I need to make a simple bandpass audio filter. Now I've used this simple C++ class: http://www.cardinalpeak.com/blog/a-c-class-to-implement-low-pass-high-pass-and-band-pass-filters
It works well and cut off the desired bands. But when I try to change upper or lower limit with small steps, on some values of limit I hear the wrong result - attenuated or shifted in frequency (not corresponding to current limits) sound.
Function for calculating impulse response:
void Filter::designBPF()
{
int n;
float mm;
for(n = 0; n < m_num_taps; n++){
mm = n - (m_num_taps - 1.0) / 2.0;
if( mm == 0.0 ) m_taps[n] = (m_phi - m_lambda) / M_PI;
else m_taps[n] = ( sin( mm * m_phi ) -
sin( mm * m_lambda ) ) / (mm * M_PI);
}
return;
}
where
m_lambda = M_PI * Fl / (Fs/2);
m_phi = M_PI * Fu / (Fs/2);
Fs - sample rate (44.100) Fl - lower limit Fu - upper limit
And simple filtering function:
float Filter::do_sample(float data_sample)
{
int i;
float result;
if( m_error_flag != 0 ) return(0);
for(i = m_num_taps - 1; i >= 1; i--){
m_sr[i] = m_sr[i-1];
}
m_sr[0] = data_sample;
result = 0;
for(i = 0; i < m_num_taps; i++) result += m_sr[i] * m_taps[i];
return result;
}
Do I need to use any window function (Blackman, etc.)? If yes, how do I do this? I have tried to multiply my impulse response to Blackman window:
m_taps[n] *= 0.42 - 0.5 * cos(2.0 * M_PI * n / double(N - 1)) +
0.08 * cos(4.0 * M_PI * n / double(N - 1));
but the result was wrong. And do I need to normalize taps?