andrewm@0: /*
andrewm@0: This code accompanies the textbook:
andrewm@0:
andrewm@0: Digital Audio Effects: Theory, Implementation and Application
andrewm@0: Joshua D. Reiss and Andrew P. McPherson
andrewm@0:
andrewm@0: ---
andrewm@0:
andrewm@0: Distortion: distortion effect using different characteristic curves
andrewm@0: See textbook Chapter 7: Overdrive, Distortion and Fuzz
andrewm@0:
andrewm@0: Code by Brecht De Man, Joshua Reiss and Andrew McPherson
andrewm@0:
andrewm@0: When using this code (or a modified version thereof), please cite:
andrewm@0:
andrewm@0: Brecht De Man and Joshua D. Reiss, "Adaptive Control of Amplitude
andrewm@0: Distortion Effects," 53rd Conference of the Audio Engineering Society,
andrewm@0: 2014.
andrewm@0:
andrewm@0: ---
andrewm@0:
andrewm@0: This program is free software: you can redistribute it and/or modify
andrewm@0: it under the terms of the GNU General Public License as published by
andrewm@0: the Free Software Foundation, either version 3 of the License, or
andrewm@0: (at your option) any later version.
andrewm@0:
andrewm@0: This program is distributed in the hope that it will be useful,
andrewm@0: but WITHOUT ANY WARRANTY; without even the implied warranty of
andrewm@0: MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
andrewm@0: GNU General Public License for more details.
andrewm@0:
andrewm@0: You should have received a copy of the GNU General Public License
andrewm@0: along with this program. If not, see .
andrewm@0: */
andrewm@0:
andrewm@0: #include "PluginProcessor.h"
andrewm@0: #include "PluginEditor.h"
andrewm@0:
andrewm@0: #if JUCE_INTEL
andrewm@0: #define JUCE_SNAP_TO_ZERO(n) if (! (n < -1.0e-8 || n > 1.0e-8)) n = 0;
andrewm@0: #else
andrewm@0: #define JUCE_SNAP_TO_ZERO(n)
andrewm@0: #endif
andrewm@0:
andrewm@0: DistortionAudioProcessor::DistortionAudioProcessor()
andrewm@0: :
andrewm@0: _numChannels (1)
andrewm@0: ,_numSamples (1) // dummy - will be set in prepareToPlay
andrewm@0: ,_sampleRate (1) // dummy - will be set in prepareToPlay
andrewm@0: ,_typeNumber (_hardClipping) // standard
andrewm@0: ,_currentTrackBuffer (1,1)
andrewm@0: ,_lastUIWidth (850)
andrewm@0: ,_lastUIHeight (650)
andrewm@0:
andrewm@0: {
andrewm@0: Reset();
andrewm@0: }
andrewm@0:
andrewm@0: DistortionAudioProcessor::~DistortionAudioProcessor()
andrewm@0: {
andrewm@0: }
andrewm@0:
andrewm@0: //-----------------------------------------------------------------------------
andrewm@0: // P R E P A R E T O P L A Y
andrewm@0: void DistortionAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
andrewm@0: {
andrewm@0: // If sample rate/block size changes or straight after construction
andrewm@0: if (_numSamples != samplesPerBlock || _sampleRate != sampleRate)
andrewm@0: {
andrewm@0: _sampleRate = sampleRate;
andrewm@0: _numSamples = samplesPerBlock;
andrewm@0: _numChannels = getNumInputChannels();
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0:
andrewm@0: //-----------------------------------------------------------------------------
andrewm@0: // P R O C E S S B L O C K
andrewm@0: void DistortionAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
andrewm@0: {
andrewm@0: float gain = powf(10.0f, _gainIndB/20.f);
andrewm@0:
andrewm@0:
andrewm@0: for (int channel = 0; channel < _numChannels; ++channel)
andrewm@0: {
andrewm@0: // Apply gain
andrewm@0: buffer.applyGain(channel, 0, buffer.getNumSamples(), gain);
andrewm@0:
andrewm@0: // Put track audio data into _currentTrackBuffer
andrewm@0: float * originalData = new float;
b@1: originalData = buffer.getWritePointer(channel);
andrewm@0:
andrewm@0: // Apply distortion (sample per sample)
andrewm@0: switch (_typeNumber) {
andrewm@0: case _hardClipping:
andrewm@0: {
andrewm@0: float threshold = 0.5f;
andrewm@0: for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
andrewm@0: {
andrewm@0: if(originalData[sample] > threshold) // positive hard clipping
andrewm@0: {
andrewm@0: originalData[sample] = threshold;
andrewm@0: }
andrewm@0: else
andrewm@0: {
andrewm@0: if(originalData[sample] < - threshold) // negative hard clipping
andrewm@0: {
andrewm@0: originalData[sample] = - threshold;
andrewm@0: }
andrewm@0: }
andrewm@0: }
andrewm@0: break;
andrewm@0: }
andrewm@0:
andrewm@0: case _softClipping:
andrewm@0: {
andrewm@0: float threshold1 = 1.0f/3.0f;
andrewm@0: float threshold2 = 2.0f/3.0f;
andrewm@0: for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
andrewm@0: {
andrewm@0: if(originalData[sample] > threshold1)
andrewm@0: {
andrewm@0: if(originalData[sample] > threshold2) // positive clipping
andrewm@0: {
andrewm@0: originalData[sample] = 1.0f;
andrewm@0: }
andrewm@0: else // soft knee (positive)
andrewm@0: {
andrewm@0: originalData[sample] = (3.0f - (2.0f - 3.0f*originalData[sample])*(2.0f - 3.0f*originalData[sample]))/3.0f;
andrewm@0: }
andrewm@0: }
andrewm@0: else
andrewm@0: {
andrewm@0: if(originalData[sample] < -threshold1)
andrewm@0: {
andrewm@0: if(originalData[sample] < -threshold2) // negative clipping
andrewm@0: {
andrewm@0: originalData[sample] = -1.0f;
andrewm@0: }
andrewm@0: else // soft knee (negative)
andrewm@0: {
andrewm@0: originalData[sample] = - (3.0f - (2.0f + 3.0f*originalData[sample])*(2.0f + 3.0f*originalData[sample]))/3.0f;
andrewm@0: }
andrewm@0: }
andrewm@0: else // linear region (-1/3..1/3)
andrewm@0: {
andrewm@0: originalData[sample] *= 2.0f;
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0: originalData[sample] /= 2.0f; // divide all by 2 to compensate for extra 6 dB gain boost
andrewm@0: }
andrewm@0: break;
andrewm@0: }
andrewm@0:
andrewm@0: case _softClippingExp:
andrewm@0: {
andrewm@0: for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
andrewm@0: {
andrewm@0: if (originalData[sample] > 0.0f) // positive
andrewm@0: {
andrewm@0: originalData[sample] = 1.0f - expf(-originalData[sample]);
andrewm@0: }
andrewm@0: else // negative
andrewm@0: {
andrewm@0: originalData[sample] = - 1.0f + expf(originalData[sample]);
andrewm@0: }
andrewm@0: }
andrewm@0: break;
andrewm@0: }
andrewm@0:
andrewm@0: case _fullWaveRectifier:
andrewm@0: {
andrewm@0: for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
andrewm@0: {
andrewm@0: originalData[sample] = fabs(originalData[sample]);
andrewm@0: }
andrewm@0: break;
andrewm@0: }
andrewm@0:
andrewm@0: case _halfWaveRectifier:
andrewm@0: {
andrewm@0: for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
andrewm@0: {
andrewm@0: originalData[sample] = 0.5f*(fabs(originalData[sample])+originalData[sample]);
andrewm@0: }
andrewm@0: break;
andrewm@0: }
andrewm@0:
andrewm@0: default:
andrewm@0: break;
andrewm@0: }
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0:
andrewm@0: //-----------------------------------------------------------------------------
andrewm@0: // R E S E T
andrewm@0: void DistortionAudioProcessor::Reset()
andrewm@0: {
andrewm@0: _gainIndB = 0.0f;
andrewm@0: _typeNumber = _hardClipping;
andrewm@0: }
andrewm@0:
andrewm@0:
andrewm@0: //-----------------------------------------------------------------------------
andrewm@0: //
andrewm@0: void DistortionAudioProcessor::releaseResources()
andrewm@0: {
andrewm@0: // When playback stops, you can use this to free up any spare memory, etc.
andrewm@0: }
andrewm@0:
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::hasEditor() const
andrewm@0: {
andrewm@0: return true; // (change this to false if you choose to not supply an editor)
andrewm@0: }
andrewm@0:
andrewm@0: AudioProcessorEditor* DistortionAudioProcessor::createEditor()
andrewm@0: {
andrewm@0: return new DistortionAudioProcessorEditor (this);
andrewm@0: }
andrewm@0:
andrewm@0:
andrewm@0: //==============================================================================
andrewm@0: void DistortionAudioProcessor::getStateInformation (MemoryBlock& destData)
andrewm@0: {
andrewm@0: // SAVE STATE INFO
andrewm@0: XmlElement xml("Distortion_XML");
andrewm@0:
andrewm@0: // Knobs
andrewm@0: xml.setAttribute("_gain" ,_gainIndB);
andrewm@0:
andrewm@0: // Combo box
andrewm@0: xml.setAttribute("_type" ,(int) _typeNumber);
andrewm@0:
andrewm@0: // then use this helper function to stuff it into the binary blob and return it..
andrewm@0: copyXmlToBinary(xml, destData);
andrewm@0: }
andrewm@0:
andrewm@0: void DistortionAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
andrewm@0: {
andrewm@0: // LOAD STATE INFO
andrewm@0: ScopedPointer xmlState (getXmlFromBinary (data, sizeInBytes));
andrewm@0:
andrewm@0: // make sure that it's actually our type of XML object..
andrewm@0: if(xmlState->hasTagName("Distortion_XML"))
andrewm@0: {
andrewm@0:
andrewm@0: // Knobs
andrewm@0: _gainIndB = (float) xmlState->getDoubleAttribute("_gain",true);
andrewm@0:
andrewm@0: //Combo box
andrewm@0: _typeNumber = (DistortionAudioProcessor::Types) xmlState->getIntAttribute("_type",true);
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0: // This creates new instances of the plugin..
andrewm@0: AudioProcessor* JUCE_CALLTYPE createPluginFilter()
andrewm@0: {
andrewm@0: return new DistortionAudioProcessor();
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getName() const
andrewm@0: {
andrewm@0: return JucePlugin_Name;
andrewm@0: }
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::silenceInProducesSilenceOut() const
andrewm@0: {
andrewm@0: return true;
andrewm@0: }
andrewm@0:
andrewm@0: int DistortionAudioProcessor::getNumParameters()
andrewm@0: {
andrewm@0: return _PARAMtotalNumParams;
andrewm@0: }
andrewm@0:
andrewm@0: float DistortionAudioProcessor::getParameter (int index) // externally accessible
andrewm@0: {
andrewm@0: switch (index)
andrewm@0: {
andrewm@0: case _PARAMdeviceReset: return 0.0f;
andrewm@0: case _PARAMgain: return (GetGainIndB());
andrewm@0: case _PARAMtype: return (GetType());
andrewm@0: default: return 0.0f;
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0: void DistortionAudioProcessor::setParameter (int index, float newValue) // externally accessible
andrewm@0: {
andrewm@0: switch (index)
andrewm@0: {
andrewm@0: case _PARAMdeviceReset:
andrewm@0: Reset();
andrewm@0: break;
andrewm@0: case _PARAMgain:
andrewm@0: SetGainIndB(newValue);
andrewm@0: break;
andrewm@0: case _PARAMtype:
andrewm@0: SetType((DistortionAudioProcessor::Types) roundFloatToInt(newValue*_numberOfTypes));
andrewm@0: break;
andrewm@0: default:
andrewm@0: break;
andrewm@0: }
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getParameterName (int index) // externally accessible
andrewm@0: {
andrewm@0: switch (index)
andrewm@0: {
andrewm@0: case _PARAMdeviceReset: return "Reset";
andrewm@0: case _PARAMgain: return "Gain (dB)";
andrewm@0: case _PARAMtype: return "Type";
andrewm@0: default: break;
andrewm@0: }
andrewm@0: return String::empty;
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getParameterText (int index)
andrewm@0: {
andrewm@0: return String (getParameter (index), 2);
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getInputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0: return String (channelIndex + 1);
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getOutputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0: return String (channelIndex + 1);
andrewm@0: }
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::isInputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0: return true;
andrewm@0: }
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::isOutputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0: return true;
andrewm@0: }
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::acceptsMidi() const
andrewm@0: {
andrewm@0: #if JucePlugin_WantsMidiInput
andrewm@0: return true;
andrewm@0: #else
andrewm@0: return false;
andrewm@0: #endif
andrewm@0: }
andrewm@0:
andrewm@0: bool DistortionAudioProcessor::producesMidi() const
andrewm@0: {
andrewm@0: #if JucePlugin_ProducesMidiOutput
andrewm@0: return true;
andrewm@0: #else
andrewm@0: return false;
andrewm@0: #endif
andrewm@0: }
andrewm@0:
andrewm@0: int DistortionAudioProcessor::getNumPrograms()
andrewm@0: {
andrewm@0: return 0;
andrewm@0: }
andrewm@0:
andrewm@0: int DistortionAudioProcessor::getCurrentProgram()
andrewm@0: {
andrewm@0: return 0;
andrewm@0: }
andrewm@0:
andrewm@0: void DistortionAudioProcessor::setCurrentProgram (int index)
andrewm@0: {
andrewm@0: }
andrewm@0:
andrewm@0: const String DistortionAudioProcessor::getProgramName (int index)
andrewm@0: {
andrewm@0: return String::empty;
andrewm@0: }
andrewm@0:
andrewm@0: void DistortionAudioProcessor::changeProgramName (int index, const String& newName)
andrewm@0: {
andrewm@0: }