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559
examples/Plugins/DSPModulePluginDemo.h
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559
examples/Plugins/DSPModulePluginDemo.h
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/*
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==============================================================================
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This file is part of the JUCE examples.
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Copyright (c) 2017 - ROLI Ltd.
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The code included in this file is provided under the terms of the ISC license
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http://www.isc.org/downloads/software-support-policy/isc-license. Permission
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To use, copy, modify, and/or distribute this software for any purpose with or
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without fee is hereby granted provided that the above copyright notice and
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this permission notice appear in all copies.
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THE SOFTWARE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES,
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WHETHER EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR
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PURPOSE, ARE DISCLAIMED.
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==============================================================================
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*/
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/*******************************************************************************
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The block below describes the properties of this PIP. A PIP is a short snippet
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of code that can be read by the Projucer and used to generate a JUCE project.
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BEGIN_JUCE_PIP_METADATA
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name: DSPModulePluginDemo
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version: 1.0.0
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vendor: juce
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website: http://juce.com
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description: Audio plugin using the DSP module.
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dependencies: juce_audio_basics, juce_audio_devices, juce_audio_formats,
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juce_audio_plugin_client, juce_audio_processors,
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juce_audio_utils, juce_core, juce_data_structures, juce_dsp,
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juce_events, juce_graphics, juce_gui_basics, juce_gui_extra
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exporters: xcode_mac, vs2017
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type: AudioProcessor
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mainClass: DspModulePluginDemoAudioProcessor
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useLocalCopy: 1
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END_JUCE_PIP_METADATA
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*******************************************************************************/
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#pragma once
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#include "../Assets/DemoUtilities.h"
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//==============================================================================
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struct ParameterSlider : public Slider,
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public Timer
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{
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ParameterSlider (AudioProcessorParameter& p)
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: Slider (p.getName (256)), param (p)
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{
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setRange (0.0, 1.0, 0.0);
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startTimerHz (30);
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updateSliderPos();
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}
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void valueChanged() override
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{
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if (isMouseButtonDown())
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param.setValueNotifyingHost ((float) Slider::getValue());
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else
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param.setValue ((float) Slider::getValue());
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}
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void timerCallback() override { updateSliderPos(); }
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void startedDragging() override { param.beginChangeGesture(); }
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void stoppedDragging() override { param.endChangeGesture(); }
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double getValueFromText (const String& text) override { return param.getValueForText (text); }
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String getTextFromValue (double value) override { return param.getText ((float) value, 1024) + " " + param.getLabel(); }
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void updateSliderPos()
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{
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auto newValue = param.getValue();
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if (newValue != (float) Slider::getValue() && ! isMouseButtonDown())
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Slider::setValue (newValue);
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}
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AudioProcessorParameter& param;
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};
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//==============================================================================
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/**
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This class handles the audio processing for the DSP module plugin demo.
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*/
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class DspModulePluginDemoAudioProcessor : public AudioProcessor
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{
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public:
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//==============================================================================
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DspModulePluginDemoAudioProcessor()
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: AudioProcessor (BusesProperties().withInput ("Input", AudioChannelSet::stereo(), true)
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.withOutput ("Output", AudioChannelSet::stereo(), true)),
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lowPassFilter (dsp::IIR::Coefficients<float>::makeFirstOrderLowPass (48000.0, 20000.0f)),
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highPassFilter (dsp::IIR::Coefficients<float>::makeFirstOrderHighPass (48000.0, 20.0f)),
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waveShapers { { std::tanh }, { dsp::FastMathApproximations::tanh } },
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clipping { clip }
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{
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// Oversampling 2 times with IIR filtering
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oversampling.reset (new dsp::Oversampling<float> (2, 1, dsp::Oversampling<float>::filterHalfBandPolyphaseIIR, false));
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addParameter (inputVolumeParam = new AudioParameterFloat ("INPUT", "Input Volume", { 0.0f, 60.0f, 0.0f, 1.0f }, 0.0f, "dB"));
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addParameter (highPassFilterFreqParam = new AudioParameterFloat ("HPFREQ", "Pre Highpass Freq.", { 20.0f, 20000.0f, 0.0f, 0.5f }, 20.0f, "Hz"));
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addParameter (lowPassFilterFreqParam = new AudioParameterFloat ("LPFREQ", "Post Lowpass Freq.", { 20.0f, 20000.0f, 0.0f, 0.5f }, 20000.0f, "Hz"));
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addParameter (stereoParam = new AudioParameterChoice ("STEREO", "Stereo Processing", { "Always mono", "Yes" }, 1));
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addParameter (slopeParam = new AudioParameterChoice ("SLOPE", "Slope", { "-6 dB / octave", "-12 dB / octave" }, 0));
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addParameter (waveshaperParam = new AudioParameterChoice ("WVSHP", "Waveshaper", { "std::tanh", "Fast tanh approx." }, 0));
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addParameter (cabinetTypeParam = new AudioParameterChoice ("CABTYPE", "Cabinet Type", { "Guitar amplifier 8'' cabinet ",
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"Cassette recorder cabinet" }, 0));
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addParameter (cabinetSimParam = new AudioParameterBool ("CABSIM", "Cabinet Sim", false));
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addParameter (oversamplingParam = new AudioParameterBool ("OVERS", "Oversampling", false));
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addParameter (outputVolumeParam = new AudioParameterFloat ("OUTPUT", "Output Volume", { -40.0f, 40.0f, 0.0f, 1.0f }, 0.0f, "dB"));
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cabinetType.set (0);
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}
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~DspModulePluginDemoAudioProcessor() {}
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//==============================================================================
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bool isBusesLayoutSupported (const BusesLayout& layouts) const override
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{
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// This is the place where you check if the layout is supported.
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// In this template code we only support mono or stereo.
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if (layouts.getMainOutputChannelSet() != AudioChannelSet::mono() && layouts.getMainOutputChannelSet() != AudioChannelSet::stereo())
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return false;
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// This checks if the input layout matches the output layout
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if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())
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return false;
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return true;
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}
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void prepareToPlay (double sampleRate, int samplesPerBlock) override
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{
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auto channels = static_cast<uint32> (jmin (getMainBusNumInputChannels(), getMainBusNumOutputChannels()));
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dsp::ProcessSpec spec { sampleRate, static_cast<uint32> (samplesPerBlock), channels };
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lowPassFilter .prepare (spec);
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highPassFilter.prepare (spec);
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inputVolume .prepare (spec);
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outputVolume.prepare (spec);
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convolution.prepare (spec);
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cabinetType.set (-1);
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oversampling->initProcessing (static_cast<size_t> (samplesPerBlock));
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updateParameters();
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reset();
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}
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void releaseResources() override {}
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void processBlock (AudioBuffer<float>& inoutBuffer, MidiBuffer&) override
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{
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auto totalNumInputChannels = getTotalNumInputChannels();
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auto totalNumOutputChannels = getTotalNumOutputChannels();
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auto numSamples = inoutBuffer.getNumSamples();
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for (auto i = jmin (2, totalNumInputChannels); i < totalNumOutputChannels; ++i)
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inoutBuffer.clear (i, 0, numSamples);
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updateParameters();
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dsp::AudioBlock<float> block (inoutBuffer);
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if (stereoParam->getIndex() == 1)
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{
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// Stereo processing mode:
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if (block.getNumChannels() > 2)
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block = block.getSubsetChannelBlock (0, 2);
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process (dsp::ProcessContextReplacing<float> (block));
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}
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else
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{
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// Mono processing mode:
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auto firstChan = block.getSingleChannelBlock (0);
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process (dsp::ProcessContextReplacing<float> (firstChan));
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for (size_t chan = 1; chan < block.getNumChannels(); ++chan)
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block.getSingleChannelBlock (chan).copy (firstChan);
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}
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}
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void reset() override
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{
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lowPassFilter .reset();
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highPassFilter.reset();
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convolution .reset();
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oversampling->reset();
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}
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//==============================================================================
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bool hasEditor() const override { return true; }
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AudioProcessorEditor* createEditor() override
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{
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return new DspModulePluginDemoAudioProcessorEditor (*this);
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}
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//==============================================================================
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bool acceptsMidi() const override { return false; }
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bool producesMidi() const override { return false; }
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const String getName() const override { return JucePlugin_Name; }
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double getTailLengthSeconds() const override { return 0.0; }
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//==============================================================================
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int getNumPrograms() override { return 1; }
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int getCurrentProgram() override { return 0; }
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void setCurrentProgram (int) override {}
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const String getProgramName (int) override { return {}; }
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void changeProgramName (int, const String&) override {}
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//==============================================================================
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void getStateInformation (MemoryBlock&) override {}
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void setStateInformation (const void*, int) override {}
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//==============================================================================
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void updateParameters()
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{
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auto newOversampling = oversamplingParam->get();
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if (newOversampling != audioCurrentlyOversampled)
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{
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audioCurrentlyOversampled = newOversampling;
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oversampling->reset();
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}
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//==============================================================================
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auto inputdB = Decibels::decibelsToGain (inputVolumeParam->get());
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auto outputdB = Decibels::decibelsToGain (outputVolumeParam->get());
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if (inputVolume .getGainLinear() != inputdB) inputVolume.setGainLinear (inputdB);
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if (outputVolume.getGainLinear() != outputdB) outputVolume.setGainLinear (outputdB);
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auto newSlopeType = slopeParam->getIndex();
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if (newSlopeType == 0)
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{
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*lowPassFilter .state = *dsp::IIR::Coefficients<float>::makeFirstOrderLowPass (getSampleRate(), lowPassFilterFreqParam->get());
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*highPassFilter.state = *dsp::IIR::Coefficients<float>::makeFirstOrderHighPass (getSampleRate(), highPassFilterFreqParam->get());
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}
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else
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{
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*lowPassFilter .state = *dsp::IIR::Coefficients<float>::makeLowPass (getSampleRate(), lowPassFilterFreqParam->get());
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*highPassFilter.state = *dsp::IIR::Coefficients<float>::makeHighPass (getSampleRate(), highPassFilterFreqParam->get());
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}
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//==============================================================================
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auto type = cabinetTypeParam->getIndex();
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auto currentType = cabinetType.get();
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if (type != currentType)
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{
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cabinetType.set(type);
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auto maxSize = static_cast<size_t> (roundToInt (getSampleRate() * (8192.0 / 44100.0)));
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if (type == 0)
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convolution.loadImpulseResponse (getAssetsDirectory().getChildFile ("Impulse1.wav"), false, true, maxSize);
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else
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convolution.loadImpulseResponse (getAssetsDirectory().getChildFile ("Impulse2.wav"), false, true, maxSize);
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}
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cabinetIsBypassed = ! cabinetSimParam->get();
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}
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static inline float clip (float x) { return jmax (-1.0f, jmin (1.0f, x)); }
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//==============================================================================
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AudioParameterFloat* inputVolumeParam;
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AudioParameterFloat* outputVolumeParam;
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AudioParameterFloat* lowPassFilterFreqParam;
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AudioParameterFloat* highPassFilterFreqParam;
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AudioParameterChoice* stereoParam;
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AudioParameterChoice* slopeParam;
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AudioParameterChoice* waveshaperParam;
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AudioParameterChoice* cabinetTypeParam;
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AudioParameterBool* cabinetSimParam;
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AudioParameterBool* oversamplingParam;
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private:
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//==============================================================================
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/**
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This is the editor component that will be displayed.
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*/
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class DspModulePluginDemoAudioProcessorEditor : public AudioProcessorEditor
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{
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public:
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//==============================================================================
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DspModulePluginDemoAudioProcessorEditor (DspModulePluginDemoAudioProcessor& p)
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: AudioProcessorEditor (&p),
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processor (p),
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inputVolumeLabel ({}, processor.inputVolumeParam->name),
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outputVolumeLabel ({}, processor.outputVolumeParam->name),
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lowPassFilterFreqLabel ({}, processor.lowPassFilterFreqParam->name),
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highPassFilterFreqLabel ({}, processor.highPassFilterFreqParam->name),
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stereoLabel ({}, processor.stereoParam->name),
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slopeLabel ({}, processor.slopeParam->name),
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waveshaperLabel ({}, processor.waveshaperParam->name),
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cabinetTypeLabel ({}, processor.cabinetTypeParam->name)
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{
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//==============================================================================
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inputVolumeSlider .reset (new ParameterSlider (*processor.inputVolumeParam));
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outputVolumeSlider .reset (new ParameterSlider (*processor.outputVolumeParam));
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lowPassFilterFreqSlider .reset (new ParameterSlider (*processor.lowPassFilterFreqParam));
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highPassFilterFreqSlider.reset (new ParameterSlider (*processor.highPassFilterFreqParam));
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addAndMakeVisible (inputVolumeSlider .get());
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addAndMakeVisible (outputVolumeSlider .get());
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addAndMakeVisible (lowPassFilterFreqSlider .get());
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addAndMakeVisible (highPassFilterFreqSlider.get());
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addAndMakeVisible (inputVolumeLabel);
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inputVolumeLabel.setJustificationType (Justification::centredLeft);
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inputVolumeLabel.attachToComponent (inputVolumeSlider.get(), true);
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addAndMakeVisible (outputVolumeLabel);
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outputVolumeLabel.setJustificationType (Justification::centredLeft);
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outputVolumeLabel.attachToComponent (outputVolumeSlider.get(), true);
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addAndMakeVisible (lowPassFilterFreqLabel);
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lowPassFilterFreqLabel.setJustificationType (Justification::centredLeft);
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lowPassFilterFreqLabel.attachToComponent (lowPassFilterFreqSlider.get(), true);
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addAndMakeVisible (highPassFilterFreqLabel);
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highPassFilterFreqLabel.setJustificationType (Justification::centredLeft);
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highPassFilterFreqLabel.attachToComponent (highPassFilterFreqSlider.get(), true);
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//==============================================================================
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addAndMakeVisible (stereoBox);
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auto i = 1;
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for (auto choice : processor.stereoParam->choices)
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stereoBox.addItem (choice, i++);
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stereoBox.onChange = [this] { processor.stereoParam->operator= (stereoBox.getSelectedItemIndex()); };
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stereoBox.setSelectedId (processor.stereoParam->getIndex() + 1);
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addAndMakeVisible (stereoLabel);
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stereoLabel.setJustificationType (Justification::centredLeft);
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stereoLabel.attachToComponent (&stereoBox, true);
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//==============================================================================
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addAndMakeVisible(slopeBox);
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i = 1;
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for (auto choice : processor.slopeParam->choices)
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slopeBox.addItem(choice, i++);
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slopeBox.onChange = [this] { processor.slopeParam->operator= (slopeBox.getSelectedItemIndex()); };
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slopeBox.setSelectedId(processor.slopeParam->getIndex() + 1);
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addAndMakeVisible(slopeLabel);
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slopeLabel.setJustificationType(Justification::centredLeft);
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slopeLabel.attachToComponent(&slopeBox, true);
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//==============================================================================
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addAndMakeVisible (waveshaperBox);
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i = 1;
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for (auto choice : processor.waveshaperParam->choices)
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waveshaperBox.addItem (choice, i++);
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waveshaperBox.onChange = [this] { processor.waveshaperParam->operator= (waveshaperBox.getSelectedItemIndex()); };
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waveshaperBox.setSelectedId (processor.waveshaperParam->getIndex() + 1);
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addAndMakeVisible (waveshaperLabel);
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waveshaperLabel.setJustificationType (Justification::centredLeft);
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waveshaperLabel.attachToComponent (&waveshaperBox, true);
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//==============================================================================
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addAndMakeVisible (cabinetTypeBox);
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i = 1;
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for (auto choice : processor.cabinetTypeParam->choices)
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cabinetTypeBox.addItem (choice, i++);
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cabinetTypeBox.onChange = [this] { processor.cabinetTypeParam->operator= (cabinetTypeBox.getSelectedItemIndex()); };
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cabinetTypeBox.setSelectedId (processor.cabinetTypeParam->getIndex() + 1);
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addAndMakeVisible (cabinetTypeLabel);
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cabinetTypeLabel.setJustificationType (Justification::centredLeft);
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cabinetTypeLabel.attachToComponent (&cabinetTypeBox, true);
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//==============================================================================
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addAndMakeVisible (cabinetSimButton);
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cabinetSimButton.onClick = [this] { processor.cabinetSimParam->operator= (cabinetSimButton.getToggleState()); };
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cabinetSimButton.setButtonText (processor.cabinetSimParam->name);
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cabinetSimButton.setToggleState (processor.cabinetSimParam->get(), NotificationType::dontSendNotification);
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addAndMakeVisible (oversamplingButton);
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oversamplingButton.onClick = [this] { processor.oversamplingParam->operator= (oversamplingButton.getToggleState()); };
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oversamplingButton.setButtonText (processor.oversamplingParam->name);
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oversamplingButton.setToggleState (processor.oversamplingParam->get(), NotificationType::dontSendNotification);
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//==============================================================================
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setSize (600, 400);
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}
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~DspModulePluginDemoAudioProcessorEditor() {}
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//==============================================================================
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void paint (Graphics& g) override
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{
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g.setColour (getLookAndFeel().findColour (ResizableWindow::backgroundColourId));
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g.fillAll();
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}
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void resized() override
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{
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auto bounds = getLocalBounds().reduced (10);
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bounds.removeFromTop (10);
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bounds.removeFromLeft (125);
|
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//==============================================================================
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inputVolumeSlider->setBounds (bounds.removeFromTop (30));
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bounds.removeFromTop (5);
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outputVolumeSlider->setBounds (bounds.removeFromTop (30));
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bounds.removeFromTop (15);
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highPassFilterFreqSlider->setBounds (bounds.removeFromTop (30));
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bounds.removeFromTop (5);
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lowPassFilterFreqSlider->setBounds (bounds.removeFromTop (30));
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bounds.removeFromTop (15);
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//==============================================================================
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stereoBox.setBounds (bounds.removeFromTop(30));
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bounds.removeFromTop (5);
|
||||
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||||
slopeBox.setBounds (bounds.removeFromTop (30));
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bounds.removeFromTop (5);
|
||||
|
||||
waveshaperBox.setBounds (bounds.removeFromTop (30));
|
||||
bounds.removeFromTop (5);
|
||||
|
||||
cabinetTypeBox.setBounds (bounds.removeFromTop (30));
|
||||
bounds.removeFromTop (15);
|
||||
|
||||
//==============================================================================
|
||||
auto buttonSlice = bounds.removeFromTop (30);
|
||||
cabinetSimButton.setSize (200, buttonSlice.getHeight());
|
||||
cabinetSimButton.setCentrePosition (buttonSlice.getCentre());
|
||||
bounds.removeFromTop(5);
|
||||
|
||||
buttonSlice = bounds.removeFromTop (30);
|
||||
oversamplingButton.setSize(200, buttonSlice.getHeight());
|
||||
oversamplingButton.setCentrePosition(buttonSlice.getCentre());
|
||||
}
|
||||
|
||||
private:
|
||||
//==============================================================================
|
||||
DspModulePluginDemoAudioProcessor& processor;
|
||||
|
||||
ScopedPointer<ParameterSlider> inputVolumeSlider, outputVolumeSlider,
|
||||
lowPassFilterFreqSlider, highPassFilterFreqSlider;
|
||||
ComboBox stereoBox, slopeBox, waveshaperBox, cabinetTypeBox;
|
||||
ToggleButton cabinetSimButton, oversamplingButton;
|
||||
|
||||
Label inputVolumeLabel, outputVolumeLabel, lowPassFilterFreqLabel,
|
||||
highPassFilterFreqLabel, stereoLabel, slopeLabel, waveshaperLabel,
|
||||
cabinetTypeLabel;
|
||||
|
||||
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DspModulePluginDemoAudioProcessorEditor)
|
||||
};
|
||||
|
||||
//==============================================================================
|
||||
void process (dsp::ProcessContextReplacing<float> context) noexcept
|
||||
{
|
||||
ScopedNoDenormals noDenormals;
|
||||
|
||||
// Input volume applied with a LinearSmoothedValue
|
||||
inputVolume.process (context);
|
||||
|
||||
// Pre-highpass filtering, very useful for distortion audio effects
|
||||
// Note : try frequencies around 700 Hz
|
||||
highPassFilter.process (context);
|
||||
|
||||
// Upsampling
|
||||
dsp::AudioBlock<float> oversampledBlock;
|
||||
|
||||
setLatencySamples (audioCurrentlyOversampled ? roundToInt (oversampling->getLatencyInSamples()) : 0);
|
||||
|
||||
if (audioCurrentlyOversampled)
|
||||
oversampledBlock = oversampling->processSamplesUp (context.getInputBlock());
|
||||
|
||||
auto waveshaperContext = audioCurrentlyOversampled ? dsp::ProcessContextReplacing<float> (oversampledBlock)
|
||||
: context;
|
||||
|
||||
// Waveshaper processing, for distortion generation, thanks to the input gain
|
||||
// The fast tanh can be used instead of std::tanh to reduce the CPU load
|
||||
auto waveshaperIndex = waveshaperParam->getIndex();
|
||||
|
||||
if (isPositiveAndBelow (waveshaperIndex, numWaveShapers) )
|
||||
{
|
||||
waveShapers[waveshaperIndex].process (waveshaperContext);
|
||||
|
||||
if (waveshaperIndex == 1)
|
||||
clipping.process (waveshaperContext);
|
||||
|
||||
waveshaperContext.getOutputBlock() *= 0.7f;
|
||||
}
|
||||
|
||||
// Downsampling
|
||||
if (audioCurrentlyOversampled)
|
||||
oversampling->processSamplesDown (context.getOutputBlock());
|
||||
|
||||
// Post-lowpass filtering
|
||||
lowPassFilter.process (context);
|
||||
|
||||
// Convolution with the impulse response of a guitar cabinet
|
||||
auto wasBypassed = context.isBypassed;
|
||||
context.isBypassed = context.isBypassed || cabinetIsBypassed;
|
||||
convolution.process (context);
|
||||
context.isBypassed = wasBypassed;
|
||||
|
||||
// Output volume applied with a LinearSmoothedValue
|
||||
outputVolume.process (context);
|
||||
}
|
||||
|
||||
//==============================================================================
|
||||
dsp::ProcessorDuplicator<dsp::IIR::Filter<float>, dsp::IIR::Coefficients<float>> lowPassFilter, highPassFilter;
|
||||
dsp::Convolution convolution;
|
||||
|
||||
static constexpr size_t numWaveShapers = 2;
|
||||
dsp::WaveShaper<float> waveShapers[numWaveShapers];
|
||||
dsp::WaveShaper<float> clipping;
|
||||
|
||||
dsp::Gain<float> inputVolume, outputVolume;
|
||||
|
||||
ScopedPointer<dsp::Oversampling<float>> oversampling;
|
||||
bool audioCurrentlyOversampled = false;
|
||||
|
||||
Atomic<int> cabinetType;
|
||||
bool cabinetIsBypassed = false;
|
||||
|
||||
//==============================================================================
|
||||
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DspModulePluginDemoAudioProcessor)
|
||||
};
|
||||
Loading…
Add table
Add a link
Reference in a new issue