Category Archives: Digital Audio

A wavetable oscillator—Part 3

In Part 2, we looked at how to use subtables designed to cover one octave each, removing upper harmonics for each higher octave to avoid aliasing. Here’s a view of the nine wavetables for our sawtooth oscillator, with the starting … Continue reading

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A wavetable oscillator—Part 2

From Part 1, we have an oscillator. But we need to broaden it to allow scaling of harmonic content based on pitch so that we have all the harmonic content we need at the low frequency end, and, as we … Continue reading

Posted in Aliasing, Digital Audio, Oscillators | 5 Comments

A wavetable oscillator—Part 1

There are many ways to make an oscillator. Without looking for further motivation, I’ll propose a wavetable oscillator. Wavetables are a fairly obvious extension of the general playback of digital audio. Such oscillators are easy to understand, and their extreme … Continue reading

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A wavetable oscillator—Introduction

Years ago, Cristoph Kemper told me how the Access Virus came to be. He had coded a filter on a DSP and wanted to test it. Of course he needed an oscillator, so he coded that…pretty soon he had a … Continue reading

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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

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Biquad formulas

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 Digital Audio, Filters, IIR Filters | 13 Comments

A biquad calculator

Something useful: a biquad filter coefficient calculator… Continue reading

Posted in Digital Audio, Filters, IIR Filters | 7 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

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A closer look at upsampling filters

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

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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

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