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The Modern Audio Analyzer

Mar 12, 2014 10:52 AM, By Bob McCarthy | Posted by Jessaca Gutierrez

Understanding dual-channel FFT audio analyzers

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Figure 1: The limitation of a single time window on the FFT analyzer. Frequency resolution is linear no matter what the time record length is, or how it is displayed. Click here to see a larger image.

It wasn’t long ago that the tuning of sound systems was seen almost exclusively in artistic terms. Engineers would play a tape or talk into a mic and walk around and make the important decisions about how the system was aimed, relative levels, crossover points, timing, equalization, and more. Systems were admittedly not as complex as now. Or were they?

It’s true that most systems had fewer channels, fewer subsystems, and a lot less signal processing than we see now, but there were other ways in which systems were much more complex. The signal path nowadays is often a one-stop flight from mix console to full-range powered speakers with a layover at the signal processor. In the old days, it was like going go coast to coast on Southwest. Console to equalizer to delay line to crossover to limiter to amplifier, and finally to the speaker. And that’s just one path to one high driver! It is the simplification and standardization of modern signal processing and speakers that make it easy for artistic sound designers to immerse themselves in soundscapes of ever-greater complexity. If the components of these super complex systems could not be optimized to work together in a predictable way, few designers would risk their shows on wayward subsystems that have creative ideas of their own. The modern audio analyzer is the tool of choice for reversing entropy in the sound system.

Analyzers have been out there for a long time, of course, but in the past they were crude tools, capable of only limited duties. The sound systems themselves were crude as well, and in many ways they have grown up together. The sound system moved out of the garage and the analyzer out of the laboratory in the 1980s and they have been on the road together ever since. An analyzer used to weigh 100lbs. and cost more than a mix console. Now they weigh nothing and cost less than a microphone. Everybody has one, but there are lots of us that don’t really know what’s going on inside of these tools and what the data really means. Our chances of making good tuning decisions increase greatly if we understand the perspective of the modern audio analyzer. Here is a look under the hood. Though we won’t dig deep enough to ready you to design your own analyzer, the goal is to make you a better driver.

Meet the Fourier Family

The modern analyzer is a digital audio device commonly known in our trade as the “Fast-Fourier-Transform” (FFT) analyzer. First let’s reverse engineer the analyzer’s name. “Transform” refers to the conversion of a sampled waveform from a timing sequence of amplitude values to a series of amplitude and phase values over frequency. A continuous stream of music goes in and is separated into slices by frequency. The simple expression for this is a transform from the time domain to the frequency domain.

“Fourier” refers to the Fourier Theorem. It is the math behind the transform, named after its originator, Jean Baptiste Joseph Fourier, in the 1800s. The Fourier Theorem goes both directions: time domain to frequency domain and back again. No waveforms were injured in the making of this transform.

The “Fast” part is where things get trickier. The Fourier Theorem is an equation that requires infinite iterations. We have a show at 8 p.m. so we can’t wait that long. The “fast” in this case means we reduce the computation to less than infinity. While this means there are some errors creeping in, they will be small enough to suit our application. It also means we can program the algorithm into our computer. Computers hate infinity!

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