/* ============================================================================== This file is part of the JUCE library. Copyright (c) 2022 - Raw Material Software Limited JUCE is an open source library subject to commercial or open-source licensing. By using JUCE, you agree to the terms of both the JUCE 7 End-User License Agreement and JUCE Privacy Policy. End User License Agreement: www.juce.com/juce-7-licence Privacy Policy: www.juce.com/juce-privacy-policy Or: You may also use this code under the terms of the GPL v3 (see www.gnu.org/licenses). JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE DISCLAIMED. ============================================================================== */ namespace juce { namespace dsp { //=============================================================================== /** A processor that performs multi-channel oversampling. This class can be configured to do a factor of 2, 4, 8 or 16 times oversampling, using multiple stages, with polyphase allpass IIR filters or FIR filters, and latency compensation. The principle of oversampling is to increase the sample rate of a given non-linear process to prevent it from creating aliasing. Oversampling works by upsampling the input signal N times, processing the upsampled signal with the increased internal sample rate, then downsampling the result to get back to the original sample rate. Choose between FIR or IIR filtering depending on your needs in terms of latency and phase distortion. With FIR filters the phase is linear but the latency is maximised. With IIR filtering the phase is compromised around the Nyquist frequency but the latency is minimised. @see FilterDesign. @tags{DSP} */ template class JUCE_API Oversampling { public: /** The type of filter that can be used for the oversampling processing. */ enum FilterType { filterHalfBandFIREquiripple = 0, filterHalfBandPolyphaseIIR, numFilterTypes }; //=============================================================================== /** The default constructor. Note: This creates a "dummy" oversampling stage, which needs to be removed before adding proper oversampling stages. @param numChannels the number of channels to process with this object @see clearOversamplingStages, addOversamplingStage */ explicit Oversampling (size_t numChannels = 1); /** Constructor. @param numChannels the number of channels to process with this object @param factor the processing will perform 2 ^ factor times oversampling @param type the type of filter design employed for filtering during oversampling @param isMaxQuality if the oversampling is done using the maximum quality, where the filters will be more efficient but the CPU load will increase as well @param useIntegerLatency if true this processor will add some fractional delay at the end of the signal path to ensure that the overall latency of the oversampling is an integer */ Oversampling (size_t numChannels, size_t factor, FilterType type, bool isMaxQuality = true, bool useIntegerLatency = false); /** Destructor. */ ~Oversampling(); //=============================================================================== /* Sets if this processor should add some fractional delay at the end of the signal path to ensure that the overall latency of the oversampling is an integer. */ void setUsingIntegerLatency (bool shouldUseIntegerLatency) noexcept; /** Returns the latency in samples of the overall processing. You can use this information in your main processor to compensate the additional latency involved with the oversampling, for example with a dry / wet mixer, and to report the latency to the DAW. Note: If you have not opted to use an integer latency then the latency may not be integer, so you might need to round its value or to compensate it properly in your processing code since plug-ins can only report integer latency values in samples to the DAW. */ SampleType getLatencyInSamples() const noexcept; /** Returns the current oversampling factor. */ size_t getOversamplingFactor() const noexcept; //=============================================================================== /** Must be called before any processing, to set the buffer sizes of the internal buffers of the oversampling processing. */ void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling); /** Resets the processing pipeline, ready to oversample a new stream of data. */ void reset() noexcept; /** Must be called to perform the upsampling, prior to any oversampled processing. Returns an AudioBlock referencing the oversampled input signal, which must be used to perform the non-linear processing which needs the higher sample rate. Don't forget to set the sample rate of that processing to N times the original sample rate. */ AudioBlock processSamplesUp (const AudioBlock& inputBlock) noexcept; /** Must be called to perform the downsampling, after the upsampling and the non-linear processing. The output signal is probably delayed by the internal latency of the whole oversampling behaviour, so don't forget to take this into account. */ void processSamplesDown (AudioBlock& outputBlock) noexcept; //=============================================================================== /** Adds a new oversampling stage to the Oversampling class, multiplying the current oversampling factor by two. This is used with the default constructor to create custom oversampling chains, requiring a call to the clearOversamplingStages before any addition. Note: Upsampling and downsampling filtering have different purposes, the former removes upsampling artefacts while the latter removes useless frequency content created by the oversampled process, so usually the attenuation is increased when upsampling compared to downsampling. @param normalisedTransitionWidthUp a value between 0 and 0.5 which specifies how much the transition between passband and stopband is steep, for upsampling filtering (the lower the better) @param stopbandAmplitudedBUp the amplitude in dB in the stopband for upsampling filtering, must be negative @param normalisedTransitionWidthDown a value between 0 and 0.5 which specifies how much the transition between passband and stopband is steep, for downsampling filtering (the lower the better) @param stopbandAmplitudedBDown the amplitude in dB in the stopband for downsampling filtering, must be negative @see clearOversamplingStages */ void addOversamplingStage (FilterType, float normalisedTransitionWidthUp, float stopbandAmplitudedBUp, float normalisedTransitionWidthDown, float stopbandAmplitudedBDown); /** Adds a new "dummy" oversampling stage, which does nothing to the signal. Using one can be useful if your application features a customisable oversampling factor and if you want to select the current one from an OwnedArray without changing anything in the processing code. @see OwnedArray, clearOversamplingStages, addOversamplingStage */ void addDummyOversamplingStage(); /** Removes all the previously registered oversampling stages, so you can add your own from scratch. @see addOversamplingStage, addDummyOversamplingStage */ void clearOversamplingStages(); //=============================================================================== size_t factorOversampling = 1; size_t numChannels = 1; #ifndef DOXYGEN struct OversamplingStage; #endif private: //=============================================================================== void updateDelayLine(); SampleType getUncompensatedLatency() const noexcept; //=============================================================================== OwnedArray stages; bool isReady = false, shouldUseIntegerLatency = false; DelayLine delay { 8 }; SampleType fractionalDelay = 0; //=============================================================================== JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling) }; } // namespace dsp } // namespace juce