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FreeDV demod: fixed filters

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This commit is contained in:
f4exb
2019-02-27 00:31:45 +01:00
parent 6c3536e0a0
commit 7759f9cebc
5 changed files with 29 additions and 30 deletions
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sdrbase/audio/audiofilter.cpp
+7 -6
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@@ -38,23 +38,23 @@ AudioFilter::~AudioFilter()
{}
void AudioFilter::setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow)
void AudioFilter::setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float fgain)
{
double fcNormHigh = fcHigh / srHigh;
double fcNormLow = fcLow / srLow;
calculate2(false, fcNormHigh, m_lpva, m_lpvb);
calculate2(true, fcNormLow, m_hpva, m_hpvb);
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);
}
void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb)
void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb, float fgain)
{
double a[22], b[22];
cheby(highPass, fc, 0.5, 2, a, b); // low-pass, 0.5% ripple, 2 pole filter
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++) {
@@ -79,7 +79,7 @@ void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb)
* Adapted from BASIC program in table 20-4 of
* https://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
*/
void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, double *b)
void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain)
{
double a0, a1, a2, b1, b2;
double ta[22], tb[22];
@@ -135,6 +135,7 @@ void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, d
}
double gain = sa/(1.0 -sb);
gain /= fgain;
for (int i=0; i<20; i++) {
a[i] /= gain;