// Diopser: a phase rotation plugin // Copyright (C) 2021-2022 Robbert van der Helm // // This program is free software: you can redistribute it and/or modify // it under the terms of the GNU General Public License as published by // the Free Software Foundation, either version 3 of the License, or // (at your option) any later version. // // This program is distributed in the hope that it will be useful, // but WITHOUT ANY WARRANTY; without even the implied warranty of // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the // GNU General Public License for more details. // // You should have received a copy of the GNU General Public License // along with this program. If not, see . #![cfg_attr(feature = "simd", feature(portable_simd))] #[cfg(not(feature = "simd"))] compile_error!("Compiling without SIMD support is currently not supported"); use nih_plug::prelude::*; use nih_plug_vizia::ViziaState; use std::simd::f32x2; use std::sync::atomic::{AtomicBool, Ordering}; use std::sync::Arc; use crate::spectrum::{SpectrumInput, SpectrumOutput}; mod editor; mod filter; mod spectrum; /// How many all-pass filters we can have in series at most. The filter stages parameter determines /// how many filters are actually active. const MAX_NUM_FILTERS: usize = 512; /// The minimum step size for smoothing the filter parameters. const MIN_AUTOMATION_STEP_SIZE: u32 = 1; /// The maximum step size for smoothing the filter parameters. Updating these parameters can be /// expensive, so updating them in larger steps can be useful. const MAX_AUTOMATION_STEP_SIZE: u32 = 512; // All features from the original Diopser have been implemented (and the spread control has been // improved). Other features I want to implement are: // - Briefly muting the output when changing the number of filters to get rid of the clicks // - A proper GUI struct Diopser { params: Arc, /// Needed for computing the filter coefficients. sample_rate: f32, /// All of the all-pass filters, with vectorized coefficients so they can be calculated for /// multiple channels at once. [`DiopserParams::num_stages`] controls how many filters are /// actually active. filters: [filter::Biquad; MAX_NUM_FILTERS], /// If this is set at the start of the processing cycle, then the filter coefficients should be /// updated. For the regular filter parameters we can look at the smoothers, but this is needed /// when changing the number of active filters. should_update_filters: Arc, /// If this is 1 and any of the filter parameters are still smoothing, thenn the filter /// coefficients should be recalculated on the next sample. After that, this gets reset to /// `unnormalize_automation_precision(self.params.automation_precision.value)`. This is to /// reduce the DSP load of automation parameters. It can also cause some fun sounding glitchy /// effects when the precision is low. next_filter_smoothing_in: i32, /// When the GUI is open we compute the spectrum on the audio thread and send it to the GUI. spectrum_input: SpectrumInput, /// This can be cloned and moved into the editor. spectrum_output: Arc, } // TODO: Some combinations of parameters can cause really loud resonance. We should limit the // resonance and filter stages parameter ranges in the GUI until the user unlocks. #[derive(Params)] struct DiopserParams { /// The editor state, saved together with the parameter state so the custom scaling can be /// restored. #[persist = "editor-state"] editor_state: Arc, /// The number of all-pass filters applied in series. #[id = "stages"] filter_stages: IntParam, /// The filter's center frequqency. When this is applied, the filters are spread around this /// frequency. #[id = "cutoff"] filter_frequency: FloatParam, /// The Q parameter for the filters. #[id = "res"] filter_resonance: FloatParam, /// Controls a frequency spread between the filter stages in octaves. When this value is 0, the /// same coefficients are used for every filter. Otherwise, the earliest stage's frequency will /// be offset by `-filter_spread_octave_amount`, while the latest stage will be offset by /// `filter_spread_octave_amount`. If the filter spread style is set to linear then the negative /// range will cover the same frequency range as the positive range. #[id = "spread"] filter_spread_octaves: FloatParam, /// How the spread range should be distributed. The octaves mode will sound more musical while /// the linear mode can be useful for sound design purposes. #[id = "spstyl"] filter_spread_style: EnumParam, /// The precision of the automation, determines the step size. This is presented to the userq as /// a percentage, and it's stored here as `[0, 1]` float because smaller step sizes are more /// precise so having this be an integer would result in odd situations. #[id = "autopr"] automation_precision: FloatParam, /// Very important. #[id = "ignore"] very_important: BoolParam, } impl Default for Diopser { fn default() -> Self { let should_update_filters = Arc::new(AtomicBool::new(false)); // We only do stereo right now so this is simple let (spectrum_input, spectrum_output) = SpectrumInput::new(Self::DEFAULT_OUTPUT_CHANNELS as usize); Self { params: Arc::new(DiopserParams::new(should_update_filters.clone())), sample_rate: 1.0, filters: [filter::Biquad::default(); MAX_NUM_FILTERS], should_update_filters, next_filter_smoothing_in: 1, spectrum_input, spectrum_output: Arc::new(spectrum_output), } } } impl DiopserParams { fn new(should_update_filters: Arc) -> Self { Self { editor_state: editor::default_state(), filter_stages: IntParam::new( "Filter Stages", 0, IntRange::Linear { min: 0, max: MAX_NUM_FILTERS as i32, }, ) .with_callback({ let should_update_filters = should_update_filters.clone(); Arc::new(move |_| should_update_filters.store(true, Ordering::Release)) }), // Smoothed parameters don't need the callback as we can just look at whether the // smoother is still smoothing filter_frequency: FloatParam::new( "Filter Frequency", 200.0, FloatRange::Skewed { min: 5.0, // This must never reach 0 max: 20_000.0, factor: FloatRange::skew_factor(-2.5), }, ) // This needs quite a bit of smoothing to avoid artifacts .with_smoother(SmoothingStyle::Logarithmic(100.0)) // This includes the unit .with_value_to_string(formatters::v2s_f32_hz_then_khz(0)) .with_string_to_value(formatters::s2v_f32_hz_then_khz()), filter_resonance: FloatParam::new( "Filter Resonance", // The actual default neutral Q-value would be `sqrt(2) / 2`, but this value // produces slightly less ringing. 0.5, FloatRange::Skewed { min: 0.01, // This must also never reach 0 max: 30.0, factor: FloatRange::skew_factor(-2.5), }, ) .with_smoother(SmoothingStyle::Logarithmic(100.0)) .with_value_to_string(formatters::v2s_f32_rounded(2)), filter_spread_octaves: FloatParam::new( "Filter Spread Octaves", 0.0, FloatRange::SymmetricalSkewed { min: -5.0, max: 5.0, factor: FloatRange::skew_factor(-1.0), center: 0.0, }, ) .with_step_size(0.01) .with_smoother(SmoothingStyle::Linear(100.0)), filter_spread_style: EnumParam::new("Filter Spread Style", SpreadStyle::Octaves) .with_callback(Arc::new(move |_| { should_update_filters.store(true, Ordering::Release) })), very_important: BoolParam::new("Don't touch this", true) .with_value_to_string(Arc::new(|value| { String::from(if value { "please don't" } else { "stop it" }) })) .with_string_to_value(Arc::new(|string| { let string = string.trim(); if string.eq_ignore_ascii_case("please don't") { Some(true) } else if string.eq_ignore_ascii_case("stop it") { Some(false) } else { None } })) .hide_in_generic_ui(), automation_precision: FloatParam::new( "Automation precision", normalize_automation_precision(128), FloatRange::Linear { min: 0.0, max: 1.0 }, ) .with_unit("%") .with_value_to_string(formatters::v2s_f32_percentage(0)) .with_string_to_value(formatters::s2v_f32_percentage()), } } } #[derive(Enum, Debug, PartialEq)] enum SpreadStyle { #[id = "octaves"] Octaves, #[id = "linear"] Linear, } impl Plugin for Diopser { const NAME: &'static str = "Diopser"; const VENDOR: &'static str = "Robbert van der Helm"; const URL: &'static str = "https://github.com/robbert-vdh/nih-plug"; const EMAIL: &'static str = "mail@robbertvanderhelm.nl"; const VERSION: &'static str = "0.2.0"; const DEFAULT_INPUT_CHANNELS: u32 = 2; const DEFAULT_OUTPUT_CHANNELS: u32 = 2; const SAMPLE_ACCURATE_AUTOMATION: bool = true; fn params(&self) -> Arc { self.params.clone() } fn editor(&self) -> Option> { editor::create(self.params.clone(), self.params.editor_state.clone()) } fn accepts_bus_config(&self, config: &BusConfig) -> bool { // The SIMD version only supports stereo config.num_input_channels == config.num_output_channels && config.num_input_channels == 2 } fn initialize( &mut self, _bus_config: &BusConfig, buffer_config: &BufferConfig, _context: &mut impl InitContext, ) -> bool { self.sample_rate = buffer_config.sample_rate; true } fn reset(&mut self) { // Initialize and/or reset the filters on the next process call self.should_update_filters.store(true, Ordering::Release); } fn process( &mut self, buffer: &mut Buffer, _aux: &mut AuxiliaryBuffers, _context: &mut impl ProcessContext, ) -> ProcessStatus { // Since this is an expensive operation, only update the filters when it's actually // necessary, and allow smoothing only every n samples using the automation precision // parameter let smoothing_interval = unnormalize_automation_precision(self.params.automation_precision.value); for mut channel_samples in buffer.iter_samples() { self.maybe_update_filters(smoothing_interval); // We can compute the filters for both channels at once. The SIMD version thus now only // supports steroo audio. let mut samples = unsafe { channel_samples.to_simd_unchecked() }; for filter in self .filters .iter_mut() .take(self.params.filter_stages.value as usize) { samples = filter.process(samples); } unsafe { channel_samples.from_simd_unchecked(samples) }; } // Compute a spectrum for the GUI if needed if self.params.editor_state.is_open() { self.spectrum_input.compute(buffer); } ProcessStatus::Normal } } impl Diopser { /// Check if the filters need to be updated beased on /// [`should_update_filters`][Self::should_update_filters] and the smoothing interval, and /// update them as needed. fn maybe_update_filters(&mut self, smoothing_interval: u32) { // In addition to updating the filters, we should also clear the filter's state when // changing a setting we can't neatly interpolate between. let reset_filters = self .should_update_filters .compare_exchange(true, false, Ordering::Acquire, Ordering::Relaxed) .is_ok(); let should_update_filters = reset_filters || ((self.params.filter_frequency.smoothed.is_smoothing() || self.params.filter_resonance.smoothed.is_smoothing() || self.params.filter_spread_octaves.smoothed.is_smoothing()) && self.next_filter_smoothing_in <= 1); if should_update_filters { self.update_filters(smoothing_interval, reset_filters); self.next_filter_smoothing_in = smoothing_interval as i32; } else { self.next_filter_smoothing_in -= 1; } } /// Recompute the filter coefficients based on the smoothed paraetersm. We can skip forwardq in /// larger steps to reduce the DSP load. fn update_filters(&mut self, smoothing_interval: u32, reset_filters: bool) { if self.filters.is_empty() { return; } let frequency = self .params .filter_frequency .smoothed .next_step(smoothing_interval); let resonance = self .params .filter_resonance .smoothed .next_step(smoothing_interval); let spread_octaves = self .params .filter_spread_octaves .smoothed .next_step(smoothing_interval); let spread_style = self.params.filter_spread_style.value(); // Used to calculate the linear spread. This is calculated in such a way that the range // never dips below 0. let max_octave_spread = if spread_octaves >= 0.0 { frequency - (frequency * 2.0f32.powf(-spread_octaves)) } else { (frequency * 2.0f32.powf(spread_octaves)) - frequency }; // TODO: This wrecks the DSP load at high smoothing accuracy, perhaps also use SIMD here const MIN_FREQUENCY: f32 = 5.0; let max_frequency = self.sample_rate / 2.05; for filter_idx in 0..self.params.filter_stages.value as usize { // The index of the filter normalized to range [-1, 1] let filter_proportion = (filter_idx as f32 / self.params.filter_stages.value as f32) * 2.0 - 1.0; // The spread parameter adds an offset to the frequency depending on the number of the // filter let filter_frequency = match spread_style { SpreadStyle::Octaves => frequency * 2.0f32.powf(spread_octaves * filter_proportion), SpreadStyle::Linear => frequency + (max_octave_spread * filter_proportion), } .clamp(MIN_FREQUENCY, max_frequency); self.filters[filter_idx].coefficients = filter::BiquadCoefficients::allpass(self.sample_rate, filter_frequency, resonance); if reset_filters { self.filters[filter_idx].reset(); } } } } fn normalize_automation_precision(step_size: u32) -> f32 { (MAX_AUTOMATION_STEP_SIZE - step_size) as f32 / (MAX_AUTOMATION_STEP_SIZE - MIN_AUTOMATION_STEP_SIZE) as f32 } fn unnormalize_automation_precision(normalized: f32) -> u32 { MAX_AUTOMATION_STEP_SIZE - (normalized * (MAX_AUTOMATION_STEP_SIZE - MIN_AUTOMATION_STEP_SIZE) as f32).round() as u32 } impl ClapPlugin for Diopser { const CLAP_ID: &'static str = "nl.robbertvanderhelm.diopser"; const CLAP_DESCRIPTION: Option<&'static str> = Some("A totally original phase rotation plugin"); const CLAP_MANUAL_URL: Option<&'static str> = Some(Self::URL); const CLAP_SUPPORT_URL: Option<&'static str> = None; const CLAP_FEATURES: &'static [ClapFeature] = &[ ClapFeature::AudioEffect, ClapFeature::Stereo, ClapFeature::Filter, ClapFeature::Utility, ]; } impl Vst3Plugin for Diopser { const VST3_CLASS_ID: [u8; 16] = *b"DiopserPlugRvdH."; const VST3_CATEGORIES: &'static str = "Fx|Filter"; } nih_export_clap!(Diopser); nih_export_vst3!(Diopser);