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Reverb

Reverb (reverberation) is a super interesting topic.  While the basic physical principle is easy to understand, implementing reverb in digital form is the subject of much ongoing research and development.

In this post, we will be making use of the comb filters that we learned about in previous posts and apply them to create an early type of reverberator, the Schroeder Reverberator.

Time Domain Representation of an Impulse with Reverb Applied
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Comb Filters

The previous post introduced the delay line which, by itself doesn’t create particularly interesting sounds. However, it is one of the fundamental building blocks for many (interesting) audio effects.

In this post, we will be using the delay line to create another family of basic building blocks also often used to create audio effects, the comb filter.

A Comb Filter Example
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Time Domain FIR Filtering

In the previous posts, we went through the basic exercise of reading audio samples through the ADC and outputting them to the DAC without manipulating the samples. This time, we’re going to kick things up a notch and add a filter to our audio processing chain.

Audio Sample with no Filtering

Audio Sample with Low Pass Filtering

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If you like math and haven’t yet seen 3blue1brown’s video on space-filling curves, I suggest you check it out.

At the start of the video, a problem is given: provided an image, map each pixel to an audible frequency. The collection of unique frequencies of each pixel would then be summed, resulting in some waveform.

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Audio Buffers

In the previous post, we created a very basic example that read a sample of audio from some input and routed it directly to output. At a sample rate of 40 kHz, this sequence of events repeated every 25 usec.

This is fine, provided that any processing you do on the sample does not last longer than 25 usec. Apply any heavier processing however and you run the risk of having your audio sound something like this:

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Coding Environment

The last post went into the hardware details for our audio project. In this post, we’re going to cover the coding environment. Aside from the basics, there are some neat features in the microcontroller that we definitely want to use and that’s going to need some extra configuration.

The extra configuration steps sent me on a bit of a wild goose chase throughout the internet so I hope that by compiling most of what I learned into one post, you can get started more quickly.

Here’s what we’re going to cover:

  • Simplicity Studio IDE Introduction
  • Creating a New Project
  • Using EMLIB libraries
  • Enabling FPU and CMSIS DSP Extensions
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Hardware Setup

There are a lot of musically savvy engineers who, after their first signal processing course, use their newfound knowledge to build an effects pedal of sorts. I definitely put myself in this category.

Over the next while, I will be messing around with different audio processing methods like filtering, reverberation, wavetables and whatever I feel like playing around with.

For this post, I am going to go through the hardware setup. The firmware setup will be covered in the next post.

The analog front end schematic and a physical implementation example is provided below.

Analog Front End Schematic
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IMG_3845

Like many others, I have been interested in visiting Japan and experiencing some of the culture. So recently, I decided to finally take a trip there. There is so much to write about Japan. So much so that condensing my two short weeks of visiting the country into one short blog post will be next to impossible. From its populous vending machines to cat cafes and lush green islands to concrete jungles, if a complete sensory overload is what you’re looking for then consider paying a visit to Japan.

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