JUCE/modules/juce_dsp/processors/juce_Oversampling.h

/*
  ==============================================================================

   This file is part of the JUCE library.
   Copyright (c) 2017 - ROLI Ltd.

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   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
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  ==============================================================================
*/

namespace juce
{
namespace dsp
{

//===============================================================================
/**
    A processing class performing multi-channel oversampling.

    It can be configured to do 2 times, 4 times, 8 times or 16 times oversampling
    using a multi-stage approach, either polyphase allpass IIR filters or FIR
    filters for the filtering, and reports successfully the latency added by the
    filter stages.

    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 N times the input signal, processing the upsampled signal
    with the increased internal sample rate, and downsampling the result to get
    back the original processing sample rate.

    Choose between FIR or IIR filtering depending on your needs in term 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 <typename SampleType>
class JUCE_API  Oversampling
{
public:
    /** The type of filter that can be used for the oversampling processing. */
    enum FilterType
    {
        filterHalfBandFIREquiripple = 0,
        filterHalfBandPolyphaseIIR,
        numFilterTypes
    };

    //===============================================================================
    /**
        Constructor of the oversampling class. All the processing parameters must be
        provided at the creation of the oversampling object.

        @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,
                                the filters will be more efficient, but the CPU load will
                                increase as well
    */
    Oversampling (size_t numChannels,
                  size_t factor,
                  FilterType type,
                  bool isMaxQuality = true);

    /** The default constructor of the oversampling class, which can be used to create an
        empty object and then add the appropriate stages.

        Note: This creates a "dummy" oversampling stage, which needs to be removed first
        before adding proper oversampling stages.

        @see clearOversamplingStages, addOversamplingStage
    */
    explicit Oversampling (size_t numChannels = 1);

    /** Destructor. */
    ~Oversampling();

    //===============================================================================
    /** Returns the latency in samples of the whole processing. Use this information
        in your main processor to compensate the additional latency involved with
        the oversampling, for example with a dry / wet functionality, and to report
        the latency to the DAW.

        Note: The latency might not be integer, so you might need to round its value
        or to compensate it properly in your processing code.
    */
    SampleType getLatencyInSamples() noexcept;

    /** Returns the current oversampling factor. */
    size_t getOversamplingFactor() 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.
    */
    dsp::AudioBlock<SampleType> processSamplesUp (const dsp::AudioBlock<SampleType>& 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 (dsp::AudioBlock<SampleType>& 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:
    //===============================================================================
    OwnedArray<OversamplingStage> stages;
    bool isReady = false;

    //===============================================================================
    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling)
};

} // namespace dsp
} // namespace juce