Performer Magazine: Summer 2023 MOBILE ISSUE

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How to Release a Song in 2023


Shape a Vocal Track with Compression and EQ


Why is Dynamic Range So Important?


COVER STORY: Behind-The-Scenes at KRK Systems

22. Why You Should Use an Audio Interface for Streaming

26. Getting Started with DI Boxes

30. Exploring Different Types of Microphones


How to Mix With Studio Monitor Headphones

38. 2023 Content Creators Buyers Guide


REVIEWS: Saramonic BlinkMe Wireless Mic, Donner PC-02, RODE NT1 5th Gen



Cover photo by Wendelin Jacober is used under a CC BY 2.0 license Cover


LETTER from the editor

Hey gang,

We’re back at it with another themed issue, this time focused on YOU, today’s content creators, and how you make music at home and on-the-go.

Inside you’ll find some helpful production tips from some of our partners and guest columnists that’ll apply no matter where you’re making your music, podcast or video content. Get some pro advice on mobile monitoring, wireless audio for your social videos, the best interface choices for streaming, an intro to DI boxes, and our annual buyer’s guide for the best gear for today’s creators and artists.

HUGE THANKS to KRK and our old friends

Audio-Technica for helping us make this issue and companion sampler a reality.

In the next issue, stay tuned for an entire home studio guide from nuts to bolts, followed up in

the winter with (what we hope) will become an annual acoustic guitar extravaganza, complete with a cassette tape sampler (we know you all have been loving those limited releases lately).

So, until then, keep creating, keep smiling, keep shining…wait, that’s what friends are for. Never mind.


Benjamin Ricci

PS – the European football season closed since we last spoke, with Spurs going trophy-less yet again, with no real end in sight for this horrendous silverware drought. They did hire a new manager (again), this time poaching the head coach from Celtic, but as of yet have made no sensible moves in the summer transfer window other than scooping up a bargain-priced James Maddison from the recently-relegated Leicester (which, if we’re being honest, does more for Maddy than Tottenham going into next season). Another year, another ambitionless vista on the horizon…

ABOUT US / Performer Magazine, a nationally distributed musician’s trade publication, focuses on independent musicians, those unsigned and on small labels, and their success in a DIY environment. We’re dedicated to promoting lesser-known talent and being the first to introduce you to artists you should know about.

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EDITORIAL SUBMISSIONS / In the words of our esteemed forefathers at CREEM: “NOBODY WHO WRITES FOR THIS RAG’S GOT ANYTHING YOU AIN’T GOT, at least in the way of credentials. There’s no reason why you shouldn’t be sending us your stuff: reviews, features, photos, recording tips, DIY advice or whatever else you have in mind that might be interesting to our readers: independent and DIY musicians. Who else do ya know who’ll publish you? We really will...ask any of our dozens of satisfied customers. Just bop it along to us to and see what comes back your way. If you have eyes to be in print, this just might be the place. Whaddya got to lose? Whaddya got?”

Volume 33, Issue 2

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William House

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Benjamin Ricci


Cristian Iancu


Bob Dobalina


Benjamin Ricci, Chris Devine, Michael St. James, Joseph Cross, Christophe Anet


William House

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© 2023 by Performer Publications, Inc. All rights reserved. No part of this publication may be reproduced by any method whatsoever without the written permission of the publisher.

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Iposted, “Few things make me as sad as seeing: ‘I just dropped my new song today.’”

Within one minute of posting that, I got flooded with countless direct messages. “What does that mean?,” “Why wouldn’t I promote myself?,” and many other comments along the same lines.

Let me give it to you straight. If the first instance I am hearing of your song is on the day of your release, you are doing it horribly wrong. It’s alright, you are one of the 50,000-60,000 artists releasing music per day who are also doing it wrong. It is no wonder why so many great tracks just fade into oblivion within a week of release.

To be fair, this used to be the way a ton of music was released not too long ago. You’d build up interest on social media, then you’d maybe do an ad campaign with a little taste of the music coupled with a pre-save campaign. But the music would be locked away on the ad, unlistenable to the fan. Then, the day it dropped, pre-saves would bump listeners, and everyone would go check it out, and you’d get a flood of streams and views.

That has all changed.

Call it the TikTokification of music if you’d like, but music fans these days want to hear the music before they really hear it. They want to use it. They want to become obsessed with the short version. They want to dance to it, lip sync to it, share it for clout. I know this sounds counterintuitive, but it’s really not much different than a major artist releasing a single to radio which is heard for over a month before the “record” comes out.

The problem is this: you are treating your music as some precious little secret that will be unveiled at the exact moment of your choosing in hopes of tons of people becoming obsessed with it and driving sales (streams). Why? Do you think by holding it back that you will prevent people from getting sick of it by hearing it too much? I have some bad news for you. If that is the case, it’s not a great song. Your song needs to make someone’s list of 50 songs they listen to every week, or better yet, one of the 20 they listen to daily, for a time, or it will go nowhere.

I get contracted to help labels and artists release music and they pay me very well for customized strategy. There are nuances and some of the steps I am giving you absolutely demand expertise and budget. But, basically, I’m going to show you how to do it, right now. It should go without saying, I assume your song is amazing. I assume it’s written, edited, performed, produced, and mixed incredibly well. Because none of this will work with a crappy song.

The dirty secret is this: even if I give you all the answers to the test (which I am doing below) the

sad fact is that most of you will not do it, most of you will try to only do it partially, and some of you won’t invest a dime in your own music business marketing. You will wonder why the song didn’t blow up, and you will eventually come back to this list and hopefully do it right next time.

So, here it is, quick and dirty, for free: How to release a song in 2023

Your song being released is part of the process, not the destination.

It goes like this: Earworm -> Single -> Video -> Multi-remix.

I will break this down quickly for you in 10 steps (without all the details on how):

1. Four weeks out from the official release date you gave to your distributor, you release your “song.” This is the :30 second hook Earworm. It is a “pre-release” Hook Announcement (it’s not that complicated, but you seed your audio clip to all platforms to use for every video/short/reel/ TikTok you produce). You may need to test a few versions out - verse build, chorus, bridge - to see what hits. Ideally, you have already done this in your “test pre-release campaign,” an entirely different article

2. You build excitement and interest over 28 days on your chosen three platforms, by video and posts about the song: why you wrote and recorded it, why you love it and others will too, why others should hear it and what they will feel when they do. When I say “you,” I do mean you, the person. The singer, the guitarist, the songwriter. You need to be in these videos, faceless won’t do. If you’re not into that, the music business is not for you (Obviously, there could be a whole course on exactly how to do this part). This is also a marketing period to stakeholders like playlisters, influencers, and brand partners, as well as audience building by engaging every single day on your chosen three platforms. The key is learning how to actually do this correctly.

3. You release the “Single” (a full version remix of the Earworm). This should be known as a “Grateful” period. On release day, you thank everyone who has already come on the journey with you over the last 4 weeks (social proof) by “letting them hear” the whole thing; NOT “my song just dropped,” NOT “listen to my new music.” You thank them in general, in public and private, using video and posts.

4. You spend 1 week thanking and promoting those who have promoted you. This includes using the graphics of playlists in your socials, thanking other artists, as well as superfans. You also release a graphic video of your remixed Earworm–the Single–as a visualizer and/or

album/song cover artwork. Then you release a lyric video, so everyone can learn the words. Break these sections apart and use them in quick 10-30 second social posts.

5. This is also the week you release the instrumental version to DSPs and YouTube. You then ask for engagement like duets, covers, etc. for 1 week. “Do a verse with me,” “Can any of my DJ friends make this a club banger?” “I keep hearing this with a country vocalist, anyone want to give it a shot?” (have stems ready) etc. You break out emotional lines or music bits to highlight your own socials. You should already be into your next single Step 1.

6. You drive listeners and fans, and appeal for shares while giving updates on the song for one week. These updates show the growth in streams, views, number of playlist adds, how many have been using your sound on TikTok and Reels. Use


those visualizers, repost any fan-made videos, etc. This period should be very video forward.

7. Then, you spend a week posting “You guys, I am so thankful, because of you and your support, we’re going to make a video!” (you already planned this video out two months ago) and then you show some behind the scenes footage, some artwork mockups, maybe some fashion choices. Introduce any other people working on it–actors, the director, your friend who brought tacos. Drive engagement by asking fans to be involved, “What do you think, should I…?” You should be promoting shows here too; as well as any interviews, guest podcast spots, writeups, etc. Then, 1 week from the video release date, you release the music video trailer. Again, as with all posts, you are reusing the sound on the platforms, not the audio in the video.

8. Release the video. Do a watch party. Tell a

story of what happened during the shoot. Do it on YouTube, Twitch, TikTok Live, IG Live, wherever is best for you. You push the video hard for at least one week by using Google Ads, retargeting, spreading out to video playlisters, and shouting out to production personnel.

9. Get someone, anyone, (even yourself) to remix the song in a different style, language, or alternative version. (Of course, you’ve already done this months ago after your testing prerelease campaign gave you the data). You then do a quick and dirty release to DSPs and YouTube. This does not need as much lead time as the other releases. The key to this action is that by doing it within this 4-8 week window, you already have subs and listeners who will be fed this without the algo or your ad pushes. It will result in multiple plays on the original.

10. Share numbers, playlists, thank fans, etc. In

addition to each of these releases, this is where you really cultivate your email list by doing a VIP giveaway. Do Cameo-type video messages to your most engaged fans. This is where you start the process of offering physical items – vinyl, a custom shirt or hat for the single, sticker giveaways, etc. Be careful with your time though, you should already be in week 4 or 5 of your next release.

So, it is never a one-day event, it is a 10-12 week schedule building momentum. That is how to “release” a single song today, in 2023.


Michael St. James is the founder and creative director of St. James Media, specializing in music licensing, publishing, production and artist development.

Tracking Tips: Shape a Vocal Track with Compression and EQ

Let’s talk about compression and EQ.

These are usually the first elements we think about when processing audio. In this approach, there are two main modes of compression and EQ: utility and effect

Focus on the “Right” Frequencies

A true “utility” EQ is generally first in the signal path, and usually subtractive. For instance, on a vocal track, we want to be sure the errant low frequencies are not triggering compression. Therefore, rolling off the extreme lows on a vocal track is always a good first step Anywhere from 50 – 80 Hz is a very safe rolloff point. Depending on a vocalist’s range,

you can often push that all the way up to 120 Hz. We’re not losing true signal here; we’re eliminating potential interferences, such as vibrations from the vocalist moving/dancing while singing, or diffusing the most powerful range of plosives.

Control Dynamic Range with Subtle Compression

A vocal that wasn’t tracked with a compressor in the signal chain may benefit some “utility” compression. What we’re looking for here is the sound equivalent of fader riding. Fast attack, slow release, and very low ratio is the best approach. To add a very subtle uniformity, use a one-millisecond attack, a one-second


release, with a 1.5:1 ratio. Tweak the compressor’s threshold until you’re getting a maximum gain reduction of 1.5 dB. This should smooth out any variations in the singer’s proximity to the microphone, and begin to tame any loud passages.

Shaping Vocals with Compression

DAWs give us the luxury of stacking compressors. The next step in processing a vocal would be an “effect” compressor. This is compression that we’re using to shape the sound, and add vibe and mojo. There aren’t really any rules here, but a good starting point is basically medium across the board. As it is an effect, we want to hear the compressor working, which necessitates a slower attack, so that part of the transient passes through before the signal is acted upon.

The faster the release time, the more you’ll hear the compressor “pump,” or engage and disengage; this is a matter of artistic preference in crafting a sound. Ratios from 2:1 to 10:1 are entirely normal. Anything 4:1 and above begins to take on a more aggressive “limiting” characteristic, whereas lower ratios will sound smoother and more transparent.

Applying EQ and Vibe

Once we have the compression jiving with the vocalist’s sound, we want to apply taste equalization, or EQ as an effect. This can involve cuts and boosts as desired. Be careful adding boosts around 200 – 400 Hz, as too much of this frequency range will sound warm when the vocal is solo’d, but tends to sound muffled and overly weighty in the context of a mix. An additional problem range exists between 8 and 11 kHz, where sibilance (the “s” sounds) can get out of control.

When boosting to try to make a vocal sound brighter, there is a huge potential here to create harshness. A boost in the 1 – 2 kHz range combined with a slight boost above 15 kHz will usually create the presence and clarity we’re looking for, with fewer side effects.

If you like the sound you’ve carved out from these four steps, but the dynamic range is still a little too wide, you can apply a brickwall limiter at the end of the chain. Despite its ancientness, the Waves L1 limiter is an all-time favorite. No frills, no learning curve. Set it to take off a maximum of 1 – 2 dB and let it do its thing.

Have you ever read the manual?

A final piece of advice: be a dedicated student of audio engineering. Put aside YouTube tutorials and shortcuts; start by reading the user manuals of your plug-ins. Get familiar with each tweakable function, and how the manufacturer divides the power amongst them. Competency on an audio


processor’s user interface will enhance your ability to put tips and tutorials to better use, and hopefully shorten your path to good results.

We hope this quick guide helps you find the sound you’re looking for. Break a leg and unleash your sound!

[Editor’s note – we wish to thank our friends at sE Electronics for granting permission to reproduce this helpful article, which originally appeared on the sE Blog. To learn more, please head to Check out the stunning sE2200 vocal mic at https:// and the indispensable RF Pro portable acoustic treatment unit at]


Why Is Dynamic Range So

Christophe Anet, QSC

So Important?


Every genre of music is associated with a certain dynamic range, meaning the difference between the loudest and softest passages. The word “dynamic” is defined as “a force that stimulates change or progress within a system of process.” It defines how versatile a piece of music or a piece of gear behaves (musical instrument, amplifier, effects pedal, effects processor, etc.). The opposite of this is “static,” defined by “lacking in movement, action or change.”

Dynamics are therefore extremely important because they instill change, progression, etc. in music and in the creative process. This article will explore the concept in detail and explain why dynamic range is so important to the enjoyment of

listening to both live and recorded music.

The Essence of Music

Dynamics are one of the essential ingredients (together with melodies, harmonies and rhythms) that make music pleasurable and compelling to listen to. A song that provides noticeable variations in level is usually more engaging than one that stays pretty much the same from beginning to end.

Pushed to the extreme, if a song has too wide a dynamic range, the quiet parts will not be heard clearly with the loudest parts being uncomfortably loud. Conversely, if the difference between loud and soft is too small, the music will sound squashed and might even be fatiguing to your ears, particularly when listened to at high levels.

In order to create drama, a musical artist, a songwriter or a producer will create arrangements that vary in volume and intensity. The variations can be subtle, like an increase in instrumentation, or they can be more obvious, like a break section (where most of the instruments stop playing) after a loud chorus.

For a drummer, the variations in dynamics between each beat are what gives the roll feeling and musicality. Singers also typically alternate between louder and softer parts from section to section, or even word by word.

What is Dynamic Range in Audio?

Simply put, dynamic range in music production is defined as the difference between the loudest peak and the quietest part, expressed in decibels (dB). Therefore, songs with a wide dynamic range will have a larger gap between the loudest sound and the quietest one.

Playback media and audio gear that reproduce music also have a dynamic range. For loudspeakers and headphones, we calculate the ratio between the loudest sound that the device can produce and the quietest one before noise becomes audible (the “noise floor”).

How much Dynamic Range can we hear?

It is worth mentioning that the human hearing is limited to a maximum dynamic range. Our auditory system has a dynamic range of about 90 dB, ranging from a 30 dB whisper to an airplane taking off at 120 dB. Beyond such levels, distortion starts to create physical pain.

In the analog audio world, the maximum dynamic range for analog equipment is somewhere between 50 and 60 dB. For digital audio, the theoretical dynamic range of un-dithered 20-bit quantization of a digital signal is 120 dB. For a 24-


bit digital audio signal, we have 144 dB of dynamic range, much of which human hearing cannot detect since our threshold is at around 120 dB. Therefore, audio compression can be a useful tool in order to help satisfy our maximum range.

Difference between Dynamic Range and Signal-to-Noise

SNR stands for Signal-to-Noise-Ratio, and it is often confused with dynamic range. While these terms are often used interchangeably, they are not necessarily the same thing. SNR expresses the difference between the standard operating level of a device and the noise floor. In very simple terms, a good signal-to-noise-ratio is when your live signal is above the noise floor.

As stated before, Dynamic Range (DNR) measures the ratio between the loudest possible peak without distortion and the quietest one before noise becomes audible (typically hum or hiss).

However, unlike SNR, dynamic range is not necessarily dependent on a signal: a dynamic range’s lowest limit is simply the softest sound that does not have a distorted output.

Additionally, the greater the dynamic range, the more headroom the device will have. The headroom covers the level range above the standard operating level, and before distortion appears. In short, for both specifications, the higher the value, the better.

How do you change Dynamic Range?

Generally, dynamic range is altered through the process of audio signal compression.

Compressors reduce the difference of the volume range between the loudest sound and quietest sound, thereby altering the overall dynamics and dynamic range of a track.

One reason for reducing the dynamic range in recorded or live music is to allow the instruments and vocals to cut better through the mix and come forward. Also, when mastering a recorded track, engineers utilize compressors and limiters (ultrapowerful compressors) to shape dynamic ranges. As a result of bringing up the softest passages in a track, compression also has the effect of making processed signals louder.

It is worth noting that there are extremes in either direction. Dynamic ranges that are too wide can be delightfully musically, but lack the loudness needed for a comfortable listening experience. On the other hand, a very narrow dynamic range could be the result of too much compression, making a song disturbingly loud and devoid of the energy and musicality from the original dynamics.

Music Genres and Dynamic Range

If all music has inherently level fluctuations, some genres tend to have broader dynamic ranges than others. While recorded pop, rock, R&B, hip-

hop and country music usually have a relatively modest dynamic range (typically around 10 dB), electronic dance music (EDM) has probably the smallest dynamic range (around 6 dB) but makes up for it by creating contrast with its almost infinite array of instrument colors and textures coming from synthesizers and samplers.

At the other end of the spectrum, jazz and classical music have considerably large differences between their quietest and loudest parts. In jazz, up-tempo songs typically vary from loud parts played on brass instruments to quiet piano and bass solos. A study of dynamic range in different musical styles conducted in 2016 revealed that dynamic ranges in jazz generally varied from 13 to 23 dB.

Classical recordings have the widest dynamic range of any genre. The same study cited above found that recorded classical music typically offers between 20 and 32 dB of dynamic range. The pinnacle is the dynamic range of a live symphony orchestra, which can reach as much as 90 dB.

Loudness and the Future of Music

Over the past 30 years, the amount of compression and limiting used in live performance and studio recordings alike has undoubtedly increased. To make music stand out on radio or in club sound systems, engineers often master songs with heavy compression in order to push tracks average levels as high as possible. This has created a split among musicians, often referred to as “the loudness wars”, and leading many to beg for their dynamic complexity back.

The cultural reduction in dynamic range, and therefore amplified loudness, has caused us to lose nuances in a song’s mix. Dynamic complexity is lost when compression is used at a higher rate. Fortunately, streaming services have introduced a feature called Loudness

Normalization, which automatically puts a ceiling on a song’s loudness. No matter how loud a recording is, it will be automatically turned down, so that it does not exceed this ceiling. As a result, when mastering for streaming, engineers no longer feel the need to reduce dynamic range to make songs louder. That has resulted in higher dynamic ranges in popular music, which are now in the 10 dB range, on average. Let’s hope that the era of squashing the life out of music is mostly over.


There is no doubt that in order to reproduce music faithfully, loudspeakers must have good dynamic range capabilities. A simple test to evaluate such capability when listening to – or selecting – a loudspeaker is to use a well-recorded drum/percussion track that features soft parts and loud peaks to assess if the full track is well reproduced. No distortions should appear at any reproduced level and all subtle musical details should be revealed during the entire length of the track.

Regarding music recording/live performance, it seems there is no magic key to finding a track’s ideal dynamic range, but understanding the relationship between audio compression and how it affects your signal, will help you find the sweet spot.

While QSC  TouchMix digital mixers have powerful, and extremely useful, on-board compressors and limiters, make sure you always use them to serve the performance or the recording with musicality and emotions in mind. Stay at the service of art.

[Editor’s note – we wish to thank the good folks at QSC Pro Audio for granting permission to reproduce this helpful article, which originally appeared on the QSC Blog. To learn more, please head to]



Benjamin Ricci

We recently had the absolute pleasure of speaking with Gibson Brands Pro Audio Director of Engineering and Sr. Manager of Product Development, Craig Hockenberry, specifically about the KRK lineup. We discussed the company’s product development process, tips for monitoring your mixes, the lifecycle of a product from conception to final production assembly, and much more. If you’re a content creator, this is incredibly interesting stuff from one of the preeminent brands on the market.

So, without further ado, please join us for a fascinating peek behind the curtain of one of our favorite pro audio brands. Special thanks to Mr. Hockenberry for being so generous with his time and expertise.

Could we get some insight from you on what the product development process looks like at KRK? A lot of artists are probably unaware of what actually goes into making a product in the MI or pro audio space.

Our R&D starts with speaking to our users.

We are fortunate to have a really cool network available to us, the KRK Kreator Alliance, which is comprised of many industry-leading professionals. We regularly call upon them to get their input on product concepts that we may have and feedback on solutions that are already on the market, as well as suggestions on marketing efforts we can take to raise awareness down the road.

In addition to the Kreator Alliance, we’ll reach out to content creators, professional musicians and engineers, our commercial partners, and our sales and marketing teams. We’ll shoot some ideas out, gather the feedback, and then go from there.

The next step is to create a “product requirements” document. Based on user feedback and what we learned from market analysis, we outline product functions and what the product should look and sound like. Even target technical specifications are captured. From there, the document goes through a stage gate process, which, once approved by stakeholders, results in a separate “technical requirements” document. The R&D team then

evaluates both documents and turns them into an architectural solution, which is what is used to develop product throughout the remainder of the process.

Once approved to begin the development process, the solution goes through several different stage gates. The team will start building preliminary schematic diagrams, the acoustic design, the mechanical design, and software, if needed. We have an amazing multidisciplinary R&D team here with mechanical, electrical, acoustics, and software engineers. Each of them have specific tasks within a project, which is outlined in the technical requirements document.

When preliminary design is complete, the product then goes through several stages of prototyping. If the first prototype works out—great, then the project advances to the next stage. If not, the problem gets reported, and we iterate prototypes until the issues are corrected.

At the prototyping stage, I assume you’re testing out things like materials and


physical builds. What would be some of the reasons that a product wouldn’t get approved at that point?

We design our own drivers from the ground up, everything from the magnetics and voice coils to cone and dome materials and designs. Let’s say we are prototyping a brand-new tweeter design and we tool up a new dome—we’ll perform a mechanical incoming inspection to ensure it meets the specs of the model that we designed. Then, we’ll also put it through detailed acoustics testing, including using our scanning vibrometer that measures and visualizes modal responses of the diaphragm. If something doesn’t match our criteria, then we’ll have to go back to the drawing board. Perhaps not to fully redesign, but maybe change the design, material, or treatment to meet the criteria. So that would be a case for when and how we’d create another prototype.

After the prototyping stage, what’s the development from there on?

The product then goes to what we call an “alpha” stage, which is basically an advanced

prototype level, but now we’re hard tooling. Since most of our designs require new tooling for custom drivers and plastic or aluminum baffles or faceplates, we’ll hard tool and put it through iterations of testing. Once all tooling is approved, we advance the project to the “beta stage,” where we have additional physical units for more complete in-house and in-field testing. At this stage, we are sending the product out to our Kreator Alliance and a team of trusted beta testers, which will put it through its paces in the field and provide feedback. If there is an issue that was missed in the other stages, we will correct it and move on to pilot production.

Pilot production is performed to verify that all production and operating procedures are good to go. It’s more of a factory-level thing, so at this point, the development is considered complete.

Is that sort of like a test run?

Yes, like a test run, exactly.

And then we’ll get those units back again to triple-check and make sure they’re OK before we pull the trigger to go into mass production.

Once the product is in production, we’re done with the development side of it. The R&D team will continue to support the manufacturing team if needed and issue ECR/ECO’s if further improvements are necessary.

What would a common timeline be like?

From say, idea generation to final assembly and shipping to customers and retail?

It really depends on the complexity and innovation that the product requires. If we have to develop a completely new, bespoke material, for instance, that could take up to two years.

Oh wow. OK.

But I would say typical design time is a year to 18 months. If we were to use a brand-new technology, it could take up to two years before we get into production.

That’s really cool information, because we talk to a lot of manufacturers, but we don’t really convey what actually goes into making a product. And I don’t think many artists really have a good grasp on the


number of steps, teams, and departments that are involved, or how much testing, refinement, and quality assurance happens before anything hits a store shelf.

There’s a team of engineers and product managers on these projects, and they’re working diligently to test the products and make sure they’re reliable. We go through very extensive extended-life testing for reliability, and we put these products through their paces to make sure they’re not going to fail in the field.

I’d love to kind of switch focus now to the artist side of things. So once the product is actually on the market, obviously it’s intended to help people make music and create content. One of the things we want to talk to our audience about is getting into the content creation game. Since you specialize in monitoring, both with headphones and speakers, let’s say I’m a musician or a content creator looking to get started for the first time and really corral

all the different equipment I need. What would you suggest for a first timer to get for monitoring playback and audio for their content creation? Would you go headphones first or would you go speakers first?

I would probably do a combination of both. I wouldn’t stick to just one; the reason being that you need to create in a very quiet environment. I know a lot of creators work from home and they need a quiet environment. Many times, if they have a family or roommates, they’ll need good ‘phones to do that. However, there are some downfalls to headphones—they don’t really create the ambience that speakers can.

Speakers will provide a better stereo image. They’re more realistic sounding than headphones because we hear sound sources out of both ears. With speakers, you can hear multiple sources out of both ears, whereas headphones only give you one source in each ear. And now that we’re speaking of the head, everybody has their own head-related transfer


function that plays into the way they hear. There are expensive headphones on the market that try to simulate that head-related transfer function, but it’s one algorithm. It’s not personalized, and everybody hears differently. The way the sound diffracts around your nose and head is different than the way the sound diffracts around my head, you know?


So, if you want true accuracy, speakers are really the way to go. However, headphones are helpful for dialing into a detailed view of the sound mix, say vocals or something like that. In that case, due to their isolation and separation, headphones might be the way to go because they’ll help you concentrate on a very particular part of your music or content.

In terms of speakers, I know there’s often some confusion when setting up studio monitors. Specifically, is there a certain distance they need to be away from walls and reflective surfaces? Do you toe-in? Do you recommend that? How do you do sound staging? Any sort of tips you can give for someone setting up a home studio for the first time?

A good place to start is our KRK Audio Tools app, which is available for iOS and is currently being optimized for the latest Android software. It’s free to download and gets you in a very good starting space with your setup. I do recommend toe-in, and it’s a 30-degree toe-in on each side. Within our app, you can set your mobile device on top of the speaker and turn it until it alerts you that you’re exactly at 30 degrees.

OK. Cool.

A good rule of thumb is to set the monitors about a meter away to get your best stereo imaging. Point the tweeter directly at your ears, not at your eyes or into your body. That’s important, especially if a monitor is more directional. But that’s a good starting spot and you’ll get very good imaging that way. As far as the room goes, we have an automatic room correction algorithm on our GoAux 4 portable solution that’s very basic, and adjusts for low frequency boundary conditions, but that’s just a starting point.


It’s also extremely important to treat the room for acoustics—this is always better than adding some sort of in-room correction. As far as boundary conditions off your wall, depending on the woofer size and your speaker size, you should try to be at least three feet off your wall. As a general rule of thumb, try to be equally

distant between the two reflecting walls on either side.

Since we’re talking about mobile monitoring and production, the KRK GoAux speakers are perfect for that application because they come with stands that enable you to point the tweeters directly at your ears.

They also come with a handy little carrying bag, which is nice because you can easily bring them to a session. If you’re recording someone at their project studio and you want to bring your own gear, it’s super easy to carry the GoAux along with you.

If we can circle back to headphones for a minute; I’m interested in how headphones are developed by a company that’s primarily known for studio monitors. In the conceptual phase, when you’re making a new headphone product, do you visualize them as a scaled-down version of a studio monitor? Or is your approach completely separate from how monitors function?

It’s separate, because the acoustic field provided by headphones versus monitors is different. The overall thought is to make the headphones as accurate as possible. Other than that, the design and the thought processes behind the design are completely different because they’re a completely different acoustic beast.

One of the questions that some newer users might have is: well, I see the specs on the headphone goes from 20Hz to 20kHz and that seems to be the full spectrum range of human hearing (and even beyond). Why? I’m looking at studio monitors and they don’t go down to 20Hz. Can you maybe explain the logistical reasons why specs on a headphone might be different than specs on a physical speaker?

It has to do with proximity to your ear. Sound is an air particle velocity being moved in a wave form. It’s not like a sine wave, which is an electrical wave form that people usually see and are familiar with. Sound is a longitudinal pressure wave that has to move a certain distance. In order for you to have a low frequency pressure wave move a lot of air volume at farther distances, you have to have a very large radiating area. So that’s why you’ll see pro audio subwoofers get down to 20 Hz, they’re larger and able to move more air.

For headphones, it’s very close proximity to your ear, so you’re able to do that with a much smaller driver because you don’t have to move such massive amounts of air to get there. It’s

just the air volume of your ear canal.

If I’m a home recording artist working on an album or a song or whatever, and I want to implement a subwoofer, how do I start? I know directionality and placement is fairly important for my main studio monitors, but when it comes to subs, is it as important where that goes in the room or how that’s positioned or angled? Or is it maybe a little less just because the waves are so much bigger?

It’s not as important; subwoofers are typically omnidirectional. However, because you hear from both your ears, you can detect directionality and where the subwoofer is positioned in the room. So just because the sub is omnidirectional, that doesn’t mean you won’t know its location, and if it is sitting off to the right side, you’re going to hear the sounds coming from the right side, especially with the initial transient response. Now, once it blooms and fills the room, it’s kind of all-encompassing, but for that initial kick drum hit, you’ll be able to tell it’s coming from that area. So, at least some placement consideration is important.

Typically, you’ll want it somewhere acoustically centered between your monitors. That could be two subwoofers that sum acoustically to the center. Or, if you have one subwoofer, then it’s typically placed in the center of your room between the left and right monitors.

My final question -- are there any tips you can give to someone just starting out assembling all of their studio pieces or content production hardware? Where would you begin? To look at studio monitors, what are some of the most important features that people should initially look for when shopping?

I would say that the most important thing is to get a monitor that’s suitable for your setup. If you have a very small room and setup, you’ll want to keep to the more compact side of the monitoring situation so they can be comfortably placed in the room where you need them. You don’t want 10-inch, three-way monitors in a small space, it’s just not going to work. So, get something suitable for your size—that’s probably No. 1. And then No. 2 is to purchase from a brand you’re very familiar with and has a good recommendation from others in the industry. Most monitors these days have good frequency response and phase response. Obviously, I’m going to recommend KRK to everybody, but as long as you are used to the monitor that you choose in your room, you really can’t go wrong.

MORE at www.

Much like having a high-quality webcam with good lighting in your room, your audio quality also has an immense impact on your brand as a streamer.

And one of the easiest, and most cost efficient ways to improve your audio quality is by investing in an audio interface.

If you’ve done any research into what you need for a streaming setup that gives you modularity for the long term, it’s likely you’ve also read about audio interfaces.

But for a device that’s commonly associated with musicians and producers, what benefit does an audio interface provide to streamers?

What is an Audio Interface?

In order to understand the various benefits that an audio interface provides, it’s important to know what it actually does.

Put simply, an audio interface converts an analog signal into a digital format that can be read by your computer.

This is what allows musicians to record vocals or instruments in a digital environment. Or in this case, streamers to use analog equipment, like microphones that use an XLR connection, in software like OBS.

An audio interface effectively bypasses your computer’s internal sound card. And any audio processing now occurs on the interface as opposed to the internal system, which segues perfectly into one of the most important benefits of using an audio interface, latency-free monitoring.

These are the main benefits to using an audio interface:

1. Latency-Free Monitoring

Monitoring is the act of listening back to, and analyzing any audio that you’re either putting out or receiving.

And any added latency can make this process incredibly frustrating, which is typically a symptom of using internal sound cards or builtin drivers native to your system while recording or streaming live.

For example, a singer will need to hear themselves in their own headphones during the recording process. If there is any delay (latency) to their signal during this process, it will be difficult to sing on tempo.

And in a situation like livestreaming, you

don’t have post production to clean up any mistakes so having real-time feedback is crucial to delivering an exceptional performance.

Put simply, if you’re streaming a video game to thousands of viewers, you’re going to want to hear your mic signal, and any other audio sources without any delay.

Latency-free monitoring means:

• No delay on what you hear on your end (this is particularly critical with vocals)

• You’ll get a good sense of what the audience is hearing

• You’ll have full control of final output of all audio sources in your headphones and can adjust the level to what’s comfortable for you

• Saves processing power on your computer since it’s occurring through the device instead of internally


On its own, a microphone produces a very low, essentially inaudible signal that’s referred to as a mic level signal.

The typical mic level signal is anywhere between -60dB and -40dB.

So in order for a listener to hear this, the signal needs to be amplified to a nominal level.

And this is precisely what a preamp does.

A preamp, or often times referred to as mic pres, simply applies gain to a signal to bring it up to a nominal level for recording or streaming.

And one of the main benefits of an audio interface is that they will be built with higherquality preamps (like the Mackie Onyx Preamps) which produce a clean signal with low noise.

A preamp can affect your audio signal in several ways:

• How “clean” the amplified signal is

• The amount of noise introduced in the signal once it’s been raised (often referred to as “hiss”)

• The tone, or “character” of your signal. Also referred to as warmth.

• How much gain you can actually apply


Let’s get one thing out of the way first — are XLR microphones inherently better than USB microphones?

Well, the answer is both yes and no.

In fact, in a lot of cases the microphones

Better Preamps XLR Microphone Connection
WhyAudio Interface

Why You Should Use an Interface for Streaming


are made with the same, or similar internal components.

The main benefit to having an XLR connection is the modularity it provides whereas using a USB microphone is all about convenience and having a minimal, low-cost setup.

The main difference with USB microphones is that they have a built-in analog to digital converter, effectively acting as the audio interface itself. This is also what allows you to monitor your computer’s audio through the USB mic.

With an XLR connection, you’ll have a much wider selection of microphones to choose from including any pieces of external

analog equipment like preamps, interfaces, and more.

For the long term, and as a streamer who may be looking to improve their setup overtime, using a standalone interface with an XLR microphone will offer you much more opportunity to expand your setup and invest in higher-quality equipment.

What if I only Want to Use a USB Microphone?

As stated earlier, a USB microphone actually acts as an audio interface itself, though the benefits it provides are not quite comparable.

The main reasons you would opt to use a USB microphone are:

• Convenience: USB microphones offer a simple, minimal solution to a setup without the need for various pieces of equipment.

• Cost: On average, USB microphones can be much cheaper that buying an XLR microphone, interface, and cables to connect everything. But again, it comes at the cost of potential sound quality and better preamps.

If you plan on adding any instruments to your streaming setup or upgrading your microphone in the future, having a standalone audio interface gives you that modularity.

However, if all you plan on using is a microphone, then a USB-only device, albeit limited for the reasons mentioned, will give you what’s needed for a simple streaming setup.

So you don’t actually need a separate interface if the USB microphone already suits your needs as a streamer.

4. Better Sound Quality

At the surface, it’s easy to explain that an audio interface offers better sound quality.

But what does “better” actually mean in this context?


First and foremost, the converters in an interface are going to be higher quality than what’s offered by a built-in sound card.

To most (without a trained ear, that is), the difference may not be noticeable, but it’s there.

An audio interface gives you the ability to record at a high bit depth.

A higher bit depth will capture a higher dynamic range of the audio being recorded.

Sample rate defines the number of samples taken from an analog signal per second. In other words, the higher the sample rate, the more the audio source is sampled per second.

To offer a real-world example, many CDs use tracks that are exported in 16/44.1 meaning a bit depth of 16 and a sample rate of 44.1kHz.

This is typically more than enough to produce a high quality sound without losing any important dynamics or audio information.

As for recording, many stick with using a bit depth of 24.

Most, if not all interfaces offer highresolution recording with 24 bits and a sample rate of up to 192kHz. However, there is a practical limit to the bit depth you use as anything higher may begin to deliver diminishing returns like large file sizes or intense loads on your CPU.

So, Which Audio Interface should I Buy?

Here are our recommendations for audio interfaces that are perfect for livestreaming and those just beginning to build their setup:

1) MProFX6v3

The Mackie ProFX6v3 features two awardwinning Onyx preamps and a versatile GigFX engine, allowing you to get colorful highresolution recordings with sample rates up to 192 kHz. Paired with the Waveform OEM software bundle, you’ll have the ultimate workstation for recording, streaming and creating content.

• High-resolution Recording via USB

• Award-Winning Onyx Preamps for Pristine Audio

• Powerful FX Engine with Customizable Presets

• 48-volt Phantom Power

• Headphone Output with Independent Volume Control

• Rugged Design

2) Onyx Artist 1x2 Audio Interface

A traditional, easy-to-use audio interface with no frills, just great sound and quality preamps.

• High resolution recording up to 24 bits and 192kHz

• Latency-free direct monitoring with headphones and/or speakers

• Mackie’s proprietary Onyx Preamps that offer high gain with ultra-low noise, perfect for vocals

• One XLR input and one Line input

• Built-in vocal processing presets to sound great with any microphone

• Built-in voice changing FX

• Bluetooth connectivity

• 48V phantom power

• And much more

Ultimately, the best audio interface for your streaming setup will depend on what you’ve decided will fit best into your setup and streaming production needs.

[Editor’s note – we wish to thank Mackie for granting permission to republish this how-to guide that originally appeared on their website. To learn more, please head to]


DI Boxes: Yourself Started

If you’ve spent any amount of time in recording or watching studio recording breakdowns, you’ve likely heard the term “DI” or “DI box” thrown around quite often. It’s so commonplace that often it’s accompanied with little-to-no explanation.

What is a DI Box?

DI boxes, or “DI’s” (standing for Direct Inject) are primarily known for use in guitar and bass applications in the studio. However, their function is to correct mismatched impedance signals between instruments and

equipment. For example, guitars have a high impedance, unbalanced output that is prone to pick up noise and degrade signal over longer distances. A DI box will convert that signal to a low impedance, balanced signal that will be compatible with outboard equipment and reduce noise for longer runs.

What is Impedance?

In simple terms, impedance is the opposition of a circuit to electrical current, or how much the circuit impedes the electrical flow. Unbalanced signals, like those in a guitar’s output jack,

would have higher impedance and thus need to be converted for use in microphone preamps or mixers without picking up a lot of undesirable noise.

High-Z vs Low-Z

“Z” is the letter and mathematical value given for impedance. When you see “High-Z” or “Low-Z” labeled on an input or output, it’s telling you the type of impedance signal coming from that respective input or output. The Hosa DIB443 Sidekick Passive DI Box, for example, has a switch on one side for you to select an instrument


Getting Started

(High-Z) or line-level (Low-Z) input, and a “Low-Z” output on the other end.

DI Box for Guitars

The most common use for DI boxes are with guitars. In a studio setting, these are most often used to track a clean guitar signal at the same time as an affected signal through an amplifier. This helps the performer play and react to the sound being recorded while preserving the original signal should that need to be altered or reamped later. This can save a lot of time without having to re-record

or perform already recorded parts.

DI Box vs Reamp Box

These two are often confused for each other, or at least why each must exist independently. The reason you want to use a designated Reamp box for any reamp functions is because it’s built to perform the opposite function, taking a low impedance signal and converting it to a high impedance signal that guitar amplifiers are supposed to receive. Reamp boxes can also be useful in very complex live setups where you send the guitar through a DI into line-level

processing that then needs to be converted back to instrument-level to feed into an amplifier on stage. With so many players opting for digital and computer processing, that kind of setup continues to grow in popularity.

What is a Wet/Dry Signal?

In guitar terminology, a “wet” signal is a fully affected sound. If you have your guitar running through pedals, effects, amplifiers, or anything that alters the original clean signal out of the guitar, that is considered a “wet” signal. The “dry” signal is the unaffected signal from your guitar.


DI boxes allow you to preserve your “dry” signal even while manipulating it with other pieces of equipment.

Passive vs Active DI Boxes

If you’ve spent any time looking at DI box options, you’ll have noticed there are “active” and “passive” options. The most common DI boxes are passive, which use an internal transformer to isolate ground-level voltages and eliminate any ground loops. The impedance will be matched to that of a low-Z microphone preamp. Passive DI boxes are straightforward and simple in design, making them the less expensive option.

Active DI boxes, on the other hand, include an active preamp, creating more headroom than passive DI boxes. This includes a “signal boost” for the preamp, which also helps preserve a stronger signal for very long cable runs, though any boost will often be accompanied by some kind of sound “coloration”. Active DI boxes can

be a favorable option for keyboard players or instruments that use active electronics. These will require power, whether in the form of a battery or external power supply.

Things to Look For in DI Boxes

DI boxes perform a relatively limited function, so there’s not much deviation between models. However, here are things you would most commonly see and what function they perform.

Ground Lift

Balanced signals are designed to cancel noise by carrying 2 duplicate signals with reversed polarity. However, introducing more equipment and even the environment itself can add hums and noise to balanced signals. The “ground lift” on a DI box disconnects pin 1 of the XLR on the output in order to break the ground loop, if necessary. As a general rule, the switch should be left in the “ground lifted” position unless otherwise needed.


Sometimes referred to as “throughput”, this output is for the unaffected signal to pass. This is necessary for any guitar tracking where you need a “wet” and “dry” signal to be recorded simultaneously, or you have multiple signal paths. On the Hosa DIB-443 Sidekick Passive DI Box this would simply be called the “output”.


This is a built-in attenuation on any DI box, reducing the levels somewhere around -20db. A pad is especially valuable to keyboard players that use line-level outputs rather than instrument level. Typically, this would mostly be reserved for any active electronics going into the DI box.

Cables You’ll Need With a DI Box

While there is some variation, most common uses begin with instrument cables. Instrument cables are shielded to resist noise and


interference since they carry unbalanced signals. If you are using the thru/bypass, you would need another instrument cable running to your signal chain.

While the outputs in a DI box are usually XLR, once the DI box has converted the high impedance signal to low impedance, you’d need a balanced interconnect with the proper connector types for your equipment.

To purchase the Hosa Sidekick Passive DI Box or learn about the functions and features, visit the product page at products/analog-audio/instrument-interfaces/ dib-443.

[Editor’s note – big thanks and appreciation to the HOSA team for allowing us to share this how-to guide article which was originally published on their website. To learn more, please head to]


Exploring the Different

What’s The Difference Between Condenser Microphones, Ribbon Microphones & Dynamic Microphones?

While you can put a condenser, ribbon or dynamic microphone on any source and get perfectly adequate recordings, there is an art to capturing noteworthy sounds in the studio. Like colors on a palette or different types of brushes, your choice of microphone imparts distinct character and textures on your sound. Each style of microphone offers its own feature set, one that enables the mic to capture a vocal or instrument in a specific way.

Ever wonder why engineers gravitate towards putting a large diaphragm condenser mic on a vocalist? Have you seen ribbons or small diaphragm condensers placed over a drummer’s kit? These standard miking techniques are classics for a reason. Each decision is based on the microphone’s design and character while also factoring in the same elements for the instrument at hand.

Throughout this blog, we’ll dive further into the world of condenser, ribbon and dynamic microphones to highlight the differences between each of these mic categories. In addition, we’ll discuss which microphones are tailored to specific instruments and explain why they make these sources shine so brightly.

Condenser Microphones

Condenser microphones may be the most logical place for us to start, as all of the mics we make are in this style. Technically speaking, these microphones employ a fairly simple system with a capsule that features a thin membrane and metal plate. When sound waves from a singer or instrument arrive at the capsule, it causes the thin membrane to move closer and farther from the backplate. This movement changes the voltage across the backplate and is converted into an electrical signal.

The next step in the process is boosting this signal. The microphone is supplied with power through its impedance converter, either a vacuum tube or FET (or Field Effect Transistor) circuit. This power is delivered by your external power supply or 48V phantom power on your mixer, mic preamp or interface. With the power supply boosting your signal via the impedance converter, your audio signal can now be delivered to the mic input on your preferred recording device.

With the technical mumbo jumbo out of the way, let’s talk about the benefits of condenser microphones. Thanks to its


Types of Microphones

Condenser, Dynamic & Ribbon Mics

unique capsule design, condenser

mics are known for being extremely detailed or what some might describe as accurate. They also offer the widest frequency response (with special regards for higher frequencies) and many employ a number of polar patterns to suit your specific source.

Within the world of condenser microphones, there are some variations in general size and impedance converters. Each of these different condensers brings something new to the table, so it’s important to understand how they are sonically unique. Let’s break it down.

Tube vs. FET

As previously mentioned, a condenser

mic uses tube and FET circuits as impedance converters to amplify its signal. Soyuz makes tube and FET versions of both our 017  and  013 Series  microphones, while the  023 Bomblet  employs a FET design. Why make two versions of the same condenser microphone? There are subtle difference between tube and FET microphones that can be used to your advantage in the studio.

Tube microphones are generally associated with a sense of warmth, as the circuit and topology of these designs generally give a smoother response that softens edges and gives your sound a nice glow. Engineers would probably reach for a tube condenser on a vocal performance due to this flattering quality. Listen to the two vocal examples and compare the sound of a tube and FET microphone in action by watching to these YouTube videos:




FET microphones are known more for their fast transient response and accuracy. This means that the microphones are capable of capturing a source with more precision than their tube brethren and tend to give you a more forward and up-front feeling. Compare the two drum overhead examples below and listen to how the FET mic perfectly captures the kit with a snappy attack.

• watch?v=YtP3SnnMNcc

• watch?v=tAPB5byhX6I

For a deeper dive into the differences between tube and FET microphones, check out our comparison at articles/tube-vs-fet-mics.

Large Diaphragm Condenser Microphones vs. Small Diaphragm Condenser Microphones

Condensers can also be divided into two groups based on body size; large diaphragm condenser microphones and small diaphragm condenser microphones. At Soyuz, we make both varieties. The 017 and 023 Bomblet fall into the large diaphragm condenser category while the 013 is a small diaphragm condenser. Aside from physical size differences (it’s easier to fit small diaphragms in tight quarters), they also have some different sonic subtleties.

Large diaphragm condenser microphones excel in the lower frequencies with a much broader and deeper response. Thus, these mics are often associated with a more full-bodied, lush sound. This is perfect for vocals and voiceovers or wherever you wish to capture the most size from a source. Small diaphragm condensers are typically known to have a superior transient response and extended high frequency response. Therefore, they are typically used for getting a more accurate and detailed sound.

Dynamic Microphones

If you’ve been involved in audio in any way, you’ve probably held a dynamic mic in your


hand. Think Shure SM57 and SM7Bs. These are industry-standard microphones because of their simple designs and ability to take all kinds of abuse.

The main kind of dynamic mic utilizes a moving coil mechanism. This design is a lot like a speaker, as a coil is glued to a membrane and a magnet surrounds the coil. As sound waves hit the microphone, the membrane moves, as well as the coil. This movement of the coil within the magnetic field creates the electrical signal needed to capture audio.

Overall, dynamic microphones have a slower transient response, so they don’t afford the same accuracy as condenser microphones. What they lack in detail, they make up for in durability, affordability and the way they handle sources with extremely high sound pressure levels (SPL). Thus, dynamics have often been used in the live sound setting and in the studio for loud sources like guitar amps, kick drums and snare drums where you may want to smooth out aggressive transients.

Ribbon Microphones

While the ribbon microphone is often given its own category, technically speaking, it’s another type of dynamic mic. Ribbon mics work like this… A ribbon (a thin, corrugated strip of metal) is suspended between two magnetic poles and acts as

a diaphragm and transducer element. As air passes through the microphone, the ribbon moves and creates a voltage.

The voltage created by the ribbon element is extremely low. Since these mics are predominantly passive, the onus is put on the impedance of the mic preamp to do the heavy lifting. By putting this much emphasis on your choice in mic preamp, a ribbon microphone’s tonality can shift based on whatever you decide to plug it into.

Ribbon microphones have a number of wonderful benefits. They deliver warm and smooth sounds with lots of low end and a natural roll-off in the higher frequencies. By design, they are bidirectional with a figure-8 polar pattern. This means a ribbon microphone is capable of picking up sounds from both sides of the microphone while excluding sounds from the sides of the mic. Therefore, ribbons make wonderful room mics and are perfect for miking drum overheads, guitar amps and more.

While Soyuz doesn’t currently make any ribbon microphones, the 023 Bomblet is in the same ballpark in several ways. This large diaphragm FET condenser microphone features a thick lowend, smooth top-end and a natural compression that will please ribbon mic enthusiasts while still being forward and present to sit in front of any mix.

If you’re looking to add flavor to your dynamic or ribbon microphone, The Launcher is an excellent resource. Our inline mic preamp adds 26 dB of gain and vibes to any dynamic or ribbon microphone that runs through the small white box. Simply, run an XLR cable from your mic into The Launcher and another XLR from the back of the unit to your mixer, preamp or interface, and you’ll benefit from a bountiful boost of gain and warm saturation.

Final Thoughts

The world of condenser microphones, ribbon microphones and dynamic microphones is a vast and wonderful place with lots of options at your disposal. While we’ve outlined these key differences in the way the mics work and sound, the beauty of recording is that your choices are totally up to your discretion. There is no wrong way to record a sound source and some of the most brilliant moments in recorded history have come when engineers bucked conventions. Try out condensers, ribbons and dynamics in your own unique way and find out what works best for you.

[Editor’s note – Soyuz Microphones has graciously granted permission to share this article in our magazine, which was originally published on the Soyuz Blog. To learn more, please head to https://]


How to Mix with Studio

Benjamin Ricci

How do I mix music with headphones? What are the best headphones for mixing and music production? And why should I be tracking with headphones in the first place? These are very common questions for first-time home studio users, and KRK and Performer Magazine have teamed up to help guide you towards the best studio monitor headphones for your recording needs.

Welcome to the content creators and mobile production issue, where we hope to provide real-world advice for setting up your first home recording studio or on-the-go rig, co-presented by Performer Magazine and our title sponsors KRK Systems. In this article, we’ll take a closer look at studio monitor headphones, specifically why you should be using a secondary monitoring source during your recording sessions, which types of studio headphones make the most sense for your applications, and how to listen to a headphone mix to make music production decisions. KRK has been kind enough to send us a number of pairs from their KNS series of studio monitor headphones over the years, including the KNS 6402 ($99), and the KNS 8402 ($149) cans.

We feel the KRK KNS lineup offers great quality at each of their respective price points, and

you can read our expert reviews of each model in previous issues.


To start, why should you bother with a set of headphones at all? One of the first purchases we see many first-time home studio users make is a pair of studio monitor speakers. And that makes sense; you want to be able to listen to your sessions as you’re tracking and mixing, and studio monitor speakers are voiced to deliver a true audio response without the coloration of, say, hi-fi speakers.

But there are a few reasons that studio monitor headphones are important. For starters, it’s always a good idea to reference your work using multiple monitor sources. What you’re hearing out of your speakers might sound great now, but it’s important to hear what you’re recording through other sources to get a sense of how the music will sound on various playback systems. And of course, one of the most popular methods for music consumption is through consumer-grade earbuds and headphones. So, auditioning your tracks through a set of headphones makes sense, not just to hear what the end-user will experience (even though studio monitor headphones are voiced a bit differently than consumer-grade models in most instances), but also because the music will exhibit a different spatial depth when listening on a closed auditory system, like headphones, than an open system

(such as speakers moving air in a room). In short, the soundstage will be more pronounced with the sound directed at both ears in true stereo, without the room environment and sound treatment affecting what you hear.

Second, we recommend studio headphone monitoring during both tracking and mixing to hear the subtler nuances of the tracks you’re working on. Especially true when layering lots of complex overdubs, it can be more difficult, at times, to truly hear each of your tracks in true separation even through the best of studio monitor speakers, if things are becoming a bit congested in the mix. At times like this, it’s especially prudent to reference your work through headphones to isolate any issues that may be causing muddiness or spatial confusion, which we feel can often be done more accurately through a pair of properly-voiced studio monitor headphones.


We’ll keep this short, while open-back designs might be great for pure audiophile listening, we don’t recommend them for home studio use, other than to audition final masters from a listener’s perspective. For tracking and mixing, however, we exclusively recommend closed-back headphones that were specifically designed and voiced for studio recording. The


KRK KNS models we’ve tested are an ideal choice for recording. They are suited especially well for mixing, specifically in the way they are voiced, and are very nicely tailored for on-the-go production in terms of comfort and portability. The KNS lineup would be ideal for tracking and more intense critical listening. And the 6402’s offer the most affordable entry point to studio monitor headphones without sacrificing quality.

Open-backed designs can introduce unwanted bleed from the room, while in turn also bleeding out audio  into the room. Neither is ideal - you don’t want any audio seeping into your brain that isn’t coming from your DAW and you don’t want loud audio from your session throwing off anyone else trying to work in the space. So open-backed designs, at least as far as we’re concerned, are a non-starter for the studio.

Of the KNS models we’ve tried before, we had this to note in our initial evaluation: “In our tests, we were pleasantly surprised at the flat response and colorless reproduction the KNS headphones had to offer. Too often at this price point, some sort of coloration seeps in and can affect the way you hear your mixes, and ultimately alter the way your tracks sound (and not always in a positive way). Most of the time, we’ve found that adds up to an increased (and often unnecessary) bass boost. Thankfully, this wasn’t the case and these new KRK studio headphones offered a “what you hear is what you get” type of vibe, exactly what you want in the studio. Bass was present and clear, without an over-emphasis on low-end frequencies. No mud, no fuss.”


There are some things to consider when it comes to specs. As we’ve mentioned, our recommendation is to stick to models specifically designed for studio usage, and not necessarily consumergrade headphones. Many of those models offer coloration to the sound being reproduced, which in a recording or mixing situation, is not ideal. Especially prevalent are overzealous bass-boost “features” that will disrupt the natural bass curve of the music coming from your DAW. Instead, focus on studio monitor headphones that not only feature comfortable earcups and headbands, but also flat frequency responses of at least 20Hz - 20kHz (many higher-end models will offer reproduction at both higher and lower frequencies, and though we won’t get into the science of hearing in this article, even frequencies that are technically outside of the audible range for humans can make a difference in what you’re hearing) and good-sized drivers.

In speakers, drivers will usually be measured in inches (in the United States), but for headphones look for specs of at least 40mm and higher. The larger the voice coil, the more air can be moved, not just resulting in louder volume (your headphone amp will play a solid role in volume, as well), but also the range of frequencies that can be accurately (and that’s the key word) reproduced. In plain English, the more accurate the sound, the better attuned to the music you can be when focusing on your session work. Trying to mix or record in an environment where you’re not hearing back a true representation of what you’re laying down in your DAW can have disastrous effects on all aspects of your project, most notably in over- and under-compensation in both high and low ends of the spectrum. The last

thing you want is to boost all the highs only to find out they were fine all along, you were just using lousy headphones and now your tracks are entirely too bright and trebly.


Again, listening to multiple sources will give you a better overall perspective and understanding of what’s going on in your music. When it comes to headphone monitoring, there are some specific items to note. To start, headphones allow you to really isolate the stereo imaging you’ve created in your tracks. Having both the left and right channels directed at each individual ear in isolation can make for perhaps an unnatural, yet revealing study of how you’re placing instruments in 3D space. Issues with panning and stereo placement can become instantly evident (even sometimes exaggerated) when listening back through proper headphones, and corrected efficiently before mastering.

Second, you’ll be able to more faithfully tune into quieter passages and layers that have been more buried in the mix. Is that synth part audible enough in the bridge? Does that acoustic guitar need to be panned in the verse so it’s not competing with the piano that’s dead-center? Or should we double a vocal line here where it’s sounding a bit thin? Choices like this can often be made more intelligently after referencing the track through speakers first, then headphones to isolate things in a more distraction-free manner.


Until now, we’ve been dealing with headphones in isolation. But they need to be plugged into something to work, right? And you might find yourself in a situation where more than one person needs to hear what’s going on at once. That’s where headphone amps come into the picture. Now, your audio interface will likely offer monitoring options either on the front panel (if they’re smart) or on the rear. But if your interface only has one headphone port, you may want to look into a dedicated studio headphone amp designed specifically for recording needs. These devices will often offer very clean power and multiple outputs and independent volume controls – meaning you can have several people listening in at once, each with their own settings.


We hope this installment has helped guide you on your way to choosing the best studio headphones for your needs. Keep in mind that this series is aimed at the beginner home studio user in an effort to dispel common myths about home recording, and to make the entire process much less intimidating than it might seem at first.

Head to to learn more and to find the products that will fit YOUR home studio needs.

May We Present: The “2023 Conty Awards” OUR ANNUAL MOBILE CREATORS BUYER’S GUIDE
Benjamin Ricci

If The Office taught us anything, it’s that we as a society love awards. I mean, who wouldn’t want a shiny new Dundee for their mantlepiece? So this summer, we’re presenting our inaugural “Conty Awards,” [‘conty’ being a take on “content,” not a nod to the recently departed manager of Tottenham Hostpur] which is our version of the Dundees, to some of the coolest products in the following categories we came up with while trying to make the next great song on our morning subway commute to the office.

So, without further ado, may I present our first-ever Conty recipients…

Our Favorite New (and Old) USB Microphone:

Audio-Technica AT2020USB-XP

The AT2020 lineup of USB mics has been our go-to for years when it comes to easy, great sounding plug-n-play recording mics for onthe-go usage. Be sure to check out the brandnew AT2020USB-XP which gives streamers, podcasters, and other creators the award-

winning AT2020 sound with plug-and-play USB-C operation and a whole lot of extras. Three selectable noise-reduction levels and an automatic gain control allow you to adjust the clear, natural sound reproduced by the mic’s high-resolution A/D converter (up to 24-bit/192 kHz). Plus, sound can be monitored directly from the mic with no latency, mixed with computer audio, and silenced with a touch of the capacitive mute button.

The original 2020+ earned our praise, as well, and still remains a killer tool at the price point. Feature-wise, it’s not overwhelming, a 1/8” connection for headphones, and two volume wheels, one for headphone volume, and the second is a mix control that allows the blend of mic vs. playback signal. Connecting is simple: plug and play. It works with Mac and Windows software easily, and with pretty much every recording application that recognizes USB devices.

It’s a condenser mic, and its frequency response is pretty flat, leaving plenty of room for external EQ (if desired). For acoustic and electric

instruments it works really well, with plenty of range and dynamics. On vocals, it doesn’t color the sound, and the ability for a singer to control their own mix and headphone levels at the mic is great, especially for those singers that want to run their own sessions without an engineer.

Overall, it’s a quality-built mic, with a quality sound, and is perfect for many uses: instruments, vocals, streaming, TikToks, voiceovers or anything else you can imagine.

Best Mobile Monitors:

KRK GoAux 4” Monitors Powered Portable Studio Monitors

Coming in as a pair, the master speaker sports 1/4-inch TRS balanced, RCA, and 1/8-inch aux inputs on the rear. These are Bluetooth-enabled for wireless connectivity and also feature a USB-B input. With additional EQ selection, there are options for Low and High frequency (flat, +2dB, and -3dB), allowing adjustability to suit the listening environment. Inside the ABS enclosure resides a four-inch Woven Glass Aramid Woofer, detailed in KRK’s trademark

Audio-Technica AT2020USB-XP KRK KNS-8402 Headphones

yellow, and a one-inch soft textile dome tweeter bringing a max peak SPL of 108dB, with a total of 100 Watts of Class D power. On the front, you’ll find a handy 1/8-inch headphone connection and an ARC (Automatic Room Correction feature) microphone input. A pair of angled stands for optimum positioning are included, as well as a padded nylon bag.

These aren’t just “throw them on the desk and go” speakers—an included ARC mic is meant to help configure the monitors to your space, with a series of sounds pumped through the speakers, and the ARC mic picks up these tones and calibrates the speakers. Perfect for home studio use AND adjusting to life on the go as a mobile producer who works in various spaces at any given time.

Considering these are meant to be portable, with the carrying case and ARC calibration feature, it’s a pretty comprehensive kit. Setting these up and tearing them down in a hotel room on a daily basis might be a bit of a pain for a band on the go, but the overall fidelity is excellent. A great way to utilize these would be tracking and mixing in your own space, then using these in different spaces, such as living rooms or

bedrooms, to listen back to mixes for reference. Content creators might want to look into a set of these when on assignment. Even better is taking them to a dedicated studio and using them to reference against various studio monitors― making them a great tool for A/B and reference mixing.

In comparison to a set of studio desktop monitors that aren’t meant to be so mobile or a set of consumer-grade desktop speakers, these certainly side closer to the studio versions by a long shot. Even if these did not leave a desktop, they’re certainly a great pair of small-format studio monitors that can deliver.

Fave “On-The-Go” Headphones:

KRK KNS-8402 Headphones

With an over the ear design, the sound isolation is great, blocking out 30dB of outside noise, making these excellent in a live tracking situation. Using the inline slider cable allows the user to control the volume into the headphones. Ever have a session where a player keeps asking for constant (and inconsistent) volume

adjustments in their cans? This will put them in the driver’s seat in controlling their own overall volume without a separate monitor mix. Speaking of volume, with a max 124 dB (which is about the same volume level as a jackhammer) there should be no issues with drivers getting damaged or fatigued in even extreme situations.

Comfort-wise, these are excellent in a physical sense, with the fantastic padding, and lightweight feel. Going into long sessions the audio fatigue was non-existent, especially with the in-line volume control, allowing the player to adjust the levels to their own taste. With a frequency response of 5hz-23kHz they can cover pretty much anything from sensitive acoustic to mega umlaut metal. The audio quality was excellent, the Neodymium drivers not only tend to make things lighter but seem to roll off harshness in the higher ranges, while still maintaining a nice and full low-end response that maintains definition.

So, for tracking, these are really nice, but one thing that always helps in a mixing setting is being able to switch between headphones and

KRK GoAux 4” Monitors Powered Portable Studio Monitors

room monitors, hunting for the audio issues one of the items is either missing or coloring in some manner. We did a few bits of back and forth between our monitors and these and found a nice balance where these seemed to have a lot of characteristics of monitors.

There’s a school of thought that mixing only on a set of headphones isn’t a good practice. However, with a set of these, since you’re getting very similar responses to monitor speakers, that school could easily be dismissed.

Overall, these are quite nice, and for tracking and personal use (our tester loved using these

with his modeling pedalboard) a set of these is a no brainer. Mixing with a set of these was a pleasure and worked nicely with monitors. There seems to be a lot of mileage to be had from putting these in your home studio, regardless of the road taken or the destination.

Best Small-Format MIDI Controller:

Donner DMK-25 PRO MIDI Controller

The Donner DMK-25 PRO MIDI Controller comes in an attractive blacked-out finish, and a two-octave keyboard that’ll get any budding producer started, whether it’s adding bass lines, melodies, drum samples or chord changes to

their tracks.

It’s all USB powered, so you can plug it into your computer without the need for big ol’ power bricks. From there, control the soft synths and drums in your DAW using the comfortable keyed, or the built-in pad controls. There are also faders and assignable sliders to help control some of your most frequently used parameters, which makes tactile control over your sound a breeze.

The OLED screen is super nice and bright, and easy to read which means navigating menus isn’t a chore. Once you’ve got your sounds loaded, both the keybed itself and the drum pads are

PreSonus Atom Controller

velocity sensitive, so they’ll respond to light and heavy touches accordingly over MIDI. Meaning you can add more dynamics to your music in the way you would on a traditional keyboard or drum set.

Finally, you’ve got the ability to use the arpeggiator, chord functions, and map the keyboard to scales as opposed to the standard chromatic keyboard, making soloing and melodic playing easy for everyone.

The new MIDI controller from Donner is ultra-affordable, compact and a joy to play. We recommend it for a backpack rig, or the starter piece for your first home studio.

Best Small-Format Podcast Audio Interface:

Focusrite Vocaster Two USB-C Podcasting Audio Interface

Podcasting started out pretty simply: a computer and a microphone. Then the ability to have someone call in, like a radio show, and the hardware and software needed for more (and better) microphones meant more gear that in many cases were overkill for most applications. Focusrite’s Vocaster Two brings a lot of streamlined functionality, and adjustable audio options in a small format.

With two XLR inputs that offer phantom power, they’re designated “guest” and “host”, along with a headphone jack for each. A 1/8” connection allows for external inputs from a phone. Taking things one step further, an additional output for a camera’s audio input now brings video options to the content creator easily. As it functions as an audio interface, there are two 1/4” speaker outputs for mixing and playback. The control surface is easy to navigate with soft touch buttons that light up when engaged, and each mic’s controls are independently adjustable. A visual gain display allows the user to see audio signal strength easily for level monitoring.

Additional functionality comes with

Focusrite’s Vocaster Hub software as a free download. This can also run the physical functionality of the Vocaster, as well as act as a mixing console, with an auto gain function that sets the levels automatically, as well as loopback functionality, meaning audio can be imported back in while the recording is happening, perfect for pre-recorded intros and outros, and hopefully when the podcast is successful, pre-recorded advertisements. With selectable EQ profiles for the microphones, it can help enhance the sound based off of the microphone being used as well as the person speaking. Ever have a dark sounding mic, with a vocalist that has a deep voice, and have to really crank the gain to get the level up? Yeah, this can help solve this while maintaining a reasonable level.

We even used the Vocaster with GarageBand and Studio One 5 with no issues at all. It can function as an interface to pretty much every DAW recording software, but the package includes trial versions of Hindenburg Lite, aCast, SquadCast Pro + Video and Ampify Studio.

This really is a plug-and-play piece of gear, with no problem getting up and running in a snap. The added audio out to a camera can make this a super easy vlogger setup that can solve so many

cabling and routing issues of getting audio into a digital video device for YouTube, live streaming and content creation.

Most Feature-Rich Drum Pad Controller for Your Backpack:

PreSonus Atom Controller

Yes, it’s another pad-style controller, but it does what it does, and it does it well. Integration with Studio One is tight as a drum, no drivers or other nonsense to get in the way (at least on our studio’s Mac). We fired up a new session, opened up some of our fave drum samples, and everything mapped and functioned instantly with no fussiness. Playing finger-style was a breeze, and we were able to instantly lay down grooves and trigger samples without even glancing at the manual. That said, there are a lot of function buttons on the front panel, and it may be a bit daunting to a new user who’s never played with a controller like this before.

The Atom Controller is lightweight (we’d be surprised if it weren’t, considering it’s just passing data back and forth), so it it’ll slide nicely into a DJ backpack for gigs or studio work. The knobs and buttons all feel nice and firm, the pads themselves offer a good amount of touch sensitivity while still being firm enough to allow for the proper pressure-sensitive playing many of us have become accustomed to over the years.

All in all, not too much more to say. The price is right, too, at just $149. For electronic artists who’re already in the PreSonus ecosphere, it’s a no brainer. We did test it out with a few other DAWs, and since everything is assignable and velocity-sensitive, we were laying down softsynth performances with relative ease after a quick initial setup. This would be great for live setups, as well, since you can trigger loops and parts easily on your laptop while performing live in real-time with other hardware synths or drum machines.

Focusrite Vocaster Two USB-C Podcasting Audio Interface Donner DMK-25 PRO MIDI Controller

SARAMONIC Blink Me 2-Person Smart Wireless Microphone System

Hello! Effswell here, and thanks to a recent delivery from Performer Magazine, I’m thrilled to share my TOP FIVE FAVORITE FEATURES of the ALL-NEW Blink Me 2-person wireless microphone system from Saramonic.

NUMBER 5: Customizable home screens on the transmitters

Not only does the Blink Me system come with TWO transmitters (TX) for double the 2.4GHz digital frequency transmission fun, but these babies can easily be customized with ANY photo or logo of your choice.

For musicians, bands, and content creators, this feature provides an extra layer of customization that can highlight product images, brand logos, a

professional headshot portrait, or even just your social media handle; for every video project you shoot! Wearing each transmitter is as simple as attaching either one of the included magnets or clips to the back of the unit and getting started.

NUMBER 4: The Compact Size of the Blink Me

The total mass of this wonderful microphone system shook me to my core. With its included carrying case, the entire Blink Me System can be easily transported, set up, used for a project, and re-packed before some competing mic systems can even press “RECORD”! This advanced portability makes the Blink Me an ideal system for recording crystal clear, 24bit/ 48kHz interviews; which, for musicians in this digital age, are one of

the most effective ways for both established and new audiences alike to familiarize themselves with your personality and unique style!

NUMBER 3: A Dual Receiver with Super Powers

Not only does the Blink Me receiver (RX) feature its own touchscreen like the two transmitters, but it also serves as the CHARGING PORT for both transmitters. With a simple magnetic connection, your transmitters will charge DIRECTLY from the body of the Receiver; with either one or both transmitters attached. When making music or content on-the- go, this is a huge benefit because if you’re anything like me, you may be prone to losing physical items often. Don’t worry -- the Blink Me keeps every piece of


equipment together and the receiver boasts 24 hours of battery life on a full charge. With 6-8 hours of battery life for each transmitter without re-charging, the Blink Me allows for plenty of time capturing ideas in the field. The best part? It’s all by design.

NUMBER 2: Multi-Use Recording Functionality

Whether you’re capturing audio for a video interview, on-street reactions or even a product demo (like this one), the Blink Me will have you COVERED! This mic system can record in three modes: Mono, Stereo, and Safety Mono; each with its own perks. Mono will mix both mics on the same output. Stereo separates each mic into Left and Right mixes that are then isolated and

can be mixed together in post-production. Safety Mono is one of my personal favorite settings because like mono it mixes both input channels into the Left output AND ALSO records a Safety track to the Right output, which automatically ensures your audio won’t suffer from clipping because it’s captured at -6dB.


Almost every wearable microphone system on the market is compatible with cameras, but the Blink Me takes its versatility a step further by allowing users to record audio into their Android or iOS devices directly*.

This is achieved by accessing either the USB-C output for devices like tablets, phones, &

computers or the 3.5mm output, which plugs into professional-grade cameras with the help of the included TRRS audio output cable. Both outputs can also record at the same time to multiple devices. How cool is that?

*For reference, I’ve recorded the entire audio segment of this review’s companion video (out now on Performer’s YouTube channel) using the Blink Me mic system. I added a bit of noise reduction in post, and the sound in my review video is the final result. Visit

PERFORMER MAGAZINE SUMMER 2023 MOBILE ISSUE 45 GEAR REVIEWS for valuable information and purchasing details.

DONNER PC-02 Podcast Workstation

In the world of content creation, sometimes simplicity is key, but when you want to up your game, there are definitely some great (and more feature-rich) options out there. Enter the new Donner PC-02. Now, obviously this unit is meant to compete with podcast solutions from the likes of RODE, Mackie and TASCAM, to name a few, so how does it stack up?

Let’s start with the layout. It’s dead-simple to navigate your way around – the front panel features a colorful main screen that’s easy to get around, control knobs that feel great, and motorized faders that move very smoothly. Plus a bunch of colorful, assignable pads that feel great if you want to punch in your show’s sound clips or samples. Hell, you could even record an entire band’s demo using just this unit, if you wanted. Or livestream a show – it’s incredibly versatile.

Around the rear is your USB-C connection along with a host of line inputs and outputs, a micro-SD card slot, headphone outputs and several XLR inputs for dynamic and condenser mics. It’s essentially everything you’d need to get a podcast with a host and multiple guests up and

running quickly.

Recording is simple, too – as you have options to go straight into a PC or tablet/phone and use either the Donner software that’s specifically set up to work with the PC-02, or your recording software of choice. You can also record a session onboard using a micro-SD card, which means you don’t even need a PC at all, and this makes it invaluable for field recording and mobile productions and podcasts. Toss this in your backpack rig, and now you’ve got a mobile studio at your fingertips no matter where life takes you. Record your band’s rehearsal and listen back to those inspiring ideas you came up with during practice.

At $599, it sits in the sweet spot, pricing-wise, as it’s not gonna break the bank, but it’s not so low that it cuts out all the useful features, either. We dug our time with the PC-02, and plan to record all of our future podcast episodes using it.

Highly recommended.

Great sound quality, dead simple to use, affordable




NT1 5th Generation Condenser Microphone

RODE’s condenser microphones are well known for getting the job done, and they haven’t sat on their laurels with their newest version: the 5th Generation of the NT1.

The complete package includes the mic, shockmount, pop filter, and cables. For ease of use the shockmount has a threaded attachment point for the pop filter, and places it right where it needs to be for a vocalist. Did we mention cables? Yes, XLR and USB cables are included, as this is a dual format microphone, and the USB-C input is tucked in between the XLR pins. The large 1” gold sputtered diameter capsule has a cardioid pattern with a frequency response of 20Hz - 20kHz and a max SPL of 142dB.

Now it can be plugged into any mixer/ DAW inputs via the usual XLR connections, like most normal microphone setups, but the USB brings a few tricks to the table. There’s some extra software to download (it is Mac & Windows compatible) but opens up the doors to the mic being its own interface. The RODE Connect Podcasting software, RODE Central configuration manager and Unify, their virtual mixing software that’s meant for streamers and content creators, gives added flexibility to users in these settings.

Inside the mic resides its own preamp and DSP, and uses RODE’s Aphex Audio processing system for 192kHz audio resolution and 32bit digital floating point output. Not all DAW software is 32-bit floating point compatible, but RODE has step-by-step walk throughs, including our DAW of choice, Studio One. For those going “old school” and sticking with XLR, you certainly aren’t slumming it, but might be missing out on some opportunities, especially with workflow.

Putting this through its paces in a broadcasting/ podcasting setting it certainly does the job nicely, no surprises there, it has a very low noise platform, and is perfect for these applications. Yes, the USB side of things lean towards the content creators and podcasters, that makes sense, with excellent audio quality, and with the complete kit, it’s kind of a nobrainer for that!

However, the previous versions of NT1’s were well known for their adaptability, and this is no exception. We used this for lead and backing vocals, acoustic guitars and electric guitar amps, and it was remarkably amazing across the board. Its EQ profile was great against some high-end frequencies, softening things up like bright acoustic guitars, and some VERY twangy clean Telecaster bits. Vocal-

was a really nice surprise. It’s not super bright, but there’s enough chime in there to cut in a mix.

For a content creator/podcaster this is a really great mic, but might be overkill, especially for a user that wants simplicity and less of a need for extra software, especially if they’re only going to be using it in USB mode. For musicians, the coin is flipped; all the reasons you’ve had an NT1 on your wish list are still there, specifically in the XLR connection mode. The added software and USB connectivity could make some sessions easier and offer up some extra tweak ability. It’s one of those mics that could be in any respectable mic locker, but chances are it won’t spend a lot of time IN the mic locker.

very adaptable, excellent musical response


Might be overkill for non-music applications. STREET



1987 Tascam Porta Two Cassette Four-Track Tape Recorder


The Porta Two is a portable cassette fourtrack tape recorder manufactured by Tascam in 1987.

Teac/Tascam are a huge influence in home recording. The Teac 144, the first cassette fourtrack recorder, was released in 1979. The 144 kicked off the cassette recording revolution. The Tascam Porta One was released in 1984 offering battery power. In 1987, The Tascam Porta Two offered a six-channel mixer, an fx loop, and battery power.


The cassette four-track allowed any musician to make quality home recordings at an affordable price. The media, cassettes, is cheap, so you could record lots of music. You had four mono tracks to work with. A typical format was to record drums or drum machine on track one, bass on track two, guitar on track three, and vocals on track four. If you needed

more than four tracks you could record tracks one, two, and three with music and bounce/ mix them onto track four. Now committing to your mix on track four, you record over tracks one, two, and three for a total of six tracks. Recording with four tracks forces you to commit to performances and mixes early on in the creative process.


Today’s engineers and artists can learn the fundamentals of music production with a four-track cassette recorder. The transport controls on the Porta Two are laid out just like a modern DAW. Recording with a four-track, you learn about microphone placement, how to achieve optimum recording levels and decision making. The Porta Two also allows you to learn about fx loops, graphic EQ, preamps, signal flow, problem solving, bouncing tracks, and mixing music. Lastly, a cassette four-track teaches you about analog recording and sound, tape machine

maintenance, and how to have fun with a pencil and a cassette tape.


Bruce Springsteen recorded Nebraska on a Teac 144. Cassette four-track recording is often referred to as the “lo-fi” movement. Other notable artists who recorded with cassette four-track machines include Guided By Voices and Ween.


I recorded Roland drum machine and synth bass parts onto my Tascam Porta Two for the song “BP Oil Rap (How Can You)” from For Real by Aslan King.


Torbin Harding is a graduate of Berklee College of Music, B.A. in Music Synthesis Production 1995, and founded Lo-ZRecords in 1997. He currently works in both the analog tape and digital recording mediums. For more info visit


Professional Performance for All Creators

The new AT2020USB-XP gives streamers, podcasters, and other creators the award-winning AT2020 sound with plug-and-play USB-C operation and a whole lot of extras. Three selectable noise-reduction levels and an automatic gain control allow you to adjust the clear, natural sound

reproduced by the mic’s high-resolution A/D converter (up to 24-bit/192 kHz). Plus, sound can be monitored directly from the mic with no latency, mixed with computer audio, and silenced with a touch of the capacitive mute button. That’s professional performance and control.

/ AT2020USB-XP Cardioid Condenser USB Microphone
Lay down tracks wherever your tracks take you. Two monitors. One travel case. Unlimited possibilities. studio anywhere

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