Blogs

This is the second post in a series of articles about deconvolving a reverberated sound to get the impulse response of natural reverberations and then using this impulse response as the artificial reverb. In the previous post we created the impulse response of an example reverb. In this post we will take a short sound – a drum hit – and we will reverberate it with the impulse response developed before. Later, we will take the reverberated drum and we will pretend that we do not know the impulse response. We will try to recreate the impulse response and see how well we do. As a reminder, it should be possible to record the natural reverb of a short sound, such as a drum hit, in a room or a hall and to deconvolve this natural reverb to create an artificial reverb.

A short drum sound may look something like this.

This should be an article about deconvolving an actual recorded reverberated sound to produce the impulse of a reverb, but we do not have all the needed tools to do so yet. To get there, we will prepare several articles. Once we are ready – and if we succeed – we should have a reasonably good imitation of the actual reverb and should be able to use the resulting impulse to simulate the actual reverb on any recorded sound.

I received, several weeks ago by now, a copy of a book by Mike Major called Recording Drums. The Complete Guide. Now that I have had the time to flip through the book, I'd say it is definitely worth having.

This post is here to present an illustrative and intuitive example of compressing drums. It includes a number of graphs that show the effect of a compressor on a drum snare. There are a number of explanations of how compressors work on the net (including on this site), but there are virtually no good intuitive examples of what compressors actually do with audio data.

Long ago I had picked up "All Blues for Jazz Guitar – comping styles, chords, and grooves" by Jim Ferguson. This book is about putting some jazz in your blues (or some blues in your jazz?). When I considered myself an "intermediate" guitar player and had the opportunity to play in jam sessions with musicians, who were well ahead of me, this book gave me plenty of good pointers. It allowed me to understand how music can become less rigid and without boundaries, how rhythm and melody can play off of each other, and how music theory can become practice.

I was prompted the other day to write a review of SongDasher – a piece of software for the iPhope or iPad from Far Out Notions, LLC. What I got in an e-mail was: "SongDasher is a streamlined song crafting tool with six tracks of audio, programmable drum beats, and flexible section-by-section composition. SongDasher allows musicians to quickly and easily capture musical ideas, and share their compositions with friends and bandmates through a variety of social media and cloud services". And so it does.

We have been collecting some posts DSP and some DSP related wiki topics. Although we have a lot, we have been putting everything together a bit haphazardly. Anton Kamenov was nice enough to organize a lot of the DSP information on this website into a book: Digital Signal Procession for Audio Applications. The book is a very solid foundation for anyone who wants to know more about how DSP in audio comes about. It is well organized and derives a lot of the DSP operations that we take for granted in a simple and transparent manner. It also adds a lot of useful information – some that we have already started adding to this site and some that is still not up.

Here are some topics of interest in the book.

Filters and other DSP operations for audio are discussed at length on this site. There are various DSP operations and the discussion below applies to all, although when engineers talk about DSP errors, they usually discuss filters – finite impulse response filters, infinite impulse response filters, etc.

I use the words "simple delay" to describe the simplest of practical recording effects – an effect that produces one repetition of the input signal with some delay in time and usually with some decay in amplitude. More complex delays can have additional parameters: feedback, delay and decay sweeps, multiple "taps" to output, etc. Even simple delays, however, can be very interesting and the following are example settings that can produce great results.

We were working recently on version 3 of Orinj. We wanted to improve the graphs of several of the DSP effects in Orinj to include the actual magnitude response of the effect. This included effects that have graphs with some equalization or some frequency type filters – the graphic and parametric equalizer, the reverb (because of its parametric equalizer), and the notch filter. This post is about computing the actual magnitude response quickly and efficiently.