Here’s a question everyone asks but is too afraid to look dumb: “Where can I actually get good mix feedback without dropping $40 on a stream submission just to play 10 seconds and get no true feedback?”

You know how it goes: you join a stream, post your track, get some shitty feedback, so you resort to posting in some random Facebook group, waiting a few days, and all you get are a few fire emojis and a “this slaps”. While that's some feedback, it's not REAL feedback; it's really people being nice online and supporting their community. It doesn’t help you fix low-end issues or figure out why your vocals keep disappearing in certain parts. The real issue isn't finding the right community online to post your mix; it'sfinding someone you can trust who actually knows what they are talking about and will give you good, useful feedback.

The question I’ve heard come up in discussions over and over is: to get real feedback, is it better to pay for mastering?

Mastering engineers are listening for different things than at the mix stage. While they are still listening to the technical aspects of the song, they are also factoring in the production and how the unresolved low end is pushing the feeling away from the vocalist. Or the snare is swallowed up by your pads.

While this inclination MAY be real, it’s really the wrong service in the wrong order.

Mastering is for enhancing a mix that is already working, not for fixing a mix. That is a critical waste of a mastering engineer’s time, and most will not put their reputation on the line for the sake of you learning how to mix.

So here's what we’re going to break down today.

Where you can actually get real feedback, for free.

What paid services are worth it, and how to prepare your mix in a way that gets you real advice, real feedback instead of ‘sounds good bro’ or a rating of 1-10.

By the end of this, you should know exactly where to drop your next mix and what to include so you actually get some useful feedback you can learn from and maximize your next efforts.

Where to get professional mix feedback without paying for mastering

Not every streamer is created equal; some of these guys are your average person who releases music and has a following.

And some of these groups are just a bunch of beginners with expensive equipment tossing around surface-level takes because they lack the vocabulary to critique what they are hearing. BUT there are a few places where these indie opinions can be helpful. MOST of the time, this is about authenticity: you’ll know if your colorbass is not ‘colorbassin’ or if your hip hop track sounds like a Dos Equis backing track. Other than that, you’ll need more structure to get more serious feedback.

BUT

Reddit’s r/mixingmastering: the most consistent free mix review online

They’ve actually got a feedback guide in their wiki that tells you how to post. This helps a lot. In this group, you’ll find real engineers and advanced producers dropping legit technical feedback IF you provide enough actually to work with.

The trick in this group is to upload something high quality, specify the genre, and tell them exactly what’s bothering you.

If you post ‘thoughts’ you’ll get some pretty useless replies, but saying something like “ I can’t get the click to punch through the low end without everything getting muddy or clipping around 80 Hz,” you’ll get some pretty detailed, helpful responses.

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Discord servers for genre-specific feedback

The r/mixingmastering Discord has a channel just for mix critiques, and honestly, the feedback there is usually better than most places.

Their Discord has more technical takes and genre-specific stuff.

Jonas Aden’s server is also good, but quality depends on who’s online (like most discords). Discord is always great for quick feedback, especially in EDM servers like Jonas.

Facebook groups are really hit or miss. I’d say they’re worth bookmarking if you’re in the right group. Groups like Airwindows Audiophiles or All Up In The Mix (Jake Richter) have a lot of active people and sometimes do events where you can get some decent feedback. The Studio One (Fender One) community even does a monthly live mix critique. These are always good back-pocket options to keep in mind, especially for checking stuff you’ve had on repeat for a while.

Paid single-track feedback services worth your money

When you need something more structured than just random forum replies, paid services are that middle ground between free feedback and hiring someone to mix your whole track. Prices are all over the place—from under $10 to a few hundred bucks—and you pretty much get what you pay for.xReflect charges $6.95 to unlock a full written report with a release verdict. It uses a combination of AI analysis and human verification, which makes it a low-risk entry point for producers who’ve never paid for mix feedback before. SNIP offers a free first session, worth taking before you commit to anything. Their paid tiers start around $30 for a written report with timestamped feedback. Both services give you a clear picture of what structured mix engineer feedback actually looks like without a significant financial commitment.

Mid-tier services with written reports and real guarantees

Drum Collective is about $25 for a written critique, and they actually give you a 30-day money-back guarantee, which is rare and shows they back up their work. Mixing Lessons is $67 for a detailed report, but they only take two mixes a month, so you might have to wait. If you want actually to talk to someone, WKND Audio Berlin does a 30-minute Google Meet for about $54, and that’s usually where you get the most specific, back-and-forth feedback on your actual session choices.

WAV over MP3: why format actually matters for critique

If you want real feedback, give the engineer something they can actually hear. MP3s mess up the high end, and the low end—exactly the stuff pros listen for first. Always send a 24-bit WAV at 44.1kHz or better. If you’ve got a version with mix bus processing and one without, send both and label them so nobody has to guess which one is your real mix.

Reference tracks and the right context to include

Always include a commercial reference track from your specified genre when you send your mix. It gives the engineer something to compare your low end, stereo width, and transients to—basically, a real-world target. If you can, send a WAV or AIFF of the reference, but a good MP3 is fine if that’s all you’ve got. Also, add a quick note with the version number, genre, and a couple specific questions you want answered. That way, whoever’s listening knows exactly what you’re shooting for and can give you feedback that actually helps.

Specific questions get specific answers.s

If you ask, “What do you think?” you’re gonna get nothing useful. But if you say, “The low-mids feel congested around 300 or 500Hz in the second verse—does it sound that way to you?” you’ll get real answers. Point out what you think is off and ask about it straight up. Engineers respect it when you’ve already tried to diagnose the problem yourself, because it shows you’re serious about fixing it, not just fishing for compliments.

Metadata exposure and how to reduce it

Audio files can have hidden info like your name, DAW version, and session details baked in. Before you share anything in public, strip that metadata out. The best way is to use FFmpeg with the-map_metadata -1flag. That wipes all the metadata without messing up your audio. If you’re sharing an unreleased track in a public spot, use a watermarked snippet or a password-protected link instead of just posting raw stems with your name hidden in the file info.

Platform terms and AI scanning risks

Some platforms have sketchy terms that let them use your uploads for machine learning. Always check the policy before you upload anything that isn’t released yet. For forums, it’s usually safer to share a full mix than stems, since stems can be picked apart and reused. If you really need feedback on stems, stick to private services with clear rules instead of public groups.Based on your situation

There’s no one-size-fits-all answer here. What’s best depends on your deadline, your budget, how deep you want the feedback, and if your track is already out. Think about all that before you post,t and you’ll save yourself a lot of headaches.

Tradeoffs: speed, credibility, depth, and cost

Free groups are quick, but the quality is all over the place. Paid feedback can run anywhere from $7 to $300, and turnaround can be a day or two or up to two weeks. If you’re in a rush and don’t want to spend much, try SNIP’s free session or Drum Collective’s guaranteed report. If you want something in between a random forum post and a full-on engineer session, The Practical Audio Lab gives you feedback from someone who actually makes beats, not just mixes them.

Build a feedback loop, not a one-time submission.

One round of feedback isn’t gonna fix your mix. Send it to one or two solid sources, make the changes, then send the new version back. Keep track of what you changed each time. Doing this will level up your ears way faster than any course, because you’re actually connecting your decisions to real results in your music. The goal isn’t to get it perfect on the first try—it’s to keep tweaking until the mix is really done.

Free places to get professional mix feedback, and when to pay

You can definitely get pro mix feedback without paying for mastering if you know where to look and how to send your stuff. Start with r/mixingmastering for free, consistent reviews. Try SNIP’s free session or Drum Collective’s guaranteed report if you want to pay a little. And every time you submit, send a 24-bit WAV, a reference track, and a couple specific questions. That combo will make your feedback way more useful.

If you want feedback from a producer who’s actually in the trenches If you want feedback from someone who’s actually in the trenches—not just some faceless service or another producer guessing—The Practical Audio Lab is made for you. No mastering upsell, no nonsense, just real notes on what’s working and what needs fixing from someone who’s actually doing this every day. Subscribe to The Practical Audio Lab to get that content directly in your inbox, concrete, no-filler guidance for independent producers who are serious about the work.

When I first got started making music, I don’t know how I got my tracks to pass the ‘car test,’ but it was crazy inconsistent. I'd play something in my studio (which wasn't treated at all), then reference it in my car and wonder where everything went, lol. Eventually I got to a point where I slapped so much processing on that I couldn't really tell what was good; I just knew it sounded kindof how I wanted it to sound, and I was getting relatively consistent requests for music.

But were my mixes really good, or just familiar? After using the VXT8 monitors for some time, I knew what I wanted to get out of them, and I was referencing in my car. But those were the only systems I was testing. What about mobile? What about laptop speakers and iPhone speakers?

I eventually decided that I needed dedicated feedback and to actually have a ‘real engineer’ handle my work, so I could get a grasp of what my mix needed to sound like to translate across every system.

Over the years, I’ve used plugins, studio engineers, and AI algorithms to better understand and affirm some principles in mixing.

How Remote Mix Feedback Services Actually Work

Before comparing prices, understand the fundamental distinction: a critique provides written or recorded feedback on the mix you submitted, but the engineer never touches your audio. A revision means an engineer actually reworks your session files. Platforms like RoastYourMix deliver critique only; marketplaces like SoundBetter connect you with engineers who do hands-on work. Confusing these two is the most common mistake producers make when shopping for a review service.

Most asynchronous platforms follow the same flow: you upload a bounced stereo mix, fill out a short form about genre and reference tracks, pay, and wait. Turnaround ranges from a few hours (AI tools) to three to seven business days (human engineers). What comes back varies just as widely — a timestamped video walkthrough, or a short paragraph of notes. The format matters as much as the content.

There’s also a smaller category of real-time review services that use tools like Audiomovers LISTENTO, where the engineer streams your mix back from their DAW while giving live notes. It’s faster and more conversational, but it costs more and requires scheduling — for most independent producers on a budget, asynchronous feedback is the more practical starting point.

Human Engineers vs. AI Mix Analyzers: What’s the Real Difference

AI mix analysis tools like Mix Check Studio give you objective data fast: integrated loudness in LUFS, true peak values, stereo correlation, frequency energy across bands, dynamic range, and mono compatibility. These tools are genuinely useful for catching technical problems before a human ever hears the mix — phase issues, spectral imbalances, and clipping typically show up in seconds, with specific numbers and targeted suggestions.

When to Use an AI Mix Checker

Run your mix through one as a first-pass diagnostic before submitting anywhere else. It catches clipping, LUFS issues, and phase problems that a human reviewer shouldn't have to spend time on. Once those are handled, the mix arrives in front of human ears clean, and the feedback you get back stays focused on what's actually creative.

When to Hire a Human Mix Engineer or Critic

The gap shows up the moment artistic judgment is required. An AI analyzer doesn't know your low-end is intentionally blown out because the genre demands it, or that the reverb tail is long on purpose. It can flag your vocal running 3 dB above the genre average — it can't tell you whether that decision serves the song. For creative decisions and nuanced feedback on arrangement, you need a human who understands what you were going for.

What Separates Useful Feedback From a Waste of Money

Generic notes like “the low end is muddy” or “the vocals need more presence” are nearly useless. Useful feedback names the frequency range, the approximate dB adjustment, the instrument causing the masking, and a concrete next step. Before committing to any service, look for samples of their actual feedback output — vague examples mean vague feedback.

Turnaround time signals something about quality, but not everything. Human engineers on platforms like SoundBetter typically deliver in three to five business days, with rush options at extra cost; critique-only services often deliver within 24–48 hours. Watch for services where a five-day wait produces three sentences — that’s not a fair trade regardless of the engineer’s experience.

On pricing: SoundBetter mixing engineers charge roughly $250–$500 per song at the mid-level tier, with revisions adding fees (rates vary significantly by engineer profile). Dedicated critique-only services are cheaper because they’re not touching your audio. Know exactly what the price buys — a full mix overhaul, a video walkthrough, or a written report — because price alone tells you very little without scope.

Why The Practical Audio Lab’s Mix Feedback Hits Different

Our feedback service is built by a working producer for working producers. It’s written in the same language used throughout the Mix Hub newsletter: specific, blunt, and focused on what you can actually fix in your next session: no jargon, no filler, no corporate filter between you and the critique.

A typical session covers frequency balance, low-end management, stereo imaging, mix-bus processing decisions, and any arrangement or dynamic issues that affect the final sound. You get actionable notes tied to specific moments in your track — not a general score or composite rating.

This is built for producers working in bedroom studios who want an honest second set of ears from someone who understands the hardware and software they’re actually using, speaks their genre, and gives a straight answer without padding the critique to justify a high price — not for major-label artists shopping the most expensive engineer on a marketplace.

How to Prep Your Mix Before Submitting for Review

For most critique-only services, a high-quality stereo bounce is all you need: WAV or AIFF, 24-bit, 44.1kHz or 48kHz. Skip the MP3 unless specifically requested. Leave a few seconds of silence at head and tail, and make sure your limiter hasn’t clipped the master bus before you bounce — a clipping mix signals that basic quality control hasn’t happened, which affects how much trust the reviewer places in your other decisions.

Don’t submit cold. Include a short note with your reference track, genre, intended loudness target (streaming, club, etc.), and the one or two problems you already suspect. “I think the snare is getting buried after the hook” is infinitely more useful context than nothing at all. The biggest submission mistake is sending a mix with a known problem, hoping the reviewer won’t notice or will fix it in the notes — always address what you already know about first.

Matching the Right Service to Your Stage and Goals

If you’re early in your mixing journey, start with an AI tool to learn the vocabulary and identify recurring technical mistakes. Once you can apply basic corrections confidently, that’s the right moment to invest in a human critique. Going straight to a human without that foundation usually means spending the session absorbing concepts a free tool could have introduced.

If you have a release deadline, prioritize turnaround speed without sacrificing specificity — a 24-hour critique-only service with detailed timestamped notes beats a seven-day wait for a vague paragraph.

A single review tells you what’s wrong with one mix. Ongoing feedback across multiple projects reveals what’s wrong with your process. If the same problems keep showing up in every mix you submit, that’s a sign you need systematic guidance, not just track-by-track critique.

Choose the Online Mix Review Service That Matches What You Actually Need

Myth: “A five-day turnaround means a more thorough review”

Turnaround time gets treated as a quality signal, but it isn’t one on its own. A service can take a week and hand back three vague sentences, or turn around in 24 hours with detailed, timestamped, actionable notes. What determines quality is specificity — named frequencies, approximate dB moves, the instrument causing the masking, a concrete next step — not how long you waited for it.

My Workflow

ZBefore bouncing a mix i always check in mono. If anything is disappearing, that means there are some phase issues, and I may have to flip the polarity of a sample, add some chorus or sample delay. Most of the time, if I'm exporting directly from the Akai Force, I’ll run through an online checker like MixCheck for some quick ‘mathematical’ feedback. This helps me make better decisions around releases. I’ll take whatever information on Phase, clipping, and frequency balance back to the mix and start adjusting if needed. It’s honestly a way I trained my ears to a different frequency range.

• Know whether you’re paying for a critique (notes only) or a revision (hands-on rework) before you submit — they’re different products at different price points.

• Run AI diagnostics first to catch technical errors, then bring a cleaned-up mix to a human for creative judgment calls a meter can’t make

  .

• Judge any service by the specificity of its feedback, not its turnaround time or price alone.

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Have you gotten mix feedback that was so vague it was useless — or so specific it changed your whole approach? Drop the story in the comments.

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The moment I realized my 2-buss chain wasn't working for me was when I realized that I had zero headroom for mastering to do anything. There is a difference between mixing a track and finalizing or mastering a track/project.

What’s up, lab tech,

If you’ve ever wondered why your mix sounds ‘almost there’ but never quite ‘pro’ the 2buss is usually where the gap resides.

We’ve seen the rise of AI driven or ‘algorithm’ driven mastering services over the years and i’ve even fallen victim to trying to use them. Trying to use the ‘robot’ mixer, if you will, left most of my tracks sounding wonky AF. It really had me wondering what the heck I was doing so freaking wrong.

I found a very useful tool from roexaudio called /mixcheckstudio.roexaudio.com/

that analyzes the mix based on genre and gives pretty clever feedback. I’d have to say most of that feedback helped reaffirm what I was experiencing when translating mixes.

Sometimes less is more, and more often than not, balancing your levels will get most of the work done. The most that needs to be happening on the stereo output is a little EQ and compression, perhaps a little tube drive to act as glue, but that’s really it.

Why Processing On The Mix Bus Isn’t Optional

That’s what you’re sending to mastering?

I don't know when this habit started, because it’s not how I first learned. But I used to start surgical EQ and compression after I got a good overall balance and then would add bus processing.

Wrong approach

As soon as I would slap the compressor or EQ on, things would get ‘interesting'; some things would be popping in and out of phase, with volume pumping, etc. It's best to mix into your compressor before you start any EQ or per-track compression.

But why even have any processing on the mix bus in the first place? Why not just send a well-balanced mix to mastering, or just crank the volume up loud enough and upload to a distributor?

Mix bus processing is more than just arbitrary effects; each insert should be doing something. Whether it's an EQ or a compressor, there is an inherent function that will improve your mix and shape your unique sonic signature.

Compression helps to glue the mix together while EQ sets the tone. Think of it like being a party planner, and you’ve got to bring a bunch of different crowds together. On your mix bus, or submix level, you’re gelling certain frequencies together and massaging them either up or down to increase the energy, vibe, punch, earth, or whatever mood you can come up with. Same thing you’d do at a party.

Bus processing is the final sonic tool for crafting a great mix, and just like party planning, if you start off thinking about and setting the bar for how your audience is going to feel, it helps to make better decisions as you’re adding/taking things away during the mix (and even the creative process)

The Correct Order: Compression → EQ → Saturation → Limiting

Signal flow is everything.

In audio, everything we’re doing is technical physics, and physics is math.

And just like math, order of operations matters. When you change the order of the operations, the outcome can be drastically different.

Compression first-
Then EQ
Some Saturation
and Limiter

What Each Stage Is Actually Doing

Ok, so let's go with the ‘flow’ and start with your compressor; this is to catch any peaks and provide the essential ‘glue’ that will define your mix. Again, using the party planner analogy, you’re bringing all the frequencies together to create a cohesive mix. This is where you’ll bring punch, a little grit, snap, or a groove.

This is also when you can tell what’s not going to make the cut, or what needs toning down, or edge.

Next is your EQ; it will set the tone of the mix. Is it bright and energetic, or dark and moody? Yes, of course, the arrangement and instruments chosen determine this; however, adding a bunch of high-end to a dark trap song could throw the entire vibe off (not to mention not leaving enough room for the artist)

Saturation is your friend, in small amounts. I’m a huge fan of using saturation as a secondary (tertiary) form of compression or congealing of frequencies.

Saturation helps roll off some high-end and provides a subtle gluing of frequencies that makes the hairs in your ears stand up.

Be cautious with this; remember less is more, so if you’re doing something edgy, it’s best to have some saturation on the 2buss but only enough to accomplish the goal. Start with highly noticeable distortion and lower the distortion until ‘audibly pleasant’

The last stage is what will ultimately save your mix from squashing. Your compressor may catch some spikes; however, the limiter is a brick wall that any strays will run directly into.

Clipping is also a form of limiting and could be applied either in front of the compressor or at the end of the chain.

The Mistake Almost Everyone Makes First

One of the number one mistakes people make is trying to use mix bus effects to fix a bad mix.

IF your mix sucks, your mix sucks. Bad performances, wrong instrument choices, and an overall lack of clarity in a mix are not on the 2-bus to fix.

Almost everyone who complains about their mix being bad is due to the mix itself being bad, and /or trying to fix it by overprocessing with compression, clipping, or limiting

Settings That Won’t Wreck Your Dynamics

Contrary to popular belief, having a fast attack on the 2-bus won’t completely ruin your dynamics.

Over the years, the “SSL Trick” has been something that can easily get more recognizable results from the 2-bus compressor; however, I’ve found that fast attack and low ratio can also yield some pretty good results.

Especially when paired with using submixes, I’ll drop a link below so you can check that out.

Your 2bus should be peaking at around -6dB, if you get a few randoms that peak at -4 its ’s not going to ruin the mix.

Attack to something fast like 5ms, and your release to something like 60ms.

This keeps the compressor ‘snappy’ and reacting fast to transients,

For 2 Bus compression, I'm a huge fan of a 1:3 to 2:1 ratio; that’s it.

I’ll typically set the threshold to around -10dB if my input is -6dB and work my way up from 1:1 until I hear what I like.

If you need something a little more aggressive, that is what the submixes are for. You can drive a little more aggressively there to make sure the signal is believably delivered yet controlled when it hits the 2bus.

For EQ, roll off a little low end (~20- 30 Hz), and the rest is subjective.

When it comes to the mix bus, it doesn't take much; try saturation at 5%.

Just make sure to gainstage in betwen plusins, The optimal level for your limiter is around -6dB however if you’re using a Maximizer plugin (which technically is a mastering plugin) then that input level may be around -12dB.

The Practical Workflow

So my workflow for client work varies a little from my personal stuff.

Anytime someone sends me something, I always load it up and listen to how they creatively heard it. I’ll make notes of what stands out to me, emotionally, vibe-wise, etc. It all depends on the track and where the artist was trying to go.

Next, I’ll listen to the reference tracks and decide which elements I’d like to incorporate, and start thinking creatively about the ‘drama’ of the song, including automation, reverbs, etc.

Then we’ll start analyzing their 2bus; ideally, if there is nothing, that's great, but if they like the sound of a particular plugin, I’ll work with that.

For myself, it's pretty straightforward: I'm either doing exactly what's above, OR I'm leveling the main aspects of the song, like the rhythm section, bass, and main melody. I’ll fix my 2bus to where I want it and start producing or adding elements from there.

One workflow starts with everything already created and ready to go; the other starts with a few critical aspects, with the rest added and evaluated for their interactions with the 2-bus compressor.

Order Matters More Than Plugin Choice — Compression, Then EQ, Then Saturation, Then Limiting.

Each Stage Should Do Less Than You Think; The Mix Bus Glues, It Doesn’t Fix.

If Something Sounds Wrong On The Bus, Check The Source Tracks Before Adding More Processing.

The Fundamental Questions

Q: What is a Mix Bus (or Master Fader), and why is it so important?

A: The Mix Bus is the singular point where all the signals from your individual channels (drums, bass, vocals, guitars, synths) are combined into a final stereo signal. It’s important because everything you apply here affects everything in the mix. Think of it as glue; it can blend disparate elements together, add color, and set the dynamic range for the final master.

Q: How much ‘mix bus processing’ should I do?

A: The golden rule is: Subtlety. While individual tracks might require aggressive 6dB cuts or boosts, a mix bus EQ or compressor typically uses tiny movements. 1dB of gain reduction is a lot on a mix bus. You are massaging, not reshaping. A lighter touch is always safer, especially if your track is going to a mastering engineer later.

Q: In what order should I use plugins on my mix bus?

A: The diagram above shows a standard and safe processing order. The flow generally goes: Subtractive processing (like surgical EQ), dynamic control (compression), color/saturation, and final peak control (limiting).

Compressor: (Box 2 in our diagram) Usually the first active element. It tames peaks, adds “glue,” and establishes the groove.

EQ: (Box 3) For global tonal balancing. Gentle high-shelf boosts or low-end management.

Saturation: (Box 4) Used optionally to add harmonic complexity, warmth, and subtle ‘analog’ glue.

Limiter: (Box 5) Always the final stage for peak protection. It ensures nothing goes over 0.0dB.

Q: How do I set up a mix bus compressor effectively (Box 1)?

A: This is the big one. As shown in the diagram, start with gentle settings:

Threshold: Set it so you are seeing just 1- 2 dB of total gain reduction (seen in the bar graph meter). It should only just kiss the signal.

Ratio: Keep it low. 1.1 or 2:1 are standard for the mix bus.

Attack: Slow. 10ms-30ms or even slower. This lets the initial ‘crack’ of the snare or kick through before the compressor clamps down, preserving impact.

Release: This depends heavily on tempo. You want the compressor to release in time with the song's beat. A common starting point is between 100ms and 300ms, or ‘Auto’ for ease.

Q: What is the purpose of the Mix Bus Limiter (Box 4)?

A: The Limiter is your safety net. On your final Master Fader, it must be the very last element in the chain. As seen in the diagram, the Ceiling is usually set to -1.0 dB. The limiter prevents transient peaks from exceeding this level (digital clipping) by instantly compressing any audio that tries to exceed it. It ensures maximum possible loudness within safety limits.

Q: When should I add these plugins to my chain?

A: There are two distinct schools of thought:

Mixing ‘Into’ the Bus: Add your compression and subtle EQ very early in the mixing process. This lets you react to how your mix interacts with the bus processing as you build the track. Many engineers prefer this as it informs their dynamic decisions.

Post-Processing: Build your entire mix without any bus processing, and apply it only once the individual tracks are perfectly balanced. This ensures a more pristine source.

There is no wrong answer, but most professionals find that mixing into a gentle compressor (set up early) yields faster, more cohesive results.

What does your current mix bus Chain look like? Comment below; I read every one of them

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🔥 What Saturation Really Does (and Why It’s Not Just Distortion)

Saturation adds harmonic content that gives sounds perceived warmth, density, and loudness.
It’s what analog gear naturally does — but digital systems don’t — which is why saturation plugins are essential for giving in-the-box mixes life.

Think of saturation as tone enhancement through controlled imperfection.
Where distortion destroys, saturation enhances — adding harmonics that give depth, cohesion, and character.

Read this first:

It provides important context and useful tips.

🧪 My Go-To Saturation Plugins (and How I Actually Use Them)

Here’s my personal saturation chain — from subtle harmonic glue to full-on analog grit — plus how I dial each one in.

(not in any specific order of importance either)

Standalone MPC/AKai Force

AIR Tube Saturation (MPC)
Your “make this sound less like a laptop” button.

This one is perfect for taking sterile digital sources and giving them some harmonic meat without turning things into fuzz. It adds that soft tube bloom that makes vocals, guitars, pads, and samples feel a bit more expensive and less brittle.

Best use: Subtle tone shaper on individual tracks.

• Vocals that feel thin or “demo-ish.”

• Guitars, synths or pads that need warmth and glue

• Clean samples that need a little “harmonic enhancement”

How to dial it in:

• Keep the Drive low and lean into the warmer tone options.

• Work in the 15–25% drive range for most sources so it sweetens without announcing itself.

• Adjust the headroom to clamp the signal for tube compression

• If the top end gets a little spicy, follow it with a gentle EQ cut in the upper highs to keep things smooth.

Soft Clipper & Diode Clipper
Your “don’t blow up the meters, but still hit hard” pair.

These two live in my “dynamic safety net” lane:

• Soft Clipper keeps things feeling clean and controlled when peaks get wild.

• Diode Clipper adds that crunchy, aggressive transient smack that just works on MPC drums and bass.

Best use:

• Soft Clipper: On drum bus, mix bus, or loud stems where you want loudness without harsh brickwall limiting.

• Diode Clipper: On kicks, snares, synth basses, or aggressive leads that need bite and character.

How to dial it in:

• Set a low threshold so it catches the hottest peaks, not the whole signal.

• Use your DAW’s parallel blend or mix control if available — clip a bit harder, then blend back for punch without wrecking the transient shape.

• Always level-match and listen for feel: louder is not automatically better.

Akai “Grimey Distortion”
That “MPC 2000 in a bad mood” energy.

This one isn’t about clean enhancement — it’s about attitude. The Grimey Distortion has that thick, slightly ugly-in-a-good-way character that feels like older MPCs getting pushed too hard. Great when you want drums or samples to feel lived-in and aggressive instead of polite.

Best use:

• Parallel distortion for drum breaks, one-shots, and chops

• Toughening up 808s or bass so they poke through small speakers

• Adding “dirt layer” to samples in boom-bap, grime, or heavy hip-hop

How to dial it in:

• Run it on an aux/parallel bus and blend it in under the clean signal.

• Turn the grime down to 1-2 and the drive below 10, make sure your input level is lower than -18dB and use drive to lift the signal into the distortion.

• Push the distortion more than you think on the aux, then tuck the mix knob down so you’re just hearing the tone and texture, not full-on meltdown.

• Use a low-pass or band-pass after it if the high fizz gets in the way.

• track in at EXTREMELY low levels and keep the signal fidelity, add small amounts of drive for more subtle distortion and timber adjustment.

MSXII LoFlyTape & LoFlyDirt
Instant “this came off a dusty cassette” vibes.

These two are pure vibe tools:

• LoFlyTape gives gentle tape-style compression, subtle wobble, and a smooth high roll-off.

• LoFlyDirt piles on the grain, dust, and harmonic grit that just screams boom-bap and soul.

Best use:

• LoFlyTape: On the mix bus, melodic buses, or keys to round off the top and pull things together. Add saturation and a bit of noise for some authentic “printed to tape” feeling.

• LoFlyDirt: On drum buses, sample chops, or chorus sections where you want extra grime and movement or over an entire mix.

How to dial it in:

• Keep LoFlyTape light on the mix bus — think “glue and soften,” not “destroy” . Be mindful some of the tape emulations get a little wobbly

• Use LoFlyDirt more aggressively on drums or featured sections, then automate it so the track gets dirtier on hooks/choruses. SP404 (v) is meant to emulate the vinyl effect on the SP404. Very useful for drums and samples.

• Don’t be afraid to filter before or after these plugins so the dirt sits in the mids and doesn’t turn into fizzy noise.

AIR Filter (VCA Distortion / Overdrive)
Sleeper hit: ignore the filter, abuse the distortion.

Even if you’re not reaching for it as a traditional filter, the VCA distortion and overdrive inside AIR Filter have a really nice tone — more musical than a lot of generic DAW overdrives, with a satisfying push that doesn’t instantly turn things to sandpaper.

Best use:

• On drums and percussion when you want more knock and hair.

• On basses and synths that need to cut without just boosting highs.

• As a character stage before compression on a subgroup.

How to dial it in:

• Set the filter to a neutral or wide-open state so it’s not really shaping the tone — you’re here for the distortion circuit, not the filter.

• Start with VCA Distortion for punch and density; switch to Overdrive when you want more obvious breakup and midrange growl.

• Push the input until you feel the tone “lean forward,” then back off a touch. Use output level to match volume and A/B until you’re getting vibe, not mud.

Daw Plugins of Choice

*below are plugins I absolutely LOVE, theyre not popular, theyre not common but I love the concept behind them and what they consistently output at different sample rates

Plogue ChipCrusher
Retro coloration done right. Bit depth, sample rate, and speaker emulation combine to recreate old-school tone.

Best Use: Great for sound design and emulating older technology like computers or keyboards.

How to dial it in:

• Choose a DAC or conversion method. I personally use mulaw or PCM and emulate older samplers like the SP12, SP1200 or MPC60/

• Choose background noise: Most older gear had imperfections in the signal path or slight noise coming from electrical components. Not only does this add to the harmonic content of the signal, but adds a level of realism or believability to whatever you pass through.
  
  
  Some of my favorites include :

  • brown_c64_x,

  • brown_c64_av_basic_screen1 / screen2

  • or c64_rf-2 / c64_rf-3

  • mac_plus_headphones_vol3

  • genesis1_rca_back-2

  • genesis1_phones_80p

  • nes_rca-1

  • nes_rca-2

  • nes_rca-3

• Pick a Cabinet: When looking to emulate actual gear in a realistic space rather than the signal coming out of the gear, this is your friend. Combining this with spacializing plugins like Panorama 7 or Soundly’s Place It, you can achieve some pretty convincing emulations of actual gear being recorded with micropones which adds a level of sonic realism to your tracks.

  • Bonus Tip: Use Airwindows uLawEncode before and after the plugin to emulate broadcast companding which was very similar to the method used to reduce signal-to-noise ratio in the MPC60.

  • Be sure to emulate ‘pre-emphasis’ by either increasing the high frequencies before chipcrusher and decreasing afterwards or a simpler method of using pre and post gain inside of the plugin. Tweak this to your liking

  • Optimal input levels for uLaw Encode are going to be around -18dB otherwise it just distorts.. dont let that input level fool you though.. it packs a punch when done right.

Airwindows BusColors4
Your “instant console” plugin.

Instead of just adding generic saturation, BusColors4 models how specific consoles actually sum and output audio — including the vibe of the output transformer. You get different characters like Rock (API-ish punch), Lush (Neve-ish thickness), Tube, and more. The big idea: choose one flavor and let it leave a consistent sonic fingerprint across your entire mix.

Best use: Place as the last plugin on your mix bus or on key subgroups (such as drums, music bus, vocals) so everything sounds like it’s passing through the same analog desk.

How to dial it in:

Pick a model:

• Lush if you want width and that soft, expensive thickness.

• Rock if you want a more forward, aggressive mix tone.

• There’s no “drive” knob, so the sound comes from how hard you hit it, similar to an actual console. Push the level going into the plugin for more color, pull it back for something cleaner. Use the output trim to match level so you’re not being fooled by loudness.
  

Airwindows Desk/Tube Desk
“Make this channel stop sounding like a cold DAW track.”

Desk is basically an analog gain-stage emulator. Think of it like giving each channel its own little piece of console behavior. It smooths out those pokey digital peaks and adds a bit of analog-style movement so things don’t feel plasticky or brittle when you start stacking tracks.

Best use: First insert on important channels or busses.

• Great on drums and percussion to tame spiky transients.

• Nice on digital synths and samples that feel a little too sharp or “fake.”

How to dial it in:
It’s a true set-and-forget plugin:

• Want more attitude? Push the signal into Desk harder.

• Want it cleaner? Back off the input level.

• Know that ‘squishy’ feeling amplifiers have when driven, increase the Power Sag to emulate the slight drop in voltage that lightly compresses or puffs up what you’re driving into it.

Tube desk just reacts to volume. No menu-diving, no overthinking — get your gain staging right and let it do its thing.

Airwindows Interstage
The “this suddenly feels 3D and expensive” button.

Interstage isn’t a distortion plugin. It’s modeling what happens between analog stages — transformers, capacitors, all the non-linear weirdness that makes hardware feel solid and “real.” The result is a slight low-mid weight, a bit of top-end smoothing, and less of that icy “in the box” sheen.

Best use: Right before a super-clean digital EQ or compressor. Or essentially anywhere in the signal chain.You’re basically feeding your clinical tools a more “analog-shaped” signal so they respond and sound less sterile. (and you can use multiple instances of them)

How to dial it in:
Just like with most of the good stuff from Airwindows, there are no knobs. It’s meant to live at 100% wet:

• Drop it in the chain.

• Add before/ after most effects.

• Bypass/enable and listen for the added “heft” and slight de-edging in the highs.

If your mix feels a little flat and too “DAW-y,” Interstage is one of those tiny moves that just adds a tiny bit of something

Airwindows MOJO
That “finished record” fairy dust.

Mojo is one of those plugins that doesn’t scream “effect” but makes things feel more expensive and glued. It’s doing a mix of saturation, low-end thickening, and high-end smoothing that mimics what happens when audio passes through a bunch of analog stages repeatedly.

Think: less brittle top, more weight in the mids, and a subtle sense of push that helps sounds sit in a mix without you knowing why.

Its not stated anywhere in the specs, but Mojo adds a bit of 2nd order harmonics similar to the SSL 9000 which is incredible for a free plugin

Best use: On individual tracks or subgroups that feel a little flat or weak.

• Great on vocals that need to feel “in front” without harshness.

• Killer on bass, keys, guitars, or samples that sound too clean or “demo-ish.”

• You can also sneak it on your mix bus at very low intensity for a global polish.

How to dial it in:

• Start with the signal into Mojo at a low levels until you start to ‘feel’ the sound getting fuller..

• LIke with all Airwindows plugins, you’ll drive the signal into this until you’re getting the sound that you want..however stay conservative with this one, the best results come from a bit of moderation.

• As always, level-match: toggle it on and off and make sure louder isn’t tricking you. You’re listening for density, warmth, and cohesion — not obvious distortion.

Airwindows uLawEncode / uLawDecode
“Companding like a busted digital box — in a good way.”

uLaw is Airwindows’ take on classic µ-law companding — the same kind of nonlinear encode/decode curve used in old telephony and some early digital systems. In practice, that means it pushes quiet details up and squashes peaks in a very specific, non-musical-on-paper but musical-in-practice way. Used right, it gives drums, samples, and vocals this compressed-but-not-a-compressor density that feels very “early digital hardware.”

Paired with bitcrushers like pockey, chipcrusher, or derez3, you can either:

• Use them together as a companding stage (encode → hit other processing → decode), or

• Use uLaw by itself as a character effect to add forward, gritty density without bitrate reduction.

Best use:

• On drum buses or individual hits when you want front-loaded punch and a weirdly satisfying digital clamp.

• On sample chops to make them feel like they came off some old converter, not a pristine 32-bit float session.

• As a companding sandwich: encode, then distort, boost clip/EQ, then decode so the overall shape feels more controlled and “hardware processed” instead of obviously mangled.

How to dial it in:

Encode for character :

• Drop uLawEncode on a drum bus, sample bus, or vocal.

• Drive the input until you hear transients get tighter and denser, not just smaller.

• Level-match and A/B — you’re listening for “forward and glued” rather than flat and smashed.

• If it’s too obvious, move it to a parallel bus:

  • Smash the parallel with uLaw.

  • Blend underneath the clean signal for a subtle “digital front” layer.

Companding chain (uLaw → FX → uLawDecode):

• Put uLaw first in the chain.

• Follow it with something that reacts nicely to the denser signal:

  • Bitcrushing like Chipcrusher, MbitFun, Derez3,

  • Slight EQ boost in the highs (before bitcrushing for extra broadcast companding emulation)

• Then put uLawDecode after those.

  • This “re-expands” the signal somewhat, but with your processing baked into the companded shape.

• Fine-tune by adjusting how hard you hit uLaw on the way in. Small changes at the encode stage can completely change how everything downstream feels.

  • Like explained previously, this plugin takes an optimal input signal around -18dB, but whatever you do in between stages will help boost the signal quite musically.

Used subtly, uLaw/uLawDecode is one of those “why does this feel more finished?” stages. Push it and it turns into a very specific crunchy, front-loaded digital weight that can be perfect for aggressive drums and chopped samples.

There is a plugin called Pockey that does something similar, however I prefer teh ability to control the type of emulation inside of the companding.

⚙️ How I Combine These in a Real Mix (Workflow)

Track-Level/Instrument-Level:

• AIR Soft Clipper or Diode for drums and bass transient control.

• AIR Tube/Filter VCA Distortion on vocals or guitars for color and presence.\

• uLawEncode/Decode combination with Chipcrusher on samples recorded into the MPC

• Grimey distortion on synths

• Tube desk + Interstage on synths or instruments recorded into MPC

Bus-Level:

• AIR Soft Clipper on drums

• LoFlyTape for glue and harmonic lift on submixes

• Black Box HG-2, busscolors or MOJO on the 2-Buss

Master Bus:

• BussColors 4,

• Mojo,

• Interstage for emulation.

• Black Box HG-2

🎚️ Quick Recipes (Try These Two Presets)

Drum Bus — Punch + Grit:
Soft Clipper (clip peaks only) → Grimey Distortion, Ulaw + Chipcrusher.

Vocal Chain — Warm Presence:
AIR Tube (low drive) → Tube Desk (Tone A, light drive) → Bus HG-2 (low drive, wet mix 10%).

🧩 Troubleshooting Saturation Problems

If it’s muddy:
Roll off low-mids before saturating (HP filter at 60–100Hz).

If it’s harsh:
Use multiband saturation and limit high-frequency drive.

If it’s too loud:
Always level-match before and after saturation — your ears perceive louder as “better.”

🎛️ How to Choose the Right Saturator for the Job

For Glue:
Use tape or tube-style saturation (Mojo, HG-2, Busscolors,LowFlyTape).

For Bite:
Go for diode or transistor types (AIR Diode, Grimey Distortion, Air VCA filter Distortion ).

For Warmth:
Try subtle harmonic saturation (Saturn 2, Tube, LoFlyDirt, Mojo, Busscolors).

🔗 Other Useful Posts

🎯 Get Free Mix Feedback

Want help balancing your saturation levels or finding the right tone for your mix?
Submit your session through the Mix Feedback Hub and get detailed notes on where saturation helps (or hurts) your mix.

Submit for Mix Feedback

💌 Join the Lab

Final Note:

Master a few saturators instead of collecting dozens. When you understand how each reacts to transients, harmonics, and tone, your mixes will sound more intentional — and more analog — than ever.

The Myth of “Unlimited Headroom”

You’ve probably heard it before:

“You don’t have to worry about clipping — it’s 32-bit float.”

That’s half true and half the reason so many of your mixes sound harsh, crunchy, smeared or simply not where you want them.

Yes, 32-bit float gives your DAW massive mathematical headroom theoretically hundreds of dB above 0 dBFS. But that protection only applies between channels, not inside your plugins. The moment your signal enters a processor that models analog behavior (compressors, EQs, saturators, limiters, etc.), you can scrap all that theoretical headroom..

Those plugins don’t care about 32-bit math. They’re built to behave like hardware , so that means they clip, they distort, fold over if your input is too hot (but they also have a sweet spot as well)

Internal Clipping Is the Silent Mix Killer

Let’s walk through a real example:

1. You’ve got a vocal that peaks around -3 dBFS.

2. You add an EQ, boost 5 kHz by +6 dB.

3. Suddenly, that boosted frequency is slamming your next plugin at +3 dBFS.

4. Your compressor now sees an overloaded signal and distorts even though your DAW’s output meter looks fine.

This is internal clipping. It’s happening inside your plugin chain long before your master fader shows red.

The fix? Gain stage between every stage not just track by track, but plugin by plugin.

Why Floating Point Doesn’t Save You

32-bit float prevents hard digital clipping at the summing bus level meaning your overall mix can temporarily exceed 0 dBFS internally without crunching. But as soon as that signal:

• hits a plugin expecting -18 dBFS nominal input, or

• hits your mix bus limiter, or

• hits your converter on export…

…it will clip.

Think of it like this: 32-bit float keeps your DAW from catching fire, but it doesn’t stop your signal processors from overcooking.

How to Spot Hidden Clipping

Here’s how to find where the damage is happening:

1. Check every plugin’s output meter. If it’s hitting red or sitting near 0 dBFS, trim it down if it doesnt have one...

2. use a VU meter after heavy processors (like saturation or compression). Aim for -18 dBFS RMS = 0 VU.(beware most of these are calibrated slightly differntly)

3. Bypass plugins one by one. If your mix suddenly clears up when something’s off, that’s your culprit.

4. Watch your mix bus gain reduction. If your limiter is eating 6–8 dB before mastering, your mix is probably too hot upstream.

The 32-Bit Truth: It’s a Safety Net, Not a Free Pass

Your DAW’s float precision is amazing but it’s not an excuse to ignore level discipline. The same rules apply today that did yesterday.
Proper headroom is psychological as much as technical. A mix that lives around -18 to -12 dBFS feels open, controlled, and musical. Push everything near 0, and even if it’s not technically clipping, your mix will sound stressed.

Remember: Headroom isn’t wasted space it’s breathing room for your transients, dynamics, and tonal balance. IF anything this gives you room to crank the FINAL plugin (if you’re still into the loudness wars thing)

Fixing the “Too Hot” Mix

If your mix keeps clipping or sounding harsh, try this:

1. Select all your tracks and pull the faders down -6 to -10 dB. I cover gainstaging here

2. Reset your mix bus plugins. Let the signal breathe.

3. Add a trim plugin (or stage your input via clip gain or whatever sound generator) and before every major processor. Keep things hovering around -18 dBFS in and out.

4. Re-evaluate tone and balance. You’ll notice your EQs and compressors suddenly sound smoother and easier to control.

Quick Reference: Safe Headroom Levels

• Individual tracks (RMS): -18 to -14 dBFS (not drums)

• Peaks/drums: -10 to -6 dBFS (but random peaks should be controlled with compression/clipping/limiting if affecting the mix negatively)

• Mix bus (pre-master): around -9 to -6 dB peak (sometimes I find my mix averaging at -12dB)

• Master output (after limiter): -1 dB ceiling (if mastering your own mix)

These aren’t concrete rules but they’re healthy ranges that keep your plugins happy and your sound consistent.

• Coming next week: Lab Report #6: Parallel Compression for Real People.
• In the Nexte coupple of weeks: Mix Notes #10: Why “Louder” Always Tricks Your Ears.

Having headroom issues or mystery distortion? Submit your latest mix to the Practical Audio Lab Mix Feedback Hub and get personalized notes on your gain structure and plugin flow.

Final Thoughts:

If you’re mixing quiet, you’re mixing smart. In the digital domain, unless you’re using a plugin that specifically adds it for emulation purposes, you don’t have to worry so much about channel noise. You can get your levels right, ‘quiet'.

“Mixing is problem solving, not perfection.” — Will (SOVLTRON)

What’s up, Lab Crew!

Like compression, reverb can be one of the more challenging aspects of a mix to get right, but when done right, it can make the entire mix sound larger, more three-dimensional, and realistic. Today, we’ll cover some of the biggest mistakes you may be making with reverb and simple ways you can fix these today.

INTRO: Why This Matters

Reverb is the rookie producer’s love affair. It’s seductive because it makes everything sound big, emotional, and spacious — at first. But without control, reverb will eat your mix alive. Mud creeps in, vocals get lost, and suddenly you’re wondering why your song sounds like it’s being played in the back of a cave.

Today’s Mix Notes aren’t tied to one subscriber’s track. Instead, I’m tackling three of the most common mistakes I see beginners make with reverb — and more importantly, how to fix them so your mix translates like a pro’s.

Want to submit your own mix for next week’s breakdown?

[Here’s how.]

1. OVERALL IMPRESSION

When I listen to beginner mixes, I often hear too much room, not enough definition. The intent is there — wanting size, wanting vibe — but execution gets in the way. Think: the vocal sounds like it’s in a cathedral, while the snare is swimming in a completely different room. Depth becomes clutter instead of clarity.

2. WHAT’S WORKING

Before I dive into mistakes, here’s what I love about beginners’ approach to reverb:

• Fearless/Pure creativity. New producers aren’t afraid to experiment with lush spaces.

• Instant vibe-setting. A little reverb can quickly turn a dry loop into an actual ‘voib’.

• Genre exploration. Whether it’s trap, house, or indie pop, beginners instinctively use reverb to make their sounds more interesting.

That instinct is fine — it just needs refining.

3. MIX CHALLENGES (aka The 3 Big Reverb Mistakes)

Mistake #1: Too Much Reverb on the Lead Vocal or Melody

The classic rookie move. You slap a hall reverb on the vocal and suddenly it sounds like it’s three zip codes away. The emotion disappears.

• Problem: Lyrics lose intelligibility, vocal feels disconnected.

  

Mistake #2: Using Only One Reverb for Everything

Different sounds need different spaces. Throwing your drums, vocals, and synths into the same hall reverb makes the mix sound flat, not cohesive.

• Problem: Instruments blur together, stereo image collapses.

  

Mistake #3: Not EQ’ing the Reverb Return

Reverb is just another signal, and it carries frequencies that can fight your mix. Low mids especially (200–500Hz) can turn into mud city if you don’t carve them out.

• Problem: The mix gets boxy and unclear, even when levels feel right.

  

(**Bonus** Mistake #4: Too Many Darn Reverbs )

Remember, Reverb is just another signal, and when you have multiple plugins that use reverb, all of those different signals will clash, creating nothing but mud.

• Problem: The mix gets unclear and congested, leading to less separation between elements.

  

4. ACTIONABLE FIXES

Here’s how to tighten up your reverb game immediately:

• Control the vocal verb. Start dry, then add reverb until you feel it just before you hear it. Automate more reverb into breakdowns or quiet sections.

• Use layered reverbs. For example: short plate on the snare, small room on the drums, hall or lush plate for vocals. Different spaces = depth.

• EQ your reverb returns. High-pass at 200Hz, low-pass around 8–10kHz to keep it tucked. Dip around 300Hz if it’s cloudy.

• Experiment with pre-delay. 20–40ms pre-delay lets the dry sound cut before the reverb blooms, keeping definition.

🎧 IF THIS WERE MY SESSION...

If I had this track in front of me, I’d start by muting all reverbs. Then I’d bring them back one at a time, shaping the return with EQ and compression just like any other instrument. I’d set different reverbs on sends — short plate for presence, hall for vibe — and automate them based on what’s happening in the overall song.

The goal isn’t less reverb. It’s intentional reverb — creating depth without drowning the mix. That’s the difference between amateur space and professional polish.

💬 DROP YOUR THOUGHTS

What’s your biggest struggle with reverb? Are you guilty of any of these three mistakes? Let’s break it down in the comments — your insight helps the whole Lab learn.

— Will (SOVLTRON)

)

Whats up Lab Crew,

Suppose you’ve ever spent way too many hours tweaking a snare EQ until you can’t even tell if it’s crispy or cardboard anymore, you know the struggle. Fresh ears = fresh perspective. That’s what this is all about.

Why Mix Feedback Matters

Mixing in isolation is a dangerous game. After a while, you stop hearing what’s actually there and start hearing what you think is there. A second set of trusted ears can:

• Point out problem areas (muddy lows, harsh highs, buried vocals).

• Reassure you on what you’re nailing.

• Teach you new ways to approach your sound.

• Help make sure your mix translates to everything from earbuds to club systems.

• Set you up for a smoother mastering process.

The goal isn’t to turn your track into my sound. It’s to help you sharpen your sound so it hits harder, cleaner, and more emotionally.

How to Submit Your Mix

Here’s how to make sure your submission actually makes it to me (and doesn’t get lost in a swamp of broken links):

1. Upload your track properly.

• Please don’t send me files directly.

• Use SoundCloud (private links are fine), Dropbox, or Google Drive.

• WAV preferred, 320kbps MP3 is the minimum.

• Make sure your link actually works before sending.

2. Include the important details.
The more context you give me, the better my feedback can be. Please send:

• Your Name / Artist Name

• Song Title

• Genre & Vibe (e.g., “Indie Pop with an 80s vibe,” “Modern Metal,” “Lo-Fi Hip-Hop for studying”)

• Current Stage (rough mix, almost final, mastering prep?)

• Your Specific Questions (don’t just ask “what do you think?” — ask about the vocal balance, the low-end, reverb choices, etc.)

• Reference Tracks (optional, but highly recommended). Nothing shows me your vision better than a pro track you’re aiming toward.

What You’ll Get Back

When I check out your track, I’ll give you feedback that’s clear and actionable, focused on:

• Balance & Panning: How everything sits in the stereo field.

• EQ & Spectrum: Clarity, mud, harshness, and overall tone.

• Dynamics & Compression: Punch, glue, breathing room, loudness potential.

• Effects: Whether reverb/delay adds space or mud.

• Overall Impact: Does the mix serve the song and move people?

I can’t guarantee a personal response to every submission, but some tracks will be highlighted in future blog posts or video critiques (with your permission). That way the whole Practical Audio Lab community can learn together.

Ready to Submit?

Click the link below to send me your mix. Don’t overthink it — whether you’re brand new or ten albums deep, everyone benefits from feedback.

Let’s sharpen your mixes and your ears, together.

Cheers,
SOVLTRON
Practical Audio Lab

🎛️ Weekly Mix Notes: Submissions Now Open

🎧 Submit Your Mix for Feedback

Every week inside The Practical Audio Lab, I’ll be going over your mixes and sharing “Mix Notes” ; short, actionable feedback to help you get better results from your mixes. Send your beats, loops, sample work etc and we’ll cover it.

Whether you’re just starting out or deep in the craft, these notes are built to be practical, honest, and encouraging.

Want your track featured?

I’m opening up submissions for upcoming Mix Notes features. If you'd like your mix reviewed, your beat broken down, or your low-end woes solved, this is your shot.

What to send:

• A streaming link (SoundCloud/private YouTube link preferred)

• A short note on what you're struggling with (optional but helpful)

• Any specific questions you'd like addressed

What you’ll get:

• Honest, constructive feedback—focused on clarity, balance, vibe, and punch

• Suggestions you can actually use

• A shout-out in the Mix Notes newsletter if selected
  (Paid subscribers will be prioritized for featured feedback)

👉 Submit your track here:

🎧 Submit Your Mix for Feedback

👉 Not ready to submit? Subscribe to follow the series and learn from others.

Let’s level up your sound—one mix at a time.

—Will
Founder, The Practical Audio Lab

What’s a tool I use in almost every session that people overlook?

Let me cut straight to it:

Channel Strips , or more specifically,

The AIR Music Tech Channel Strip.

It’s not on anyone’s “Top 10 Must-Have Plugins” list. But I’ve been mixing records for a while now and —and I still reach for this strip in almost every session. Why? Because it works.

Why I Use It (Still)

We’ve seen the plugin boom—every flavor of vintage modeled this, analog-emulated that.

And somewhere along the way, folks forgot how powerful simple tools can be.

The AIR Channel Strip does what a good engineer always aims to do:
Fix the problems. Shape the tone. Move the hell on.

Let me break it down:

• Gate/Expander: I use it to clean up room bleed, noisy backgrounds, or tighten up live drums—quickly. Fairly simple, but for the most part.
  
  

• Compressor: Is it fancy? No. Is it fast, reliable, and musical and get the job done? yes
  
  

• EQ: 3 bands and a sweepable mid. That’s more than enough for 80% of what you need in a mix.
  
  

  
  

Where It Lives in My Chain

For me, this depends, sometimes it lives right at the top of the insert stack—slot one.
Essentially this way its acting as the ‘input’ stage of the console.

I might swap it out later in the mix if I need surgical precision or color. But in the early stages, this strip lets me clean, control, and shape the sound before any later processing.

Most of the time it lives before any additional compression, but sometimes I’ll place the channel strip as the last form of tone control before the 2bus or a submix.

Why Most People Skip It

Because they’re too busy chasing character instead of results.

Look—I love a tube emulation as much as the next guy. But the truth is, most records don’t fail because you didn’t have enough vibe. They fail because no one took the time to just clean the signal, control dynamics, and carve space.

This strip does all that in one move. And it doesn’t slow you down.

Try This Today:

Drop it on your lead vocal.
– Gate out the room or background air.
– Hit it with 2–3dB of compression.
– Sweep the mids. Pull out the mud.

Suddenly your vocal is sitting better. Cleaner. Tighter.
And you didn’t load up five different plugins or lose the groove scrolling presets.

Or try it out on drums, cut out below ~25Hz boost around 60hz for punch, cut the same region in your bass to help it ‘fit’ .. boom..

Veteran’s Note:

If you know how to listen, you don’t need a million tools.
You need one tool that gets out of the way.

The Channel Strip does just that. It’s not trying to impress you, nor are you trying to impress anyone.
It’s trying to help you finish the damn record.

Yes, there are a million versions of channel strips, but unless you’re going for a specific sound it’s best to stick with just one that you’re comfortable with

Lab Notes

🧪 Try stacking two strips—one for utility, one for tone. That’s old-school console thinking.
🎧 On drums, don’t gate too hard. Just enough to clean tails without killing the groove.

I’ve mixed on million-dollar desks. I’ve used racks taller than most engineers. But when I’m in the box, this little channel strip keeps showing up. Not because it’s magic. But because it’s honest.

That’s what The Practical Audio Lab is about.
No fluff. Just tools that work.

—

👉 Got a sleeper tool you lean on? Share it in the comments.
👉 Want more straight-talk mix drops like this? Hit subscribe. Let’s keep it practical.

Just in case you didn’t know, there are a few ‘soft’ clippers inside of the MPC/Force OS that don’t get much shine.

(sorry if i ramble on in the video, it was kindof off the cuff)

TL:DR:DW: - Drop your volume to about -18dB and run your sounds through Distortion Custom or Distortion Grimey for two extra clippers.

While AIR Distortion, Lo-Fi, Transient Designer, and a few of the Old School emulations can also qualify there are another three that you might want to take a peek at as well.

Outside of the lack of clarity in the current user guides for the MPC and Force, and the seemingly novice-focused marketing, I love what the developers at Akai/Inmusic have done. Digging back into the MPC4000 and MPC5000 manuals, I’ve discovered some gems in plugin coding that don’t seem to be widely known.

A few days ago a gentemlman posted in the MPC users Facebook group about getting his drums thicker.

There are many approaches to this but I feel MPC/Force needs a straightforward way of getting great results. So I set on a quest.

Now when making music, I like to think of myself as the intermediary between a great performance and a great recording. So with that mindset, I’ve started digging further into the MPC effects to find more unique ways of shaping the sound in simple ways.

It’s widely known that what we see affects how our ears perceive sound. Since the Akai insert effects are lacking the skeuomorphism we’re accustomed to with modern VST’s it’s easy to perceive them as archaic, useless, and lacking modern features. If you happen to believe that, to your credit you may be right But that doesn’t mean they aren’t useful tools.

In fact, outside of harmonic modeling, most plugins happen to be doing the same thing, albeit in different ways. Some EQ’s or compressors might have different curves, however, with a bit of training you can essentially replicate any sound processing. In the digital domain that is.

That might feel hard to believe, especially when juxtaposing the AIR plugins against developers like UAD or Slate Digital. But no matter the GUI or plugin design, an EQ is an EQ a compressor is a compressor.

Ultimately there are only a few ways that audio is processed. Linear Processing and non-linear Processing. Another dimension is whether the processing is static meaning it does not change depending on the signal, or non-static adapting or varying based on the signal.

With a simple understanding of these concepts or even a quick reference at the chart, you’ll quickly learn that the AIR plugins are relatively straightforward about how they process audio.

1. AIR Compressor as a limiter/Soft Clipper

As I explained in the video most compressors can also act as limiters depending on your ability to adjust the ratio (100:1) if you add in a knee parameter and adjust the attack and release to as fast as possible, you effectively get a soft clipper.

This is one way of getting thicker drums, louder mixes, etc.

Another way is using distortion. It’s pretty well known that AIR distortion is also a soft/hard clipper but what about the Akai Plugins?

Well, turns out there are two pretty good ones inside of Akai’s distortion models as well.

2. Distortion Grimy as a soft clipper

Luckily I found the MPC 5000 manual which explains how some of the distortions (alongside other plugins) are performing under the hood.

While other distortions are focusing on the frequencies entering the effect, either boosting or attenuating, there was one word that stuck out in two distortion effects “clip”

Since clipping requires a threshold, passing a signal significantly beneath the threshold will in turn enable more control over the distortion effect.

In Distortion Grimey’s case, there happens to be a sort of notch filter that when opened completely gives a boost in the 9kHz range. For most material, this filter is open enough to consider slightly warm and transparent.

This is where gain-staging comes in, if there is a clipper, what is its threshold? Essentially the plugin will peak at about -5dB to -6dB. This means that any signal below that should pass through relatively unaffected.

I found that about -18dB to -24dB is the Goldilocks zone.

In a nutshell if you drop your volume super low and use this plugin to bring it up, you can thicken up your sound without adding a ton of gain.

Even though the loudness wars are over, plugins like this not only get your sound louder with more headroom, but also improve the stereo image and depth in the mix.

Example Track “A”

Example Track “B”

Though the tracks sound relative in volume, IMHO one has more punch and clarity while still maintaining ~-5dB of headroom.

See if you can accurately guess which one.

3. Distortion Custom is also a clipper

But its a little less straightforward to use in this manner. Again you want to start with a relatively low signal and drive the signal subtly.

The MPC5000 explains that the distortion effect can be controlled starting at the zero crossing which is similar to crossover distortion found in amplifiers.

The goal is to let as much as the signal through (under the threshold) and shape the sound subtly using the amount of clipping added to the signal.

honestly, you can take the settings from the image above and apply them. It can be tedious for someone not familiar with the plugin and at the wrong levels it’s more a sound design tool than a mixing tool.

I do like how this plugin adds a bit more harmonic content to low frequencies, however it’s a very strong distortion effect. Better suited for percussive elements or a flat vocal recording.

Note from the Editor:

In some weird way I find myself rejecting the idea that the manufacturers are lying to and/ or are trying to deceive us and believe that the truth lies somewhere in the middle. For those who care, i believe that all of the plugin manufacturers /DAW manufacturers aren’t being 100% transparent about what is happening inside of the box.

I believe some of the DSP is not that complicated at its core, that some of the DSP is based on open source code, and that we’re paying for fancy GUIs (and backend trademark licensing) when we buy some of these plugins.

But things have always been that way, when looking at samplers and workstations from the past, certain Motorolla chips or voltage filters were used across multiple brands and products with the final differentiating factor being the combination of such and the coding which bind the system together cohesively.

Another thing that has also remained constant is that a great mix is a great mix. regardless of what it’s tracked/recorded on. Learning about how audio works, frequency and how to mix will benefit just as much as learning about how to program drums or flip a sample.

Rather than join the bandwagon on why something is not as good as the past, I prefer to embrace current challenges with insights from the past and bridge the gap between the trying something new and whats tried and true.

This is a tricky question..

TLDR:

Honestly, i think that adding a bit of sample offset and randominzing that will get you pretty close to what you may get from an older machine. just a small amount will add a bit of fluctuation of when the note hits just by a few samples..

Backstory

When I first started making tracks I had spent a lot of time in Chuck Levins music store toying with the MPC5000 and hanging out with some local artists in Woodbridge who were using an MPC2000XL. I promising myself one day I would own an official beat machine "instead of an institutional edition of Logic Pro 8 I ripped from the Art Institute while I was a student.

Really loving the feel of the MPC I got an MPK49 and started to look for ways I could get that same feel using what I already had..

Logic Pro has a function where you can take the groove of a MIDI file and use it as a template for other MIDI performances. Luckily I was able to locate a few collections of MIDI files from all of the more popular MPC series and turn them into groove templates.

One of the things I noticed using the templates was that my tracks sounded more ‘natural’ which was a good and bad thing. In hindsight using those templates made my life so much harder.

If I quantized anything with the MPC template, I’d have to forfeit quantization elsewhere as the timing would feel all over the place. But the tracks still felt, natural and human in a unique way.

After a while, I started acting like my softsynths were ‘real’ synthesizers and I began recording them live or resampling directly into logic. This gave me so much experience in production but was a bit cumbersome as well. (I also was using cracked plugins/DAW with no recall so I essentially NEEDED to have audio files of all my work.)

I continued to work this way inside of a DAW for years out of necessity but also because I wanted to get an MPC one day so why not practice.

But why does any of this matter?

Well in 2012 I got my hands on an MPC1000 and really felt familiar with the timing. It was similar to the 2000XL but completely different from the 5K

no it wasn’t the same as the Groove templates in the DAW, however the machine felt familiar.

What makes the MPC’s before the Rennaisance so special?

The timing on the MPC was so interesting as it wasn’t perfect. There were small fluctuations in the note timing even with quantize or Time Correction on that made even simple drum patterns groove in a way.

Im speculating here, but I think there also is a small delay between when a pad is pressed and a sample plays that add to the overall ‘punch’ of the MPC. Especially kick drums.

Either way in the simplest way, the timing of these machines really could dictate how you produced on them. You had to be musical or really ‘feel’ the groove or the loop. Night and day to today’s drag-and-drop style of producing.

Besides each model (MPC60, 3k, 2kXL, 4k) every unit had a specific feel to it and and synced with other clocks in different ways. Some swear by the MPC60’s ability to clock with external sources better, others the MPC3000. Either way all of that character and jitter was thrown out of the window with the introduction of MPC software and the MPC Rennaisance.

Let’s Really Look At This ‘Jitter’ Thing

First here is the video that really inspired me to make this post and drop this gem.

But thats Tony’s analysis. There used to be some jitter tests on the web, im unable to find them however Here is a short video of the MPC3k 'groove' template that i was working with a few years back..

Essentially this is 1/8th notes quantized to the natural 50% TC setting inside of the MPC3k. In the video I’m scrolling though midi files of notes quantized at 1/8 with swing applied all the way from 50% up to 75%.

Every other note is the ‘swing note’ so the downbeat note should be relatively still. however, You can see the 'jitter' or randomization of the start points of the MIDI notes on the downbeat.

Roger Linn swing was simply a delay on the 16th note of every 8th note ..Basically the 'and' in counting beats or the syncopated beat. however..

You can see every downbeat outside of the "1" is a little off , either before or after.. This is because of the jitter.

This is what made the machine sound human Even at quantized values . AND on top of this , MPC60 and 3000 were 96PPQN so there was a bit of quantization happening regardless just due to hardware restrictions.

96ppqn + jitter = MPC Feel.. (and punch)

below is an example of everything quantized to the ‘robotic’ or perfect DAW standard. every note is exactly where it is ‘supposed’ to be.

Emulating MPC Quantization Jitter/Groove

So if you want to emulate that MPC Jitter which ultimately contributed something to the overall ‘groove’ , then add a bit of offset to your samples.

Randomizing Offset Value

After you add a bit of offset then randomize the offset for that pad. Shouldnt be too great of value but about 0-10 should get the job done. You can slowly tweak this until you hear what you’re looking for.

Hear it for yourself

Demo # 1

Demo #2

Conclusion

For me, this feature helps to bridge the gap between the old and the new. Instead of adding swing or avoiding quantization, you can add the exact amount of timing to each note, especially if you prefer they hit a little late. This enables you to import a stale MIDI file and turn it into something that feels more human. Without touching any MIDI.

So try these simple techniques to add some vintage MPC feel to your tracks. Its obviously not going to be the same as a real MPC3k or 60, however you can use this to sound less robotic without adjusting your MIDI.

You can 100% get REAL analog warmth and punch right on the MPC..

But first things first,

This only works on MPC X, MPC Key 61, and Akai Force units..

These are the only units that have a built-in discrete mic preamp.

If you don’t have these devices, you won’t be able to use the built-in mic preamp technique I described above. However, you can achieve similar warmth and character by using an external hardware preamp, audio interface with analog circuitry, or analog summing box in your signal chain. There are also some quality plugin emulations and software solutions that can add analog-style saturation and punch to your tracks if you are working in the box.

Second, you’ll need to grab an 1/4” to XLR (male) cable.

Per the manual, using an XLR cable activates the onboard mic pre.

In any mic preamp, the components that drive the signal level are always analog.

So by using a 1/4” to XLR male cable , you can resample your audio out of the unit back into the mic preamps and add some low-noise character and punch.

This is how I’ve got mine set up.

Here are two audio demos… There's no compression on the master, just the raw beat…

Sending out through the main outs and gain-staged to avoid clipping on the inputs.

Sample 1

Sample 2

have you ever finished producing a song and wondered why it was not popping to life like your favorite songs on the radio? OR have you ever wondered what the heck your mix engineer does with your trackouts? Lucky for you I’ve discovered the trick.

But realistically,

There are a shit ton of plugin designers out there that talk about re-creating the warmth and depth and spacial orientation of analog recording equipment. While I do believe that a select few of these do their job, most of what we are looking for in a recording lies in some of the skilled mixing techniques (in addition to great sounding gear).

Without getting too technical, our ears love hearing complex sounds, especially harmonics, which often are amplified with analog recording gear. Harmonics in a nutshell make things more interesting, so one of the first things you want to do is…

Use interesting sounds

And when I say interesting you can accomplish this in a few ways. I’ll cover reverb and delay later in the article, but one of the first ways you can add depth and dimension to your music is by making your core sound bed or chord progression interesting.

One simple way of making your sound bed interesting is taking a small portion of the lead melody or vocal. This can be a few hundred ms OR a 1/4 note. Take this sound and repeat it throughout your song. you can push the sound further in the background by blending the start and endpoint with a subtle reverb.

What this does is create a ‘sound bed’ in the fundamental key of your track. This can help to separate the foreground sounds from the background and help push the musical progression forward. Just that bit of separation and movement helps create depth in your music.

Another way of approaching this would be to layer your sounds. This can range from your percussion to the sound bed, chord progression, and lead sound. A great layering tip would be to think of your layers in terms of frequency, finding separate sounds for lower, middle, and high-frequency ranges. Generally speaking, lower and mid-range tones would be a little lower in volume.

Interesting sounds help to naturally bring out the frequencies and harmonic interactions our ears love while creating unique sounds that are hard to pinpoint. I love to think of sound as a flavor and how when we taste something unique and new, our brain is analyzing so many elements that we forget to classify it as ‘bad or good’ it simply is interesting until we’ve had enough or want more.

Like flavor, some frequencies or elements need to be muted a bit to have cohesiveness. So one major thing is to use…

Smart “Doppler” Leveling

Not just balancing the faders, but thinking of the frequency or the ‘notes’ that need to be either forward or backward in the mix.

Have you ever heard an ambulance drive by and the tone change as it approaches then drive away? This is because of the different frequencies and the wavelengths of the frequencies that make up the siren. High frequency waves have short lengths and low frequency waves have long waves.

Most simply, when we hear the ambulance coming towards us, we are getting more of the high-frequency information because the high-frequency waves are shorter and perceived as louder (plus the ambulance is getting closer). As the siren moves away, the tone goes deeper as the lower-frequency waves still reach us while the high-frequency waves aren’t still reaching our ears.

Treat your audio the same way, if you want something up front bring its level up OR raise some high-frequency information. Something further away will be lower in volume BUT you’ll want to shave a bit of high-frequency information off to make the sound more natural.

instead of just a steep cut, try using bells and shelves to lower and/or raise frequencies.

When you’re mixing you always want to consider the space that the mix is happening in. Not just your mixing environment, but where IS this mix? Does the song take place in a bedroom, a club, arena? You decide, this will set the tone of the mix and help bring everything together.

I love to have my track leveled and panned so everything. where it needs to be in the ‘world’ before we get to the room. Then I’ll send all instruments 100% to the reverb room. This gives me a great idea of how to ‘set’ the reverb. The best advice I can give about that is that setting the reverb is simply…

making your reverb a real space

This means creating a realistic space based on the principles of how audio moves around real spaces. This is kindof where things get a little tricky.

We want to recreate this ‘fake’ room with several reverbs and use our knowledge of how sound travels in space. We’ll set up One reverb for the front of the room, one for the middle, one for the back of the room, and one for the overall room tone.

Each reverb is essentially the same ‘room’ with slightly different decay, early reflections, and room size.

If you were a dolphin, your life would depend on the ability to locate food/predators in space based on reflections. Think like a dolphin and think about your room and where each instrument is placed in that room. If this was a real room, your brain could tell where things were placed by hearing early reflections similar to the dolphin with echolocation. The early reflections tell us where the reflective parts of the room are. The time between the initial sound and the sound hitting those reflective parts of the room is called the pre-delay.

Things that are up closer or in the Front of the room have a longer pre-delay time as it takes more time to reach the first reflective parts of the room. Subsequently, since there are fewer parts of the room for the sound to bounce off of the ‘size’ of the room will be smaller. . Things closer to the back have a shorter pre-delay time as they essentially are the first bunch of ‘things’ that the sound will bounce off of. Your room size further back will be based on how ‘big’ you want to define your space.

(of course, you can use moderate settings in the mid-range)

This video from mastering.com covers the concept well and is partly the inspiration for this post.

But essentially you can get a bit creative with your approach to your room. One key consideration is keeping the reverb tail at musical intervals. This calculator will help you convert your BPM into milliseconds for your specific reverb. Longer tails always work for the back of the room and 1/4 note seems to be a sweet spot for the front and middle.

Even with your reverbs you can apply your knowledge of smart leveling. If something is further away, you can roll off more of your top end and add more early reflections. Something closer has a little less bass more pre-delay and fewer early reflections.

I have a great time matching my reverbs with my delays. this way the sound of the ‘room’ is consistent. So say something is closer, using shorter delay times that match the room and/or the tempo to create a sense of width up front. This plus that small reverb or even sample delay can ‘pull’ or ‘pop’ elements out in the stereo field. Even using a HAAS-type delay can bring elements forward and make them fill up the stereo field.

And speaking of HAAS…

Use HAAS Panning

Panning is one of the simplest ways to increase the stereo field, however, by itself, it adds no dimension, just left and right balance. Using a HAAS-type effect on your panned signal can add some depth by imitating the Interaural Time Differences (IDT) that we experience when we hear something in the real world that is to our left or right.

Simply enough, we rarely ever hear sound in one ear. In the real world, when we hear a directional sound, we hear a small amount of sound in the opposing ear. Because we all have big heads, the sound we hear in the opposing ear is a little later in time and lower in amplitude (ILD). So when panning in your DAW (or MPC) try mimicking these very phenomena.

For the MPC/ Akai Force system, all you need to do is grab a diffuser delay or the AIR Delay. Turn off the delay sync and set your delay time to about 6ms. Also turn the feedback down, diffusion up, and set the high damping somewhere in the middle. Pan your delay to the opposite side of your original panned signal and boom. You can adjust your high damp and mix to your liking but this should immediately sound more real.

This can sound good and in combination with your reverbs you can essentially ‘place’ things inside of your mix with some pretty good three-dimensional accuracy. The downside to processing like this is that there may be a bit of phase issues if the mix is summed to mono. In that case, try this.

Create an aux track for your desired panned signal. Add a sample delay onto the track, link the left and right signal, and create a delay of about 6ms. Pan the aux track to the opposite of your panned signal. Add a pitch shifter or chorus right after the Haas effect and detune by about 3-4 cents. This way the secondary signal has a slightly different frequency and won’t cause phase issues when summed to mono.

Honestly, I’d use this on background instruments or things that shouldn’t be in the foreground, otherwise, you may find a lot of life sucked out of your mix.

And especially for things that are in the background…

Manage Your Transients

I understand that everyone wants a punchy, in-your-face mix however, your mix may have the opposite effect and be ‘flat’ if there is no dynamic range. Things that are further away from us have less of a ‘smack’ than things close up.

Imagine someone clapping their hands right in front of you. You’re going to hear that loud clap really clearly (and maybe a bit of reverb tail depending on how large the space is). Now that same person is 100 yards away in an arena clapping their hands. you’re going to hear more of the reverberation of their clap and less of the transient.

So treat your audio like this. While compression is great at enhancing transients, it also can do the opposite and tame transients of very dynamic elements.

Either a compressor or transient shaper can tame the attack of a sound and push it further back into the mix or enhance it and bring it forward.

Again, if you combine this element with your smart (EQ) leveling, reverb, and delay, you’ll get a more realistic placement of your instruments in your mix.

Think of it like this.

Fast attack/Slow release = further away
Slow attack/Fast Release = closer

Instead of reaching for a plugin from some ‘reputable’ developer, or the one that looks the coolest or even has your favorite engineer’s name on it. First, instead, learn some of how audio is manipulated.

I’m of a strong belief that most plugins are simply handling audio in the few ways that audio can be manipulated.

Most DAWs give you everything you’ll need to mix and release quality music, so learning these tools will make everything you do with the fancy stuff even better because you know what you’re doing and why.

Can you directly apply the techniques used on the SSL console to get great punchy-sounding mixes with any plugin?

I believe that you can,

Now of course I don’t have an SSL G series console lying around for comparison, so we’ll just have to accept the results for what they are.

But even without the gear, our ears have been used to hearing some of these compression styles, even outside of music not mixed on SSL boards or with Their Duende line of plugins.

Now why does this even matter?

There seems to be an entire sect of the MPC community that is completely focused on the conversion process of digital audio to analog audio. I think the group is really looking for something like this :

While reasonable to think that paying over $1k for a sampler would demand the sampler to have some unique sonic quality, Im a huge proponent of two things.

1. use what you’ve got and be glad you’ve got that

2. If the thing makes your tracks sound good by itself, imagine what someone who knows what they’re doing can create with it.

The focus on “analog sounds better” or “what converters are in the new MPCs” even the whole, companies are lying to us, etc. is completely garbage and uninformed.

A lot of mix engineers have transitioned from analog gear and copied their mixing techniques from their outboard gear to their virtual counterparts and are successfully combining both worlds when needed.

and we all know the digital sound domain is clearly not the same as analog, however, you can still get great results using proper skills that result in punchy, loud, full mixes.

So, using what I have, I applied some of these techniques, especially the ‘SSL 4000’ compression technique to the AIR Channel Strip which has a similar signal flow to the SSL.

Here is a brief video on the trick, There are several out there however this is the one I based my experiment on.

For those without ten minutes to watch, essentially you’ll start with a somewhat neutral compressor.

Since we’re not working with an actual SSL emulation, we’ll want to set your release settings to about 400ms and your attack either as fast as possible OR about 15ms.

I prefer a bit of both depending on the source, for vocals and instruments I’m looking for a fast attack, drums I’ll use a little slower attack to let the transients pass through. Try not to go above 30ms.

Next, take your threshold and set it to about -20dB

adjust your ratio to 1:1 so that whatever passes through the compressor is not affected.

Now as you play your audio, start to increase your compressor ratio until you’re achieving about -6dB to -10dB of compression.

I know the original trick is kindof locked in at -6dB, however, use your ears for this stage as well you may be going for a little more compression based on the source.

Finally, let’s level-match your output to match the source and do a little A/B comparison.

What you should find is a more punchy, in-your-face-sounding sound.

I love what the AIR channel strip does with the stereo image/spacing in the source. the sounds are more percussive and in your face and seem to fill up more of the stereo field.

You can try this ‘trick’ or technique with your favorite compressor AND it works pretty much on any source..

BONUS GAME

I saw this video on adding width to a mix with just two plugins.. Well, kindof..

In the analog domain, harmonic distortion and crosstalk are really the magical elements that give analog a slight sonic advantage over digital. These harmonic excitements and imperfections are pleasant on the ears so when needed, we can use them as a subtle mix tool.

Most of the time we’ll reach for a distortion plugin to add some character to a sound, however, distortion is a very powerful mix tool for several different reasons.

1. Distortion sounds good.

2. Distortion adds harmonics which help our brains locate sounds easier.

3. Distortion can add a bit of glue to a mix

4. Distortion can tame transients to an extent

5. Distortion can improve the stereo image.

so using a little bit of magic IE skill, we can accomplish a few things at once with just a bit of distortion before our compressor

That’s right, BEFORE!

unlike a guitar pedal, we’re not looking to even out the sound ior change the tone, this is MIXING..

on an analog mixing console, the signal would have been run through multiple transistors and/or capacitors before reaching the final bus compressor. So placing a distortion or saturation plugin before your compressor (or even on your channels) can give the effect of mixing on a large console.

I love the AIR Creative FX collection and use AIR Tube drive but you can place any saturation plugin before your 2-Buss compressor and it should still do the trick.

Try not to ‘drive’ the signal per se, just use enough saturation to bring the sound forward without too much harmonic noise..

Here are the settings I use for Air Tube Drive.

Remember, a great mix isn’t made just because you used the converters on the MPC3000, transistors on the SSL board or these nifty tricks I’ve shown you here.

All of these tips are tools in your arsenal to use when you need them, and ONLY when you need them, not just because you can. Thats the quickest way to muck something up.

SOVL

So recently and for the past 20 years there has been a debate on whether FL studio 21 or FL studio 11 ‘hits harder’.

Even Image Line joined in the debate releasing this video explaining how FL studio’s audio engine is the same in every release.

I firmly believe in the simple (kindof unknown) fact that gear sounds good not just because of what it is but what it is doing.

So let’s look at what FL studio did great and let’s emulate that with plugins.

Does FL Studio 11 knock harder than FL 20??? 👀 🤯

”When you open FL 11, there is a limiter on the master just like FL 20. But when you look at the limiter, the gain knob is set to 5.5dB of gain.

In this two minute video Decap masterfully covers what Imageline took tein minutes to explain in their video.

If what Decap AND Imageline are saying is true, emulating this on any system shouldn’t be that hard technically.

Looking at the settings it seems that there is a slight envelope on the gain reduction. We can assume this is a kind of ‘soft clipping’ effect.

Clipping is not something you want to totally avoid, it can actually help add punch and dynamics to your drums in a mix without raising the dB too high.

But before we just run to slam our mix into a limiter, we know that after a few dB of limiting, we’re going to run into clipping in a bad way and destroy the mix.

Of course, we could just throw a limiter on, add 5.5 dB, and adjust the lookahead to 3-5 to match FL studio’s curve settings. But that would be boring lol..

Remember we’re talking about mixing, so at this point, we’re throwing your ‘engineering hat’ on.

this way you can use this trick in all kinds of mixing or mastering applications


so first gain stage the mix so that each track is hitting the digital analog sweet spot (-18dB..) then of course push things forward or backward to your taste..

If we’re talking about that “Southside/ 808 Mafia type sound” then your Bassdrum (kick ) and 808 can be a little louder in the mix.
(Try to have a bare minimum -10dB of headroom on your master fader)

I highly suggest watching this video below.

So after watching the video you might be tempted to place your clipper before your limiter in place of AD conversion using the limiter to bring things up right?

But this is why we place our engineering hats on, FL is actually doing things a little bit differently.

So knowing what we know now, we’re going to put a clipper AFTER our limiter in the signal flow..



we want to reinforce the ‘pocket’ in the beat and have more of a polished release sound.

So first I grab my AIR Distortion Plugin.
You can copy these settings here

We’ll add a bit of soft clipping to the signal before after we use limiter to add more consistent gain reduction.

Since we don’t want to completely slam everything remember the principle of mastering -2dB is pretty much the maximum amount we want to limit.

So this is where you just adjust your gain, aiming for -2dB MAX gain reduction on your 2 Bus.

If you need MORE punch I’ve found adding an MPC3000 emulation (or bit crusher) before or after limiting sounds pretty good too, it’s like 2 layers of clipping

Just know that if you’re sending it to Mastering it will have distortion and it’s LOUD.

And that’s kindof where their argument about ‘sound’ gets hilarious..

if you want that “sound” just turn the volume up.

Well, it used to be that simple in the analog domain, however, it’s the same concept in a DAW.

Just add a signal flow limitation that adds some great harmonics and drives that signal up until it sounds good.

So of course here are some examples

Track 1.

I use Air Enhancer to bring up the highs and the MPC 3000 emulation before the soft clipper to add more punch.

Track 2.

I turn off the ‘soft clipper’ and add 5dB to the limiter. While this mix is ‘cleaner’ it lacks that ‘finished’ sound of Track 1.

So in Track 1 you can hear that the soft clipper is pushing things up and in your face with that ‘finished’ or ‘polished’ sound.

For me this is the sound im looking for, if I wanted more clarity and space, I’d send in option #2.

So try this out, make sure to gain stage your tracks, even if you’re working at 32bit float, plugins may operate differently

Also these are the types of mixes that mastering engineers HATE to get

At this point you’re kindof acting as the final engineer, you’re determining how loud the final product is.

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Whats up Forum,

King petty here. IDK why but I cannot get this damn sonics and converters conversation off my mind.

I think what really grinds my gears is that I’m seeing that most of the people complaining about the new MPCs are not using them the same way they would use their older MPCs.

understandably, the newer MPC’s give you more options and of course virtual instrument options.

So What’s My Take on It ?🪙

Honestly, I have a love/hate relationship with all VST’s. My questions are always :

1. What is the sound I’m looking for and is it giving me the sound I want?

2. Is it using up unnecessary processing power?

Especially when im re-creating samples of yester year, im thinking to myself “how can I do this with only what the producers of yesteryear had?”. I’m always revisiting production, sound design, and engineering techniques, that they used.

What’s amazing to find is that signal processing doesn’t really change, we just like adding convenient (sometimes processor-heavy) VST’s to do the dirty work. But it’s really just frequency, time, or pitch/ manipulation and distortion that we love the most.

When it comes to the stuff we made on the samplers of yesteryear, HOW you sample something (especially from vinyl) can make ALL the difference in the result..

And NO i haven’t found a plugin that can accurately emulate this sound because its ACTUAL manipulation, not simulated..

So my take on this is actually this entire breakdown of this process. Really to illustrate two points and ask a question:

1. You get completely different results when you use the processes the engineers/producers used back in the day

2. doing all of that manual processing is a TON of work.

*I got today’s inspiration from this video. 👇🏾

Emulating Sampling at 78RPM on MPC60

I’m working in Ableton today but essentially EVERY step of this should be possible in any program.

My goal with this is to go through the process like I would if I had a turntable and a rudimentary sampler to load up with sounds.

This is BASICALLY the same thing I’d have to do on my MPC1000..(minus converting to stereo)

SO first im just finding any stock sample, but you can do this with anything you want.

Next, We’re going to load that to another track

And then we’ll freeze that track..

Then drag that audio over to an empty audio track.

So now when you click on that clip you’ll see some information about the clip.

You’ll want to change the warp settings to ‘Re-Pitch’, this will ensure that your audio is pitched up relative to the tempo.

So before adjusting the tempo head over to this Pitch Correction Calculator to check how many semitones the sample is raised when adjusting to the desired turntable speed.

Essentially the main thing you’ll need to remember is the total number of semitones that the sample is raised. You’ll enter that number on the Tempo and Pitch calculator.

Once you’ve got your new tempo for your project you can now adjust your tempo.

At this point, we’ll switch to the sampler. Im using the Akai Force but you can do the same thing by resampling in Ableton,edison or whatever DAW you’re using.

So before I record the sample into the MPC/Akai Force, If i’m looking to add some realistic vinyl noise, I’ll add some vinyl texture on the frozen track.

I always make sure to have my output levels peaking at about 0dB VU on the meter.. this is the ‘analog sweet spot’ (same for the inputs on the plugin)

Next is to set my monitor on the output plugin to the Right Channel and do the same for my MPC input. (make sure input is MONO)

That should leave us with a beautiful mono track like this one.

We’ll do the same thing for the Left channel and add them both to a pad.

We’ll edit the Sample a few ways,
1. Reduce the sample by 14.78 semitones
2. Hard pan the Left and Right channels.

and to prevent phase issues make sure both samples start at the same time.

Yeah. That was a lot of work, right?

Remember, back before onboard memory was virtually limitless, the samplers had a limited amount of processing power.

So all of these were necessary to get music INTO the sampler let alone create stereo samples (MPC60).

So now that we’ve broken our backs with all of this processing let’s listen to the sound before our processing

And after we’ve repitched the sample, hard-panned the left and right channels and truncated any extra space at the beginning of the sample.

taking it a step further you can even lower the bitrate to 12 or 11 bits to get more of a gritty sound.

Try it out for yourself.

If you’ve had an older MPC then you’re probably familiar with this approach.

Final Thoughts💭

I think that the work involved is really fun because the results have so much ear candy and are interesting enough to make some really good-sounding music.

However, there comes a point where the grit doesn’t really sound so great coming through a hi-fi system unless the sample is great.

This always brings me back to the point that good music sounds good not solely because of grit, or any special sauce added to it or the converters that it was run through.

IMHO, It sounds good because the energy behind it is good, and the music itself reflects that…

Running it through some signal processing is just amplifying the harmonics of something that was good to start off with.

BUT thats my opinion,

LMK what you think in the comments

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What’s up forum!!

So let’s be frank,

Most of these drum sounds included in a majority of these expansions and whatnot are essentially recycled AF.

I can’t tell you how many expansions I have right now, that have the SAME EXACT kicks/snares and vocal samples.

I would be super annoyed but .. ya know

some we paid for.

others we got discounted

IYKYK…

So what I did notice is that after a while not only did using the same drum sounds over and over make my patterns sound stale, BUT

There were some simple production things I was leaving out of my tracks that hurt my beats..

So for my guy Lanse S., this one is for you too.

I was inspired to write this post after you posted this:

So here we go!

Step #1.

Gain an understanding of three elements that make great drum progressions work.

Groove 〰️

- This has to be the most important element. This isn’t swing, it’s more the flavor and the style of the beat. Groove CAN be created ‘on the grid’ however you want to get away from quantizing too much when creating a great groove.

Dynamics 🔊

- Genre dependent, dynamics are just as important to drum programming as groove.

EDM, hop-hop, etc. have very punchy kick drums and in-your-face snares and hats, whereas alternative and other acoustic forms of music typically take a softer approach to their dynamics on the drums. Pay attention to these nuances in different genres of music.

Space 🖼️

- So in digital genres like hip hop and bro step, the space that the drums occupy is pretty dry. With acoustic drums, they are typically mixed up together and bleed, and the sound of the room contributes to the space already. Elements like reverb, and foley will help define the ‘space’ your drums are sitting in. Absent this your drum programming will sound flat af.

Step #2.

Find great drum samples and organize them into kits 🧰

It’s possible to recycle your drums over and over just as long as you’re creating some kind of interesting variation or combination of samples. These samples should also be the basis of your groove, do you have a punchy kick on the downbeat or is it soft? how hard is the snare hitting about the kick? These choices also factor into step 3.

Step #3.
Know what genre you’re producing for and be creative with your approach 🕺🏾

Are you driving the track or simply the accompanying rhythm and beat? How do your favorite songs approach the rhythm section? this is where all three fundamental elements of a drum groove come together as well as where the the three steps to making a great drum groove converge.

Here is a beat breakdown by DJ Dahi.

One thing that is instantly noticeable about the drums is :

• His initial approach to creating the rhythm is completely creative. The entire track is based around this (and he resamples!!!)

• The rest of the sound selection is really good… not everything needs a kick drum.

• he’s vocalizing the ‘rappers bounce’ or cadence while programming drums.

• the creative process is relatively quick

The entire process didn’t really take a long time. He had a great idea of where the sounds he wanted to use were (step 2), created a groove (step 1) knew the direction he was creating (step 3 ish), and dove right in with confidence.

*BONUS*

So here are some other tips for your drums…

in some varying form, I use these methods on most of my tracks.

Part of getting great drums is getting the sound glued together, compression can help bring your drums to life adding punch, improving the groove, and controlling the dynamics.

• Flip(automate) the panning on your drums on certain sections

• Use a transient designer to add punch

• Use a clipper to get extra gain without sacrificing quality

• Use silence to make things way more interesting.

• use distortion to make things a bit louder

• Create interesting sections with a sound designer like RC-Retrocolor, Air Sprite, Tape Mello-Fi, or Air Lofi Pro.

• Get it right quickly and move on

Check THIS out.. (MPC users )

So, have you ever loaded an instance of this EQ and gotten some CRAZY sounding results when you’ve tried to eq something..

yeah me too..

Its hard to really hear whats happening, sometimes it feels like the darn thing is broken..

Well not quite..

It seems that this was developed without an anti aliasing filter or something.

after running it through a EQ Curve Analyzer I Found these results below …

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So it seems that as you increase the Q, the amount of gain applied reaches the Nyquist frequency causing aliasing or the ‘mirroring’ of frequencies back into the signal.

So if you want to avoid errors in your audio or EQ settings then you’ll need to adjust how you’re approaching your EQ..

*Treat the Q gently*

NOW…

If you want to get even more precise/surgical EQ you’re going to need to recognize a few things..

Since there is no apparent anti-aliasing filter inside of the plugin, you’ll need to know a bit about how the Q on EQ works.

Really what this means is that the higher the Q, the more narrow the bandwidth..

that’s about the extent of my knowledge (for now)

lol

But nonetheless

With PEQ4 and PEQ 2Band 2 Shelf, increasing the Q value is also increasing the gain…

However, this isn’t globally..

The plugin seems to let more low-frequency information in when trying to cut.

SO you’ll need to dial back a bit on the gain and Q to get the settings you want..

That sounds like a lot..

however, I actually kindof like this EQ..

It reminds me of old busted hardware lol

.

So to sum up how to use it based on these minor discoveries

Use -1.1dB for precise cuts at Q values greater than 50.

Use -2.2dB for deep cuts at a Q value of 50

and Ideally, you can ‘kind of’ view it as if you are half the q value then you can add a little more gain reduction.

OR you can just follow this simple ‘PEQ4 CHeat sheet lol “

PEQ4 CHEAT SHEET

Hope this helps..

Again this should work on both PEQ 4 and PEQ 2band 2 shelf

Till next time

Recently I’ve been working on getting the best mixes possible with the current gear that you have.

I’m under the belief that you can get a phenomenal mix without the assistance of a stacked channel of plugins or the latest and greatest from *(insert company)*

But to do so, FIRST, you must have good sounds and a great rough mix to start with,

And ONLY when you have those things do you need to worry about step two:

So, gain staging is one of the FIRST things you want to do when you start mixing ..

Why?

Let’s resume the age-old ‘analog vs. digital’ debate, shall we?

ok, not really but in a way these are related.

When working on any analog system, there is what’s called a ‘sweet spot’

This is basically where each individual compressor, amplifier, or console circuitry works the best.

The manufacturers/designers of the circuitry (back in the day) optimized the circuitry inside of the equipment based on a certain signal or input level.

Whether it’s the ‘optimal’ input level OR the slightly louder level where a small amount of harmonic distortion is added to the signal, audio engineers found that this produced a very desirable sound in the end result.

So fast forward to more modern times, plugin developers started to design their analog modeled VST’s after the circuitry in the older hardware machines.

Above you can see how Slate Digital explains their VU meter modeling.

The importance of this is simple and completely related to the analog sweet spot the plugins are modeled after.

So where does that leave you

Lets look at your mix channel by channel…

So i’ve got a simple mix pulled up on MPC software however, what i’m going to explain applies on DAW AND standalone systems lie the MPC, Maschine, Akai Force and others.

If you look closely you can see we’re clipping on the master..

So GENERALLY, in the analog domain clipping is okay, however, digital clipping is highly undesirable.

So there arises the need to ‘stage’ the gain in digital audio.

The first thing that most of us do is immediately lower the fader so there is no clipping on the channel and /or master.

If you’ve got a 100% RAW mix and plan on exporting that not using ANY plugins whatsoever then that’s fine..

The SECOND that you think about adding a plugin you’ll need to assume your gainstaging.

Q: Do I have to Gain Stage all of my channels?

A: NO! depending on if you’re adding dynamics processing or how much headroom you might not BUT its a good practice to do so anyway.

So Before you start lowering any fader or adding any plugin, you’ve got to figure out how to lower the gain of the signal.

I highly recommend adding a VU meter to the master fader just for monitoring purposes. This wont add any extra gain or coloration all this does is convert your dBFS reading into a dBVU reading.

As you can see my VU meter is off the charts too, You’ll see that the reading is a bit different as the max in the DAW was about 6dB, and on the VU meter its reading is 15.1dBVU

So,

Next and this is critical, remember when I said you need to lower the gain BEFORE it gets to the first plugin stage. There is a reason for this and its the general signal flow when it comes to DAW audio.

Here is a diagram from Producerhive.com

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You can see that the sound source comes first THEN the insert effects..

this is how you get EVERYTHING to sound better, I cover it here in my reverb post

Gainstaging essentially happens here..

right before any vst…

*Plus if you were to use the volume faider, how ill you get fine gain adjustments.. The fader is based on a logarithmic scale… Basically small movements lower in the fader equal bigger shifts in gain.. just look at the fader.

So any way doing THIS is essentially MIX gain staging.

(Recording gain staging is essentially the same except you’re doing this for your mic preamps or compressors)

In Logic Pro there is a small section above the channel insert slots that is the channel gain, this is completely separate from the fader because it’s BEFORE the insert slots.

On different systems like the MPC or Akai force, you may have to adjust the gain with either a plugin OR at the sample /instrument level.

Same thing for MIDI tracks, On the plugin instrument, adjust the volume fader until the VU meter is averaging around 0dB.

OR until your ear is telling you that it’s the ‘sweetest’ dB.

So essentially that’s gain staging.

BUT

There are some critical things to remember first.

1. Your main instrument or desired focus should stand out, so start with this element. So if this is your kick, or your 808, sample, or vocal determine this FIRST and the rest should follow in order of importance
   

2. Mix in mono, this will give you a great idea of what the level balance is of the mix. believe it or not, mono mixes translate the best as they are fully balanced between left and right.

3. Use a reference track, without this you’re essentially mixing with zero target.
   

4. Solo the track at its loudest point, this will ensure that you have no rogue peaks that your bus compressor has to tame.

5. VU meters are HORRIBLE for drums or materials with short transients, aim for a level way below 0dB on the VU meter, You’ll need to trust your ears, saturation on drums is always nice, BUT avoid creating issues for your mastering engineer.

Gain staging alone isn’t the key to a great mix, it takes starting off with GREAT material.

Essentially your mix should already be pretty decent to start with and all of your efforts will bring the audio under whatever focus you desire.

Cool beans.

What’s even doper is that all of this can be done on ANY system that you’re on.

Q: Ive heard that -18dBFS is basically the same as 0dB on a VU meter.

A: Wonderful in theory, try it in practice with a sine wave THEN other percussive material, you’ll find its not a one trick pony..The best judge is not a blanket standard but your own ears.

Over the past 15 years or so, analog emulation has been a hot topic.

I can remember when UAD came out with their SSL emulation, and shortly afterward Native Instruments, Waves, and all sorts of companies were attempting to emulate the analog sound.

Then came the second wave of analog nostalgia, the Lo-Fi craze..

RC20 really changed the game, however, what it really takes advantage of is signal flow, distortion, and modulation.

understanding these concepts will allow you to create essentially any Lo-Fi or degraded effect you want.

BUT

We’re lazy,

So here are some of my absolutely essential/day-to-day retro/analog/Lo-Fi plugins that I use to make tracks sound like this

Air Flavor Pro

This is a plugin that I know I prayed to the Akai Professional Gods of Development for. After poking around in the Mellotron effects plugin section (Akai Force/MPC series) I realized we were missing something really big as an effect.

I probably tweeted my excitement a few more times than I’m proud of.. however.

Shortly afterward they released the flavor plugin and finally the paid version

Flavor Pro.

This is a unique one and I’ve found the vinyl emulation to be a little less than adequate at times, however,

everything else is a go.

Arturia - Tape Mello Fi

Tape Mello Fi is one of my favorite tape plugins on the market.

The tape drive is perfect for emulating the limiting effect of tape mastering.

When I’m using this, I’m typically re-creating the types of effects you would expect in vintage Japanese anime.

I’ll combine anything from Airwindows before this in the signal flow and drive it a bit for some beautiful-sounding distorted FX and samples.

Ploque Arts - Chip Crusher

Years ago I found this plugin while trying desperately to emulate the MPC sound.

Yeah, back in 2008-2014 I was obsessed with sounding like my beats were made on the MPC.

From resampling and degrading, I can personally say I was kind of on the Lofi tip before it was really a ‘thing’

So this is one of my secret sauce plugins.

Especially for sound design or just adding something different to sample or mix elements.

The cabinets are perfect for unique-sounding reverb effects and the different DAC encodings add some subtle harmonics that sound a bit beautiful when driven.

JMG Sound - Retronaut

Retronaut is another secret sauce plugin,

you might notice that the plugin setting is on its ‘init’ setting. that’s because this thing is good right out of the box.

I haven’t really dove into the features and ‘how’ of this plugin really.

It just sounds good, and has some pretty good presets..

Definitely worth a deep dive in the future.

XLN Audio - RC-20 Retro Color

RC-20 is one of the more popular plugins out there. I personally like maybe 5 or so presets and that’s about it.

To really get the most out of this plugin you need to spend some time with it and think outside of the box.

Plugin Alliance - Black Box Analog Design HG-2

Depending on the source material I may put this at the beginning or end of the signal flow.

This is an all-around ‘analogifier’ and brings an element of dimension and power to the sound.

This sounds great on an overall mix AND individual samples.

Personally, I just turn a few things on and use very subtle adjustments to the sound until it’s popping how I want.

Now for my super secret Gem .. only for premium hehe lol

Baby Audio - Super VHS

This is one of my favorite plugins honestly..

This things is pure magic.

I really think it sounds like the VHS era of things and love running synths through a chain with this at the end.

It sounds great right out of the box but when you turn on the ‘Magic’ button and add a bit of static you’ll see how this plugin shines.

So I DO use these plugins in ‘standalone’ mode, however, I’m typically running some VST through the FX chain and sampling into the Force.

This way the effects are ‘printed’, they sound great, and I can mix and match arrangements of fx (through the different samples) without any CPU interference.

This is actually a workflow I picked up working in a DAW trying to emulate the MPC2000XL workflow.

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These FX are great by themselves but wonderful when you combine them.

Remember, sound design is about signal flow, so re-arrange your plugins and experiment with different/random preset combinations in your sampling.

Be creative!

And of course if you find value in these types of posts consider subscribing, it helps support my publication (like the affiliate links above!😇)