Dr. AIX's POSTS — 29 May 2014

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Yesterday, I explained that analog to digital conversion using PCM requires the use of a Low Pass Filter to remove any frequency higher than the Nyquist frequency, which is exactly half of the sampling rate. Bad things happen in the audible band if frequencies greater than SR/2 are allowed to get into the system. But you might be surprised that this isn’t always a bad thing.

I studied electronic and computer music during my years of music school at CSUN, Cal Arts and UCLA. In fact, my doctoral dissertation was the first electronic music composition ever granted by UCLA. During the 80’s digital synthesis and sampling were becoming mainstream. There was the introduction of the Yamaha DX-7, which used FM as the basis for its amazing repertoire of sounds. Dr. John Chowning, who has become a close friend over the past few years, was the head of Stanford’s computer music research facility and developed the instrument. To this day, the DX-7 and FM synthesis have generated more license revenue for Stanford university than any other commercialized IP the university has ever invented.

Digital sampling was also a new technology of the time. The idea behind digital samplers is to capture all of the notes (or a representative group) of an instrument and assign them to the equivalent notes on a keyboard. The sampler would reproduce that actual sound of the sampled instrument at the corresponding pitch allowing composers to have an “orchestra” in their own private studios. Today, sampling is one of the most widely used sound producing techniques available.

Roger Linn used this technology to create the Linn Drum, which applied the sampling techniques describe above to a drum kit and percussion instruments. This revolutionized music production. Instead of having to hire a real drummer and mic up an entire drum set, the Linn Drum made it possible to include great drum sounds and rhythmic “loops” that humans couldn’t possible play. For better or worse drum machines and samplers dominate commercial music production.

So how does foldover or aliasing play a role in digital sampling? How could the unwanted frequencies associated with this PCM problem possibly benefit sampling? Imagine the sound of cymbals. They are very rich sources of high frequencies. Musicians refer to these densely packed high tones as “inharmonic” partials or overtones. The term “inharmonic” means that these tones don’t follow the natural overtone series, which are the basic acoustic building blocks of musical tones. Cymbals and other percussion instruments generate sound that is full of noise…random collections of frequencies.

Sounds like a real opportunity to use the “aliases” as a source of random high frequency sounds when trying to enrich the sound of a cymbal sample. Imagine flooding the AD converter while recording a sample with frequencies that are greater then SR/2, the Nyquist frequency. This means leaving off the LPF that we know is typically required.

All of these high frequency tones “foldover” the Nyquist and appear as a dense cloud of high frequency partials in the audio band. Presto, one of the shortcomings of PCM digital audio can be used to produce better samples without additional hardware of memory capacity.

However, it can also be a difficult problem when we want to record accurate highs. If a signal contains lots of ultrasonic noise and that noise is sent to other components that can’t handle that noise (amplifiers and speakers for example) or lacks the ability to roll it off, your ears will hear it as spurious noise. This actually happens. I heard about this very circumstance in the case of a new cell phone and a problem in the Qualcomm chip set that failed to address the ultrasonic noise resulting is terrible audio. Luckily, an audiophile at the company listened to the demo tracks that were derived from source DSD master (converted to PCM) and alerted the QC department before the new device made it to the press and public. It was a close call.

Another vote to avoid DSD, in my opinion. It all comes down to the quality of the content.

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About Author

Dr. AIX

Mark Waldrep, aka Dr. AIX, has been producing and engineering music for over 40 years. He learned electronics as a teenager from his HAM radio father while learning to play the guitar. Mark received the first doctorate in music composition from UCLA in 1986 for a "binaural" electronic music composition. Other advanced degrees include an MS in computer science, an MFA/MA in music, BM in music and a BA in art. As an engineer and producer, Mark has worked on projects for the Rolling Stones, 311, Tool, KISS, Blink 182, Blues Traveler, Britney Spears, the San Francisco Symphony, The Dover Quartet, Willie Nelson, Paul Williams, The Allman Brothers, Bad Company and many more. Dr. Waldrep has been an innovator when it comes to multimedia and music. He created the first enhanced CDs in the 90s, the first DVD-Videos released in the U.S., the first web-connected DVD, the first DVD-Audio title, the first music Blu-ray disc and the first 3D Music Album. Additionally, he launched the first High Definition Music Download site in 2007 called iTrax.com. A frequency speaker at audio events, author of numerous articles, Dr. Waldrep is currently writing a book on the production and reproduction of high-end music called, "High-End Audio: A Practical Guide to Production and Playback". The book should be completed in the fall of 2013.

(9) Readers Comments

  1. Sounds interesting. Have you tried this on cymbals or something else inharmonic? I’d be curious to hear the results.

    • I can’t say that I have…it’s hard to get equipment to behave this way these days.

  2. This was a great post. Always interesting when “accidental” techniques present some interesting/beneficial outcomes. For instance, the way overdriving the inputs to a mixer, thus producing richer harmonics at the output of a VCA, can be thought of as pleasant sounding. I wish CSU Dominguez Hills had a computer music and electronic music degree program, but I’m glad that the Audio program includes recording and advanced music synthesis classes, etc.

  3. DSD was never meant for the quality of content. DSD itself was created as a cheaper alternative to PCM (Yes, supposedly, DSD uses less components on signal path thus cheaper). But as technology advanced and the chips are now several cents, this necessity has been eliminated.

  4. There is a famous, now retired, electronics engineer named Dick Burwen. He sells a Windows Media Player plug in called Burwen Bobcat that adds high frequency harmonics to the playback. (I do not own his products or suggest that you own his products) I wonder whether what he is doing is analogous to what Mark described with aliasing in the above post? When I listen to a demonstration of the effect I here a more diffuse sound. I think the sound would alleviate listener fatigue and reduce my listening induced tinnitus. Happier ears is the claim Burwen is making for his plug in…obviously I’m believing him, but I would like to hear what others think. It would be interesting to run a spectrogram on the processed and unprocessed clips to see how much Burwen’s plug in adds in high harmonics. You can listen for yourself on this page (might be limited to Windows users): http://www.burwenbobcat.com/BBTR_DEMO.html

    My hypothesis is that live music naturally contains many of the harmonics artificially created by Burwen’s plug in. When Mark & Co. conduct experiments regarding the effect of ultrasonics on listeners, I think they will discover the response of listener brains to music reproduced with ultrasonics is more similar brain response to live music than listener brain response music reproduced without ultrasonics. I’m not saying I am right, but I am saying I would like science to prove or refute that hypothesis.

    • I will look into this.

    • I remember the Burwen Bobcat! Amazingly ‘thick’ sounding. Didn’t really understand it.

    • I know only Gary Langan

  5. Thick, as Grant described it, is what I heard from the Bobcat. My use of diffuse to describe it isn’t quite right.

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