Fig. 1
Our columnist breaks down the science behind the dB unit specific to digital systems, and divulges a perennial question about comparative amp loudness.
Hello, and welcome to another Dojo. Last month, I focused on the history and development of the VU meter, and then defined some of the more confusing terms regarding decibels (dB, dBu, dBA, and dBSPL), as well as various ways to measure loudness and amplitude. I also asked: “Is a 100-watt amp twice as loud as a 50-watt amp?”—the answer shall be revealed presently. But first, I’m going to focus on a relative newcomer to the scene: dBFS. Tighten up your belts, the Dojo is now open.
More dB Terms?
What is dBFS? It is relegated to the digital realm, and whenever you hear this term, I want you to remember the following joke: “What sounds better than digital distortion? Everything!”
You see, dBFS (decibels full scale), is the unit of measurement for amplitude within a digital system (such as your DAW), and all digital systems have a maximum peak level before clipping (overload) occurs. A reading of 0 dBFS is the highest signal level achievable for a digital audio. Unlike the analog world, where brief moments of being “in the red” of the VU meter won’t adversely affect the audio, dBFS digital “overs” are squared off (or, mercilessly guillotined), and clipping occurs. It sucks, and is to be avoided at all times. The audio irony is that, when or if you see those two teeny-tiny rectangles above your master fader turn red (Fig.1), you’ve got big problems, and need to turn everything down in order to avoid clipping when you bounce/render your mix.
Is a 100-watt amp twice as loud as a 50-watt amp?
I’m going to answer this by helping you learn some more about how we perceive loudness, and I’ll be using dBA as our mode of measurement. Because, as you’ll recall from last month, using dBFS treats all frequencies equally, and that’s not how the human ear works. In other words, our sensitivity to frequency, sound level, and duration vary across our general range of hearing (20 Hz to 20 kHz). Specifically, our hearing has evolved to be most sensitive between the approximate ranges of 2 kHz and 5 kHz.
Using this range as a yardstick, audiologists agree that 3 dB (up or down) is the minimal detectable change the average person can hear. In my experience, in a critical listening environment, I—and anyone else in the studio—can easily hear 1 dB changes—this especially happens when mixing and mastering.
“The audio irony is that, when or if you see those two teeny-tiny rectangles above your master fader turn red, you’ve got big problems.”
However, to increase the sound, 3 dB requires twice the power (intensity). An increase of 6 dB requires twice the amplitude (voltage), and an increase of 10 dB sounds to the human ear twice as loud. Stated another way: +3 dB is 2x the acoustic power, +6 dB is 4x the acoustic power, and +10 dB is 10x the acoustic power. For those who are mathematically inclined (I am not), here is the decibel formula: dB = 10log10 (S1/S2), where S1 and S2 are the intensity of the two sounds.
Remember when I said last month that decibels are based on a ratio, and they are logarithmic? Now we want to look at the relationship of decibels and power. What is the international unit of measurement for power? It’s the watt (W), named after Scottish inventor James Watt (1736–1819). Check out these relationships:
| Number of Decibels | Relative Increase of Power |
| 0 dB | 1x |
| 3 dB | 2x |
| 10 dB | 10x |
| 20 dB | 100x |
| 30 dB | 1,000x |
| 50 dB | 100,000x |
| 100 dB | 10,000,000,000x |
So, let’s say we have a lovely vintage 4x12 cabinet, and we have two amps that we are going to run through it—a 50-watt head and a 100-watt head (both with matching impedance). Using this relationship, we can answer our previous question! Our 50-watt head is the starting point and baseline for measurement.
Let’s say that, as we play our latest epic jam through our dimed 50-watt head, our drummer comes over and says, “It’s not loud enough! They can’t hear it in the parking lot of the Waffle House across from the stadium!” So, we switch to the 100-watt head thinking it will be twice as loud. Wrong! Go back and look at the chart above. We’ve doubled the power (2x) and that only gained us a paltry increase of 3 dB! That’s right, a 100-watt amp is only 3 dB louder than a 50-watt amp. So, what would be twice as loud? Do the math—we’d need a 10x increase in watts (50 W × 10), so a 500-watt amplifier (or 10 dimed 50-watt amps, simultaneously) would be needed! (Yes, but do they go to 11?)
Until next time, namaste.
Fig. 1
Using templates when recording makes a big difference in streamlining your workflow, and will leave you more time to get creative.
Hello and welcome to another Dojo! This time I’d like to focus on the benefits of using templates in your recording and mixing process. I’ll also show you some ways in which you can increase your productivity by using customized templates for your particular workflow regardless of what DAW(s) you use. Whether you’re recording a live band or a solo artist, you can create templates that include the necessary tracks, processing, and routing setups to meet your unique requirements. Tighten up, the Dojo is now open.
Over the last 30 years, digital audio workstations (DAWs) have revolutionized the way music is produced and recorded, making it easier to create high-quality recordings from the comfort of your own home. With so many options now available, it can be challenging to streamline the recording process and maintain consistency across multiple sessions. This is where templates—pre-configured session setups that can be customized and reused to simplify the recording process—come in.
The main point here is to create a template that works for you. I have found that the more specialized the template, the less flexible it becomes for use in other scenarios. For example, a 48-channel mixing template with specific plugins, buses, and other routing assignments won’t be a first choice when recording a power trio. I think the important thing is to recognize the type(s) of work you do and make different levels of templates accordingly. By creating various kinds of templates that include all the necessary tracks, plugins, and settings, you can ensure that each recording or mix session starts with a consistent foundation, allowing you to focus on the creative process rather than technical setup.
“By sharing templates, you can ensure that everyone is working with the same setup and settings, making it easier to collaborate and share ideas.”
Saving Time
Creating a new tracking session in your DAW from scratch can be a time-consuming process, especially if you’re working with a large number of tracks or complex routing setups. Using templates allows you to quickly set up your session and get to work, without having to waste time configuring settings or searching for the right plugins. I find this particularly useful when starting a new project that involves recording multiple songs with the same artist or band.
Typically, I create the session’s tracks and buses, assign, route, and organize my signal flow, in-the-box or outboard (Fig. 1), and get sound levels from each musician by making adjustments at the mic first, then add EQ and compression as needed. Once all that is done, I save the session as a “tracking template” with the artist/band name and date. When we’re ready to move on to the next song, I pull up the “tracking template” and save it as a “new session”! Now I have the same organization of track count, routing, etc., and I am able to repeat the process for each song moving forward.
Mixing It Up
The same logic applies when moving to the mixing stage. I’ll create a new template focused on advanced signal routing and incorporate things like console and tape emulation (if it wasn’t tracked through a console), side-chain options, routing folders, and instrument groups specific to that project. I found that using one-size-fits-all, highly specialized mixing templates end up being overbuilt and I waste time parsing out only what is necessary, as well as making sure that it is not draining my RAM and CPU resources.
Collaboration
Using templates can also be beneficial when collaborating with other musicians or engineers. By sharing templates, you can ensure that everyone is working with the same setup and settings, making it easier to share ideas and tracks. This can be especially important when working remotely, as it can help ensure that everyone is on the same page, even if they are not in the same physical location.
Creating templates can also help future-proof your recording process, ensuring that your recordings remain consistent and of high quality as your needs change over time. By creating templates that can be easily updated or modified, you can adapt to new recording technologies or workflows without having to start from scratch. This can help you stay ahead of the curve and ensure that your recordings are always of the highest quality.
Finally, you can create templates that use console emulation on every channel, aux, and mix bus. There’s Universal Audio’s LUNA API Vision Console Emulation Bundle ($559 street), Neve and API summing plugins ($149 street) and many other possibilities from Waves NLS, and Slate Digital’s Virtual Console Collection ($149).
Regardless of the DAW you use, taking the time to create some different types of templates will save you time and help keep you and everyone involved in the creative state of mind. Until next time, keep creating! Namaste.
From left to right: a 1/4" tip-sleeve instrument and speaker cable, a 1/4" tip-ring-sleeve headphone cable, an XLR, two RCA, and a 3.5 mm tip-ring-sleeve headphone cable.
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!
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Midrange is the guitar’s magic zone. An EQ pedal will help you sculpt a mix-ready tone before you hit record.
Hello, and welcome to another Dojo. This time I want to shine some light on a secret to great tone: midrange! I’ll be approaching this from the front end of the recording process, using an EQ pedal, but these ideas can be easily applied further downstream in your DAW by using outboard EQs, or EQ plugins. I encourage you to record your experiments so you can hear them and evaluate the differences. The Dojo is now open.
Let’s define midrange, loosely. Midrange frequencies are wide-ranging and are often divided into three sub-categories: low-mids, mids, and high-mids. Basically, it’s between 200 Hz and 4 kHz. That’s huge! It spans the range the human ear is most sensitive to in frequency (even though we can hear approximately from 20 Hz to 20 kHz). So, where exactly do the low-mids start and the high-mids end? What are the crossover frequency points between each band? Those questions are best debated over beer and pizza and will involve the EQ’s circuit design, like where the center frequencies are for each band and how narrow or wide each band is (aka the Q). For comparison, think of the color spectrum and then go and ask a group of painters when red fully transitions into orange and then to yellow, and you’ll get the idea.
For a standard-tuned guitar, I’ve found frequencies between 400 Hz and 2.6k Hz are adjusted the most often and where most of my tone sculpting takes place.
We should all be deeply familiar with the inherent timbral characteristics of single-coil (super articulate and responsive) and humbucker (full-bodied and powerful) pickups. At some point, you’ve most likely wished that your humbucker-loaded guitars could sound more like your single-coil guitars and vice versa. What if a simple 5- to 7-band EQ pedal could get us closer to dialing in the tone we’re seeking and offer more flexibility in the long run? That’s exactly why there are so many different types of EQ pedals on the market—each created exactly for these kinds of purposes.
For a standard-tuned guitar, I’ve found frequencies between 400 Hz and 2.6k Hz are adjusted the most often, and where most of my tone sculpting takes place.
Why not just use my amp? The mids in classic tube amp circuit designs are blunt instruments and don’t offer the surgical precision of a multiband EQ. In fact, many classic Fender amps (tweed Deluxe, Princeton, and Deluxe) are completely devoid of a mid control. One exception is the hallowed 1959 4x10 Bassman, with its mid frequency centered around 500 Hz. A Marshall plexi’s mid knob is centered around 800 Hz.
Before we start focusing in on midrange frequencies, you may be wondering about the most clearly audible range of the guitar. The low E (open 6th string) is 82.41 Hz and the highest fretted note on a Gibson Les Paul (22nd fret of 1st string) is around 1174.66 Hz. But there’s also an insane amount of frequencies above 1.2 kHz that really define the guitar’s clarity, presence, articulation, and sense of “air.” They are immensely important. Play your guitar and shave off everything above 1.2k Hz and you’ll immediately hear what I’m talking about.
Let’s quickly shape some tone. I’m going to make my Telecaster’s bridge pickup sound as close as possible to my Les Paul’s bridge pickup and vice versa. (Photo 1) shows I adjusted 400 Hz (+11 dB), 800 Hz (+8 dB), 2 kHz (+6 dB), and 4 kHz (+8 dB). This gave me the fatness and articulation of my Les Paul’s bridge pickup and sounded really close. To get my Les Paul’s bridge pickup to sound more like my Tele’s [Photo 2], I adjusted 400 Hz (-7 dB), 800 Hz (-4 dB), 1.6 kHz (-3 dB), 2 kHz (-6 dB), 2.5 kHz (+7 dB), and 4 kHz (+5 dB). This gave me the spank and chime of my Tele’s bridge pickup. Bonus: I like to reduce 400 Hz to 800 Hz when playing rhythm on my Les Paul’s neck pickup anyway. It really cleans out the bottom end clutter that never sits right in the mix.
Here are some additional thoughts for EQ pedal experimentation:
• Humbuckers have more low-mid information than single coils (300 Hz to 900 Hz).
• Single-coils have much more high-mids (2 kHz to 4.5 kHz).
• To increase pick articulation (1 kHz to 2 kHz).
• To reduce muddiness (250 Hz to 350 Hz).
• To reducing harshness (2.3 kHz to 2.7 kHz).
Until next time, Namaste!