0 * TRIG_SIGNAL_LENGTH_MS) / 1000.0).ceil() as u32,
            scount: 0,
        }
    }

    pub fn reset(&mut self) {
        self.scount = 0;
    }

    pub fn set_sample_rate(&mut self, srate: f32) {
        self.length = ((srate * TRIG_SIGNAL_LENGTH_MS) / 1000.0).ceil() as u32;
        self.scount = 0;
    }

    #[inline]
    pub fn trigger(&mut self) { self.scount = self.length; }

    #[inline]
    pub fn next(&mut self) -> f32 {
        if self.scount > 0 {
            self.scount -= 1;
            1.0
        } else {
            0.0
        }
    }
}

impl Default for TrigSignal {
    fn default() -> Self { Self::new() }
}

#[derive(Debug, Clone, Copy)]
pub struct Trigger {
    triggered:  bool,
}

impl Trigger {
    pub fn new() -> Self {
        Self {
            triggered: false,
        }
    }

    #[inline]
    pub fn reset(&mut self) {
        self.triggered = false;
    }

    #[inline]
    pub fn check_trigger(&mut self, input: f32) -> bool {
        if self.triggered {
            if input <= 0.25 {
                self.triggered = false;
            }

            false

        } else if input > 0.75 {
            self.triggered = true;
            true

        } else {
            false
        }
    }
}

#[derive(Debug, Clone, Copy)]
pub struct TriggerPhaseClock {
    clock_phase:    f64,
    clock_inc:      f64,
    prev_trigger:   bool,
    clock_samples:  u32,
}

impl TriggerPhaseClock {
    pub fn new() -> Self {
        Self {
            clock_phase:    0.0,
            clock_inc:      0.0,
            prev_trigger:   true,
            clock_samples:  0,
        }
    }

    #[inline]
    pub fn reset(&mut self) {
        self.clock_samples = 0;
        self.clock_inc     = 0.0;
        self.prev_trigger  = true;
        self.clock_samples = 0;
    }

    #[inline]
    pub fn sync(&mut self) {
        self.clock_phase = 0.0;
    }

    #[inline]
    pub fn next_phase(&mut self, clock_limit: f64, trigger_in: f32) -> f64 {
        if self.prev_trigger {
            if trigger_in <= 0.25 {
                self.prev_trigger = false;
            }

        } else if trigger_in > 0.75 {
            self.prev_trigger = true;

            if self.clock_samples > 0 {
                self.clock_inc =
                    1.0 / (self.clock_samples as f64);
            }

            self.clock_samples = 0;
        }

        self.clock_samples += 1;

        self.clock_phase += self.clock_inc;
        self.clock_phase = self.clock_phase % clock_limit;

        self.clock_phase
    }
}

#[derive(Debug, Clone, Copy)]
pub struct TriggerSampleClock {
    prev_trigger:   bool,
    clock_samples:  u32,
    counter:        u32,
}

impl TriggerSampleClock {
    pub fn new() -> Self {
        Self {
            prev_trigger:   true,
            clock_samples:  0,
            counter:        0,
        }
    }

    #[inline]
    pub fn reset(&mut self) {
        self.clock_samples = 0;
        self.counter       = 0;
    }

    #[inline]
    pub fn next(&mut self, trigger_in: f32) -> u32 {
        if self.prev_trigger {
            if trigger_in <= 0.25 {
                self.prev_trigger = false;
            }

        } else if trigger_in > 0.75 {
            self.prev_trigger  = true;
            self.clock_samples = self.counter;
            self.counter       = 0;
        }

        self.counter += 1;

        self.clock_samples
    }
}

/// Default size of the delay buffer: 5 seconds at 8 times 48kHz
const DEFAULT_DELAY_BUFFER_SAMPLES : usize = 8 * 48000 * 5;

#[derive(Debug, Clone)]
pub struct DelayBuffer {
    data:   Vec<f32>,
    wr:     usize,
    srate:  f32,
}

impl DelayBuffer {
    pub fn new() -> Self {
        Self {
            data:   vec![0.0; DEFAULT_DELAY_BUFFER_SAMPLES],
            wr:     0,
            srate:  44100.0,
        }
    }

    pub fn new_with_size(size: usize) -> Self {
        Self {
            data:   vec![0.0; size],
            wr:     0,
            srate:  44100.0,
        }
    }

    pub fn set_sample_rate(&mut self, srate: f32) {
        self.srate = srate;
    }

    pub fn reset(&mut self) {
        self.data.fill(0.0);
        self.wr = 0;
    }

    /// Feed one sample into the delay line and increment the write pointer.
    /// Please note: For sample accurate feedback you need to retrieve the
    /// output of the delay line before feeding in a new signal.
    #[inline]
    pub fn feed(&mut self, input: f32) {
        self.data[self.wr] = input;
        self.wr = (self.wr + 1) % self.data.len();
    }

    #[inline]
    pub fn cubic_interpolate_at(&self, delay_time: f32) -> f32 {
        let data   = &self.data[..];
        let len    = data.len();
        let s_offs = (delay_time * self.srate) / 1000.0;
        let offs   = s_offs.floor() as usize % len;
        let fract  = s_offs.fract();

        let i = (self.wr + len) - offs;

        // Hermite interpolation, take from 
        // https://github.com/eric-wood/delay/blob/main/src/delay.rs#L52
        //
        // Thanks go to Eric Wood!
        //
        // For the interpolation code:
        // MIT License, Copyright (c) 2021 Eric Wood
        let xm1 = data[(i - 1) % len];
        let x0  = data[i       % len];
        let x1  = data[(i + 1) % len];
        let x2  = data[(i + 2) % len];

        let c     = (x1 - xm1) * 0.5;
        let v     = x0 - x1;
        let w     = c + v;
        let a     = w + v + (x2 - x0) * 0.5;
        let b_neg = w + a;

        let fract = fract as f32;
        (((a * fract) - b_neg) * fract + c) * fract + x0
    }

    #[inline]
    pub fn nearest_at(&self, delay_time: f32) -> f32 {
        let len  = self.data.len();
        let offs = (delay_time * self.srate).floor() as usize % len;
        let idx  = ((self.wr + len) - offs) % len;
        self.data[idx]
    }

    #[inline]
    pub fn at(&self, delay_sample_count: usize) -> f32 {
        let len  = self.data.len();
        let idx  = ((self.wr + len) - delay_sample_count) % len;
        self.data[idx]
    }
}

/// Default size of the delay buffer: 1 seconds at 8 times 48kHz
const DEFAULT_ALLPASS_COMB_SAMPLES : usize = 8 * 48000;

#[derive(Debug, Clone)]
pub struct AllPass {
    delay: DelayBuffer,
}

impl AllPass {
    pub fn new() -> Self {
        Self {
            delay: DelayBuffer::new_with_size(DEFAULT_ALLPASS_COMB_SAMPLES),
        }
    }

    pub fn set_sample_rate(&mut self, srate: f32) {
        self.delay.set_sample_rate(srate);
    }

    pub fn reset(&mut self) {
        self.delay.reset();
    }

    #[inline]
    pub fn next(&mut self, time: f32, g: f32, v: f32) -> f32 {
        let s = self.delay.cubic_interpolate_at(time);
        self.delay.feed(v + s * g);
        s + -1.0 * g * v
    }
}

#[derive(Debug, Clone)]
pub struct Comb {
    delay: DelayBuffer,
}

impl Comb {
    pub fn new() -> Self {
        Self {
            delay: DelayBuffer::new_with_size(DEFAULT_ALLPASS_COMB_SAMPLES),
        }
    }

    pub fn set_sample_rate(&mut self, srate: f32) {
        self.delay.set_sample_rate(srate);
    }

    pub fn reset(&mut self) {
        self.delay.reset();
    }

    #[inline]
    pub fn next_feedback(&mut self, time: f32, g: f32, v: f32) -> f32 {
        let s = self.delay.cubic_interpolate_at(time);
        self.delay.feed(v + s * g);
        v
    }

    #[inline]
    pub fn next_feedforward(&mut self, time: f32, g: f32, v: f32) -> f32 {
        let s = self.delay.cubic_interpolate_at(time);
        self.delay.feed(v);
        v + s * g
    }
}

// one pole lp from valley rack free:
// https://github.com/ValleyAudio/ValleyRackFree/blob/v1.0/src/Common/DSP/OnePoleFilters.cpp
#[inline]
/// Process a very simple one pole 6dB low pass filter.
/// Useful in various applications, from usage in a synthesizer to
/// damping stuff in a reverb/delay.
///
/// * `input`  - Input sample
/// * `freq`   - Frequency between 1.0 and 22000.0Hz
/// * `israte` - 1.0 / samplerate
/// * `z`      - The internal one sample buffer of the filter.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut z    = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let s_out =
///            process_1pole_lowpass(*s, freq, 1.0 / 44100.0, &mut z);
///        // ... do something with the result here.
///    }
///```
pub fn process_1pole_lowpass(input: f32, freq: f32, israte: f32, z: &mut f32) -> f32 {
    let b = (-std::f32::consts::TAU * freq * israte).exp();
    let a = 1.0 - b;
    *z = a * input + *z * b;
    *z
}

// one pole hp from valley rack free:
// https://github.com/ValleyAudio/ValleyRackFree/blob/v1.0/src/Common/DSP/OnePoleFilters.cpp
#[inline]
/// Process a very simple one pole 6dB high pass filter.
/// Useful in various applications.
///
/// * `input`  - Input sample
/// * `freq`   - Frequency between 1.0 and 22000.0Hz
/// * `israte` - 1.0 / samplerate
/// * `z`      - The first internal buffer of the filter.
/// * `y`      - The second internal buffer of the filter.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut z    = 0.0;
///    let mut y    = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let s_out =
///            process_1pole_highpass(*s, freq, 1.0 / 44100.0, &mut z, &mut y);
///        // ... do something with the result here.
///    }
///```
pub fn process_1pole_highpass(input: f32, freq: f32, israte: f32, z: &mut f32, y: &mut f32) -> f32 {
    let b  = (-std::f32::consts::TAU * freq * israte).exp();
    let a  = (1.0 + b) / 2.0;

    let v =
          a  * input
        - a  * *z
        + b  * *y;
    *y = v;
    *z = input;
    v
}

// one pole from:
// http://www.willpirkle.com/Downloads/AN-4VirtualAnalogFilters.pdf
// (page 5)
#[inline]
/// Process a very simple one pole 6dB low pass filter in TPT form.
/// Useful in various applications, from usage in a synthesizer to
/// damping stuff in a reverb/delay.
///
/// * `input`  - Input sample
/// * `freq`   - Frequency between 1.0 and 22000.0Hz
/// * `israte` - 1.0 / samplerate
/// * `z`      - The internal one sample buffer of the filter.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut z    = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let s_out =
///            process_1pole_tpt_highpass(*s, freq, 1.0 / 44100.0, &mut z);
///        // ... do something with the result here.
///    }
///```
pub fn process_1pole_tpt_lowpass(input: f32, freq: f32, israte: f32, z: &mut f32) -> f32 {
    let g = (std::f32::consts::PI * freq * israte).tan();
    let a = g / (1.0 + g);

    let v1 = a * (input - *z);
    let v2 = v1 + *z;
    *z = v2 + v1;

    // let (m0, m1) = (0.0, 1.0);
    // (m0 * input + m1 * v2) as f32);
    v2
}

// one pole from:
// http://www.willpirkle.com/Downloads/AN-4VirtualAnalogFilters.pdf
// (page 5)
#[inline]
/// Process a very simple one pole 6dB high pass filter in TPT form.
/// Useful in various applications.
///
/// * `input`  - Input sample
/// * `freq`   - Frequency between 1.0 and 22000.0Hz
/// * `israte` - 1.0 / samplerate
/// * `z`      - The internal one sample buffer of the filter.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut z    = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let s_out =
///            process_1pole_tpt_lowpass(*s, freq, 1.0 / 44100.0, &mut z);
///        // ... do something with the result here.
///    }
///```
pub fn process_1pole_tpt_highpass(input: f32, freq: f32, israte: f32, z: &mut f32) -> f32 {
    let g  = (std::f32::consts::PI * freq * israte).tan();
    let a1 = g / (1.0 + g);

    let v1 = a1 * (input - *z);
    let v2 = v1 + *z;
    *z = v2 + v1;

    input - v2
}

/// The internal oversampling factor of [process_hal_chamberlin_svf].
const FILTER_OVERSAMPLE_HAL_CHAMBERLIN : usize = 2;
// Hal Chamberlin's State Variable (12dB/oct) filter
// https://www.earlevel.com/main/2003/03/02/the-digital-state-variable-filter/
// Inspired by SynthV1 by Rui Nuno Capela, under the terms of
// GPLv2 or any later:
/// Process a HAL Chamberlin filter with two delays/state variables that is 12dB.
/// The filter does internal oversampling with very simple decimation to
/// rise the stability for cutoff frequency up to 16kHz.
///
/// * `input` - Input sample.
/// * `freq` - Frequency in Hz. Please keep it inside 0.0 to 16000.0 Hz!
/// otherwise the filter becomes unstable.
/// * `res`  - Resonance from 0.0 to 0.99. Resonance of 1.0 is not recommended,
/// as the filter will then oscillate itself out of control.
/// * `israte` - 1.0 divided by the sampling rate (eg. 1.0 / 44100.0).
/// * `band` - First state variable, containing the band pass result
/// after processing.
/// * `low` - Second state variable, containing the low pass result
/// after processing.
///
/// Returned are the results of the high and notch filter.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut band = 0.0;
///    let mut low  = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let (high, notch) =
///            process_hal_chamberlin_svf(
///                *s, freq, 0.5, 1.0 / 44100.0, &mut band, &mut low);
///        // ... do something with the result here.
///    }
///```
#[inline]
pub fn process_hal_chamberlin_svf(
    input: f32, freq: f32, res: f32, israte: f32, band: &mut f32, low: &mut f32)
    -> (f32, f32)
{
    let q      = 1.0 - res;
    let cutoff = 2.0 * (std::f32::consts::PI * freq * 0.5 * israte).sin();

    let mut high  = 0.0;
    let mut notch = 0.0;

    for _ in 0..FILTER_OVERSAMPLE_HAL_CHAMBERLIN {
        *low  += cutoff * *band;
        high = input - *low - q * *band;
        *band += cutoff * high;
        notch = high + *low;
    }

    //d// println!("q={:4.2} cut={:8.3} freq={:8.1} LP={:8.3} HP={:8.3} BP={:8.3} N={:8.3}",
    //d//     q, cutoff, freq, *low, high, *band, notch);

    (high, notch)
}

/// This function processes a Simper SVF with 12dB. It's a much newer algorithm
/// for filtering and provides easy to calculate multiple outputs.
///
/// * `input` - Input sample.
/// * `freq` - Frequency in Hz.
/// otherwise the filter becomes unstable.
/// * `res`  - Resonance from 0.0 to 0.99. Resonance of 1.0 is not recommended,
/// as the filter will then oscillate itself out of control.
/// * `israte` - 1.0 divided by the sampling rate (eg. 1.0 / 44100.0).
/// * `band` - First state variable, containing the band pass result
/// after processing.
/// * `low` - Second state variable, containing the low pass result
/// after processing.
///
/// This function returns the low pass, band pass and high pass signal.
/// For a notch or peak filter signal, please consult the following example:
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples   = vec![0.0; 44100];
///    let mut ic1eq = 0.0;
///    let mut ic2eq = 0.0;
///    let mut freq  = 1000.0;
///
///    for s in samples.iter() {
///        let (low, band, high) =
///            process_simper_svf(
///                *s, freq, 0.5, 1.0 / 44100.0, &mut ic1eq, &mut ic2eq);
///
///        // You can easily calculate the notch and peak results too:
///        let notch = low + high;
///        let peak  = low - high;
///        // ... do something with the result here.
///    }
///```
// Simper SVF taken from baseplug (Rust crate) example svf_simper.rs:
// implemented from https://cytomic.com/files/dsp/SvfLinearTrapOptimised2.pdf
// thanks, andy!
#[inline]
pub fn process_simper_svf(
    input: f32, freq: f32, res: f32, israte: f32, ic1eq: &mut f32, ic2eq: &mut f32
) -> (f32, f32, f32) {
    let g = (std::f32::consts::PI * freq * israte).tan();
    // XXX: the 1.989 were tuned by hand, so the resonance is more audible.
    let k = 2f32 - (1.989f32 * res);

    let a1 = 1.0 / (1.0 + (g * (g + k)));
    let a2 = g * a1;
    let a3 = g * a2;

    let v3 = input - *ic2eq;
    let v1 = (a1 * *ic1eq) + (a2 * v3);
    let v2 = *ic2eq + (a2 * *ic1eq) + (a3 * v3);

    *ic1eq = (2.0 * v1) - *ic1eq;
    *ic2eq = (2.0 * v2) - *ic2eq;

    // low   = v2
    // band  = v1
    // high  = input - k * v1 - v2
    // notch = low + high            = input - k * v1
    // peak  = low - high            = 2 * v2 - input + k * v1
    // all   = low + high - k * band = input - 2 * k * v1

    (v2, v1, input - k * v1 - v2)
}

/// This function implements a simple Stilson/Moog filter with 24dB.
/// It provides multiple outputs for low, high and band pass and a notch
/// output.
///
/// * `input` - Input sample.
/// * `freq` - Frequency in Hz.
/// otherwise the filter becomes unstable.
/// * `res`  - Resonance from 0.0 to 0.99. Resonance of 1.0 is not recommended,
/// as the filter will then oscillate itself out of control.
/// * `israte` - 1.0 divided by the sampling rate (`1.0 / 44100.0`).
/// * `b0` to `b4` - Internal values used for filtering.
///
///```
///    use hexodsp::dsp::helpers::*;
///
///    let samples  = vec![0.0; 44100];
///    let mut b0   = 0.0;
///    let mut b1   = 0.0;
///    let mut b2   = 0.0;
///    let mut b3   = 0.0;
///    let mut b4   = 0.0;
///    let mut freq = 1000.0;
///
///    for s in samples.iter() {
///        let (low, band, high, notch) =
///            process_stilson_moog(
///                *s, freq, 0.5, 1.0 / 44100.0,
///                &mut b0, &mut b1, &mut b2, &mut b3, &mut b4);
///
///        // ... do something with the result here.
///    }
///```
// Stilson/Moog implementation partly translated from SynthV1 by rncbc
// https://github.com/rncbc/synthv1/blob/master/src/synthv1_filter.h#L103
// under GPLv2 or any later.
//
// It's also found on MusicDSP and has probably no proper license anyways.
// See also: https://github.com/ddiakopoulos/MoogLadders
// and https://github.com/ddiakopoulos/MoogLadders/blob/master/src/MusicDSPModel.h
#[inline]
pub fn process_stilson_moog(
    input: f32, freq: f32, res: f32, israte: f32,
    b0: &mut f32, b1: &mut f32, b2: &mut f32, b3: &mut f32, b4: &mut f32
) -> (f32, f32, f32, f32) {
    let cutoff = 2.0 * freq * israte;

    let c = 1.0 - cutoff;
    let p = cutoff + 0.8 * cutoff * c;
    let f = p + p - 1.0;
    let q = res * (1.0 + 0.5 * c * (1.0 - c + 5.6 * c * c));

    let inp = input - q * *b4;
    let t1 = *b1; *b1 = (inp + *b0) * p - *b1 * f;
    let t2 = *b2; *b2 = (*b1 + t1)  * p - *b2 * f;
    let t1 = *b3; *b3 = (*b2 + t2)  * p - *b3 * f;
                  *b4 = (*b3 + t1)  * p - *b4 * f;

    *b4 = *b4 - *b4 * *b4 * *b4 * 0.166667; // clipping

    *b0 = inp;

    let band = 3.0 * (*b3 - *b4);

    // low, band, high, notch
    (*b4, band, inp - *b4, band - inp)
}

// translated from Odin 2 Synthesizer Plugin
// Copyright (C) 2020 TheWaveWarden
// under GPLv3 or any later
#[derive(Debug, Clone)]
pub struct DCBlockFilter {
    xm1:    f64,
    ym1:    f64,
    r:      f64,
}

impl DCBlockFilter {
    pub fn new() -> Self {
        Self {
            xm1: 0.0,
            ym1: 0.0,
            r:   0.995,
        }
    }

    pub fn reset(&mut self) {
        self.xm1 = 0.0;
        self.ym1 = 0.0;
    }

    pub fn set_sample_rate(&mut self, srate: f32) {
        self.r = 0.995;
        if srate > 90000.0 {
            self.r = 0.9965;
        } else if srate > 120000.0 {
            self.r = 0.997;
        }
    }

    pub fn next(&mut self, input: f32) -> f32 {
        let y = input as f64 - self.xm1 + self.r * self.ym1;
        self.xm1 = input as f64;
        self.ym1 = y;
        y as f32
    }
}



#[cfg(test)]
mod tests {
    use super::*;

    #[test]
    fn check_range2p_exp() {
        let a = p2range_exp(0.5, 1.0, 100.0);
        let x = range2p_exp(a, 1.0, 100.0);

        assert!((x - 0.5).abs() < std::f32::EPSILON);
    }

    #[test]
    fn check_range2p() {
        let a = p2range(0.5, 1.0, 100.0);
        let x = range2p(a, 1.0, 100.0);

        assert!((x - 0.5).abs() < std::f32::EPSILON);
    }
}