In this edition of Recording Dojo, our columnist sheds light on the differences between decibel measurements, and offers a breakdown of real-life equivalents of basic sound level values.
Starting this month, I’d like to bring some clarity to some widely misused audio terms that engineers struggle with, especially when it comes to mixing and mastering. Do you know what a dB is and/or how it got its name, or understand the difference between dB, dBu, dBA, dBSPL, and dBFS and how they inherently apply to your recordings? Tighten up your belts, the Dojo is now open.
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.
Learning the differences between various cables can greatly improve the quality of your recordings.
Hello, and welcome to another Dojo session! This time I’d like to drill down to some audio bedrock and unearth the differences between balanced and unbalanced cables. I want to help you understand the differences and give you some strategies to greatly reduce noise (hums, buzzes, and static) in your recordings. Tighten up, the Dojo is now open.
There are many different connection types and gauges of balanced and unbalanced audio cables, and both are used to transmit audio signals from one device to another. However, they differ in their construction and performance, and understanding these differences is essential for achieving optimal audio quality.
Tipping the Scales
Unbalanced audio cables are the most common type of cable used in consumer audio equipment. This includes our beloved 1/4" TS (tip-sleeve) instrument and speaker cables, RCA, and TRS (tip-ring-sleeve) 3.5 mm and 1/4" headphone cables. The first two kinds of cables consist of two wires—a signal wire and a ground wire, while the headphone cables are in stereo, with three wires: left, right, and ground. Signal wires carry the audio signal, while the ground wire acts as a reference point. At the cable’s end, the tip (and in the case of headphone cables, the ring) of the plug carries the signal, while the sleeve is the ground connection.
Unbalanced cables are very limited in the distance they can transmit audio signals cleanly (preferably less than 20 feet). The longer the cable, the less high frequencies, and the more susceptible it is to noise and interference from external sources—like electromagnetic fields created by other electronic devices nearby (amps, synths, drum machines, outboard gear, cell phones, computers, televisions, etc.) and radio frequency interference.
Balancing the Scales
Balanced audio cables, on the other hand, which include XLR and balanced 1/4" TRS types of connectors, are designed to reduce interference and improve audio quality. They always consist of three wires—two signal wires and a ground wire. Note that while some unbalanced cables have three wires, the two signal wires in balanced cables carry the same audio signal, with one flipped 180 degrees out of phase, making them balanced mono as opposed to unbalanced stereo. Balanced cables are ideal for use in recording studios and live sound because they are capable of transmitting audio signals over longer distances (several hundred feet) without introducing noise or hum.
How? Without getting too technical, when the audio signal is split into two separate, identical paths across the two signal wires (with one being out of phase), and then recombined in phase once again, the resultant signal is amplified, and any noise that was present is canceled out. This includes 60 Hz buzz, hum (ground loops), white noise (thermal sound), digital clock jitter, and more.
They Look the Same, but Are They?
One mistake that’s easy to make is to confuse an unbalanced stereo headphone cable with a balanced mono TRS cable, as they both look the same and both have three wires. But if you tried to connect your unbalanced stereo cable output from your smartphone or tablet to a balanced input of a mixer, anything from the center of the stereo field (most likely the main vocals, kick, snare, and bass instruments) will be canceled, because the balanced input will sum both the left and right from the stereo cable, and anything common to both will be 180 degrees out of phase. Essentially, the balanced input will treat the center image as “noise” and remove it.
Can I Convert Balanced Into Unbalanced and Vice Versa?
Yes, you can, and that’s exactly what DI (direct injection) boxes and reamp boxes do. A DI box will convert unbalanced instrument level signals to balanced line level signals and reamp boxes do the opposite—balanced line levels to unbalanced instrument levels. If you’re unfamiliar with these devices and how they work, check out my Dojo video on how to reamp your guitar.
How to Reamp Your Guitar | Recording Dojo
Until next time, namaste and keep making your music!
Your favorite stomps are real-time, tactile sound processors. Plug them in and expand your DAW’s options.
Welcome to another Dojo. This time I want to help supercharge your creative process by advocating for a hybrid approach to effects processing. Specifically, I want you to embrace using stomp pedals as real-time, tactile effects processors and combine them with your favorite DAW effects and plugins.
You should be deeply familiar with how to insert plugins directly on your DAW tracks’ aux sends (for serial processing with modulation and time-based effects, like reverbs and delays) or aux buses (for parallel processing, like compressors). But what about guitar pedals? Yes, they’re typically used in live performance settings and get a lot of abuse on the stage and studio floor, but with the explosion of modern, programmable, MIDI-capable pedals on the market and their ever-increasing processing power, “lowly” stompboxes are long overdue to be elevated to the same level as rack effects and kept within arm’s reach on your mixing desk.
Pedals can add analog warmth, hands-on control, and modularity, are easy on your computer’s RAM and CPU resources, are always OS compliant, and retain their value.
Turning Knobs vs. Mouse-Clicking
When it comes to tracking and mixing, turning knobs on a physical, controllable surface, such as a pedal, mixing console, or MIDI controller, can provide a more tactile and intuitive way to make adjustments to your sound, compared to using a mouse to click and drag on virtual knobs and sliders within your DAW. Let’s face it, in the heat of a session this can be tedious—especially if you run out of trackpad or mousepad space while recording or mixing.
But what about exploring some hybrid approaches that take advantage of both formats? After all, plugins provide flexibility, precision, consistency, automation, portability (nothing to lug around), cost-effectiveness (cheaper than outboard gear), and compatibility (the same plugins can work on multiple DAWs). Pedals can add analog warmth, hands-on control, and modularity, are easy on your computer’s RAM and CPU resources, are always OS compliant, and retain their value (how much are original Klons going for now?!). They can also help you achieve a unique and personalized sound that can be difficult to replicate with digital plugins and their stock presets. Many modern foot pedals can also handle both line level and instrument level inputs.
Builders—Strymon, Eventide, Boss, EarthQuaker, Empress, Meris, Chase Bliss, and many more—have a wide range of pedals that are MIDI capable and, quite frankly, have processing power that far surpasses many classic rack effects units.
So, I’d like to offer some creative ways to use pedals, in addition to your regular plugins, in your next session or mix:
Getting Ready
Before starting, remember to be aware that some pedals are looking for instrumentlevel and not line level inputs (the latter is what typically is output from your interface). You can find helpful info by reading my September 2022 Dojo column, “What You Should Know Before Using Guitar Pedals with Other Instruments.”
To start, duplicate the track(s) you want to process with your effects pedal(s) in your DAW and route the output of those tracks to one or two of your line outputs on your interface. Depending on the pedals in question, you may have options for mono in and out, mono in/stereo out, or stereo in and out. Connect all relevant cables and connect the output of the last pedal to the input of your interface. If your incoming signal is low, switch from line to mic on your interface for each input.
Next, in your DAW, create one mono or one stereo track, depending upon how you are going to return the processed signal from your interface and record-enable the track(s). Now you’re ready to record new, processed material (from one pedal or your entire pedal board!) in real-time and take advantage of every parameter on each pedal.
You can now use your pedals to adjust distortion levels, reverb, and delay times in real-time (with all the glorious artifacts, glitches, and smears), as well as adjust tremolo rates and chorus depths on the fly. Get creative! Take chances and invite any and all happy accidents!
One particular approach I love is throwing loop pedals into this equation, after all the other pedals, for some wild, abstract processing. My signal flow usually goes from overdrive to mod-based effects (chorus, phaser, tremolo) to time-based effects (delays and reverbs) followed by a looper. At present, my favorite looper pedal for this by far is Habit by Chase Bliss ($399 street). It has three minutes of loop time and can take user-definable snippets of your loop, play them back asynchronously, feed that back into the loop itself, and record all modifications as well (and this is just scratching the surface). Highly recommended!
Combine this “out-of-the-box” technique along with your normal “in-the-box” workflow and you should be creating some pretty amazing sounds. Let me know if you find a cool approach! I’ll share it in the Dojo channel.
Until next time, blessings, and continue to share your gifts with the world. It matters, and you matter!