AudioFilter Class Reference

#include <audiofilter.h>

Collaboration diagram for AudioFilter:

Public Member Functions
	AudioFilter ()
	~AudioFilter ()
void	useHP (bool useHP)
bool	usesHP () const
void	setDecimFilters (int srHigh, int srLow, float fcHigh, float fcLow, float gain=1.0f)
float	run (const float &sample)
float	runHP (const float &sample)
float	runLP (const float &sample)

Private Member Functions
void	calculate2 (bool highPass, double fc, float *a, float *b, float fgain)
void	cheby (bool highPass, double fc, float pr, int np, double *a, double *b, float fgain)
void	cheby_sub (bool highPass, double fc, float pr, int np, int stage, double &a0, double &a1, double &a2, double &b1, double &b2)

Private Attributes
IIRFilter< float, 2 >	m_filterLP
IIRFilter< float, 2 >	m_filterHP
bool	m_useHP
float	m_lpva [3]
float	m_lpvb [3]
float	m_hpva [3]
float	m_hpvb [3]

Static Private Attributes
static const float	m_lpa [3] = {1.0, 1.392667E+00, -5.474446E-01}
static const float	m_lpb [3] = {3.869430E-02, 7.738860E-02, 3.869430E-02}
static const float	m_hpa [3] = {1.000000e+00, 1.667871e+00, -7.156964e-01}
static const float	m_hpb [3] = {8.459039e-01, -1.691760e+00, 8.459039e-01}

Detailed Description

By default this is a 2 pole lowpass Chebyshev (recursive) filter at fc=0.075 using coefficients found in table 20-1 of http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf

At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice

a0= 3.869430E-02 a1= 7.738860E-02 b1= 1.392667E+00 a2= 3.869430E-02 b2= -5.474446E-01

given x[n] is the new input sample and y[n] the returned output sample:

y[n] = a0*x[n] + a1*x[n] + a2*x[n] + b1*y[n-1] + b2*y[n-2]

This one works directly with floats

It can be generalized using the program found in tables 20-4 and 20-5 of the same document. This form is used as a decimation filter and can be set with the setDecimFilters method

Definition at line 44 of file audiofilter.h.

Constructor & Destructor Documentation

AudioFilter()

AudioFilter::AudioFilter

(

)

Definition at line 32 of file audiofilter.cpp.

                         :
        m_filterLP(m_lpa, m_lpb),
        m_filterHP(m_hpa, m_hpb),
        m_useHP(false)
{}

~AudioFilter()

AudioFilter::~AudioFilter

(

)

Definition at line 38 of file audiofilter.cpp.

{}

Member Function Documentation

calculate2()

void AudioFilter::calculate2 ( bool  highPass, double  fc, float *  a, float *  b, float  fgain  )

private

Definition at line 54 of file audiofilter.cpp.

References cheby(), and i.

Referenced by setDecimFilters().

{
    double a[22], b[22];

    cheby(highPass, fc, 0.5, 2, a, b, fgain); // low-pass, 0.5% ripple, 2 pole filter

    // Copy to the 2-pole filter coefficients
    for (int i=0; i<3; i++) {
        vb[i] = a[i];
        va[i] = b[i];
    }

    va[0] = 1.0;

    qDebug() << "AudioFilter::calculate2:"
        << " highPass: " << highPass
        << " fc: " << fc
        << " a0: " << va[0]
        << " a1: " << va[1]
        << " a2: " << va[2]
        << " b0: " << vb[0]
        << " b1: " << vb[1]
        << " b2: " << vb[2];
}

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cheby()

void AudioFilter::cheby ( bool  highPass, double  fc, float  pr, int  np, double *  a, double *  b, float  fgain  )

private

Definition at line 83 of file audiofilter.cpp.

References b1, b2, cheby_sub(), and i.

Referenced by calculate2().

{
    double a0, a1, a2, b1, b2;
    double ta[22], tb[22];

    std::fill(a, a+22, 0.0);
    std::fill(b, b+22, 0.0);
    a[2] = 1.0;
    b[2] = 1.0;

    for (int p = 1; p <= np/2; p++)
    {
        cheby_sub(highPass, fc, pr, np, p, a0, a1, a2, b1, b2);

        // Add coefficients to the cascade
        for (int i=0; i<22; i++)
        {
            ta[i] = a[i];
            tb[i] = b[i];
        }

        for (int i=2; i<22; i++)
        {
            a[i] = a0*ta[i] + a1*ta[i-1] + a2*ta[i-2];
            b[i] = tb[i] - b1*tb[i-1] - b2*tb[i-2];
        }
    }

    // Finish combining coefficients
    b[2] = 0;

    for (int i=0; i<20; i++)
    {
        a[i] = a[i+2];
        b[i] = -b[i+2];
    }

    // Normalize the gain
    double sa = 0.0;
    double sb = 0.0;

    for (int i=0; i<20; i++)
    {
        if (highPass)
        {
            sa += i%2 == 0 ? a[i] : -a[i];
            sb += i%2 == 0 ? b[i] : -b[i];
        }
        else
        {
            sa += a[i];
            sb += b[i];
        }
    }

    double gain = sa/(1.0 -sb);
    gain /= fgain;

    for (int i=0; i<20; i++) {
        a[i] /= gain;
    }
}

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cheby_sub()

void AudioFilter::cheby_sub ( bool  highPass, double  fc, float  pr, int  np, int  stage, double &  a0, double &  a1, double &  a2, double &  b1, double &  b2  )

private

Definition at line 150 of file audiofilter.cpp.

References b1, cos(), exp(), log(), M_PI, sin(), sqrt(), and tan().

Referenced by cheby().

{
    double rp = -cos((M_PI/(np*2)) + (stage-1)*(M_PI/np));
    double ip = sin((M_PI/(np*2)) + (stage-1)*(M_PI/np));

    // Warp from a circle to an ellipse
    double esx = 100.0 / (100.0 - pr);
    double es = sqrt(esx*esx -1.0);
    double vx = (1.0/np) * log((1.0/es) + sqrt((1.0/(es*es)) + 1.0));
    double kx = (1.0/np) * log((1.0/es) + sqrt((1.0/(es*es)) - 1.0));
    kx = (exp(kx) + exp(-kx))/2.0;
    rp = rp * ((exp(vx) - exp(-vx))/2.0) / kx;
    ip = ip * ((exp(vx) + exp(-vx))/2.0) / kx;

    double t = 2.0 * tan(0.5);
    double w = 2.0 * M_PI * fc;
    double m = rp*rp + ip*ip;
    double d = 4.0 - 4.0*rp*t + m*t*t;
    double x0 = (t*t)/d;
    double x1 = (2.0*t*t)/d;
    double x2 = (t*t)/d;
    double y1 = (8.0 - 2.0*m*t*t)/d;
    double y2 = (-4.0 - 4.0*rp*t - m*t*t)/d;
    double k;

    if (highPass) {
        k = -cos(w/2.0 + 0.5) / cos(w/2.0 - 0.5);
    } else {
        k = sin(0.5 - w/2.0) / sin(0.5 + w/2.0);
    }

    d = 1.0 + y1*k - y2*k*k;

    a0 = (x0 - x1*k + x2*k*k)/d;
    a1 = (-2.0*x0*k + x1 + x1*k*k - 2.0*x2*k)/d;
    a2 = (x0*k*k - x1*k + x2)/d;
    b1 = (2.0*k + y1 + y1*k*k - 2.0*y2*k)/d;
    b2 = (-(k*k) - y1*k + y2)/d;

    if (highPass)
    {
        a1 = -a1;
        b1 = -b1;
    }
}

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run()

float AudioFilter::run

(

const float &

sample

)

Definition at line 196 of file audiofilter.cpp.

References m_filterHP, m_filterLP, m_useHP, and IIRFilter< Type, Order >::run().

Referenced by AudioResampler::downSample(), AudioResampler::upSample(), and AudioNetSink::write().

{
    return m_useHP ? m_filterLP.run(m_filterHP.run(sample)) : m_filterLP.run(sample);
}

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runHP()

float AudioFilter::runHP

(

const float &

sample

)

Definition at line 201 of file audiofilter.cpp.

References m_filterHP, and IIRFilter< Type, Order >::run().

{
    return m_filterHP.run(sample);
}

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runLP()

float AudioFilter::runLP

(

const float &

sample

)

Definition at line 206 of file audiofilter.cpp.

References m_filterLP, and IIRFilter< Type, Order >::run().

Referenced by AudioNetSink::write().

{
    return m_filterLP.run(sample);
}

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setDecimFilters()

void AudioFilter::setDecimFilters	(	int	srHigh,
		int	srLow,
		float	fcHigh,
		float	fcLow,
		float	gain = 1.0f
	)

Definition at line 42 of file audiofilter.cpp.

References calculate2(), m_filterHP, m_filterLP, m_hpva, m_hpvb, m_lpva, m_lpvb, and IIRFilter< Type, Order >::setCoeffs().

Referenced by AudioResampler::setAudioFilters(), and AudioNetSink::setDecimationFilters().

{
    double fcNormHigh = fcHigh / srHigh;
    double fcNormLow = fcLow / srLow;

    calculate2(false, fcNormHigh, m_lpva, m_lpvb, fgain);
    calculate2(true, fcNormLow, m_hpva, m_hpvb, fgain);

    m_filterLP.setCoeffs(m_lpva, m_lpvb);
    m_filterHP.setCoeffs(m_hpva, m_hpvb);
}

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useHP()

void AudioFilter::useHP ( bool  useHP )

inline

Definition at line 49 of file audiofilter.h.

References useHP().

Referenced by useHP().

{ m_useHP = useHP; }

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usesHP()

bool AudioFilter::usesHP ( ) const

inline

Definition at line 50 of file audiofilter.h.

References b1, and b2.

{ return m_useHP; }

Member Data Documentation

m_filterHP

IIRFilter<float, 2> AudioFilter::m_filterHP

private

Definition at line 63 of file audiofilter.h.

Referenced by run(), runHP(), and setDecimFilters().

m_filterLP

IIRFilter<float, 2> AudioFilter::m_filterLP

private

Definition at line 62 of file audiofilter.h.

Referenced by run(), runLP(), and setDecimFilters().

m_hpa

const float AudioFilter::m_hpa = {1.000000e+00, 1.667871e+00, -7.156964e-01}

staticprivate

Definition at line 71 of file audiofilter.h.

m_hpb

const float AudioFilter::m_hpb = {8.459039e-01, -1.691760e+00, 8.459039e-01}

staticprivate

Definition at line 72 of file audiofilter.h.

m_hpva

float AudioFilter::m_hpva[3]

private

Definition at line 67 of file audiofilter.h.

Referenced by setDecimFilters().

m_hpvb

float AudioFilter::m_hpvb[3]

private

Definition at line 68 of file audiofilter.h.

Referenced by setDecimFilters().

m_lpa

const float AudioFilter::m_lpa = {1.0, 1.392667E+00, -5.474446E-01}

staticprivate

Definition at line 69 of file audiofilter.h.

m_lpb

const float AudioFilter::m_lpb = {3.869430E-02, 7.738860E-02, 3.869430E-02}

staticprivate

Definition at line 70 of file audiofilter.h.

m_lpva

float AudioFilter::m_lpva[3]

private

Definition at line 65 of file audiofilter.h.

Referenced by setDecimFilters().

m_lpvb

float AudioFilter::m_lpvb[3]

private

Definition at line 66 of file audiofilter.h.

Referenced by setDecimFilters().

m_useHP

bool AudioFilter::m_useHP

private

Definition at line 64 of file audiofilter.h.

Referenced by run().

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The documentation for this class was generated from the following files:

• sdrbase/audio/audiofilter.h
• sdrbase/audio/audiofilter.cpp