// 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 atomic_float::AtomicF32; use nih_plug::prelude::*; use nih_plug_vizia::ViziaState; use std::simd::f32x2; use std::sync::atomic::{AtomicBool, Ordering}; use std::sync::{Arc, Mutex}; 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; /// The maximum number of samples to iterate over at a time. const MAX_BLOCK_SIZE: usize = 64; /// The filter frequency parameter's range. Also used in the `SpectrumAnalyzer` widget. pub(crate) fn filter_frequency_range() -> FloatRange { FloatRange::Skewed { min: 5.0, // This must never reach 0 max: 20_000.0, factor: FloatRange::skew_factor(-2.5), } } // 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 pub struct Diopser { params: Arc, /// Needed for computing the filter coefficients. Also used to update `bypass_smoother`, hence /// why this needs to be an `Arc`. sample_rate: Arc, /// 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], /// When the bypass parameter is toggled, this smoother fades between 0.0 and 1.0. This lets us /// crossfade the dry and the wet signal to avoid clicks. The smoothing target is set in a /// callback handler on the bypass parameter. bypass_smoother: Arc>, /// 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>, } #[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, /// If this option is enabled, then the filter stages parameter is limited to `[0, 40]`. This is /// editor-only state, and doesn't affect host automation. #[persist = "safe-mode"] safe_mode: Arc, /// This plugin really doesn't need its own bypass parameter, but it's still useful to have a /// dedicated one so it can be shown in the GUI. This is linked to the host's bypass if the host /// supports it. #[id = "bypass"] bypass: BoolParam, /// 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 sample_rate = Arc::new(AtomicF32::new(1.0)); let should_update_filters = Arc::new(AtomicBool::new(false)); let bypass_smoother = Arc::new(Smoother::new(SmoothingStyle::Linear(10.0))); // 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( sample_rate.clone(), should_update_filters.clone(), bypass_smoother.clone(), )), sample_rate, filters: [filter::Biquad::default(); MAX_NUM_FILTERS], bypass_smoother, should_update_filters, next_filter_smoothing_in: 1, spectrum_input, spectrum_output: Arc::new(Mutex::new(spectrum_output)), } } } impl DiopserParams { fn new( sample_rate: Arc, should_update_filters: Arc, bypass_smoother: Arc>, ) -> Self { Self { editor_state: editor::default_state(), safe_mode: Arc::new(AtomicBool::new(true)), bypass: BoolParam::new("Bypass", false) .with_callback(Arc::new(move |value| { bypass_smoother.set_target( sample_rate.load(Ordering::Relaxed), if value { 1.0 } else { 0.0 }, ); })) .make_bypass(), 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, // This value is also used in the spectrum analyzer to match the spectrum analyzer // with this parameter which is bound to the X-Y pad's X-axis filter_frequency_range(), ) // 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_with_note_name(0, true)) .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", 0.0, FloatRange::SymmetricalSkewed { min: -5.0, max: 5.0, factor: FloatRange::skew_factor(-1.0), center: 0.0, }, ) .with_unit(" octaves") .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 = env!("CARGO_PKG_HOMEPAGE"); const EMAIL: &'static str = "mail@robbertvanderhelm.nl"; const VERSION: &'static str = env!("CARGO_PKG_VERSION"); const DEFAULT_INPUT_CHANNELS: u32 = 2; const DEFAULT_OUTPUT_CHANNELS: u32 = 2; const SAMPLE_ACCURATE_AUTOMATION: bool = true; type BackgroundTask = (); fn params(&self) -> Arc { self.params.clone() } fn editor(&self, _async_executor: AsyncExecutor) -> Option> { editor::create( editor::Data { params: self.params.clone(), sample_rate: self.sample_rate.clone(), spectrum: self.spectrum_output.clone(), safe_mode: self.params.safe_mode.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 .store(buffer_config.sample_rate, Ordering::Relaxed); true } fn reset(&mut self) { // Initialize and/or reset the filters on the next process call self.should_update_filters.store(true, Ordering::Release); self.bypass_smoother .reset(if self.params.bypass.value() { 1.0 } else { 0.0 }); } 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()); // The bypass parameter controls a smoother so we can crossfade between the dry and the wet // signals as needed if !self.params.bypass.value() || self.bypass_smoother.is_smoothing() { // We'll iterate in blocks to make the blending relatively cheap without having to // duplicate code or add a bunch of per-sample conditionals for (_, mut block) in buffer.iter_blocks(MAX_BLOCK_SIZE) { // We'll blend this with the dry signal as needed let mut dry = [f32x2::default(); MAX_BLOCK_SIZE]; let mut wet = [f32x2::default(); MAX_BLOCK_SIZE]; for (input_samples, (dry_samples, wet_samples)) in block .iter_samples() .zip(std::iter::zip(dry.iter_mut(), wet.iter_mut())) { 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. *dry_samples = unsafe { input_samples.to_simd_unchecked() }; *wet_samples = *dry_samples; for filter in self .filters .iter_mut() .take(self.params.filter_stages.value() as usize) { *wet_samples = filter.process(*wet_samples); } } // If the bypass smoother is activated then the bypass switch has just been flipped to // either the on or the off position if self.bypass_smoother.is_smoothing() { for (mut channel_samples, (dry_samples, wet_samples)) in block .iter_samples() .zip(std::iter::zip(dry.iter_mut(), wet.iter_mut())) { // We'll do an equal-power fade let dry_t_squared = self.bypass_smoother.next(); let dry_t = dry_t_squared.sqrt(); let wet_t = (1.0 - dry_t_squared).sqrt(); let dry_weighted = *dry_samples * f32x2::splat(dry_t); let wet_weighted = *wet_samples * f32x2::splat(wet_t); unsafe { channel_samples.from_simd_unchecked(dry_weighted + wet_weighted) }; } } else if self.params.bypass.value() { // If the bypass is enabled and we're no longer smoothing then the output should // just be the origianl dry signal } else { // Otherwise the signal is 100% wet for (mut channel_samples, wet_samples) in block.iter_samples().zip(wet.iter()) { unsafe { channel_samples.from_simd_unchecked(*wet_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 sample_rate = self.sample_rate.load(Ordering::Relaxed); 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 = 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(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);