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HexoDSP/src/dsp/helpers.rs
Weird Constructor dae7c512eb I think the reverb is now mostly done!
2021-08-08 13:43:51 +02:00

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Rust
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// Copyright (c) 2021 Weird Constructor <weirdconstructor@gmail.com>
// This file is a part of HexoDSP. Released under GPL-3.0-or-later.
// See README.md and COPYING for details.
use std::cell::RefCell;
use num_traits::{Float, FloatConst, cast::FromPrimitive, cast::ToPrimitive};
macro_rules! trait_alias {
($name:ident = $base1:ident + $($base2:ident +)+) => {
pub trait $name: $base1 $(+ $base2)+ { }
impl<T: $base1 $(+ $base2)+> $name for T { }
};
}
trait_alias!(Flt = Float + FloatConst + ToPrimitive + FromPrimitive +);
/// 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];
#[allow(rustdoc::private_intra_doc_links)]
/// 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) {
self.sm = SplitMix64::new(seed);
}
#[inline]
pub fn next(&mut self) -> f32 {
self.sm.next_open01() as f32
}
#[inline]
pub fn next_u64(&mut self) -> u64 {
self.sm.next_u64()
}
}
thread_local! {
static GLOBAL_RNG: RefCell<Rng> = RefCell::new(Rng::new());
}
#[inline]
pub fn rand_01() -> f32 {
GLOBAL_RNG.with(|r| r.borrow_mut().next())
}
#[inline]
pub fn rand_u64() -> u64 {
GLOBAL_RNG.with(|r| r.borrow_mut().next_u64())
}
// 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()))
}
pub fn new_time_seed() -> Self {
use std::time::SystemTime;
match SystemTime::now().duration_since(SystemTime::UNIX_EPOCH) {
Ok(n) => Self::new(n.as_secs() as u64),
Err(_) => Self::new(123456789),
}
}
#[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())
}
}
/// Linear crossfade.
///
/// * `v1` - signal 1, range -1.0 to 1.0
/// * `v2` - signal 2, range -1.0 to 1.0
/// * `mix` - mix position, range 0.0 to 1.0, mid is at 0.5
#[inline]
pub fn crossfade<F: Flt>(v1: F, v2: F, mix: F) -> F {
v1 * (f::<F>(1.0) - mix) + v2 * mix
}
/// Constant power crossfade.
///
/// * `v1` - signal 1, range -1.0 to 1.0
/// * `v2` - signal 2, range -1.0 to 1.0
/// * `mix` - mix position, range 0.0 to 1.0, mid is at 0.5
#[inline]
pub fn crossfade_cpow(v1: f32, v2: f32, mix: f32) -> f32 {
let s1 = (mix * std::f32::consts::FRAC_PI_2).sin();
let s2 = ((1.0 - mix) * std::f32::consts::FRAC_PI_2).sin();
v1 * s2 + v2 * s1
}
const CROSS_LOG_MIN : f32 = -13.815510557964274; // (0.000001_f32).ln();
const CROSS_LOG_MAX : f32 = 0.0; // (1.0_f32).ln();
/// Logarithmic crossfade.
///
/// * `v1` - signal 1, range -1.0 to 1.0
/// * `v2` - signal 2, range -1.0 to 1.0
/// * `mix` - mix position, range 0.0 to 1.0, mid is at 0.5
#[inline]
pub fn crossfade_log(v1: f32, v2: f32, mix: f32) -> f32 {
let x =
(mix * (CROSS_LOG_MAX - CROSS_LOG_MIN) + CROSS_LOG_MIN)
.exp();
crossfade(v1, v2, x)
}
/// Exponential crossfade.
///
/// * `v1` - signal 1, range -1.0 to 1.0
/// * `v2` - signal 2, range -1.0 to 1.0
/// * `mix` - mix position, range 0.0 to 1.0, mid is at 0.5
#[inline]
pub fn crossfade_exp(v1: f32, v2: f32, mix: f32) -> f32 {
crossfade(v1, v2, mix * 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>
//
// Original source seems to be musicdsp.org, Author: Bram de Jong
// see also: https://www.musicdsp.org/en/latest/Effects/41-waveshaper.html
// Notes:
// where x (in [-1..1] will be distorted and a is a distortion parameter
// that goes from 1 to infinity. The equation is valid for positive and
// negativ values. If a is 1, it results in a slight distortion and with
// bigger a's the signal get's more funky.
// A good thing about the shaper is that feeding it with bigger-than-one
// values, doesn't create strange fx. The maximum this function will reach
// is 1.2 for a=1.
//
// f(x,a) = x*(abs(x) + a)/(x^2 + (a-1)*abs(x) + 1)
/// Signal distortion by Bram de Jong.
/// ```text
/// gain: 0.1 - 5.0 default = 1.0
/// threshold: 0.0 - 100.0 default = 0.8
/// i: signal
/// ```
#[inline]
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
/// ```
#[inline]
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))
}
#[inline]
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;
macro_rules! fc {
($F: ident, $e: expr) => { F::from_f64($e).unwrap() }
}
#[inline]
fn f<F: Flt>(x: f64) -> F { F::from_f64(x).unwrap() }
fn fclamp<F: Flt>(x: F, mi: F, mx: F) -> F { x.max(mi).min(mx) }
fn fclampc<F: Flt>(x: F, mi: f64, mx: f64) -> F { x.max(f(mi)).min(f(mx)) }
#[derive(Debug, Clone, Default)]
pub struct DelayBuffer<F: Flt> {
data: Vec<F>,
wr: usize,
srate: F,
}
impl<F: Flt> DelayBuffer<F> {
pub fn new() -> Self {
Self {
data: vec![f(0.0); DEFAULT_DELAY_BUFFER_SAMPLES],
wr: 0,
srate: f(44100.0),
}
}
pub fn new_with_size(size: usize) -> Self {
Self {
data: vec![f(0.0); size],
wr: 0,
srate: f(44100.0),
}
}
pub fn set_sample_rate(&mut self, srate: F) {
self.srate = srate;
}
pub fn reset(&mut self) {
self.data.fill(f(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: F) {
self.data[self.wr] = input;
self.wr = (self.wr + 1) % self.data.len();
}
/// Combines [DelayBuffer::cubic_interpolate_at] and [DelayBuffer::feed]
/// into one convenient function.
#[inline]
pub fn next_cubic(&mut self, delay_time_ms: F, input: F) -> F {
let res = self.cubic_interpolate_at(delay_time_ms);
self.feed(input);
res
}
/// Combines [DelayBuffer::linear_interpolate_at] and [DelayBuffer::feed]
/// into one convenient function.
#[inline]
pub fn next_linear(&mut self, delay_time_ms: F, input: F) -> F {
let res = self.linear_interpolate_at(delay_time_ms);
self.feed(input);
res
}
/// Shorthand for [DelayBuffer::cubic_interpolate_at].
#[inline]
pub fn tap_c(&self, delay_time_ms: F) -> F {
self.cubic_interpolate_at(delay_time_ms)
}
/// Shorthand for [DelayBuffer::cubic_interpolate_at].
#[inline]
pub fn tap_n(&self, delay_time_ms: F) -> F {
self.nearest_at(delay_time_ms)
}
/// Shorthand for [DelayBuffer::cubic_interpolate_at].
#[inline]
pub fn tap_l(&self, delay_time_ms: F) -> F {
self.linear_interpolate_at(delay_time_ms)
}
/// Fetch a sample from the delay buffer at the given time.
///
/// * `delay_time_ms` - Delay time in milliseconds.
pub fn linear_interpolate_at(&self, delay_time_ms: F) -> F {
let data = &self.data[..];
let len = data.len();
let s_offs = (delay_time_ms * self.srate) / f(1000.0);
let offs = s_offs.floor().to_usize().unwrap() % len;
let fract = s_offs.fract();
let i = (self.wr + len) - offs;
let x0 = data[i % len];
let x1 = data[(i + 1) % len];
x0 + fract * (x1 - x0)
}
/// Fetch a sample from the delay buffer at the given time.
///
/// * `delay_time_ms` - Delay time in milliseconds.
#[inline]
pub fn cubic_interpolate_at(&self, delay_time_ms: F) -> F {
let data = &self.data[..];
let len = data.len();
let s_offs = (delay_time_ms * self.srate) / f(1000.0);
let offs = s_offs.floor().to_usize().unwrap() % 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) * f(0.5);
let v = x0 - x1;
let w = c + v;
let a = w + v + (x2 - x0) * f(0.5);
let b_neg = w + a;
let fract = fract as F;
(((a * fract) - b_neg) * fract + c) * fract + x0
}
#[inline]
pub fn nearest_at(&self, delay_time_ms: F) -> F {
let len = self.data.len();
let offs =
((delay_time_ms * self.srate)
/ f(1000.0))
.floor().to_usize().unwrap() % len;
let idx = ((self.wr + len) - offs) % len;
self.data[idx]
}
#[inline]
pub fn at(&self, delay_sample_count: usize) -> F {
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, Default)]
pub struct AllPass<F: Flt> {
delay: DelayBuffer<F>,
}
impl<F: Flt> AllPass<F> {
pub fn new() -> Self {
Self {
delay: DelayBuffer::new_with_size(DEFAULT_ALLPASS_COMB_SAMPLES),
}
}
pub fn set_sample_rate(&mut self, srate: F) {
self.delay.set_sample_rate(srate);
}
pub fn reset(&mut self) {
self.delay.reset();
}
#[inline]
pub fn delay_tap_n(&self, time_ms: F) -> F {
self.delay.tap_n(time_ms)
}
#[inline]
pub fn next(&mut self, time_ms: F, g: F, v: F) -> F {
let s = self.delay.linear_interpolate_at(time_ms);
let input = v + g * s;
self.delay.feed(input);
input * -g + s
}
}
#[derive(Debug, Clone)]
pub struct Comb {
delay: DelayBuffer<f32>,
}
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 delay_tap_c(&self, time_ms: f32) -> f32 {
self.delay.tap_c(time_ms)
}
#[inline]
pub fn delay_tap_n(&self, time_ms: f32) -> f32 {
self.delay.tap_n(time_ms)
}
#[inline]
pub fn next_feedback(&mut self, time: f32, g: f32, v: f32) -> f32 {
let s = self.delay.cubic_interpolate_at(time);
let v = v + s * g;
self.delay.feed(v);
v
}
#[inline]
pub fn next_feedforward(&mut self, time: f32, g: f32, v: f32) -> f32 {
let s = self.delay.next_cubic(time, 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
}
#[derive(Debug, Clone, Copy, Default)]
pub struct OnePoleLPF<F: Flt> {
israte: F,
a: F,
b: F,
freq: F,
z: F,
}
impl<F: Flt> OnePoleLPF<F> {
pub fn new() -> Self {
Self {
israte: f::<F>(1.0) / f(44100.0),
a: f::<F>(0.0),
b: f::<F>(0.0),
freq: f::<F>(1000.0),
z: f::<F>(0.0),
}
}
pub fn reset(&mut self) {
self.z = f(0.0);
}
#[inline]
fn recalc(&mut self) {
self.b = (f::<F>(-1.0) * F::TAU() * self.freq * self.israte).exp();
self.a = f::<F>(1.0) - self.b;
}
pub fn set_sample_rate(&mut self, srate: F) {
self.israte = f::<F>(1.0) / srate;
self.recalc();
}
#[inline]
pub fn set_freq(&mut self, freq: F) {
if freq != self.freq {
self.freq = freq;
self.recalc();
}
}
#[inline]
pub fn process(&mut self, input: F) -> F {
self.z = self.a * input + self.z * self.b;
self.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
}
#[derive(Debug, Clone, Copy, Default)]
pub struct OnePoleHPF<F: Flt> {
israte: F,
a: F,
b: F,
freq: F,
z: F,
y: F,
}
impl<F: Flt> OnePoleHPF<F> {
pub fn new() -> Self {
Self {
israte: f(1.0 / 44100.0),
a: f(0.0),
b: f(0.0),
freq: f(1000.0),
z: f(0.0),
y: f(0.0),
}
}
pub fn reset(&mut self) {
self.z = f(0.0);
self.y = f(0.0);
}
#[inline]
fn recalc(&mut self) {
self.b = (f::<F>(-1.0) * F::TAU() * self.freq * self.israte).exp();
self.a = (f::<F>(1.0) + self.b) / f(2.0);
}
pub fn set_sample_rate(&mut self, srate: F) {
self.israte = f::<F>(1.0) / srate;
self.recalc();
}
#[inline]
pub fn set_freq(&mut self, freq: F) {
if freq != self.freq {
self.freq = freq;
self.recalc();
}
}
#[inline]
pub fn process(&mut self, input: F) -> F {
let v =
self.a * input
- self.a * self.z
+ self.b * self.y;
self.y = v;
self.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 implemented from
// https://cytomic.com/files/dsp/SvfLinearTrapezoidalSin.pdf
// Big thanks go to Andrew Simper @ Cytomic for developing and publishing
// the paper.
#[inline]
pub fn process_simper_svf(
input: f32, freq: f32, res: f32, israte: f32, ic1eq: &mut f32, ic2eq: &mut f32
) -> (f32, f32, f32) {
// XXX: the 1.989 were tuned by hand, so the resonance is more audible.
let k = 2f32 - (1.989f32 * res);
let w = std::f32::consts::PI * freq * israte;
let s1 = w.sin();
let s2 = (2.0 * w).sin();
let nrm = 1.0 / (2.0 + k * s2);
let g0 = s2 * nrm;
let g1 = (-2.0 * s1 * s1 - k * s2) * nrm;
let g2 = (2.0 * s1 * s1) * nrm;
let t0 = input - *ic2eq;
let t1 = g0 * t0 + g1 * *ic1eq;
let t2 = g2 * t0 + g0 * *ic1eq;
let v1 = t1 + *ic1eq;
let v2 = t2 + *ic2eq;
*ic1eq += 2.0 * t1;
*ic2eq += 2.0 * t2;
// 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 low pass filter with 24dB.
/// It provides only a low pass 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 `b3` - Internal values used for filtering.
/// * `delay` - A buffer holding other delayed samples.
///
///```
/// 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 delay = [0.0; 4];
/// let mut freq = 1000.0;
///
/// for s in samples.iter() {
/// let low =
/// process_stilson_moog(
/// *s, freq, 0.5, 1.0 / 44100.0,
/// &mut b0, &mut b1, &mut b2, &mut b3,
/// &mut delay);
///
/// // ... do something with the result here.
/// }
///```
// Stilson/Moog implementation partly translated from here:
// https://github.com/ddiakopoulos/MoogLadders/blob/master/src/MusicDSPModel.h
// without any copyright as found on musicdsp.org
// (http://www.musicdsp.org/showone.php?id=24).
//
// It's also found on MusicDSP and has probably no proper license anyways.
// See also: https://github.com/ddiakopoulos/MoogLadders
// and https://github.com/rncbc/synthv1/blob/master/src/synthv1_filter.h#L103
// 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,
delay: &mut [f32; 4],
) -> f32 {
let cutoff = 2.0 * freq * israte;
let p = cutoff * (1.8 - 0.8 * cutoff);
let k = 2.0 * (cutoff * std::f32::consts::PI * 0.5).sin() - 1.0;
let t1 = (1.0 - p) * 1.386249;
let t2 = 12.0 + t1 * t1;
let res = res * (t2 + 6.0 * t1) / (t2 - 6.0 * t1);
let x = input - res * *b3;
// Four cascaded one-pole filters (bilinear transform)
*b0 = x * p + delay[0] * p - k * *b0;
*b1 = *b0 * p + delay[1] * p - k * *b1;
*b2 = *b1 * p + delay[2] * p - k * *b2;
*b3 = *b2 * p + delay[3] * p - k * *b3;
// Clipping band-limited sigmoid
*b3 -= (*b3 * *b3 * *b3) * 0.166667;
delay[0] = x;
delay[1] = *b0;
delay[2] = *b1;
delay[3] = *b2;
*b3
}
// translated from Odin 2 Synthesizer Plugin
// Copyright (C) 2020 TheWaveWarden
// under GPLv3 or any later
#[derive(Debug, Clone, Copy)]
pub struct DCBlockFilter<F: Flt> {
xm1: F,
ym1: F,
r: F,
}
impl<F: Flt> DCBlockFilter<F> {
pub fn new() -> Self {
Self {
xm1: f(0.0),
ym1: f(0.0),
r: f(0.995),
}
}
pub fn reset(&mut self) {
self.xm1 = f(0.0);
self.ym1 = f(0.0);
}
pub fn set_sample_rate(&mut self, srate: F) {
self.r = f(0.995);
if srate > f(90000.0) {
self.r = f(0.9965);
} else if srate > f(120000.0) {
self.r = f(0.997);
}
}
pub fn next(&mut self, input: F) -> F {
let y = input - self.xm1 + self.r * self.ym1;
self.xm1 = input;
self.ym1 = y;
y as F
}
}
// PolyBLEP by Tale
// (slightly modified)
// http://www.kvraudio.com/forum/viewtopic.php?t=375517
// from http://www.martin-finke.de/blog/articles/audio-plugins-018-polyblep-oscillator/
//
// default for `pw' should be 1.0, it's the pulse width
// for the square wave.
#[allow(dead_code)]
fn poly_blep_64(t: f64, dt: f64) -> f64 {
if t < dt {
let t = t / dt;
2. * t - (t * t) - 1.
} else if t > (1.0 - dt) {
let t = (t - 1.0) / dt;
(t * t) + 2. * t + 1.
} else {
0.
}
}
fn poly_blep(t: f32, dt: f32) -> f32 {
if t < dt {
let t = t / dt;
2. * t - (t * t) - 1.
} else if t > (1.0 - dt) {
let t = (t - 1.0) / dt;
(t * t) + 2. * t + 1.
} else {
0.
}
}
/// This is a band-limited oscillator based on the PolyBlep technique.
/// Here is a quick example on how to use it:
///
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// // Randomize the initial phase to make cancellation on summing less
/// // likely:
/// let mut osc =
/// PolyBlepOscillator::new(rand_01() * 0.25);
///
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
/// let pw = 0.2; // Pulse-Width for the next_pulse()
/// let waveform = 0; // 0 being pulse in this example, 1 being sawtooth
///
/// let mut block_of_samples = [0.0; 128];
/// // in your process function:
/// for output_sample in block_of_samples.iter_mut() {
/// *output_sample =
/// if waveform == 1 {
/// osc.next_saw(freq, israte)
/// } else {
/// osc.next_pulse(freq, israte, pw)
/// }
/// }
///```
#[derive(Debug, Clone)]
pub struct PolyBlepOscillator {
phase: f32,
init_phase: f32,
last_output: f32,
}
impl PolyBlepOscillator {
/// Create a new instance of [PolyBlepOscillator].
///
/// * `init_phase` - Initial phase of the oscillator.
/// Range of this parameter is from 0.0 to 1.0. Passing a random
/// value is advised for preventing phase cancellation when summing multiple
/// oscillators.
///
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///```
pub fn new(init_phase: f32) -> Self {
Self {
phase: 0.0,
last_output: 0.0,
init_phase,
}
}
/// Reset the internal state of the oscillator as if you just called
/// [PolyBlepOscillator::new].
#[inline]
pub fn reset(&mut self) {
self.phase = self.init_phase;
self.last_output = 0.0;
}
/// Creates the next sample of a sine wave.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
///
/// // ...
/// let sample = osc.next_sin(freq, israte);
/// // ...
///```
#[inline]
pub fn next_sin(&mut self, freq: f32, israte: f32) -> f32 {
let phase_inc = freq * israte;
let s = fast_sin(self.phase * 2.0 * std::f32::consts::PI);
self.phase += phase_inc;
self.phase = self.phase.fract();
s as f32
}
/// Creates the next sample of a triangle wave. Please note that the
/// bandlimited waveform needs a few initial samples to swing in.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
///
/// // ...
/// let sample = osc.next_tri(freq, israte);
/// // ...
///```
#[inline]
pub fn next_tri(&mut self, freq: f32, israte: f32) -> f32 {
let phase_inc = freq * israte;
let mut s =
if self.phase < 0.5 { 1.0 }
else { -1.0 };
s += poly_blep(self.phase, phase_inc);
s -= poly_blep((self.phase + 0.5).fract(), phase_inc);
// leaky integrator: y[n] = A * x[n] + (1 - A) * y[n-1]
s = phase_inc * s + (1.0 - phase_inc) * self.last_output;
self.last_output = s;
self.phase += phase_inc;
self.phase = self.phase.fract();
// the signal is a bit too weak, we need to amplify it
// or else the volume diff between the different waveforms
// is too big:
s * 4.0
}
/// Creates the next sample of a sawtooth wave.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
///
/// // ...
/// let sample = osc.next_saw(freq, israte);
/// // ...
///```
#[inline]
pub fn next_saw(&mut self, freq: f32, israte: f32) -> f32 {
let phase_inc = freq * israte;
let mut s = (2.0 * self.phase) - 1.0;
s -= poly_blep(self.phase, phase_inc);
self.phase += phase_inc;
self.phase = self.phase.fract();
s
}
/// Creates the next sample of a pulse wave.
/// In comparison to [PolyBlepOscillator::next_pulse_no_dc] this
/// version is DC compensated, so that you may add multiple different
/// pulse oscillators for a unison effect without too big DC issues.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
/// * `pw` - The pulse width. Use the value 0.0 for a square wave.
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
/// let pw = 0.0; // 0.0 is a square wave.
///
/// // ...
/// let sample = osc.next_pulse(freq, israte, pw);
/// // ...
///```
#[inline]
pub fn next_pulse(&mut self, freq: f32, israte: f32, pw: f32) -> f32 {
let phase_inc = freq * israte;
let pw = (0.1 * pw) + ((1.0 - pw) * 0.5); // some scaling
let dc_compensation = (0.5 - pw) * 2.0;
let mut s =
if self.phase < pw { 1.0 }
else { -1.0 };
s += poly_blep(self.phase, phase_inc);
s -= poly_blep((self.phase + (1.0 - pw)).fract(),
phase_inc);
s += dc_compensation;
self.phase += phase_inc;
self.phase = self.phase.fract();
s
}
/// Creates the next sample of a pulse wave.
/// In comparison to [PolyBlepOscillator::next_pulse] this
/// version is not DC compensated. So be careful when adding multiple
/// of this or generally using it in an audio context.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
/// * `pw` - The pulse width. Use the value 0.0 for a square wave.
///```
/// use hexodsp::dsp::helpers::{PolyBlepOscillator, rand_01};
///
/// let mut osc = PolyBlepOscillator::new(rand_01() * 0.25);
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
/// let pw = 0.0; // 0.0 is a square wave.
///
/// // ...
/// let sample = osc.next_pulse_no_dc(freq, israte, pw);
/// // ...
///```
#[inline]
pub fn next_pulse_no_dc(&mut self, freq: f32, israte: f32, pw: f32) -> f32 {
let phase_inc = freq * israte;
let pw = (0.1 * pw) + ((1.0 - pw) * 0.5); // some scaling
let mut s =
if self.phase < pw { 1.0 }
else { -1.0 };
s += poly_blep(self.phase, phase_inc);
s -= poly_blep((self.phase + (1.0 - pw)).fract(),
phase_inc);
self.phase += phase_inc;
self.phase = self.phase.fract();
s
}
}
// This oscillator is based on the work "VECTOR PHASESHAPING SYNTHESIS"
// by: Jari Kleimola*, Victor Lazzarini†, Joseph Timoney†, Vesa Välimäki*
// *Aalto University School of Electrical Engineering Espoo, Finland;
// †National University of Ireland, Maynooth Ireland
//
// See also this PDF: http://recherche.ircam.fr/pub/dafx11/Papers/55_e.pdf
/// Vector Phase Shaping Oscillator.
/// The parameters `d` and `v` control the shape of the sinus
/// wave. This leads to interesting modulation properties of those
/// control values.
///
///```
/// use hexodsp::dsp::helpers::{VPSOscillator, rand_01};
///
/// // Randomize the initial phase to make cancellation on summing less
/// // likely:
/// let mut osc =
/// VPSOscillator::new(rand_01() * 0.25);
///
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
/// let d = 0.5; // Range: 0.0 to 1.0
/// let v = 0.75; // Range: 0.0 to 1.0
///
/// let mut block_of_samples = [0.0; 128];
/// // in your process function:
/// for output_sample in block_of_samples.iter_mut() {
/// // It is advised to limit the `v` value, because with certain
/// // `d` values the combination creates just a DC offset.
/// let v = VPSOscillator::limit_v(d, v);
/// *output_sample = osc.next(freq, israte, d, v);
/// }
///```
///
/// It can be beneficial to apply distortion and oversampling.
/// Especially oversampling can be important for some `d` and `v`
/// combinations, even without distortion.
///
///```
/// use hexodsp::dsp::helpers::{VPSOscillator, rand_01, apply_distortion};
/// use hexodsp::dsp::biquad::Oversampling;
///
/// let mut osc = VPSOscillator::new(rand_01() * 0.25);
/// let mut ovr : Oversampling<4> = Oversampling::new();
///
/// let freq = 440.0; // Hz
/// let israte = 1.0 / 44100.0; // Seconds per Sample
/// let d = 0.5; // Range: 0.0 to 1.0
/// let v = 0.75; // Range: 0.0 to 1.0
///
/// let mut block_of_samples = [0.0; 128];
/// // in your process function:
/// for output_sample in block_of_samples.iter_mut() {
/// // It is advised to limit the `v` value, because with certain
/// // `d` values the combination creates just a DC offset.
/// let v = VPSOscillator::limit_v(d, v);
///
/// let overbuf = ovr.resample_buffer();
/// for b in overbuf {
/// *b = apply_distortion(osc.next(freq, israte, d, v), 0.9, 1);
/// }
/// *output_sample = ovr.downsample();
/// }
///```
#[derive(Debug, Clone)]
pub struct VPSOscillator {
phase: f32,
init_phase: f32,
}
impl VPSOscillator {
/// Create a new instance of [VPSOscillator].
///
/// * `init_phase` - The initial phase of the oscillator.
pub fn new(init_phase: f32) -> Self {
Self {
phase: 0.0,
init_phase,
}
}
/// Reset the phase of the oscillator to the initial phase.
#[inline]
pub fn reset(&mut self) {
self.phase = self.init_phase;
}
#[inline]
fn s(p: f32) -> f32 {
-(std::f32::consts::TAU * p).cos()
}
#[inline]
fn phi_vps(x: f32, v: f32, d: f32) -> f32 {
if x < d {
(v * x) / d
} else {
v + ((1.0 - v) * (x - d))/(1.0 - d)
}
}
/// This rather complicated function blends out some
/// combinations of 'd' and 'v' that just lead to a constant DC
/// offset. Which is not very useful in an audio oscillator
/// context.
///
/// Call this before passing `v` to [VPSOscillator::next].
#[inline]
pub fn limit_v(d: f32, v: f32) -> f32 {
let delta = 0.5 - (d - 0.5).abs();
if delta < 0.05 {
let x = (0.05 - delta) * 19.99;
if d < 0.5 {
let mm = x * 0.5;
let max = 1.0 - mm;
if v > max && v < 1.0 {
max
} else if v >= 1.0 && v < (1.0 + mm) {
1.0 + mm
} else {
v
}
} else {
if v < 1.0 {
v.clamp(x * 0.5, 1.0)
} else {
v
}
}
} else {
v
}
}
/// Creates the next sample of this oscillator.
///
/// * `freq` - The frequency in Hz.
/// * `israte` - The inverse sampling rate, or seconds per sample as in eg. `1.0 / 44100.0`.
/// * `d` - The phase distortion parameter `d` which must be in the range `0.0` to `1.0`.
/// * `v` - The phase distortion parameter `v` which must be in the range `0.0` to `1.0`.
///
/// It is advised to limit the `v` using the [VPSOscillator::limit_v] function
/// before calling this function. To prevent DC offsets when modulating the parameters.
pub fn next(&mut self, freq: f32, israte: f32, d: f32, v: f32) -> f32 {
let s = Self::s(Self::phi_vps(self.phase, v, d));
self.phase += freq * israte;
self.phase = self.phase.fract();
s
}
}
// Adapted from https://github.com/ValleyAudio/ValleyRackFree/blob/v1.0/src/Common/DSP/LFO.hpp
//
// ValleyRackFree Copyright (C) 2020, Valley Audio Soft, Dale Johnson
// Adapted under the GPL-3.0-or-later License.
/// An LFO with a variable reverse point, which can go from reverse Saw, to Tri
/// and to Saw, depending on the reverse point.
#[derive(Debug, Clone, Copy)]
pub struct TriSawLFO<F: Flt> {
/// The (inverse) sample rate. Eg. 1.0 / 44100.0.
israte: F,
/// The current oscillator phase.
phase: F,
/// The point from where the falling edge will be used.
rev: F,
/// Whether the LFO is currently rising
rising: bool,
/// The frequency.
freq: F,
/// Precomputed rise/fall rate of the LFO.
rise_r: F,
fall_r: F,
/// Initial phase offset.
init_phase: F,
}
impl<F: Flt> TriSawLFO<F> {
pub fn new() -> Self {
let mut this = Self {
israte: f(1.0 / 44100.0),
phase: f(0.0),
rev: f(0.5),
rising: true,
freq: f(1.0),
fall_r: f(0.0),
rise_r: f(0.0),
init_phase: f(0.0),
};
this.recalc();
this
}
pub fn set_phase_offs(&mut self, phase: F) {
self.init_phase = phase;
self.phase = phase;
}
#[inline]
fn recalc(&mut self) {
self.rev = fclampc(self.rev, 0.0001, 0.999);
self.rise_r = f::<F>( 1.0) / self.rev;
self.fall_r = f::<F>(-1.0) / (f::<F>(1.0) - self.rev);
}
pub fn set_sample_rate(&mut self, srate: F) {
self.israte = f::<F>(1.0) / (srate as F);
self.recalc();
}
pub fn reset(&mut self) {
self.phase = self.init_phase;
self.rev = f(0.5);
self.rising = true;
}
#[inline]
pub fn set(&mut self, freq: F, rev: F) {
self.freq = freq as F;
self.rev = rev as F;
self.recalc();
}
#[inline]
pub fn next_unipolar(&mut self) -> F {
if self.phase >= f(1.0) {
self.phase = self.phase - f(1.0);
self.rising = true;
}
if self.phase >= self.rev {
self.rising = false;
}
let s =
if self.rising {
self.phase * self.rise_r
} else {
self.phase * self.fall_r - self.fall_r
};
self.phase = self.phase + self.freq * self.israte;
s
}
#[inline]
pub fn next_bipolar(&mut self) -> F {
(self.next_unipolar() * f(2.0)) - f(1.0)
}
}
#[macro_export]
macro_rules! fa_distort { ($formatter: expr, $v: expr, $denorm_v: expr) => { {
let s =
match ($v.round() as usize) {
0 => "Off",
1 => "TanH",
2 => "B.D.Jong",
3 => "Fold",
_ => "?",
};
write!($formatter, "{}", s)
} } }
#[inline]
pub fn apply_distortion(s: f32, damt: f32, dist_type: u8) -> f32 {
match dist_type {
1 => (damt.clamp(0.01, 1.0) * 100.0 * s).tanh(),
2 => f_distort(1.0, damt * damt * damt * 1000.0, s),
3 => {
let damt = damt.clamp(0.0, 0.99);
let damt = 1.0 - damt * damt;
f_fold_distort(1.0, damt, s) * (1.0 / damt)
},
_ => s,
}
}
//pub struct UnisonBlep {
// oscs: Vec<PolyBlepOscillator>,
//// dc_block: crate::filter::DCBlockFilter,
//}
//
//impl UnisonBlep {
// pub fn new(max_unison: usize) -> Self {
// let mut oscs = vec![];
// let mut rng = RandGen::new();
//
// let dis_init_phase = 0.05;
// for i in 0..(max_unison + 1) {
// // randomize phases so we fatten the unison, get
// // less DC and not an amplified signal until the
// // detune desyncs the waves.
// // But no random phase for first, so we reduce the click
// let init_phase =
// if i == 0 { 0.0 } else { rng.next_open01() };
// oscs.push(PolyBlepOscillator::new(init_phase));
// }
//
// Self {
// oscs,
//// dc_block: crate::filter::DCBlockFilter::new(),
// }
// }
//
// pub fn set_sample_rate(&mut self, srate: f32) {
//// self.dc_block.set_sample_rate(srate);
// for o in self.oscs.iter_mut() {
// o.set_sample_rate(srate);
// }
// }
//
// pub fn reset(&mut self) {
//// self.dc_block.reset();
// for o in self.oscs.iter_mut() {
// o.reset();
// }
// }
//
// pub fn next<P: OscillatorInputParams>(&mut self, params: &P) -> f32 {
// let unison =
// (params.unison().floor() as usize)
// .min(self.oscs.len() - 1);
// let detune = params.detune() as f64;
//
// let mix = (1.0 / ((unison + 1) as f32)).sqrt();
//
// let mut s = mix * self.oscs[0].next(params, 0.0);
//
// for u in 0..unison {
// let detune_factor =
// detune * (((u / 2) + 1) as f64
// * if (u % 2) == 0 { 1.0 } else { -1.0 });
// s += mix * self.oscs[u + 1].next(params, detune_factor * 0.01);
// }
//
//// self.dc_block.next(s)
// s
// }
//}
#[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);
}
}
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