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JUCE/modules/juce_dsp/processors/juce_Oversampling.h

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/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2017 - ROLI Ltd.
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 5 End-User License
Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
27th April 2017).
End User License Agreement: www.juce.com/juce-5-licence
Privacy Policy: www.juce.com/juce-5-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 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 type the type of filter design employed for filtering
during oversampling
@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