CUGL 2.3
Cornell University Game Library

#include <CUBiquadIIR.h>
Public Types  
enum class  Type : int { UNDEFINED = 0 , LOWPASS = 1 , HIGHPASS = 2 , BANDPASS = 3 , ALLPASS = 4 , NOTCH = 5 , PEAK = 6 , LOWSHELF = 7 , HIGHSHELF = 8 , RESONANCE = 9 } 
Public Member Functions  
BiquadIIR ()  
BiquadIIR (unsigned channels)  
BiquadIIR (unsigned channels, Type type, float frequency, float gainDB, float qVal=INV_SQRT2)  
BiquadIIR (const BiquadIIR ©)  
BiquadIIR (BiquadIIR &&filter)  
~BiquadIIR ()  
unsigned  getChannels () const 
void  setChannels (unsigned channels) 
void  setCoeff (const std::vector< float > &bvals, const std::vector< float > &avals) 
const std::vector< float >  getBCoeff () const 
const std::vector< float >  getACoeff () const 
void  setBCoeff (float b0, float b1, float b2) 
void  setACoeff (float a1, float a2) 
void  setType (Type type, float frequency, float gainDB, float qVal=INV_SQRT2) 
void  step (float gain, float *input, float *output) 
void  calculate (float gain, float *input, float *output, size_t size) 
void  clear () 
size_t  flush (float *output) 
Static Public Member Functions  
static float  db2gain (float gainDB) 
static float  gain2db (float gain) 
static float  bandwidth2q (float width) 
static float  q2Bandwidth (float qVal) 
Static Public Attributes  
static bool  VECTORIZE 
This class implements a biquad digital filter.
This is the most efficient filter acceptable for a parametric equalizer. As such, this filter has several types for quick creation of parameteric components. However, in most settings Butterworth filters are preferred because they have better roll off.
Frequencies are specified in "normalized" format. A normalized frequency is frequency/sample rate. For example, a 7 kHz frequency with a 44100 Hz sample rate has a normalized value 7000/44100 = 0.15873. However, filters are not intended to be model classes, and so it does not save the defining frequency.
This class supports vector optimizations for SSE and Neon 64. In timed simulations, these optimizations provide at least a 34x performance increase (and for 4 or 8 channel audio, much higher). These optimizations make use of the matrix precomputation outlined in "Implementation of Recursive Digital Filters into Vector SIMD DSP Architectures".
https://pdfs.semanticscholar.org/d150/a3f75dc033916f14029cd9101a8ea1d050bb.pdf
The algorithm in this paper performs extremely well in our tests, and even outperforms Apple's Acceleration library. However, our implementation is limited to 128bit words as 256bit (e.g. AVX) and higher show no significant increase in performance.
For performance reasons, this class does not have a (virtualized) subclass relationship with other IIR or FIR filters. However, the signature of the the calculation and coefficient methods has been standardized so that it can support templated polymorphism.
This class is not thread safe. External locking may be required when the filter is shared between multiple threads (such as between an audio thread and the main thread).

strong 
The underlying type of the biquad filter.
Most biquad filters are intended for a parameteric equalizer, and so will have one of the filters types below. If the coefficients of the biquad filter are set directly, it will have type UNDEFINED.
cugl::dsp::BiquadIIR::BiquadIIR  (  ) 
Creates a secondorder passthrough filter for a single channel.
cugl::dsp::BiquadIIR::BiquadIIR  (  unsigned  channels  ) 
Creates a secondorder passthrough filter for the given number of channels.
channels  The number of channels 
cugl::dsp::BiquadIIR::BiquadIIR  (  unsigned  channels, 
Type  type,  
float  frequency,  
float  gainDB,  
float  qVal = INV_SQRT2 

) 
Creates a special purpose filter of the given type
In addition to the type, the filter is defined by the target frequency and the gain for that frequency (which may be negative). This gain will be applied to the target frequency, but will rolloff or attenuate for other frequencies according to the type. The gain is specified in decibels, not as a multiplicative factor.
Frequencies are specified in "normalized" format. A normalized frequency is frequency/sample rate. For example, a 7 kHz frequency with a 44100 Hz sample rate has a normalized value 7000/44100 = 0.15873.
The Q factor is the inverse of the bandwidth, and is generally only relevant for the BANDPASS and BANDSTOP filter types. For the other types, the default value of 1/sqrt(2) is generally sufficient. For BANDPASS and BANDSTOP, the bandwidth2q()
value produces the appropriate value for proper filter design.
Filters are not intended to be model classes, and so it does not save the defining frequency, type, gain, or other values.
If the type is undefined, the frequency and peakGain will be ignored, creating a passthrough filter.
channels  The number of channels to process 
type  The filter type 
frequency  The (normalized) target frequency 
gainDB  The gain at the target frequency in decibels 
qVal  The special Q factor 
cugl::dsp::BiquadIIR::BiquadIIR  (  const BiquadIIR &  copy  ) 
Creates a copy of the biquad filter.
copy  The filter to copy 
cugl::dsp::BiquadIIR::BiquadIIR  (  BiquadIIR &&  filter  ) 
Creates a biquad filter with the resources of the original.
filter  The filter to acquire 
cugl::dsp::BiquadIIR::~BiquadIIR  (  ) 
Destroys the filter, releasing all resources.

static 
Returns the q value for the given the filter bandwidth (in octaves)
The filter bandwidth is generally only relevant for the BANDPASS and BANDSTOP filter types. For the other types, the q value of 1/sqrt(2) is generally sufficient.
width  The filter bandwidth in octaves 
void cugl::dsp::BiquadIIR::calculate  (  float  gain, 
float *  input,  
float *  output,  
size_t  size  
) 
Performs a filter of interleaved input data.
The output is written to the given output array, which should be the same size as the input array. The size is the number of frames, not samples. Hence the arrays must be size times the number of channels in size.
To provide real time processing, the output is delayed by the number of acoefficients. Delayed results are buffered to be used the next time the filter is used (though they may be extracted with the flush
method). The gain parameter is applied at the filter input, but does not affect the filter coefficients.
gain  The input gain factor 
input  The array of input samples 
output  The array to write the sample output 
size  The input size in frames 
void cugl::dsp::BiquadIIR::clear  (  ) 
Clears the filter buffer of any delayed outputs or cached inputs

static 
Returns the gain factor for given value in decibels.
The factor is the amount to multiply the amplitude signal. The provided value is represented in decibels, so there may be some roundoff error in conversion.
gainDB  The audio gain in decibels 
size_t cugl::dsp::BiquadIIR::flush  (  float *  output  ) 
Flushes any delayed outputs to the provided array
The array size should be twice the number of channels. This method will also clear the buffer.

static 
Returns the decibel gain for given factor.
The factor is the amount to multiply the amplitude signal. The returned result is in decibels, so there may be some roundoff error in conversion.
gain  The audio gain factor 
const std::vector< float > cugl::dsp::BiquadIIR::getACoeff  (  )  const 
Returns the lower coefficients for this IIR filter.
This filter implements the standard difference equation:
a[0]*y[n] = b[0]*x[n]+...+b[nb]*x[nnb]a[1]*y[n1]...a[na]*y[nna]
where y is the output and x in the input.
const std::vector< float > cugl::dsp::BiquadIIR::getBCoeff  (  )  const 
Returns the upper coefficients for this IIR filter.
This filter implements the standard difference equation:
a[0]*y[n] = b[0]*x[n]+...+b[nb]*x[nnb]a[1]*y[n1]...a[na]*y[nna]
where y is the output and x in the input.

inline 
Returns the number of channels for this filter
The data buffers depend on the number of channels. Changing this value will reset the data buffers to 0.

static 
Returns the filter bandwidth (in octaves) for the given q value
The filter bandwidth is generally only relevant for the BANDPASS and BANDSTOP filter types. For the other types, the q value of 1/sqrt(2) is generally sufficient.
qVal  The special Q factor 
void cugl::dsp::BiquadIIR::setACoeff  (  float  a1, 
float  a2  
) 
Sets the lower coefficients.
Setting this leaves the upper coefficients unchanged.
a1  The lower firstorder coefficient 
a2  The lower secondorder coefficient 
void cugl::dsp::BiquadIIR::setBCoeff  (  float  b0, 
float  b1,  
float  b2  
) 
Sets the upper coefficients.
Setting this leaves the lower coefficients unchanged.
b0  The upper zeroorder coefficient 
b1  The upper firstorder coefficient 
b2  The upper secondorder coefficient 
void cugl::dsp::BiquadIIR::setChannels  (  unsigned  channels  ) 
Sets the number of channels for this filter
The data buffers depend on the number of channels. Changing this value will reset the data buffers to 0.
channels  The number of channels for this filter 
void cugl::dsp::BiquadIIR::setCoeff  (  const std::vector< float > &  bvals, 
const std::vector< float > &  avals  
) 
Sets the coefficients for this IIR filter.
This filter implements the standard difference equation:
a[0]*y[n] = b[0]*x[n]+...+b[nb]*x[nnb]a[1]*y[n1]...a[na]*y[nna]
where y is the output and x in the input. If a[0] is not equal to 1, the filter coeffcients are normalized by a[0].
All bcoefficients and acoefficients after the third are ignored. If any coefficients are missing, they are replaced with 1 for b[0] and a[0], and 0 otherwise.
bvals  The upper coefficients 
avals  The lower coefficients 
void cugl::dsp::BiquadIIR::setType  (  Type  type, 
float  frequency,  
float  gainDB,  
float  qVal = INV_SQRT2 

) 
Sets this filter to the special purpose one of the given type
In addition to the type, the filter is defined by the target frequency and the gain for that frequency (which may be negative). This gain will be applied to the target frequency, but will rolloff or attenuate for other frequencies according to the type. The gain is specified in decibels, not as a multiplicative factor,
Frequencies are specified in "normalized" format. A normalized frequency is frequency/sample rate. For example, a 7 kHz frequency with a 44100 Hz sample rate has a normalized value 7000/44100 = 0.15873.
The Q factor is the inverse of the bandwidth, and is generally only relevant for the BANDPASS and BANDSTOP filter types. For the other types, the default value of 1/sqrt(2) is generally sufficient. For BANDPASS and BANDSTOP, the bandwidth2q()
value produces the appropriate value for proper filter design.
Filters are not intended to be model classes, and so it does not save the defining frequency, type, gain, or other values.
If the type is undefined, the frequency and peakGain will be ignored, creating a passthrough filter.
type  The filter type 
frequency  The (normalized) target frequency 
gainDB  The gain at the target frequency in decibels 
qVal  The special Q factor 
void cugl::dsp::BiquadIIR::step  (  float  gain, 
float *  input,  
float *  output  
) 
Performs a filter of single frame of data.
The output is written to the given output array, which should be the same size as the input array. The size should be the number of channels.
To provide real time processing, the output is delayed by the number of acoefficients. Delayed results are buffered to be used the next time the filter is used (though they may be extracted with the flush
method). The gain parameter is applied at the filter input, but does not affect the filter coefficients.
gain  The input gain factor 
input  The input frame 
output  The frame to receive the output 

static 
Whether to use a vectorization algorithm (Access not thread safe)