# Category Archives: Filters

I don’t want to get into the business of teaching people how to code—there are a huge number of free resources available on the internet to that do that. But I’ll give a small taste for those trying to get … Continue reading

Posted in Biquads, Digital Audio, Filters, IIR Filters, Source Code | 64 Comments

## Convolution—in words

Convolution is a convoluted topic—and that’s what it means (convoluted, from Merriam-Webster : “Extremely complex and difficult to follow. Intricately folded, twisted, or coiled.”). Really, it’s more difficult to explain why you would want to use convolution than it is … Continue reading

For fixed filters, we can plug biquad coefficients into our programs. But often, we need to calculate them on the fly, to user settings or changes in sample rate. As a companion to the biquad calculator, here are the formulas … Continue reading

Posted in Biquads, Digital Audio, Filters, IIR Filters, Widgets | 36 Comments

## Sample rate conversion: down

In doubling the sample rate, we inserted zeros between existing samples, then used a lowpass filter to remove the resulting alias in the audio band. To resample at half the current rate, we use a lowpass filter to remove audio … Continue reading

## A closer look at upsampling filters

Interpolation type:NoneZero-order holdLinearSinc 1Sinc 2Sinc 3  Show impulse response Sweep! In this demonstration, we generate a sine wave sweep from low in the audio band to near the Nyquist Frequency, which is half the sample rate. You can view it … Continue reading

## Sample rate conversion: up

Once we have a suitable set of FIR filter coefficients from our windowed sinc calculator, it’s time to apply them. Again, our recipe for doubling the sample rate: 1) Insert a zero between existing samples. (This is the upsampling step, … Continue reading

## Building a windowed sinc filter

As promised, here’s our windowed sinc calculator for building a 2x oversampling filter:  Factor  Length  Rejection  Gain Notes: Use the Tab or Enter keys to effect changes (most browsers), or press Calculate. The frequency axis is in multiples of the … Continue reading

## Towards practical resampling

In a previous article, we looked at sample rate conversion in the frequency domain. Let’s take a quick second look in the time domain as reinforcement of principles behind sample rate conversion, before developing a practical rate convertor. In an … Continue reading

## The bilinear z transform

The bilinear transform is the most popular method of converting analog filter prototypes in the s domain to the z domain so we can implement them as digital filters. The reason we are interested in these s domain filters is … Continue reading

Posted in Digital Audio, Filters, IIR Filters | 23 Comments