// Copyright (c) 2021 Weird Constructor // 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; #[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) } } } /// A simple amplifier #[derive(Debug, Clone)] pub struct Sampl { phase: f64, srate: f64, trig: Trigger, is_playing: bool, last_sample: f32, decaying: f32, } impl Sampl { pub fn new(_nid: &NodeId) -> Self { Self { phase: 0.0, srate: 44100.0, trig: Trigger::new(), is_playing: false, last_sample: 0.0, decaying: 0.0, } } pub const freq : &'static str = "Sampl freq\nPitch input for the sampler, giving the playback speed of the \ sample.\nRange: (-1..1)\n"; pub const trig : &'static str = "Sampl trig\nThe trigger input causes a resync of the playback phase \ 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"; pub const dcms : &'static str = "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"; pub const sample : &'static str = "Sampl sample\nThe audio sample that is played back.\nRange: (-1..1)\n"; 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 \ 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' \ this will enable short fade in and out ramps.\n\ This if useful if you don't want to add an envelope just for \ getting rid of the clicks if spos and epos are modulated."; pub const sig : &'static str = "Sampl sig\nSampler audio output\nRange: (-1..1)\n"; pub const DESC : &'static str = r#""#; pub const HELP : &'static str = r#""#; } impl Sampl { #[allow(clippy::many_single_char_names)] #[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; } 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 } #[allow(clippy::float_cmp)] #[inline] fn play(&mut self, inputs: &[ProcBuf], nframes: usize, 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); 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; 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; 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; 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; self.decaying = self.last_sample; is_playing = true; } let s = if is_playing { let freq = denorm_offs::Sampl::freq( freq, det.read(frame), frame); let playback_speed = freq / 440.0; 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 } else { false }; 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; } self.last_sample = s; out.write(frame, s); if !do_loop && prev_phase > self.phase { // played past end => stop playing. is_playing = false; } s } else { 0.0 }; 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; out.write(frame, s); } self.is_playing = is_playing; } } impl DspNode for Sampl { fn outputs() -> usize { 1 } fn set_sample_rate(&mut self, srate: f32) { self.srate = srate.into(); } fn reset(&mut self) { self.trig.reset(); } #[inline] fn process( &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); 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); } self.last_sample = 0.0; return; } self.play( inputs, ctx.nframes(), &sample_data[..], out, pmode.i() == 0, dclick.i() == 1); } else { for frame in 0..ctx.nframes() { out.write(frame, 0.0); } self.last_sample = 0.0; } let last_frame = ctx.nframes() - 1; ctx_vals[0].set(out.read(last_frame)); } }