bandpass FIR filter

I need to make a simple bandpass audio filter. Now I've used this simple C++ class: http://www.cardinalpeak.com/blog/ac-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?

---

I found a good free implementation of FIR filter: http://www.iowahills.com/A7ExampleCodePage.html

...This Windowed FIR Filter C Code has two parts, the first is the calculation of the impulse response for a rectangular window (low pass, high pass, band pass, or notch). Then a window (Kaiser, Hanning, etc) is applied to the impulse response. There are several windows to choose from...

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y[i] = waveform[i] × (0.42659071 – 0.49656062cos(w) + 0.07684867cos(2w))

where w = (2)i/n and n is the number of elements in the waveform

Try this I got the code from: http://zone.ni.com/reference/en-XX/help/370592P-01/digitizers/blackman_window/

I hope this helps.

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