Hooking up a mixer to a compressor or external preamp

I've been recording music for the past 3 years with a bizarre device, something I picked up online called the Peavey 10AT mixer. It's an audio mixer that has "AutoTune" built-in.

The Peavey 10AT mixer

Going back to the early days of my audio engineering interest, back in 2001 when I picked up my first recording device, a Fostex-MR8 digital multi-tracker, I didn't know what I was doing. I jumped into the home recording hobby at a time when digital recording was very new. Analog was inaccessible, because I knew nothing about tape machines, or even DAT, so I was forced to figure out digital recording. Without any real mentor figure, I had to teach myself everything. It was almost 100% trial and error.

I can remember often hearing advice, going back to the early days of my interest in sound recording, which was always, "You need a mixer." But, I didn't understand. How could I hook up an analog mixer to record the sound into a digital file on my computer? It didn't make sense.

Remember, this was 2001, and so the concept of an affordable audio interface (or even a well known one) wasn't yet possible. Hence why I was directed towards the Fostex MR-8 machine, a consumer grade, "Digital multi-tracker" that essentially recorded audio files onto an SD card that was inserted into the device.

The Fostex MR-8 Digital multi-tracker

The Fostex MR-8 forced you to "Bounce" your mixes into a final version before you were able to transfer them via USB to your desktop computer. This mix-down process was shaky, and I remember the set of steps required was possible, but not automatic. You had to make sure the file sizes and the lengths of the audio were proper, otherwise you'd be unable to mix-down and you'd be left with files stuck on the physical machine, unable to proceed with the USB transferring. It was less-than-ideal, to say the least. Regardless, I was able to do this even back then, when I was only 13 years old.

I guess, for the sake of thoroughness, I can talk about three other devices I've personally spent time using to record, in between the Fostex MR-8 device (in 2001) and the Peavey 10AT (2017-Present).

The Presonus Firepod

In 2006 I bought and used a Presonus Firepod. This was a unit that allowed for 8 microphones to be connected simultaneously, and each of the audio tracks to be recorded individually into a DAW (Digital Audio Workstation). So, this bridged many gaps for me at the time. First off, having eight microphone inputs was desirable. As a neophyte drum recordist, I saw plenty of videos about how and why you needed separate microphones for every drum on the drum kit (Although now I favor just using a stereo pair). I used the Firepod actively until about the fall of 2013, so it lasted me 7 years. The device suffers from preamps that contain some sort of digital "sheen" artifact that's unpleasant, although I will mention, if you record in a good environment and with the leveling properly set, you'll notice this "sheen" disappears entirely. Still, the Firepod didn't offer the most sterile input signal processing. Which is why....

The Focusrite Saffire Pro 14

In 2014, I bought a Focusrite Saffire Pro 14. This device solved the issues of the Firepod, because the input signal it offers is very dry and sterile, almost completely transparent. I can't say much about this device, as I ultimately found it sort of lifeless and less-than-inspiring. I guess the most interesting part of my ownership of the device was the fact I had to risk my life to pick it up. I walked on foot to a very remote FedEx pickup location, and found myself in a bad part of town. At one point I was surrounded by a gang, who were riding ATVs around the streets, and this was in the inner-city, in a very remote part of New York City. I felt like I was in an episode of the Sopranos, when they take a guy to the remote part of Coney Island, for instance, and all of the sudden he realizes what's about to happen.

The Focusrite ISA One Preamp

I did, however, attempt to liven up the Focusrite, with another piece of Focusrite gear, an ISA One analog Preamp. This was the first (and only) ever dedicated pre-amplifier I've owned and used. It's one piece of gear I somewhat regret letting go of, because it was pretty cool. I will mention, though, that it's only a one-channel device, so it's somewhat flawed in that sense. In order to hook up this preamp to the Focusrite Saffire Pro 14, I didn't have to try too hard, because I asked for help at the local Guitar Center (near Union Square in Manhattan) on what cables I'd need. The salesperson seemed to be knowledgeable, and let me know the exact cables I'd need. The ISA One included very easy-to-understand instructions as well, which helped me hook it up to the Saffire Pro 14 audio interface seamlessly. The ISA One performed well in that it livened up the input signal for the Focusrite Saffire Pro 14's AD conversion, and it also provided  more clarity to the signal, which I mostly utilized on the vocal tracks on the music I was recording at the time. It was pretty good, but I wasn't that into it.

From Wikipedia: In electronics, an analog-to-digital converter (ADC, A/D, or A-to-D) is a system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal.

The Teac A-3340S (Pictured without reels)

The other major device I used was a TEAC A-3340S. This was my first (and only) foray into fully analog recording. This device was interesting. I bought it off Craigslist and spent hours cleaning the heads with rubbing alcohol and cotton swabs, before running some brand new (at $65 a reel) tape through the machine. I was surprised to find out it actually worked, and seemed to be calibrated too. I used an analog mixer with this tape machine, namely a Yamaha MG series mixer.

Yamaha MG Series analog mixer

So at one point, I was able to record to 1/4" tape with the TEAC A-3340S, and then playback the tape through the Saffire Pro 14's AD converter and into my computers DAW, for a fully analog to digital experience. This was back in 2015, more or less.

Eventually, I lost my job and looked around at the extensive amount of equipment I had acquired and decided I ought to part ways with most of it, and ultimately I sold all of my gear except for an acoustic guitar and a pair of cheap Shure PGA57 dynamic microphones, for the sake of minimalism and simplicity. That was years ago.

The Peavey 10AT mixer

Obviously since then, I've been able to re-assess my recording interest and eventually found another job that paid well, and I was able to buy new and different recording equipment, and that's where I decided on the Peavey 10AT. And that's where everything came full circle.

The Peavey 10AT is not only a traditional analog mixer, it also allows you to stream the audio directly to your computer via USB (without any convoluted mix-down steps) with a built in AD converter. However, the 10AT has it's limitations, namely an inability to separate input signal feeds into separate tracks post AD conversion. Anyway, I was recently interested in adding some outboard gear to the Peavey 10AT and that's where I started to investigate how to hook up an external preamp or compressor to this mixer.

A block diagram of the signal flow for the Peavey 10AT

I've noticed the Peavey 10AT gives me a tonality that I've been able to work with, for sure, but how about the idea of using this Peavey 10AT as a traditional AD converter, much in the way I used my old Saffire Pro 14? How would I connect outboard gear to the mixer at the correct part of the signal path? That's where I checked out the above block diagram and considered the below schematic.

Schematic on how to connect a Warm WA12 MKII 500 preamplifier to an audio interface.

 
So if we inspect the schematic provided by Warm Audio, to show how to connect their WA12 MKII 500 preamplifier to an audio interface, we see it says, "Just plug it right into the front of your interface, into the first channel."

Well, upon closer inspection, it also says, "Buy our leveling amplifier and patch that between our preamp and your interface," but I'll disregard the inclusion of the leveling amp for now to avoid an in-depth discussion. But anyway, a leveling amplifier is a device that reduces the dynamic range of a signal, so the quieter parts become louder and the loudest parts become quieter. This effect creates a fuller or "richer" tonality to the signal, something I probably could've used with the Saffire Pro 14. But continuing on with the wiring discussion...

Inspecting the block diagram of the Peavey 10AT, you'll notice if I were to follow the above schematic and wired the Warm Audio output into the first channel input of my 10AT, I'd be running my microphone signal through TWO preamplifiers... the Warm Audio WA12 MKII 500 preamp AND the Peavey 10AT's built-in preamplifier , which obviously will muddy up the signal with excess gain. So, how to address this?

Looking at the Peavey 10AT block diagram, using channels 5 and 6 will allow us to bypass the Peavey 10AT's built-in preamplifier. This ought to work, however, we'll need to use the correct cables, and I believe these are referred to as "Stereo 1/4 inch" cables, or, rather, TRS cables, and also "Balanced 1/4 inch" cables. So you'll need this type of cable...

1/4" TRS male to XLR female cable

Just to double check, I read the Peavey 10AT official user's manual and found this:

"MIC/LINE INPUT (CH.1-8)
This combination input jack accepts a ¼” or XLR balanced plug. The XLR balanced input is optimized for a microphone or other low impedance source. The ¼” input is a TRS balanced type, and also accepts ordinary TS guitar cables."

Beyond 5/6's inputs, there are more channels on the Peavey 10AT mixer to consider: channels 9 and 10, and the AUX channel.

It seems like (again referencing the Peavey 10AT Block Diagram), channels 9/10 allow you to bypass the built-in EQ, but also disallow adding any built-in Digital FX to the signal, or applying AUX sends. So, it seems like using channels 9/10 isn't really designed exclusively for connecting outboard gear, but rather, running an external audio source into the mixer, like streaming audio from a radio, or an iPod, or something like that. Though, technically it can replicate the functionality of using channels 5 and 6, while also allowing you to explicity bypass the 10AT's built-in EQ. Channels 9/10 are basically just like Channels 5/6 minus a robust channel strip with Gain, L/R Panning, and Input Volume adjustment. With Channel 9/10 you're limited to a single master volume fader.

I guess, with the AUX channel, you are sending variable levels of each channel with an AUX dial to the AUX send outputs, which is useful for live mixing when your drummer asks for "More guitar" in his monitor. His monitor would be wired to the AUX send outputs, and the guitar's input, for instance, wired DI into Channel 4. By turning up the AUX dial on Channel 4, the drummer would hear his requested, "More guitar," in his monitor.

Of course, you can replace the drummer's monitor in the above example with, say, a piece of outboard gear, perhaps a rack-mounted EQ unit. Considering this idea, you are sending the guitar signal into the input of this rack-mounted EQ, instead of to a monitor.

By then connecting the output of that rack-mounted EQ back into a different input on the mixer, you are using the AUX function as an Effects Send/Return, or an Effects Loop. By turning the AUX dial on the guitar (Channel 4) up, you will send more of the guitar into the rack-mounted EQ, and subsequently back out of the rack-mounted EQ and into that separate channel on the mixer for playback.

So, bearing all this in mind, it seems Channels 5/6 are probably the way to go. So we've figured out how to hook up an external preamp to the Peavey 10AT, now what about a compressor as well? If we look at the back of this device below, FMR Audio's RNC 1773 compressor, we are given some clues on how this might work.

FMR Audio's Really Nice Compressor 1773 model, with unbalanced inputs/outputs

So this device has simple unbalanced 1/4" inputs and outputs. Now what?

If we follow the logic we found in the Peavey 10AT's block diagram, we notice on Channels 1/2, the 10AT's built-in preamplifier is first in the signal chain, and comes before the Peavey 10AT's built-in compressor. So, following that logic, with our outboard gear, we'd want to put the FMR RNC 1773 compressor AFTER the Warm Audio WA12 MKII 500 preamplifier in the signal chain.

Male to Male TRS (balanced) cable

Remember, we decided to connect the Warm Audio WA12 MKII 500 to channels 5/6 of the Peavey 10AT. So, the FMR Audio RNC 1773 will sit between the Warm Audio Preamp, and channels 5/6 of the Peavey 10AT. And... it seems as though, because the Warm Audio device offers us balanced XLR outputs, we're basically just patching the FMR compressor in with an additional male to male TRS cable. Pretty easy huh?

But wait, we're connecting cables between the balanced output of the Warm Audio preamp, and the unbalanced inputs/outputs of the FMR compressor to our balanced Peavey 10AT inputs. Now what? According to an article I found online, Mike Rivers, an audiologist tries to squash some rumors about balanced versus unbalanced connections, he tackles the issue head on:

• "If you connect a balanced output to an unbalanced input you'll lose half the level. Sometimes, sometimes not. It depends on the output topology."

Much to my surprise, there are full-blown threads online with debates on what cables to use to connect this FMR Audio RNC Compressor much in the same way I'm suggesting here. One person even alleges that the creator of the FMR Audio compressor has told him in confidence the I/O for the 1773 is in fact balanced, despite the official documentation saying otherwise.

The "Quick start" guide for the RNC delivers official information:

"The RNC can be hooked into Tip-Ring-Sleeve (TRS) insert points found on many consoles where: TIP = Console insert out, RING = Console insert in, SLEEVE = Ground reference. This means that each channel of the RNC can be hooked to a each channel of a TRS console insert with a single TRS cable."
 
Needless to say, I'd probably go with strictly (balanced) TRS connections between the Warm Audio preamp, the RNC compressor and the Peavey 10AT mixer. Boom!!
 
Lastly, the Warm Audio preamp discussed is only one-channel, whereas the FMR compressor offers two-channels. So, ideally you'd want a 2-channel preamp, or you could just buy two (2) of the Warm audio preamps, so you can record in stereo and apply uniform compression to your stereo signal. Shazam!!

That concludes this discussion and history of my own recording experiences and tutorial/explanation on how to potentially patch in outboard gear with your digital mixer or AD converter. Until next time. - Mike.

The pseudo-science of digital audio recording

It wasn't so long ago that digital recording didn't exist or wasn't accessible at all to anyone with any conservative budget. Because it's so new, it should've been a red flag to me to see so much "expertise" in the area of advice to young "recording engineers" (of which I've been) and I should've done a bit more research into the history of the art-form before jumping right in. I started recording music in 2001 or so and I started digitally. My first recording device was a Fostex MR-8 Digital Multitracker. Only last year did I ever experiment with using analog tape to record live music and the experience has changed my approach, to say the least. Not because it sounds "better"... but because the way you get signals to tape (or in the case of digital, hard-drive) is, in my opinion, overlooked.

Let me elaborate... When you record to tape the audio signal needs to live on the physical magnetic tape, whereas with digital the signal is converted to digital information (i.e. 0s and 1s) and this difference has an effect on how you should be recording. Now, the size (width) of the tape is finite and so is the size of the digital storage space, but the sizes are different.

The big mistake I was making was that if you read any beginners guide to recording they tell you this: Get the loudest signal you can get without going into the red (clipping) for each component of your music, then once it's all recorded, go back and "mix" your audio and adjust the volumes. I blindly followed this advice for years but now, I don't anymore and this is why...

When you record to tape, if the signal is too hot you will achieve tape "saturation" which is generally an unpleasant sound of distortion, but before hitting this "point of no return" you will hear the tape gladly accepting the signal and letting it fill out all of the space it is alotted: generally this is what's referred to as "warmth," i.e. natural tape-based signal compression. Now the physical size (i.e. width) of the tape is only 2 inches or less, so it's generally a small space to live. I think the idea of capturing  the most digital info as possible in terms of the various components to a mix is mis-guided, almost by definition. This is because the point isn't to hoard all the sonic information just because you can, the point is to record the sound you want to hear on playback. The luxury of "unlimited tape" blinded the neophyte digital recordists ability to properly engineer mixes. The analog workflow has proven to be essential to a natural sounding recording. When you dive deeper into this subject it becomes a combination of philosophical decisions and assessment of history, but ultimately it's about one thing... what sounds good?

Of course we aren't taught this concept of commitment and technique to recreate analog warmth on a digital audio capturing device when it comes to digital recording expert advice and that's where my issue is. If you apply this strategy of get-the-loudest-signals-you-can-without-clipping-and-then-adjust-in-mixing to your digital recording, you'll make a recording where it will come out sounding not necessarily over-saturated, but somehow wrong. It's too many hot signals competing with one another... and only one or two break through at a time. It's a lot for the ear to handle. The processing power of digital recording is mis-used in this context and the results are less than good typically. At long last, clarity?

So how do you achieve tape warmth on a digital recording interface? Change the way you take in audio signals. Just take in the signal at the volume it will be heard at in the mix (i.e. an appropriate and accurate reproduction of what you're actually hearing in the room) and don't go as high as the red, don't even go as high as the yellow. Don't have to adjust any volume knobs at all in your DAW. They should all always be at zero. I personally record to peak signals somewhere between -24dB and -12dB depending on the instrument. I assess this input level as I mostly focus on what I'm hearing through the monitors. If you're using a mixer, put your faders all just below 0 and adjust your input signal volumes with gain (and EQ) only. Your results will be better because you are following an analog workflow for your digital session, and the unity gain simulates the concept of natural tape compression in that it limits the input depth? Something like that... I'm still working on articulating this properly.