Fig. 1
I’m going to start with our old beloved friend, the VU (volume unit) meter. Officially introduced in 1939, the purpose of the VU meter was to provide a standardized way of measuring and representing audio signal levels, and it immediately gave audio engineers and producers an increased ability to make broadcasts and recordings with substantially more consistent loudness levels.
Once it was standardized by ASA (later, ANSI—American National Standards Institute), the VU meter became a cheap and indispensable tool for measuring how much signal was being sent to broadcast transmitters. Essentially, the VU meter works by having a calibrated response time (ballistic) of 300 ms and reflects an inferred average of any peak that occurred within that time window. While it reflects speech in an intuitive way, it falls short with accurately registering very fast transients (like drum hits, claps, fast staccato attacks, etc.). Thus, every decibel of change is not accurately reflected by the meter but averaged (Fig. 1). It wasn’t long until engineers realized that short bursts of +3 VU on analog gear wouldn’t trigger distortion or affect the overall perception of loudness.
In fact, the VU meter was really designed to help the engineer get their audio signals to hover around 0 VU (which equals to +4 dBu, or 1.228V RMS) and what most of us call “0 dB.” This is further reinforced by the design of the meter’s scale range since standardized VU meters range from -20 VU to +3 VU (23 VU entries, in all). However, the majority of the meter’s real estate deals with the last six entries at the top of the scale (-3 VU to +3 VU).
“It immediately gave audio engineers and producers an increased ability to make broadcasts and recordings with substantially more consistent loudness levels.”
Decibels, Decibels, Everywhere!
Let me define the decibel—it is one 10th of a “bel,” named after Alexander Graham Bell (which is why the “B” in “dB” is capitalized). But what’s a bel? It is just a logarithmic unit of measurement relative to something else. We don’t actually hear decibels; we measure them because they approximate the human ear’s logarithmic perception of amplitude (aka loudness). This is known as dBA.
So this is what we mean when we’re talking about how “loud” or “noisy” something is, and also what audiologists use when measuring how well we can hear. What?!
Here’s some basic values for you:
20 dBA: whisper
40 dBA: home computer fan
60 dBA: normal conversation
80 dBA: inside an airplane
90 dBA: lawnmower, hair dryer, blender
95 dBA: prolonged exposure can cause slight hearing loss and tinnitus
100 dBA: motorcycle, construction site, normal stereo at max volume
110 dBA: rock concert, jackhammer
125 dBA: pain threshold; prolonged exposure will cause hearing loss
135 dBA: air raid siren
140 dBA: pain threshold; permanent hearing damage possible
150 dBA: handgun
180 dBA: possible death, rocket launch
But wait, isn’t this also known as dBSPL? I wish! They’re so close but so far away. You see, sound pressure relates to the variations in atmospheric pressure caused by the sound, and SPL (sound pressure level) is the pressure level of that sound measured in decibels. The crucial difference is that dBSPL treats all frequencies equally because it uses Mother Earth’s atmosphere as the gauge for measurement, and dBA doesn’t. It focuses on frequencies that humans most easily perceive—thus, it uses our outer and inner ear to gauge and measure the SPL. To put a finer point on this, using dBSPL will give much different readings if there are frequencies below 1000 Hz, whereas they are both very similar for any frequencies above 1000 Hz.
Next month, I’ll continue down this path and we’ll be using decibels to measure watts, volts, and SPL to definitively answer the age-old question: Is a 100-watt amp twice as loud as a 50-watt amp? See you next time. Namaste.
I've found germanium-based fuzz pedals possess a creamier type of distortion, overdrive, and fuzz. I've also found them to be more responsive and finicky to changes in dynamics (volume rides, light-to-heavy picking, etc.) than my silicon fuzz pedals. Regardless, the recording tips I'm going to give you, apply to any type of fuzz or distortion.
Okay, the dojo is open.
Tip 1: Fuzz directly into a preamp or DAW. Don't worry if you don't have a vintage EMI console lying about, ready for you to plug into and overdrive the mic preamp. Plug your fuzz directly into a DAW input. Crank it up and enjoy the angry beehive. This can get you very close to that classic Beatles' “Revolution" sound, or, more recently, U2's “The Miracle (Of Joey Ramone)." For the latter, just add a little slapback delay (around 120 ms). You might notice that you have to play your guitar a bit harder to have consistent amounts of fuzz, or you can use a compressor before the fuzz pedal and that will even things out.
Photo 2
Fuzz pedals have always been hit or miss with me (as well as wah pedals). They're either too shrill or have a huge spike somewhere in our ear's most sensitive zone, which lies between approximately 1 kHz and 6 kHz. So how can we tame that and sculpt it into something different?
Tip 2: EQ after the fuzz. Say you have a particularly shrill fuzz and you want to reign it in with some EQ. You could use your pickup selector switch, and/or tone knob, and/or adjust the tone settings on your amp. But my experience is that none of these choices can really tame those offending frequencies without adversely affecting the others you love. Having a dedicated multiband EQ pedal can do the trick deftly, and it's useful in other areas as well.
Fig. 1
Most offending frequencies in fuzz pedals hover around the 2 kHz to 4 kHz zone. Sculpting these via a multiband EQ pedal can greatly improve your fuzz sound. The Boss GE-7 ($119 street) is a worthy addition to any pedalboard, and you can tailor the sound to your liking. If you're not sure which frequency band is the one that's driving you (or your audience) crazy, simply boost each band one by one until it jumps out at you, then turn that band down. It usually doesn't take more than 6 dB of reduction to get the tone just right.
Compare Photo 1, a stock GE-7, with Photo 2, an XAct Tone Solutions (XTS) modded GE-7 ($189 street). The stock GE-7 has the following frequency bands: 100 Hz, 200 Hz, 400 Hz, 800 Hz, 1.6 kHz, 3.2 kHz, and 6.4 kHz. However, 200 Hz lies in bass territory and 6.4 kHz is in hi-hat territory. Thus they are not very useable, giving you only five bands to reliably work with. Knowing this, some companies have been making mods to this stock pedal. The XTS GE-7 has: 400 Hz, 800 Hz, 1.2 kHz, 1.6 kHz, 2 kHz, 2.5 kHz, and 4 kHz. These frequencies are much more “musically" centered in the sweet spot of the guitar's range and where it can sit in the mix.
Fig. 2
If you don't have a pedal, you can use an EQ plug-in. Start by boosting and sweeping with a narrow Q [Fig.1], and once you locate the offending frequency, notch it out to taste [Fig. 2].
For an audio/video example of this month's article, I invite you to watch my “Bryan Clark: FUZZ FLAVORS" video, where I do both Tip 1 and Tip 2 scenarios.
Until next month, namaste.
Any room with rectangular walls exhibits standing waves (aka eigenmodes), resulting in an uneven response to what is played. There are usually three axial modes between walls, floor, and ceiling, and it's easy to calculate them. Here's an example: The speed of sound is 1,130 feet per second. Dividing that speed by twice the room size gives you the frequency of the lowest mode. So, walls at a distance of 10 feet apart exhibit their lowest mode at around 56 Hz. And now? Well, you've now got the mode for an ideal, empty room with 100 percent reflective walls and no doors or windows! You could build tuned absorbers for this specific frequency, but it's not clear whether the end result will work in real life.
Another increasingly popular method is to use software, which is available for almost any mobile device, to analyze a room's acoustic qualities. For most of us, it's information overload and can be especially misleading for a novice. When you start to take speaker placement into account, it gets even more complicated. Plus, you'll have to find correct places for the amps and cabinets after the gig, and let's not forget having to retune the absorbers once the new, big sofa enters the room. What most rehearsal rooms need are broadband absorbers, and instead of tuning them, use many. The absorbers for the low end are called “bass traps."
To see how well these steps can work when done right, take a look at Fig. 1. It's a 3-D waterfall plot where the red region shows the undamped room. There is a sharp, vertical exaggeration of a few frequencies with echos and reverb of up to almost a half-second. The blue region—which represents the addition of bass traps—not only exhibits a far more even response: It significantly decays echos, ringing, and reverb time.
Typical absorbers for the higher range are flat and wall-mounted, but the bass range needs more. An absorber consists of an open-cell material that dissipates the air particles' movements into heat. Yes, there are also systems that actively phase out a room's modal resonance or an even broader range of frequencies, but in our case, a passive absorber is the way to go.
An absorber's placement does depend on the specific room. Generally speaking, however, classic absorbers are placed at the center of the walls or as clouds on the ceiling filtering out mid or high reflections. But traps for the bass range are most often installed in a room's corners. Bass traps use similar open-cell foam, fiberglass, or mineral wool, but they differ in placement, thickness, and concept. (Typical bass traps use 2"- to 4"-thick absorbing material and a distance from the corners of 1.5 to 2 feet.) Fig. 2 shows the absorption coefficient at different frequencies of three different bass-trap constructions. For our purposes, the broadband version is the most effective in terms of tone and budget.
Fig. 2 — Here's the absorption coefficient of different absorber concepts and constructions.
So, why should bass traps be placed in the corner? Some say it's because corners bundle up the bass energy like a horn speaker, just reversed, as if the corners are the place where all the played notes meet. It's a nice picture to paint, but not quite the real reason. Once we have a standing wave, this means that our source signal is reflected from the wall and interferes with the next incoming wave front.
A standing wave has certain fixed characteristics at specific places in the room. The sound pressure directly at the wall is at a maximum, while at the same point the reflecting particle velocity is zero. (There is a wall!) Pressure has its next maximum in half a wavelength's distance from the wall, and its first minimum at a quarter of the wavelength, which is exactly where particle velocity has its first maximum. Pressure and velocity are always quarter-wavelengths (or 90 degrees out of phase). And the location of maximum particle velocity is where the absorber has to go.
The main point here is to leave a certain distance between the absorber and the reflecting wall. It doesn't hurt to fill these spaces, too, but it's a less effective use of material. And by placing bass traps in the corner, we'll not only get more depth and an absorption of lower frequencies, but varying gap size can give us a broader range of wavelengths.