Category Archives: Filters

A one-pole filter

Posted on December 15, 2012 by Nigel Redmon

Here’s a very simple workhorse of DSP applications—the one-pole filter. By comparison, biquads implement two zeros and two poles. You can see that our one-pole simply discards the zeros (the feed-forward delay paths) and the second pole (feedback path): We … Continue reading →

Posted in DC Blocker, Digital Audio, IIR Filters | 6 Comments

Biquad C++ source code

Posted on November 26, 2012 by Nigel Redmon

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 | 10 Comments

Convolution—in words

Posted on March 5, 2012 by Nigel Redmon

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 →

Posted in Convolution, Digital Audio, FIR Filters, Impulse Response, Reverb | Leave a comment

Biquad formulas

Posted on January 2, 2011 by Nigel Redmon

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 | 26 Comments

A biquad calculator

Posted on December 20, 2010 by Nigel Redmon

Something useful: a biquad filter coefficient calculator… Continue reading →

Posted in Biquads, Digital Audio, Filters, IIR Filters | 11 Comments

Sample rate conversion: down

Posted on December 20, 2010 by Nigel Redmon

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 →

Posted in Aliasing, Digital Audio, Filters, Sample Rate Conversion | Leave a comment

A closer look at upsampling filters

Posted on December 11, 2010 by Nigel Redmon

Interpolation type: None Zero-order hold Linear Sinc 1 Sinc 2 Sinc 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 … Continue reading →

Posted in Aliasing, Digital Audio, Filters, FIR Filters, Impulse Response, Sample Rate Conversion | Leave a comment

Sample rate conversion: up

Posted on December 5, 2010 by Nigel Redmon

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 →

Posted in Aliasing, Convolution, Digital Audio, Filters, FIR Filters, Sample Rate Conversion | 4 Comments

Building a windowed sinc filter

Posted on December 5, 2010 by Nigel Redmon

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 →

Posted in Digital Audio, Filters, FIR Filters, Impulse Response, Sample Rate Conversion | 2 Comments

Towards practical resampling

Posted on November 23, 2010 by Nigel Redmon

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 →

Posted in Digital Audio, Filters, FIR Filters, Impulse Response, Sample Rate Conversion | 4 Comments