215 lines
9.5 KiB
C
215 lines
9.5 KiB
C
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/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2020 - Raw Material Software Limited
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JUCE is an open source library subject to commercial or open-source
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licensing.
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By using JUCE, you agree to the terms of both the JUCE 6 End-User License
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Agreement and JUCE Privacy Policy (both effective as of the 16th June 2020).
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End User License Agreement: www.juce.com/juce-6-licence
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Privacy Policy: www.juce.com/juce-privacy-policy
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Or: You may also use this code under the terms of the GPL v3 (see
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www.gnu.org/licenses).
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JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
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EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
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DISCLAIMED.
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==============================================================================
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*/
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namespace juce
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{
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namespace dsp
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{
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//===============================================================================
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/**
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A processor that performs multi-channel oversampling.
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This class can be configured to do a factor of 2, 4, 8 or 16 times
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oversampling, using multiple stages, with polyphase allpass IIR filters or FIR
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filters, and latency compensation.
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The principle of oversampling is to increase the sample rate of a given
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non-linear process to prevent it from creating aliasing. Oversampling works
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by upsampling the input signal N times, processing the upsampled signal
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with the increased internal sample rate, then downsampling the result to get
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back to the original sample rate.
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Choose between FIR or IIR filtering depending on your needs in terms of
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latency and phase distortion. With FIR filters the phase is linear but the
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latency is maximised. With IIR filtering the phase is compromised around the
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Nyquist frequency but the latency is minimised.
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@see FilterDesign.
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@tags{DSP}
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*/
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template <typename SampleType>
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class JUCE_API Oversampling
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{
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public:
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/** The type of filter that can be used for the oversampling processing. */
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enum FilterType
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{
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filterHalfBandFIREquiripple = 0,
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filterHalfBandPolyphaseIIR,
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numFilterTypes
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};
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//===============================================================================
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/** The default constructor.
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Note: This creates a "dummy" oversampling stage, which needs to be removed
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before adding proper oversampling stages.
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@param numChannels the number of channels to process with this object
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@see clearOversamplingStages, addOversamplingStage
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*/
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explicit Oversampling (size_t numChannels = 1);
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/** Constructor.
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@param numChannels the number of channels to process with this object
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@param factor the processing will perform 2 ^ factor times oversampling
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@param type the type of filter design employed for filtering during
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oversampling
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@param isMaxQuality if the oversampling is done using the maximum quality, where
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the filters will be more efficient but the CPU load will
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increase as well
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@param useIntegerLatency if true this processor will add some fractional delay at the
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end of the signal path to ensure that the overall latency of
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the oversampling is an integer
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*/
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Oversampling (size_t numChannels,
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size_t factor,
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FilterType type,
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bool isMaxQuality = true,
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bool useIntegerLatency = false);
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/** Destructor. */
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~Oversampling();
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//===============================================================================
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/* Sets if this processor should add some fractional delay at the end of the signal
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path to ensure that the overall latency of the oversampling is an integer.
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*/
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void setUsingIntegerLatency (bool shouldUseIntegerLatency) noexcept;
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/** Returns the latency in samples of the overall processing. You can use this
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information in your main processor to compensate the additional latency
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involved with the oversampling, for example with a dry / wet mixer, and to
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report the latency to the DAW.
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Note: If you have not opted to use an integer latency then the latency may not be
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integer, so you might need to round its value or to compensate it properly in
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your processing code since plug-ins can only report integer latency values in
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samples to the DAW.
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*/
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SampleType getLatencyInSamples() const noexcept;
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/** Returns the current oversampling factor. */
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size_t getOversamplingFactor() const noexcept;
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//===============================================================================
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/** Must be called before any processing, to set the buffer sizes of the internal
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buffers of the oversampling processing.
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*/
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void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling);
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/** Resets the processing pipeline, ready to oversample a new stream of data. */
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void reset() noexcept;
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/** Must be called to perform the upsampling, prior to any oversampled processing.
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Returns an AudioBlock referencing the oversampled input signal, which must be
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used to perform the non-linear processing which needs the higher sample rate.
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Don't forget to set the sample rate of that processing to N times the original
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sample rate.
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*/
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AudioBlock<SampleType> processSamplesUp (const AudioBlock<const SampleType>& inputBlock) noexcept;
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/** Must be called to perform the downsampling, after the upsampling and the
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non-linear processing. The output signal is probably delayed by the internal
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latency of the whole oversampling behaviour, so don't forget to take this
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into account.
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*/
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void processSamplesDown (AudioBlock<SampleType>& outputBlock) noexcept;
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//===============================================================================
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/** Adds a new oversampling stage to the Oversampling class, multiplying the
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current oversampling factor by two. This is used with the default constructor
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to create custom oversampling chains, requiring a call to the
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clearOversamplingStages before any addition.
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Note: Upsampling and downsampling filtering have different purposes, the
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former removes upsampling artefacts while the latter removes useless frequency
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content created by the oversampled process, so usually the attenuation is
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increased when upsampling compared to downsampling.
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@param normalisedTransitionWidthUp a value between 0 and 0.5 which specifies how much
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the transition between passband and stopband is
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steep, for upsampling filtering (the lower the better)
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@param stopbandAmplitudedBUp the amplitude in dB in the stopband for upsampling
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filtering, must be negative
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@param normalisedTransitionWidthDown a value between 0 and 0.5 which specifies how much
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the transition between passband and stopband is
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steep, for downsampling filtering (the lower the better)
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@param stopbandAmplitudedBDown the amplitude in dB in the stopband for downsampling
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filtering, must be negative
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@see clearOversamplingStages
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*/
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void addOversamplingStage (FilterType,
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float normalisedTransitionWidthUp, float stopbandAmplitudedBUp,
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float normalisedTransitionWidthDown, float stopbandAmplitudedBDown);
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/** Adds a new "dummy" oversampling stage, which does nothing to the signal. Using
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one can be useful if your application features a customisable oversampling factor
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and if you want to select the current one from an OwnedArray without changing
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anything in the processing code.
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@see OwnedArray, clearOversamplingStages, addOversamplingStage
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*/
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void addDummyOversamplingStage();
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/** Removes all the previously registered oversampling stages, so you can add
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your own from scratch.
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@see addOversamplingStage, addDummyOversamplingStage
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*/
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void clearOversamplingStages();
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//===============================================================================
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size_t factorOversampling = 1;
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size_t numChannels = 1;
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#ifndef DOXYGEN
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struct OversamplingStage;
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#endif
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private:
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//===============================================================================
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void updateDelayLine();
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SampleType getUncompensatedLatency() const noexcept;
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//===============================================================================
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OwnedArray<OversamplingStage> stages;
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bool isReady = false, shouldUseIntegerLatency = false;
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DelayLine<SampleType, DelayLineInterpolationTypes::Thiran> delay { 8 };
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SampleType fractionalDelay = 0;
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//===============================================================================
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JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling)
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};
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} // namespace dsp
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} // namespace juce
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