4 Techniques That Will Change Your Outlook On Mixing

Throughout the process of learning how

to mix music, many people hit a wall

where they know something's wrong, but

they can't quite [music] name it.

Whether the kick is fighting the bass or

the vocal is just sitting on top of

everything instead of being inside the

mix as a cohesive piece. Whatever it is,

you end up turning the same knobs back

and forth for about an hour, and somehow

[music] you end up worse than where you

started. If you've done that, leave a

comment below. I definitely have done

that, and I'm sure we're not alone. It's

not that you're doing something wrong.

What's missing [music] is the connection

between what you're hearing and what you

can actually do about it using the tools

available.

That connection

>> [music]

>> is what a good education gives you, but

it's harder to find good education

>> [music]

>> than it should be. A lot of mixing

videos on YouTube, my own included, tend

to teach you how tools work,

>> [music]

>> which is helpful information, but it's

less common to find videos that tell

[music] you how to translate what you're

hearing into adjustments on those tools.

Today, I want to walk you through four

concepts, one for each of the primary

tools you're probably already using: EQ,

compression, reverb, and delay. Each one

is a specific idea that gave me a

framework for when to choose the tool

and what to listen for once the tool

[music] is in the signal chain. These

come from the mixing and ear training

program at Audio University, which is

taught by myself and [music] Gabe

Herman, who's been teaching audio

engineering and mixing professionally

for over 20 years. I'm going to pull

some clips from the course so you can

hear Gabe explain these ideas directly,

>> [music]

>> and if any of this lands for you, the

program is linked in the show notes

below.

Let's start with EQ because I think it's

where most people spend most of their

time, and it's also where most people

are working much harder than they really

need to. The typical approach goes

something like this. You hear something

off in the mix, you open an EQ plugin,

you sweep a boost around until you find

a frequency that sounds bad, and then

you cut it. And the logic here makes

sense, and it is really a good tool and

a good technique that I still use at

times.

The problem is that it can lead to more

problems than it solves. If you apply a

big boost while you sweep through the

frequency spectrum, you can make almost

anything sound terrible at almost any

frequency, which means you end up

cutting things that weren't actually

problems in the first place. There's a

faster way to get your bearings with EQ

that one of my teachers taught me.

Certain frequencies correspond to vowel

sounds. Your mouth is a resonant cavity,

and when you shape it to say a vowel

sound, you're resonating at a specific

frequency. Once you make that connection

between what frequency corresponds to

what vowel sound, you have a reference

point every time you open an EQ.

Around 250 hertz, you get the "oo" sound

like the word boot.

Move up to 500 hertz, and it shifts to

an "o" sound like boat.

At 1 kilohertz, you're in "ah" territory

like the word father.

2 kilohertz is more like "a" as in face.

And 4 kilohertz is an "e" sound like the

word meet.

Above [snorts] 4 kilohertz, you start to

get into the sibilance range. 8

kilohertz gives you a clean "s" sound.

Higher than that, it gets sharper,

almost like an "s" mixed with a "t" at

16 kilohertz.

Below 250 hertz, you can focus on the

physical sensation of the sound in the

body. 125 hertz, you tend to feel in

your chest, and 63 hertz, you feel

lower, especially when you're listening

through full-range speakers or a system

with subwoofers.

What this gives you is a way to connect

what you're hearing to where you need to

go on an EQ. A vocal that sounds boxy

might have too much energy around 500

hertz or 1 kilohertz, somewhere between

"o" and "ah".

A mix that sounds harsh usually is a

buildup of energy sitting between 2

kilohertz and 4 kilohertz, between "a"

and "e".

I'd encourage you to download the free

ear training guide linked below this

video for a quick reference for each of

these reference points I've just shared

with you. You can also download it at

audio

universityonline.com/eartrainingguide.

This was a huge revelation for me when I

learned it, and even if many engineers

don't use this exact method, they

probably have some tricks of their own,

reference points for what they're

hearing.

Here's Gabe describing how he approaches

an electric guitar in one of the mix

sessions within our course. This guitar

really speaks from kind of the back of

the mouth. [music]

It's got like an

kind of [music]

vibe to it, and I use my mouth and I use

the sounds in my [music] mouth to EQ all

the time because it helps to kind of

place things in my body, and then I can

internalize them better.

>> That instinct, translating what you hear

into something your body already knows

and understands, is a foundational

element of ear training that actually

sticks with you. The key is repetition.

You hear it, identify it, apply a

change, and check the result. And over

time, you stop guessing and start using

EQ with a goal in mind.

Okay, compression. Most people learn to

think of a compressor as a dynamics

tool. It catches loud moments and pulls

them back, and that's accurate, but it

misses something else that's important

about what compressors actually do to

the sound. And after watching Gabe teach

this, it really started to click for me.

Here's how Gabe puts it in the course.

Compressors make square waves.

They are uh

really not generating the the effect

that most people associate with dynamic

range control. While they do control

dynamics, it's the interaction of their

dynamic range control

um devices and and components that

generate um harmonic saturation and

harmonic distortion

in a way that is different than just

dynamic range control. And so we need to

be able to think about dynamic range

control separate from harmonic

distortion, and you might get further

learning how to use a compressor not

just by hearing how the dynamics of the

signal change under compression, but at

the same time learning to hear what the

harmonic qualities are of too much

compression

um or poorly timed compression, etc.

You should focus on the intent. Why do

we use a compressor? What changes do we

want to see in our sound? What should

stay the same?

Think of distressers uh uh think of

compressors as distortion boxes and pair

them with complementary source material.

So, if you have um a uh

a very fast transient in a snare drum

signal, say you've recorded your snare

with an API uh 512 mic pre with an op

op-amp circuit, um those do a great job

of preserving transients. It may be ad-

uh advantageous for you to uh then use a

compressor paired with that signal

that's able to handle very fast

transients.

Um so, think about the um the the tool

as being um the way that tool is

designed to be applied to a specific

type of sound, and that might help you

develop uh a better strategy, and we

will discuss strategies as another part

in the series.

Lastly, I want you to remember, and this

is the most important thing, and you'll

maybe get tired of me hammering on them

so hard throughout this, but compressors

are not intelligent. You are

intelligent. Compressors as distortion

devices, not just as dynamic

controllers. Every time a compressor

acts on a signal, it's not just turning

down the loud parts, it's changing the

shape of the waveform, and that change

produces harmonic distortion. The

character of that distortion comes

directly from your attack and release

settings. A fast attack means the

compressor clamps down quickly and

starts reshaping the waveform almost

immediately. And a fast release means it

recovers very quickly. The more

aggressively the waveform gets reshaped,

the more harmonic distortion is

produced, and the more the character of

the sound changes. Slower settings mean

less reshaping, less distortion, and a

cleaner sound overall. That's a big part

of why two compressors with different

time constants can sound completely

different at the same ratio and

threshold. The timing settings aren't

just controlling dynamics, they're also

controlling character.

Here's Gabe working on a kick drum using

compression specifically to shape the

tone.

I want to talk about setting attack and

release times and how I might use a

compressor to shape the tone of a kick

drum.

In this session, I've got just one track

we're going to be listening to, which is

the kick in mic. Let's give it a listen.

In our previous session on EQ, I

discussed how I used this frequency

curve to enhance the sound of the kick

drum. I'm going to bypass that so we can

hear the difference of the signal

without the EQ versus with the EQ.

Without it.

With it.

What I'd like to point out is that we

used an EQ in in previous session to

enhance the low frequency content and to

make the uh articulation of the kick

drum a little stronger while reducing

some of the frequencies that were

shadowing the upper mid-range um closer

to around 200 hertz. We call this the

mud range.

Um today when we install a compressor

behind this signal we're going to be

changing the envelope and introducing

harmonic distortion to that signal.

When I think about what I want to change

about this signal, I have to be very

careful.

If I were looking at the frequency

responses drawn on the waveforms of our

edit window, you'll see very fast

transients followed by some pretty quiet

resonances.

Uh of course, this is what the waveform

looks like before the EQ, not after.

If I wanted to do that, if I wanted to

see what it looks like, I would have to

commit the audio up to this level

which I'm going to do right now.

As I do this, what I'm going to see is a

change in the waveform that will

actually represent what's different now

that I have

um made the uh EQ rendered.

What we see now is that before the body

of the kick drum, which was a little bit

weak, has been made a little bit

stronger.

But we still have these very strong

transients

followed by some relatively quiet

resonance of the kick drum.

What I'd like to do is use a compressor

to tame the transients and make these

sections here after the transient a

little bit louder.

If I use my cursor tool to select just

this amount of time that is my transient

for this kick drum hit I can look up and

see that it's about 1 millisecond of

time here from start to end.

A difference of .096

uh milliseconds.

Or a tenth of a millisecond I should

say.

So when I set up my compressor, what I

want to be very careful of

is that I'm not making the time too

long. If I make the attack time too long

I'll uh end up setting up a compressor

that's too slow to be able to react to

the signal.

And that means I'll end up compressing

what's happening on the back end of my

signal, this later part which is the

body

and the resonance in the decay of the

tone of the of the envelope of the

signal.

That wouldn't be good at all because

what I really want to do is squash the

transient, not the body. So I need to

keep my attack times pretty quick.

Lucky for me this stock compressor that

comes with Pro Tools is very quick. I'm

going to set the attack time to 10

picoseconds. That's way fast enough to

be able to uh uh

uh get out in front of this transient.

At the same time I want to make sure

that my release time is also very fast

because I want the compressor to be able

to stop compressing relatively quick.

I need to reduce the volume here. If I

go with something strong like 6 to 1

compression that's going to create many

more harmonic uh

uh uh uh

auto order harmonic uh elements in my

signal than I think I really want.

I like the way this sounds right now.

It's warm, it's punchy, it's thick. If I

start introducing too many auto order

harmonics to the signal, it's going to

change the timbre of the signal to be

more crunchy, more gritty, maybe a

little bit more aggressive, which is

fine for some music, not in this case.

So I'm going to set my attack time to be

very fast, my release time to be very

very fast. I'm going to set my ratio to

something a little bit more prudent, say

2 to 1.

I'm going to be very judicious with my

threshold. I want to make sure that the

signal is just barely coming up over the

threshold. Remember the further down I

bring the threshold, the more of the

signal will be squared off.

So I really don't want to be too

aggressive with this or else I might end

up having it sound a little bit crunchy.

Just to demonstrate what that sounds

like though, I am going to show you what

it sounds like when we go a little too

far.

But let's start with just trying to find

a good starting point. So I'm going to

bypass

uh this

compressor. I'm going to select a little

bit of uh signal here and we're going to

loop it.

>> [music]

>> Now we have our

our kick drum soloed.

I'm looking at the meter and I'm seeing

that the threshold the way it's set

right now

the needle just basically pops right

over the edge of this threshold line

just when the transients hit.

I can also see that it rebounds back

pretty fast.

I'm going to up my threshold just a

little bit more.

And I'm going to slowly bring it down up

until I start to see

some orange flashes that come all the

way down to close to about -3.

More than that and I'm going to start to

really be introducing a lot of harmonic

distortion.

The other thing I need to be careful of

is that there are some hits that are a

little bit stronger than others. There's

no consistency here.

So maybe my goal is to make all the

transients consistent

and then work from there.

So I'll bring down my threshold

to where all of my transients are

basically getting hit a little bit.

This will make sure that all of these

tops are getting brought down.

Now

I'm going to put in uh take the bypass

out and we'll hear what that sounds

like.

Ah instantly

we can hear that squaring off of that

waveform, that harmonic distortion.

>> [music]

>> Too much.

I'm going to give back a little bit.

>> [music]

>> That's a little better but I can still

hear some crackle. Now I'll start to

slope slow down the release time a

little bit.

Now I should be able to back my

threshold in a little bit more without

hearing as much

of that distortion because my compressor

isn't acting as strong. It's not

creating a sharp a square wave.

That sounds better. Now if I bypass it

and with the compressor in

it's quieter. The transients aren't as

loud and if we listen very carefully, we

might actually even be able to start to

hear some of those harmonics creeping

into the tone of the signal.

I'm going to make this a little bit

louder though. I'm going to use my

makeup gain now

to make the overall signal louder.

That should help to bring out these

resonance that I'm trying to bring out

in my tone. But as I do, you'll also

start to hear the harmonic distortion

come out from the compressor.

>> [music]

[music]

>> Without the compression

with it.

In my description earlier

uh in my presentation, I discussed that

we should think about compressors as

time-based processors.

What I think we can hear

to hearing a sound that's actually quite

long in its resonance. It takes a lot

longer for the sound to go away.

It's because I'm compressing the signal

and I'm bringing up I'm compressing the

transients and I'm bringing up all the

noise floor stuff that's happening

later.

So what's happening is you could think

of it as as as controlling dynamic

range, but what I'm really doing is

making the sound last longer

which is a time domain process, right?

I'm making the sound last longer,

therefore we're going to hear it a

little bit clearer. It's going to sound

louder to our ear.

The other thing is I've introduced a lot

of harmonic distortion and we can hear

that the the kick drum has gone from

being kind of low and punchy and uh

ticky to being kind of from from from uh

loose and flappy.

These are bad descriptive words but

they're the best we can do when we're

talking about subjective uh qualitative

uh assessment here. Um I can measure the

harmonics but I can describe it as being

different in some sort of way that's not

just the harmonics but the quality of

the sound.

So we have to figure out a way to

balance the uh uh uh

uh balance the the the issues here. We

we have On the one hand I'm I'm

succeeding at compressing and and

suppressing the transient of the signal.

On the other hand

as I'm doing that I'm also bringing up

the harmonics in the back end of the

signal and that's not very good.

So we have to find a way to kind of come

to terms here. So I'm going to uh

uh introduce a compromise.

Perhaps what I want here is not as much

compression.

Maybe I can take 1 and 1/2

to 1

instead of 2 to 1.

That should help. I can also soften my

knee.

By softening the knee, I'm making it so

that I'm not squaring off the

compressor. The com- The compressor is

not squaring off the waveform quite as

aggressively. It's going to curve around

that compression point, which should

help to preserve some of those

transients that had that sine wave type

element to them, so that they're not

suddenly becoming flat mesas.

Um that should also help. So, let's try

introducing some knee and we backed off

the ratio a little bit.

>> [music]

>> Uh that's much better.

>> [music]

>> Without the compression.

With it.

>> [music]

>> Here we're hearing a little bit less of

an aggressive harmonic tonality to the

compression. If compression adds

harmonic distortion every time it acts

on a signal, then using it to fix uneven

dynamics means adding coloration to

every moment you're trying to correct.

The compressor doesn't know it's dealing

with a word that got swallowed or a

guitar phrase that was played softer. It

reacts to level and it reacts the same

way every single time. You're smarter

than a compressor, though. For the

dynamic problems that are really about

performance, a phrase that's too quiet

or a syllable that got lost, you might

be better off going in manually. That

could mean adjusting the clip or region

gain, which happens before the signal

hits the compressor. That way, you

smooth out the performance before the

compressor even sees the signal. Then

the compressor can focus on what it's

actually there for, adding texture and

character, if you want texture and

character.

Here's how Gabe explains it. You should

try to get 90% of the way there with

your tracking technique, your clip gain

automation or with volume automation,

and then use the last 10% of that

control. Um maybe that's done with a

compressor.

Okay? And And this is the number one

trick that once I show my students how

this works, like they're suddenly

compression just becomes a lot more

intuitive.

It's funny. There's things about dynamic

range that have to do with performance

and not um harmonic distortion. So, if

you want to control the dynamics of a

signal,

you shouldn't reach for compressor to do

that.

Reach for um if you're in the tracking

stage,

repositioning the microphone. Maybe it's

pulling the microphone further away, so

that you don't have quite as much of a

dynamic range between loud bits and

smaller bits, because everything's less

focused. That can still sound really

great. In fact, pulling the mic back um

might help you

um be more successful at using

compression while tracking, because the

compressor isn't trying to do the heavy

lifting of pulling the levels down when

it goes significantly above the

threshold, but most of that's just been

taken care of naturally by the uh the

inverse square law and the fact that

you're pulling the microphone away,

you're pulling the mic away, so you get

less proximity effect in the vocal. Um

the dynamic range of the vocal will get

naturally more even. And then when the

compressor is activated, it's working on

a much smaller percentage of the signal

than if it's right up in the mouth of

the performer, where it's having to do

an awful lot. You're just going to get a

lot of harmonic distortion from that,

not necessarily dynamic range control

that you might want in the presentation

of a performance.

So, um

my suggestion is to try to get

the dynamic range part of your signal

under control using automation and be

intelligent about which syllables you

want to emphasize. Put a little flourish

on the last line of every vocal. Um

uh think about words like uh and the and

from and the small syllables that

sometimes get lost. Try to boost those

up with gain control, um gain

automation, so that they're more

present, because a compressor isn't

going to be listening to the lyrics and

deciding, I didn't understand what they

said right there. I should have

compressed that better.

Compressors are not smart. They do not

take intelligent uh

intelligence into into consideration

when they make their changes. They act

aggressively and consistently, and they

will not bring humanity into your

tracks.

The goal isn't to avoid compressors.

It's to use them for what they do well

and stop asking them to solve problems

they can't fully understand.

You're probably already familiar with

the standard use of reverb in a mix. You

put something in a space to make it feel

more real or you add a shared room to a

mix, so everything feels like it belongs

together. Both of those are useful, but

when I watched Gabe explain another

situation where he uses reverb, it took

me by surprise.

He thinks of reverb in some cases as an

EQ tool.

Not just a metaphor for EQ, but as an

actual way to control the tonal balance

of the full sound.

When a room resonates, it reinforces

certain frequencies and lets others

decay faster. So, by choosing the type

of reverb, shaping its frequency content

and controlling its decay time, you can

augment the tonal character of the

instrument in a way that feels more

tonal than spatial. Here's Gabe working

on an acoustic guitar that's been EQ'd

and compressed and is sitting reasonably

well in the mix, but still feeling a bit

pointed and edgy.

He goes to reverb in this case.

People would think of reverb as being

spatial, but I'm actually going to use

this to create timbre. Make the the body

of the guitar feel a little bit more

mellow. As I'm listening to this, it

just feels like the guitar is very kind

of pointy and edgy. So, what I've done

here is I've created a an aux return and

a send

that's called a GT verb and I'm going to

send this acoustic guitar to this

return.

And on this return, I'm going to insert

a reverb and I don't want anything too

special. It doesn't have to be something

fantastical. In this case, I'm using

Dverb.

I'm going to turn the gain reduction all

the way off, so that it's not pulling

anything out of there and I'm going to

set the room parameter to small.

And I really don't want much here, maybe

about 178 milliseconds.

That's all I really need to do. And then

I want to pull the high frequencies down

to maybe oh, I don't know, 1.6k.

So, I'm going to put the fader up here.

Let's hear what this sounds like. I'm

just going to solo the acoustic guitar

and we'll see what this reverb sounds

like.

>> [music]

>> If I mute the reverb,

that's pretty cool.

It gives it a room. It gives it a space.

It definitely makes it feel more stereo.

And I'm starting to get into the

direction of it sounding warmer, but I

haven't really gotten there. So, I'm

going to go into like my EQ here after

this reverb and I'm just going to make a

bump at around 200 hertz. What I'm also

going to do, so that it doesn't get in

the way of the bass, is I'm going to

pre-delay this signal maybe about 10

milliseconds or so. And I'll stop at 12.

That seems That seems feasible. And I'm

probably going to want to de-emphasize

the high frequencies as much as I can.

So, I might just throw in a a nice

gentle shelf here

to further that along. So, I'm really

tilting the the tonality of this and

this this hopefully it makes the room

reflect the low harmonics that are

bouncing around inside the reverb

program. That's going to make the guitar

feel thicker without adding low

frequencies via EQ into the sound

directly.

>> [music]

>> And I feel a little space between where

the reverb starts and where the guitar

ends.

So, you can hear how the guitar is

starting [music] to take on that darker

tone.

Now, I'm going to do one more thing. I'm

going to compress the reverb. What we're

going to try to

achieve is take all of the fast-acting

stuff that's in there and kind of pull

it out and let it release really slow.

We're going to set the ratio of

compression to maybe 5 6 to 1 and a

really soft knee

like this. And I'm just going to pull

back the threshold, so we're burying the

reverb kind of in constant compression.

Then I'm going to use the output gain to

bring it back up in my mix.

And you'll feel [music] that instantly

my guitar sounds and feels warmer, like

a warmer bodied instrument.

If I took this out, [music]

not only do we lose the space, but you

can sense that [music] the low harmonics

just kind of disappear out of it.

The guitar gets warmer and thicker

without any change to the dry signal

itself. He's not EQ'ing the guitar. He's

EQ'ing the room and what it gives back.

The same principle applies across

different instruments. Here's Gabe again

on a snare drum, deciding he wants the

snare to sound thicker and fuller.

>> [music]

>> Okay, now just a short reverb.

You can almost hear how it's giving it

of a flamy feel to it.

And again, this is just the dry snare by

itself.

>> [music]

>> Now with our processing.

>> [music]

[music]

>> So hopefully you can hear that there's a

change not just in the presentation of

the stereo image or the space around the

snare, but also a temporal shift in the

snare.

I could have made the snare feel thicker

with EQ,

but the room can do a lot of that work

for me as well. And rather than try to

make that individual track uh thicker or

uh try to push more low frequencies into

say a compressor which might react

negatively to it, instead I'll just do

some parallel processing in the reverb

and try to get the reverb to add

harmonics to my dry signal and hopefully

make the snare uh sound more full and

lush.

Different instrument, same idea.

Once you start thinking about the reverb

return as something you can shape

tonally,

the way you use reverb will change.

The central idea Gabe comes back to

throughout the delay section of the

course is that time and space are the

same thing. Delay is how we perceive

distance. When a sound arrives at your

left ear slightly before it arrives at

your right ear, your brain doesn't hear

two different sounds, it hears one sound

coming from the left side. And that tiny

timing difference is how you locate

things in the world around you. And

you've been doing it your whole life

whether you thought about it that way or

not. Think about how you judge distance

with your eyes.

Close one eye and you lose depth

perception.

The brain needs two slightly different

views of the same thing to build a 3D

picture. And hearing works in a similar

way.

Delay is one of the most important tools

we have for creating dimension and

depth.

What makes this practical is

understanding that different amounts of

delay do completely different things.

Gabe organizes them into three time

windows. The first is the binaural

window from roughly 1 to 10

milliseconds. In this range, your brain

doesn't separate two instances of the

same sound into two distinct events. It

sums them together and reads the timing

as a localization cue. A few

milliseconds of delay on the right side,

for example, makes the whole thing feel

like it's coming from the left with no

panning required. Some engineers use

this concept called the Haas effect to

push things wider than the physical

speaker boundaries. The catch is mono

compatibility though. When you sum a

binaural delay to mono, the two versions

of the signal interfere with each other.

And what might have sounded huge in

stereo sounds phasey in mono. So it's

always worth checking your mix in mono

before committing. The second window is

early reflections from about 15 to 35

milliseconds. This is where delay stops

being a positioning tool and starts

being an acoustic tool. At 15

milliseconds, the brain has just enough

processing time to register the delayed

signal as something separate from the

original. And done intentionally, it can

register to the listener as a reflection

of the nearby wall. A helpful guideline

is that 1 foot of distance equals

roughly 1 millisecond of delay. So 15

milliseconds implies a 15-foot path.

That means delay can be used to define

the size of the space you're trying to

create within your mix. You push it

higher in this range and the implied

space gets larger. This is how you make

a dry recording feel like it was

captured somewhere. And the concept can

also be implemented using the pre-delay

setting that you might find in a reverb

plugin. The third window is late

reflections from 35 to 60 milliseconds.

This range adds a deeper sense of room,

less about where you are and more about

the character of the space itself. The

back wall, the texture of the surfaces,

mono compatibility is less of a problem

here because the signals have enough

time between them that they're not

fighting each other as much. And the

level between the direct and indirect

delays tends to be wider and that also

results in less interference. Beyond 60

seconds, the brain stops reading the

second sound as connected to the first.

It becomes its own event. That's the

territory of rhythmic echoes and

slapback creative effects rather than

acoustic simulation. When you understand

how the brain interprets different delay

times,

you get a sense for what's possible with

delay. And the next time you want to add

dimensionality or special effects to a

mix, you can get close to the settings

you need without digging through

presets. So those are the four. EQ

through the lens of vowel sounds so you

have a reference every time you open an

EQ. Compression as something that shapes

the tone of a signal just as much as the

dynamics with clip gain handling the

outliers in the performance before the

signal reaches the compressor. Reverb as

a tonal tool, not just a spatial tool.

And delay as a set of time zones, each

one producing a different effect on how

the listener experiences the space

around the sound. There's a point Gabe

makes at the end of the course that I

don't think enough of us focus on while

we're mixing. It's one skill to use

compression and EQ to make individual

instruments sound good.

It's a different skill entirely to use

those same tools to make instruments

work together. So everything has its own

space and so that the whole thing feels

like one cohesive sound.

That second skill is what the course is

actually building toward. Information

about mixing or anything for that matter

isn't hard to come by anymore.

And that's exciting.

But the challenge is still to build a

skill that will actually stick with you.

And that takes structured practice,

feedback on your work, and time spent in

the material with other people working

through the same things. The Audio

University membership is built around

that exactly. The courses are taught by

myself, Gabe Herman, and other guest

instructors with decades of experience

mixing and teaching. And beyond the

courses, both Gabe and I are active in

the membership. So you can leave

comments directly on the lessons or ask

questions when something doesn't click.

And you can get a response from one of

us or from both of us and we'll each

share our perspective. We also run live

coaching events periodically where you

can ask questions face-to-face and

there's also a mix review feature where

you can submit a mix and get expert

feedback from Gabe himself. A lot of

people in the membership are working

through the same problems and hearing

responses to other people's questions

tends to move you forward in ways that

watching videos won't do on its own. You

still have to put the practice in

yourself though to actually hear

results. But having that support

structure around you changes how fast

that happens. The membership includes

the ear training lab and downloadable

multi-tracks so you can put the theory

into practice. If you're looking for

more than just videos to watch and

you're willing to put in the work, the

link is in the show notes below.

I'll see you in the next video.