HexoDSP/src/dsp/node_sampl.rs

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// 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.
use crate::nodes::{NodeAudioContext, NodeExecContext};
use crate::dsp::{NodeId, SAtom, ProcBuf, DspNode, LedPhaseVals};
use crate::dsp::{out, at, inp, denorm, denorm_offs}; //, inp, denorm, denorm_v, inp_dir, at};
use super::helpers::Trigger;
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#[macro_export]
macro_rules! fa_sampl_dclick { ($formatter: expr, $v: expr, $denorm_v: expr) => { {
let s =
match ($v.round() as usize) {
0 => "Off",
1 => "On",
_ => "?",
};
write!($formatter, "{}", s)
} } }
#[macro_export]
macro_rules! fa_sampl_pmode { ($formatter: expr, $v: expr, $denorm_v: expr) => { {
let s =
match ($v.round() as usize) {
0 => "Loop",
1 => "OneShot",
_ => "?",
};
write!($formatter, "{}", s)
} } }
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/// A simple amplifier
#[derive(Debug, Clone)]
pub struct Sampl {
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phase: f64,
srate: f64,
trig: Trigger,
is_playing: bool,
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last_sample: f32,
decaying: f32,
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}
impl Sampl {
pub fn new(_nid: &NodeId) -> Self {
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Self {
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phase: 0.0,
srate: 44100.0,
trig: Trigger::new(),
is_playing: false,
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last_sample: 0.0,
decaying: 0.0,
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}
}
pub const freq : &'static str =
"Sampl freq\nPitch input for the sampler, giving the playback speed of the \
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sample.\nRange: (-1..1)\n";
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pub const trig : &'static str =
"Sampl trig\nThe trigger input causes a resync of the playback phase \
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and triggers the playback if the 'pmode' is 'OneShot'";
pub const offs : &'static str =
"Sampl offs\nStart position offset.\nRange: (0..1)\n";
pub const len : &'static str =
"Sampl len\nLength of the sample, after the offset has been applied.\nRange: (0..1)\n";
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pub const dcms : &'static str =
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"Sampl dcms\nDeclick fade time in milliseconds.\nNot audio rate!\nRange: (0..1)\n";
pub const det : &'static str =
"Sin det\nDetune the oscillator in semitones and cents. \
the input of this value is rounded to semitones on coarse input. \
Fine input lets you detune in cents (rounded). \
A signal sent to this port is not rounded.\n\
Note: The signal input allows detuning over +- 10 octaves.\
\n\nKnob Range: (-0.2 .. 0.2)\n\
Signal Range: (-1.0 .. 1.0)\n";
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pub const sample : &'static str =
"Sampl sample\nThe audio sample that is played back.\nRange: (-1..1)\n";
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pub const pmode : &'static str =
"Sampl pmode\nThe playback mode of the sampler.\n\
- 'Loop' constantly plays back the sample. You can reset/sync the phase \
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using the 'trig' input in this case.\n\
- 'OneShot' plays back the sample if a trigger is received on 'trig' input.\n";
pub const dclick : &'static str =
"Sampl dclick\nIf this is enabled and the 'pmode' is 'OneShot' \
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this will enable short fade in and out ramps.\n\
This if useful if you don't want to add an envelope just for \
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getting rid of the clicks if spos and epos are modulated.";
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pub const sig : &'static str =
"Sampl sig\nSampler audio output\nRange: (-1..1)\n";
}
impl Sampl {
#[inline]
fn next_sample(&mut self, sr_factor: f64, speed: f64, sample_data: &[f32]) -> f32 {
let sd_len = sample_data.len();
if sd_len < 1 { return 0.0; }
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let i = self.phase.floor() as usize + sd_len;
// 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 = sample_data[(i - 1) % sd_len];
let x0 = sample_data[i % sd_len];
let x1 = sample_data[(i + 1) % sd_len];
let x2 = sample_data[(i + 2) % sd_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 f = self.phase.fract();
self.phase = (i % sd_len) as f64 + f + sr_factor * speed;
let f = f as f32;
(((a * f) - b_neg) * f + c) * f + x0
}
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#[inline]
fn play(&mut self, inputs: &[ProcBuf], nframes: usize,
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sample_data: &[f32], out: &mut ProcBuf, do_loop: bool,
declick: bool)
{
let freq = inp::Sampl::freq(inputs);
let trig = inp::Sampl::trig(inputs);
let offs = inp::Sampl::offs(inputs);
let len = inp::Sampl::len(inputs);
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let dcms = inp::Sampl::dcms(inputs);
let det = inp::Sampl::det(inputs);
let sample_srate = sample_data[0] as f64;
let sample_data = &sample_data[1..];
let sr_factor = sample_srate / self.srate;
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let ramp_time = denorm::Sampl::dcms(dcms, 0) as f64 * self.srate;
let ramp_sample_count = (ramp_time / 1000.0).ceil() as usize;
let ramp_inc = 1000.0 / ramp_time;
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let mut is_playing = self.is_playing;
if do_loop {
is_playing = true;
}
let mut prev_offs = -10.0;
let mut prev_len = -10.0;
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let mut start_idx = 0;
let mut end_idx_plus1 = sample_data.len();
for frame in 0..nframes {
let trig_val = denorm::Sampl::trig(trig, frame);
let triggered = self.trig.check_trigger(trig_val);
if triggered {
self.phase = 0.0;
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self.decaying = self.last_sample;
is_playing = true;
}
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let s =
if is_playing {
let freq =
denorm_offs::Sampl::freq(
freq, det.read(frame), frame);
let playback_speed = freq / 440.0;
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let prev_phase = self.phase;
let sd_len = sample_data.len();
let cur_offs =
denorm::Sampl::offs(offs, frame).abs().min(0.999999)
as f64;
let recalc_end =
if prev_offs != cur_offs {
start_idx =
((sd_len as f64 * cur_offs)
.floor() as usize).min(sd_len);
prev_offs = cur_offs;
true
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} else {
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false
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};
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let cur_len =
denorm::Sampl::len(len, frame).abs().min(0.999999)
as f64;
if recalc_end || prev_len != cur_len {
let remain_s_len = sd_len - start_idx;
end_idx_plus1 =
((remain_s_len as f64 * cur_len)
.ceil() as usize).min(remain_s_len);
prev_len = cur_len;
}
let sample_slice =
&sample_data[start_idx..(start_idx + end_idx_plus1)];
// next_sample mutates self.phase, so we need the current phase
// that is used for looking up the sample from the audio data.
let sample_idx = self.phase.floor() as usize;
let mut s =
self.next_sample(
sr_factor,
playback_speed as f64,
sample_slice);
if declick {
let samples_to_end = sample_slice.len() - sample_idx;
let ramp_atten_factor =
if sample_idx < ramp_sample_count {
sample_idx as f64 * ramp_inc
} else if samples_to_end < ramp_sample_count {
samples_to_end as f64 * ramp_inc
} else {
1.0
};
s *= ramp_atten_factor as f32;
}
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self.last_sample = s;
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out.write(frame, s);
if !do_loop && prev_phase > self.phase {
// played past end => stop playing.
is_playing = false;
}
s
} else {
0.0
};
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let s =
if !declick || self.decaying.abs() < 0.00001 {
self.decaying = 0.0;
s
} else {
self.decaying *= 0.98;
(s + self.decaying).clamp(-1.0, 1.0)
};
self.last_sample = s;
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out.write(frame, s);
}
self.is_playing = is_playing;
}
}
impl DspNode for Sampl {
fn outputs() -> usize { 1 }
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fn set_sample_rate(&mut self, srate: f32) { self.srate = srate.into(); }
fn reset(&mut self) {
self.trig.reset();
}
#[inline]
fn process<T: NodeAudioContext>(
&mut self, ctx: &mut T, _ectx: &mut NodeExecContext,
atoms: &[SAtom], _params: &[ProcBuf], inputs: &[ProcBuf],
outputs: &mut [ProcBuf], ctx_vals: LedPhaseVals)
{
let sample = at::Sampl::sample(atoms);
let pmode = at::Sampl::pmode(atoms);
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let dclick = at::Sampl::dclick(atoms);
let out = out::Sampl::sig(outputs);
if let SAtom::AudioSample((_, Some(sample_data))) = sample {
// 3 is for sample-sample-rate and at least 2 audio samples.
if sample_data.len() < 3 {
for frame in 0..ctx.nframes() {
out.write(frame, 0.0);
}
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self.last_sample = 0.0;
return;
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}
self.play(
inputs,
ctx.nframes(),
&sample_data[..],
out,
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pmode.i() == 0,
dclick.i() == 1);
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} else {
for frame in 0..ctx.nframes() {
out.write(frame, 0.0);
}
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self.last_sample = 0.0;
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}
let last_frame = ctx.nframes() - 1;
ctx_vals[0].set(out.read(last_frame));
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}
}