Resampling a sound sample, what filter do I use?

Question

I am trying to resample a signal (sound sample) from one sampling rate, to a higher sampling rate. Unfortunately it needs some kind of filter, as some 'aliasing' appears to occur, and I'm not familiar with filters. Here is what I came up with:

int i, j, a, b, z;

a = 44100;
b = 8363;

// upsample by a
for(i = z = 0; i < samplen; i++)
    for(j = 0; j < a; j++)
        cbuf[z++] = sampdata[i];

// some filter goes here???

// downsample by b
for(j = i = 0; i < z; i += b)
    buf[j++] = cbuf[i];

The new sample is very similar to the original, but it has some kind of noise. Can you please tell me what filter I need to add, and preferably some code related to that filter?

Original sound: http://www.mediafire.com/?9gnga1in52d6t4x Resampled sound: http://www.mediafire.com/?x34h7ggk8n9k8z1

Answer 1

Don't use linear interpolation unless both sample rates (source and destination) are well above the highest frequency in your data. It's a very poor low-pass filter.

What you want is an interpolating low pass filter with a stop-band starting below half the lower of the two sample rates you are dealing with. Common methods of implementing this are upsampling/downsampling using IIR filters, and using poly-phase FIR filters. A windowed Sinc interpolator also works well for this if you don't need real-time performance, and don't want to upsample/downsample. Here's a Windowed Sinc interpolating low-pass filter in Basic, that should be trivial to convert into C.

If you want to use IIR filtering, here's the canonical Cookbook for biquad IIR filters.

If you want the best explanation of audio resampling theory, here's Stanford CCRMA's Resampling page.