Step 1

Fire up Reason and create a new project. I’m going to open a Thor and a Redrum and throw a reverb unit on the mixer. It’s not necessary to do this to learn the technique, but it’s better to hear the changes as they’re occurring so you get a grip on the tools you’re learning about.

If you want to follow along, right click the empty rack and create a Thor Polysonic Synthesizer. Right click the empty rack again and create a Redrum Drum Computer. Finally, I’m going to right-click the Mixer, head down to the Create submenu, and load an RV7000 Advanced Reverb unit — this has nothing to do with the tutorial but I can’t stand the sound of dry programmed drums.

Step 2

Switch into Edit Mode, or turn on your MIDI controller, and record a few clips that you can line up in the sequencer repeatedly as you audition tempo changes. There doesn’t need to be much; a section as long as one bar is fine.

Try and make it somewhat pleasant to listen to because you’ll be hearing this for a while as you explore tempo and time signature automation.

Step 3

The first thing to do is set your baseline tempo. Baseline tempo is the tempo that will Reason will apply where no modifying automation data has been set. To save yourself a lot of work and clip drawing, this should be the tempo that the majority of the song takes place in.

To do this, locate the tempo setting on the Transport bar, in between the click track settings and the transport locators. You can use the arrows next to the number to change the value, or you can click the tempo number and type a new one in.

For my purposes, 120 BPM is as good a tempo as any, so I’ll leave this as is.

Step 4

In the list of tracks in your sequencer window, the top should always be the Transport track. Locate this and use the arrow on the right of it to expand it. There’s not really anything to be revealed by expanding it yet, but we’re about to change that.

Right click on the Transport track and click on the menu option Parameter Automation. When the dialogue box opens up, check both Time Signature and Tempo, and press OK.

Now, when expanded, the Transport track has to lanes available for editing:

Step 5

Draw a clip with the Pencil tool in the part of the song where you want to modify the tempo. I recommend having to Snap to Grid enabled with the snap setting on Bar, to ensure your clips begin and end on the bar.

You can select the clip and press Enter or Return to edit the tempo with the Pencil tool in Arrange mode, but you’ll find this method fairly inaccurate to work with. I recommend flipping over to Edit mode and double-clicking on the clip there to edit it.

Step 6

Once in Edit mode you can still draw crazy tempo patterns with the pencil tool like so:

Of course these are rarely practical. In most instances, you’ll want to set the tempo for the duration of the clip. Matching clips to sections of the song is also handy later if you chose to move that entire section — all you’ll need to do is move the tempo clip rather than program a new one in the section’s new location.

The quickest way to set the tempo of the clip is to use the Pencil to place a single dot in the clip. It doesn’t matter where you place the dot because Reason changes the tempo at the start of the clip based on the value of the first dot, even if the first dot doesn’t appear until later in the clip.

Click on the dot (you can tell which dot is selected as it is totally black, rather than white with a black outline) in order to change its value. On the toolbar above the edit window, there’s a setting labeled Value. This is the value of the automation point you have selected, and this allows you to set your tempo precisely — of course, tempo needs to be precise, so drawing and dragging is rarely suitable.

Step 7

Now let’s automate the time signature. As with the tempo, the first thing you’ll want to do is set the baseline time signature. The project time signature is located underneath the tempo setting on the Transport bar and works in much the same way:

The default time signature is 4/4, and the vast majority of modern songs are in this time signature. I’m going to leave the project setting as is and move on to automation.

Step 8

Time signature automation also works on clips, but differs slightly in that there’s no drawing done within the clip. Each clip represents one time signature change. Drop a clip in the location you’d like to see a change:

Step 9

You’ll find that you can’t double click on this clip to edit it. Whether you’re in Edit mode or Arrange mode, you’ll see a dropdown menu instead. It has several preset common time signatures in it, and an “Other” option. The signature I want is not in the common list, so I’m going to click on Other and set my own.

Click OK when finished, and it’s set — wherever I place that clip, the song will change to 7/8 time.

Step 1

Create a new project, and set up a Subtractor, a Dr:REX, a Spider CV Merger & Splitter and a Thor synthesizer, in that order. When you create the Spider CV unit make sure to hold down Shift so that it isn’t auto-routed. 95% of the time you’ll want to make that a habit, since auto-routing a Spider unit is rarely useful.

Right click the Subtractor unit and select Initialize Patch. This gives you a completely blank synth to work with. Do the same with the Thor.

Step 2

Fire up your MIDI controller or use the sequencer’s Edit view so you can hit keys and listen to the sound of your modifications. Select the Subtractor in the sequencer list so that anything coming from your MIDI controller is going through that device.

Find the Subtractor’s LFO 1 settings section and set them like this:

• Sync on — this synchronizes the waveform to the song’s tempo
• Rate and Amount each at halfway, pointing at 12 o’clock
• Waveform set to Sawtooth (the third option)

Have a listen to the difference between the initialized patch and the modified LFO 1. You don’t have to use these settings, but know the difference between the blank patch and your changes so that you can hear them later in our combined CV signal.

Step 3

Head down to your Dr:REX and locate the LFO on the front panel. I’ve selected the Subtractor and the Dr:REX just to show you that you can use the CV parameters from just about any device; it doesn’t really matter what you usually do with them.

Here are the settings I’ve dialed in here:

• Sync off
• Rate and Amount at roughly 1 o’clock
• Waveform set to Square (number four) — a waveform with very abrupt changes

Step 4

Now it’s time to wire up the Spider CV unit. Flip Reason around by pressing Tab and locate the back of the Subtractor. Under the heading Modulation Out, the Subtractor will have an LFO 1 port.

Before we wire this, locate the Spider CV. On the left half of the Spider’s back panel are the ports for the merger. The right hand side is the splitter. The first four ports are inbound CV signals to be merged, and the dials above the ports act much like audio level faders — they determine how much of that CV signal will be sent through to effect the sound, and you can mix the different CV signals as much as you like this way.

Drag the Subtractor’s LFO 1 modulation output down to the first input of the Spider’s merger side.

The Dr:REX has a similar back panel with a Modulation Output section — connect the LFO output to the Spider’s second CV merge port.

Step 5

Head down to the Thor and play a few notes through it so you know what the blank patch sounds like. Now go back to your Spider CV unit and connect the Merge output to the Thor’s Filter 1 Freq input, which can be found under the Modulation Input list. Your connection should look like this:

If you listen to the sound of the Thor now, it’s totally different, and you can hear both modulations at work: one regular, smoother and repetitive modulation, and a second erratic, abrupt modulation. One sounds like it’s in time and the other owes its erratic sound due to the lack of timing—that’s because we’ve set Sync on for one and not the other.

Download audio file (1.mp3)

Step 6

Now’s a good time to experiment with mixing the CV signals. Try turning one right down so you can hear the effects of the other alone and vice versa. This will give you an idea of which one you’d rather make the predominant signal.

You’ll find that mixing CV signals is pretty much the same as mixing audio levels; you’re pulling some things back while making others more prominent, while trying to achieve an overall balance.

Step 7

Now let’s take a look at splitting CV. Essentially this allows you to take one inbound CV signal and share it among several devices. We’re going to take the combined CV we already have and split it.

First of all, disconnect the Merge output from the Thor’s modulation input and reconnect it to the first input port in the Spider splitter:

I’m now going to create an additional two Thor devices and load them up with basic patches — one will use the default Epic Poly and the other will use Underwater Asgard Pad. I’ve used these because they’re easy for you to find in the default patch folder (and it makes my job slightly quicker!).

Now you want to do what you did before with the Thor device, but three times over, and using the Split A’s outputs instead of the merger’s single output. Connect the three outputs to the three consecutive Thor Filter 1 Freq modulation inputs.

If you create a clip on the first Thor and copy it to the other two, you can hear the CV working in unison across all of them. Here’s what it sounds like when you “synchronize CV”:

Download audio file (1.mp3)

The Spider CV Merger & Splitter is a simple device — but the possibilities for your compositions are almost endless.

In the last installment of Effective Drum Mixing we went through the rough levels of each instrument, checking the phase of the snare and general panning. Therefore, the faders are mostly where we want them, the snare is punching through without phase problems and every drum is panned in the stereo field.

Listen to what we’ve done so far.

Download audio file (part1.mp3)

Step 1: Kick Drum

I like having my kick drum sound as good as possible before I go adding other elements into the mix. I feel the kick drum is the driving force behind the drum kit and if it sounds weak the whole kit sounds weak beside it. Let’s gate, compress and EQ the kick drum until it sounds as good as possible.

Gate

There’s a lot of bleed from the other drums in the kick drum mic, therefore I’m going to slap a gate on it to get most of that unwanted sound out. We need a gate that opens up really quickly and closes just as fast without killing the sound of the bass drum. After tweaking Logic’s gate I came up with these parameters:

I turned the threshold down until I stopped hearing the other drums. Notice that the attack is really fast, we don’t have a lot of Hold and the Release is fast enough to close before we hear the snare drum, but slow enough so we hear the full oomph of the kick drum.

Compression

Now for some compression. The eternal debate is always on if you should compress before you EQ, or compress after you EQ. I have a tendency to compress before I EQ since I want my sounds to be even when I’m tweaking their “sonic colors”.

Someone told me a good rule about this chain of command. Compress before you EQ if you are going to EQ drastically, and compress after you EQ if you are going to compress drastically. That way, if you do a lot of boosts in the EQ department and compress after it, you are doing more compression on your boosts than the actual signal. And if you EQ after you compress you have a steady signal that you can trust your EQing better.

This drummer has a fairly steady foot, so we’re only leveling things out a little bit. Putting medium ratio of 5:1 and pushing the threshold up until it’s reducing about 3-4 dBs on average. Since we’re cutting about 3 dBs from the signal we’ll make up for it by adding 3.5 dBs of makeup gain.

A note on attack and release: Attack determines how fast the compressor starts chomping down on a signal, and the release determines how fast it stops working on it. Therefore, if you want all the punch of the kick to come through we need the compressor to start working after the initial transient of the drum. Therefore, the attack needs to be a little slower, and if it’s a fast kick drum pattern we need the release to be fast so the next transient won’t get affected by the compressor too.

So you see how it works here. I dialed the attack all the way down where it was working on the complete transient and started dialing it up until the click of the kick came through. We need the compressor to stop working before the next transient, and since it’s such a fast double kick bass drum pattern the release is very fast. Just by tweaking the attack and release time we can get very different results from our compressor.

Let’s compare the setting above, and then change the attack to 0.0 ms (very fast) and the Release to 410ms, or slow.

This is what we’ve got in the tutorial so far:

Download audio file (kickcomp.mp3)

And this is the different Attack/Release settings:

Download audio file (kickcomp2.mp3)

EQ

Now, let’s go for some equalization. We want the rock kick to come out and play so there are a few things that are typical to that sort of sound. Obviously, we’ll be needing a thick kick full of low end. We need to get rid of the boxiness factor that is a little noticeable, and lastly we need to bring out the beater a little bit. But first, I’m going to filter out everything below 30 Hz. Why? Because there’s not a lot of information below 30 Hz that’s useful to us. Most people don’t notice when it’s gone, and any typical laptop or computer speaker doesn’t even come close to reproduces those lows. So a high pass filter aimed at 30 Hz and we’re good to go.

I know for a fact that the bass guitar on this track is going to be a pretty thick and bassy one, so I’m going to leave the deep bass for that instrument. I like to think of the bass guitar and bass drum interacting between 80-100 Hz and if I want the bass to dominate the lower frequencies I’ll boost it down nearer to 80 Hz than 100 Hz. Because I know the character of the bass I’ll see what a 100 Hz boost on the kick drum will sound like.

This is the untreated kick.

Download audio file (kickeqbypass.mp3)

And this is the same with a 5 dB boost on 102 Hz

Download audio file (kickeq100hz.mp3)

There is a nice thick punch coming from us boosting the low end. Since we have a shelving EQ on it we might need to cut some frequencies below when we try to fit a bass guitar in, but that’s for another tut.

Boxiness can be a nuisance in the sound of a kick drum. Sometimes our kick drums sound like a cardboard box and when that happens we need to be aware of where we can fix it. Boxiness can be found in the higher-lower mids (if that makes sense) around the 300-600 Hz area. The best way to find a problematic frequency is by boosting with a very narrow Q and then sweeping around the spectrum until you hear the annoying frequency pop out.

Listen to this audio sample of where I do exactly that. I boost the kick with a narrow Q and then pin-point the frequency I want to cut. You can hear the different sets of frequencies as I sweep and then I settle on the one which I cut.

Download audio file (kickeqsweep.mp3)

My cut might be interpreted as a little wide, but we need that wide cut in the lower mid range for that rock kick drum.

This is the kick with a 16 dB cut at 360 Hz.

Download audio file (kickeq360hz.mp3)

Now for the last advice. We need to give the kick a little punch in the higher frequencies and no better way to do it than bringing out the beater of the drum. Let’s boost a nice 6 dBs at 2.5 kHz in order to bring out a little of the click from the beater.

Now we’ve transformed, with gating, compression and EQ, this normal bass drum:

Download audio file (kickdry.mp3)

Into this kick drum ready to rock!

Download audio file (kickwet.mp3)

Step 2: Overheads

Now that we’ve got the kick drum nice and tight, I’d like to focus my attention on the overheads. Maybe you’d like to keep going, adding the snare, hi-hat and so on, but I feel that by adding the overheads on top of the kick drum gives you a nice palette to paint from. You have the base from the kick drum, and then you have the overheads covering everything else. We’ll be cutting a few things out of the overhead tracks, as well as accenting the cymbals themselves.

We’ll start by filtering out the overheads. Since we have such a good kick drum sound, we don’t really need to hear the kick drum in the overheads. In reality I’m going to use the overheads tracks for very little, only a little ambience and for the cymbals. The snare drum likes to reside in the 500-600 Hz area so we’ll be cutting that out as well. Then we’ll boost the overheads a little in the highest frequencies, lifting up the cymbals and giving them some air frequencies.

Some might say that the filtering is a bit excessive, and I could agree. But since I really only want to bring out the cymbals and a little ambience, I think this is enough. Besides, my teacher in school, we called him Dr. Filter, and he taught us a lot.

The full overheads that you can listen to here:

Download audio file (overheadsdry.mp3)

Are far different than the over-filtered and high-end boosted overheads you can find here:

Download audio file (overheadswet.mp3)

But I assure you that when all the drums are in place, every element has a place carved out and the overhead tracks aren’t trying to overcrowd other drums. We’re not going to compress these overheads. Let’s just leave them as is for now and concentrate on adding the snare into the mix.

Step 3: Snare

The snare is the constant pulse of the drum kit. If the kick is the driving force of the track, the snare is the march commander, constantly marching in time. We need a snare to sound punchy in our rock track. We need to EQ it so that it crackles and pops as well as compress it so that it breathes in time with the track.

We have two snare tracks, the over and under-snare. We already had rough levels for them in the previous tutorial so now let’s see how we go about EQ’ing and compressing them. Let’s start with the over snare.

Compression

After going through all of Logic’s compressor setting I finally reverted back to the Platinum type compression. I tried the Classics, the VCA, FET and Opto to see if anyone of those had anything interesting to offer. But alas, no. I’m sticking with the Platinum setting, which you need to be careful with.

I feel that Logic’s compressors can get a bit aggressive and you don’t much to completely over-compress and suck the life out of a particular instrument. But I think I did a pretty good job of it. I just want to add a little bit of compression, riding the peaks and controlling the level. Therefore I’m only looking for a gain reduction of about 1-2 dB.

You see that the threshold is fairly low, or at -8.5 dB. As said before, we want to control the peaks and just give the snare a little boost and color. Again, the attack and release settings are really important here as a really fast attack can suck the life out the transient, and a really long release never gives the snare room to breathe.

We want the compressor and snare to breathe in time with the track. Set your attack to really fast and then dial down until you hear the snare pop out before the compressor clamps down on it. We want that bite of the snare to pop out. Then time the release so it stops compressing in time with the snare. Watch the reduction meter and look at how the blue bar extends and retracts in time with the song. That’s how you make a snare compressor breathe. By finely tuning the attack and release in time with the song.

This is how the snare drum sounds now:

Download audio file (snarecomp.mp3)

By comparison, listen to how destructive compression can be in the wrong hands. Here I have a sample where the snare is over-compressed and completely lifeless and squashed. Just by upping the threshold, ratio and turning the attack to the fastest we get a squashed sound. Then adding a long release creates a pumping effect that really never gives the snare drum time to breathe. Don’t compress like this please.

Download audio file (snareovercomp.mp3)

EQ

Let’s equalize the snare drum by using the same techniques as the one we used on the kick drum. I’m going to filter out the unneeded low end, cut out some ringing overtones and add a little bite to the high mids.

The best way to know how much you can filter out is to put your filter all the up to about 400 Hz or so where you really notice the sound getting filtered out. Then slowly work your way back down until you don’t notice any changes in the sound. From 150 Hz and down you don’t really notice anything being added to the snare drum sound, so find a good place where you think you are not losing any sonic information from the snare. I filtered my snare from 130 Hz and down.

I feel that snares have a tendency to have annoying overtones that constantly ring when the snare is hit. With surgical EQ, you can get rid of these tones easily. Just do what we did before, a narrow high Q boost all the up and sweep around until you find the ringing spot. Then cut down on that spot, removing these frequencies. I did that two times on this snare, removing a 205 Hz boomy tone, and a 540 Hz ringing overtone. I boosted the 160 Hz area to give the snare a bit more meat, and gave it a wide boost in the 3 kHz area for a little point and bite.

Listen to the drastic difference EQ’ing can make. With just a little EQ you can transform a ringing, boomy snare drum into a nice punchy snare.

Download audio file (snareeq.mp3)

What we’ve got so far is this:

Download audio file (drumsfinal.mp3)

Until the Conclusion

Looks like we’re out of space so we’ll postpone the conclusion until next time. Stay posted for the third and final installment in the Effective Drum Mixing series. I’ll be continuing down the drum-kit, adding the other snare to the mix. Also, I’ll be using expansion on the toms, as well as compressing and EQing them. Finally we’ll mix the full kit by itself, using reverb and compression to make it really come to life.

Although the Spider Audio Merger & Splitter is listed among the effects, it’s not one. It’s a utility device that allows you to do two things: combine multiple audio signals into one signal, and split audio signals into multiple signals.

There are many practical reasons to use a device like this. You might want to save mixer channels by grouping a few signals onto one fader. You might want to split a mono sound into a stereo one and have different effects on the left and right signals. The list goes on.

It helps to be familiar with routing in Reason 4 before you use these devices — absolutely everything you do with them occurs around the back of the rack.

Setting Up Your Project

To begin with, open a new project in Reason which should have a mixer, mastering suite and hardware interface by default. Add two sound modules (I’ve added a Thor and a Malstrom) and a Spider Audio Merger & Splitter. Your project will look just like this:

We’re going to use the same Spider device to practice merging and splitting. Throw two effect devices on the rack — hold Shift while you’re creating them so they don’t get auto-routed. I’ve gone for a reverb unit and a delay unit.

Merging Signals

Press Tab to flip around the back of Reason. At the moment the effect devices can be safely ignored, and you should see a rack backend that looks just like this:

The only routing here is the main outputs of both sound devices being sent to the inputs on two channels of the mixer. To merge sounds, you basically need the outputs of sound devices to be sent to corresponding stereo inputs on the Spider Audio device, so drag the left output from each device to the left inputs on the Spider Audio (the right channel will be auto-routed):

Every time a signal is generated by one of these devices, it’ll be sent to the Spider Audio and turn into a single signal. If you want to find out what this sounds like, put some notation on each track, and drag the outputs of the merger to mixer channel one. The outputs are the two ports connected to the merger ports by arrows:

If you need to balance levels between the signals in one Spider merge, you’ll need to do it by tweaking the output of each device itself, instead of using each mixer channel as you normally would.

Splitting Signals

The splitter is the second slab of ports on the back of the unit, works in reverse: instead of taking up to four stereo signals and merging them into one, it takes one stereo signal and splits it up into four stereo signals.

You can route the output of any device (even a mono device by using the Left port) into the splitter and then route the outgoing signals using the ports right of the arrows, but in this case we’re going to do something more interesting: we’re going to take the merged signal we just created and split it. Disconnect the merger outputs from the mixer you used to test the sound before, and send it to the input of the splitter:

We can now take the first set of outputs and connect it to the Reverb unit, and the second set of outputs can be hooked up to the delay. If you’re using the same effects as I am, note that the delay won’t auto-route your stereo signals so remember to manually connect the right signal as well.

What we’ve got now is a very messy looking backend:

There are cords going everywhere, and hopefully you can make out enough detail there to troubleshoot any problems you might be having at this stage!

You won’t be hearing a thing at this stage. We still need to route the outputs from the effect units. Hook the first device up to mixer channel one and the second device to mixer channel two (hopefully you can make everything out in this image — it would be great if Propellerhead found a way to retain the realistic design style while infusing some clarity in future versions of the software):

Press Tab to flip the rack back around. One reason to split an audio signal is to provide different effects for left and right signals, so pan one hard left and one hard right:

You can use aux effects for this process as well — simply set up send/return effects on your mixer (this tutorial shows you how) and send the split signals straight to mixer channels.

Conclusion

Congratulations! You’ve just merged your first audio signal, and subsequently split it up again, effected different parts of it, and then re-united them as one sound with contrasting effects on each side of the stereo spectrum. Not too shabby for twenty minutes worth of work. I wouldn’t use the setup I’ve used to teach you how to work with the Spider Audio in a production setting, but you should definitely know your way around one of Reason’s most neglected and deceptively simple devices and be able to come up with hundreds, if not thousands, of interesting and insane effects.

Or, you could just make the mixing process a little bit simpler. It’s up to you.

In true Propellerhead fashion, as they’ve shown with both Reason and the newer Record software, their design ethos is to imitate not just the look and feel but the functionality of physical audio equipment. If you’re used to working with other DAWs, Reason’s mixer might be more unfamiliar to you than if you’d gotten used to a physical mixer. In fact, people who’ve read my Plus tutorial on understanding mixers may not need to read much of this tutorial at all!

The Reason Channel Strip

The Mixer is made up of 14 stereo channels, each with identical parameters. The beauty of digital mixers is that it doesn’t matter how many channels each device supports, because you can add more! We’ll discuss how you can do this later. Here’s what a channel strip, without any settings altered from the defaults, will look like:

There are five main elements of the channel strip.

The Master Fader: As far as most people are concerned, the most important part of the channel strip. The master fader allows you to control the level of each track, and is the main tool used in creating a mix and balancing all the sounds together. A flat value in Reason is marked at 100 (with a minimum of 0 and a maximum of 127, as with all MIDI-based values) where other DAWs would be set to 0dB.

The Pan Control: This is the knob above the master fader, marked with an L and R (for left and right). This allows you to choose a location for the sound in the stereo field. This helps make the music sound more natural, as if the instruments are all part of a real acoustic environment.

The Mute & Solo Controls: Most DAWs have mute and solo buttons on both the track headers in Arrange mode (or Sequencer mode) and the same buttons in their Mixer. With Reason, the Mixer mute and solo devices affect the output of the attached device while the sequencer mutes or solos MIDI data. The result is the same, but the method is slightly different.

Basic EQ: Generally you’ll want to achieve the equalized sound you want using an equalizer on the device, or alternately in a ReWired DAW, but maybe you just want to shave off a little low end without compromising an EQ you’ve worked hard on. That’s why the Reason mixer has two pots to trim EQ — one for treble and one for bass.

Aux Levels: You can wire send/return effects through Reason’s Mixer, and the aux levels allow you to determine how much — if any — of the track’s signal you want to send to the effect device. There is a pre-fader toggle for the fourth aux.

Understanding the Back Panel

The back panel of the Mixer allows you to change the default wiring Reason implements when you create new devices. There are many reasons you’d want to do this, from creative reasons to signal flow reasons. Here’s what it looks like:

There are three sections to the back panel. The first is the routing for each of the channel strips. To the right of that, we’ve got the Aux Send Out and Aux Return assembly. And on the far right, there’s master left and right outputs.

The channel strips feature left and right stereo inputs, and control voltage inputs and sensitivity trim pots for Level and Pan. When connecting a mono sound source to the Mixer, attach the device’s output to the Mixer’s left input. What you use to modulate the Level and Pan parameters, if you do at all, is up to you and your creativity.

The aux connections section allows you to set up send/return effects, and this is something I’ll show you how to do manually in a moment (though you won’t often need to do this manually — Reason’s auto-routing is fine in most cases).

There are also chaining Send In and Masters for adding further mixers, with the most popular configuration making one mixer the master mixer and the rest simply providing extra channels.

The Master Outs simply send your left and right master signals to hardware — usually with a detour through the MClass Mastering Suite Combinator.

Setting Up & Using Send/Return Effects

To show you how to create a send/return effect in Reason, we’ll start with a disconnected reverb unit:

First, you need to tell Reason where to send our instrument signals through the Aux Send Out ports. Click and drag on the Left (Mono) port under the column 1 on your Mixer and connect it to the left Audio Input of the effect device. Reason automatically wires the right cable for you.

Now, drag a cable from the left Audio Output of the effect device, and connect it to the left input in column 1 under the Return section.

Press Tab to flip the rack around, and set the effect levels using the labeled pot under the RETURN label on the right side of the Mixer.

Each channel strip will have an Aux knob with a corresponding number to control the track’s outgoing level. Tweak this until you have the desired amount of the effect in the mix.

Setting Up & Using More Mixers

I’ve got a second mixer in my setup now and I want to control its master fader with my first mixer’s master fader, and I want to make the same effects available through the Aux channels. Here’s what I’m starting with:

To connect the master faders so that the first mixer controls the second mixer’s fader, simply connect the Master Outputs of Mixer 2 to the Chaining Master of Mixer 1:

To make Mixer 1’s send/return effects available to the channels on Mixer 2, connect the Aux Returns of Mixer 2 to the Chaining Aux Send Ins of Mixer 1:

Propellerhead have really made this an easy process. Even better is the fact that by creating a Mixer, it’ll automatically be chained to the one prior to it, but in order to tweak this setup you need to know how to create it yourself and why certain cables connect to certain ports.

The downside is that there is no way to cause the mute and solo buttons on one Mixer to correctly effect the playback of the others.

Creating Submixes

If you want to create submixes, one method is to use a Mixer for each submix, get the mixes right, fold them up so they’re out of sight, and then tweak the main levels on the master mixer as necessary (you can easily go back and change things in the submixes as you go). With submixes, we can chain the aux effects without chaining the output.

If you want to make the same effects from Mixer 1 available to your submix, chain the mixer effects as shown in the previous section by connecting the submix’s Aux Returns to the main mixer’s Chaining Aux Send Ins. If not, just create send/return effects on that mixer as you normally would.

Then you simply need to take the master output of your submix mixer and route it to one of the main mixer’s channel inputs:

EQ Modes Or, Why Do My Older Songs Sound Wrong?

If you’ve been using Reason for a considerably long time, you might’ve wondered why you’re older songs can sound a bit strange now that you’ve upgraded. This is because at some point in Reason development the EQ algorithms were changed to improve their quality in general. To play your older songs normally, simply press Tab to flip to the back panel and change the switch in the lower left corner from Improved EQ to Compatible EQ.

Get five mixing engineers in a room and they will probably give
you five different tips on mixing your drums. Some want it au natural,
making it breathe and sound live. Others want it pumped in your
face with crushed compression kicking you to the ground. And many
like the happy middle, applying compression when needed but still
retaining the natural feel of the drum kit. No matter what your
preference is, there are certain guidelines that are good to follow
when starting to mix your drums.

In the following tutorial I’m going to walk you through the
basics of mixing together a drum kit. We’ll be taking a few bars of
multi-tracked drums, mixing and sculpting with various things like EQ,
compression, levels and panning before we get it sounding just right.
Take note that these tracks are hard rock drums, and I will be mixing
them in that genre. That means that I will probably apply more
compression than you purists think, and I will EQ to get a proper
rock sound instead of a natural sounding jazz sound for instance.

Ok let’s dive in. This is what we’re working with.

Download audio file (drumsdry.mp3)

A few bars of a decently recorded drum track, maybe a little roomy
but nothing we can’t make sound great.

11 drum tracks. Some would argue that’s too much, some would argue
that’s way too little. I think Metallica’s black album was made with
around 30 microphones just for the drums. But I wouldn’t know where
to start a tut if we had 30 microphones to work from. So we just have
the bare essentials.

• One mic for the kick drum.
  
• Two for the snare. One on top, one underneath.
  
• One mic for the hihat.
  
• One for each tom, 4 in total.
  
• Three overhead microphones, two over the kit, one in the
  room.
  

Step 1 – Levels

I’m having every track at unity gain (0.0 db) in the screenshot
above, but there are a few methods for getting the correct levels. I
like getting the overheads into the mix first before I tackle
anything else. Getting them at a good level and then adding other
elements into the mix is an easier way to gauge the effect each track
has on the overall mix. Let’s pull everything down except the
overheads.

By listening to the overheads we get an overall feel for the
drum kit. We can hear every drum, although it lacks all of it’s punch.
I also pulled down the room mic since we’ll have a different role for
it later on.

Download audio file (overheadsonly.mp3)

Now let’s add other elements into the mix. Starting with the kick
and snare we move the faders up until they are loud enough to fill in
the punch we need from that specific drum without being way louder
than the overheads.

Now we have a pretty good level on the kick, snare and hihat. The
kick gives a nice punch to the track, while the combined faders of
the snare give a nice cracked accent to the dull snare that was
present in the overheads. But there might be something wrong with the
snare? If you are mixing tracks that you didn’t record then you
always have to keep phase relationships between instruments in mind,
and seeing as the snare has two microphones, you wonder if they are
in phase. Let’s check.

By zooming in on the two snare tracks you can see black on white
how incredible out of phase these two tracks are to each other. We
want the waveforms to go in the same direction and complement each
other, not in the opposite direction and cancel each other out. There
are various ways to flip the phase of a signal but I’m going to
Logic’s sample editor and flipping the phase on the bottom snare
microphone. Double click the region and the sample editor pops up.

Go to FUNCTIONS>INVERT and Logic flips the phase of the whole
waveform for you. We want the two waveforms to go up and down in sync
with each other and now it looks much better. And sounds better too!

Now let’s solo these two snare tracks and listen to a before and
after sample. Before phase reversal:

Download audio file (snarephase.mp3)

And now hear the snare in phase. A much thicker, fuller sounding
snare:

Download audio file (snareinverse.mp3)

To finish off getting rough levels let’s focus on getting the toms
right. I’m soloing a area where the drummer hits the toms and getting
those specific hits loud enough in the mix without making them jump
out.

Let’s hear the tom break here with a roughly leveled drum track.

Download audio file (roughlevels.mp3)

Step 2 – Panning

Now that we’ve gotten our rough levels, it’s time for the
next step. Since drums are a stereo instrument and tend to fill up
the entire stereo spectrum, we need to put them in stereo. We
can’t have all the drums panned in the center. We want to spread the
drums out a little bit and feel like we are watching a drummer play
in front of us.

A note on perspective: Some
people like to pan from the audience perspective, and others like to
pan from the drummers perspective. Meaning that when listening to the
drums you either feel like you are behind the drum kit playing it, or
sitting in front of it watching it.

Since I’m an awful drummer I’m
panning from the sound engineers perspective. That means that the
hihat goes on the right, the overhead above the ride cymbal goes to
the left and the toms roll from right to left.

The general rule is that kick
and snare stays in the middle. Kick and snare is what drives the beat
forward and thus it needs to be anchored in the middle of the stereo
spectrum. Some people pan the snare a little to the right, or maybe
just one of the snare mics to the right, but we’ll be keeping it in
the center.

Take the overheads and pan them hard left and hard right,
filling out the edges of the spectrum. This is what will give our
drum kit width in the mix as by panning those two elements across the
entire spectrum we automatically position each drum relative to
itself. But since we have more elements to play with let’s pan the
rest of the drums.

Pan the hihat to the right. Pan
the toms in a way that when the drummer makes a tom fill you can hear
it sound from right to left. In the audio sample below you can hear a
small tom fill that goes from right to left, just like you would
experience if you were watching a live drummer play.

Download audio file (panned.mp3)

That sounds a little more realistic
now as we get the feeling of more depth to the drum tracks. Go back
and compare this tiny level and pan change to the original
unprocessed track and see how radically different it has become by
only doing a tiny bit of processing.

Step 3 – Ambience

You might have noticed I was
leaving the room mic out of the equation completely in the last few
steps. The fact is that it’s a room mic recorded in a not-so-great
sounding room, but it can illustrate a point that many engineers use
when recording drums.

Metallica’s 30 drum microphone Black album
wasn’t recorded with all of the mics close up on the kit. Many of those
microphones were positioned around the room, capturing sweet spots in
the acoustically treated drum room. That’s the reason Enter Sandman
sounds so huge, cause it has all these microphones capturing the huge
sound the drum kit makes in a big room.

But we only have one room
microphone. So let’s add him in gradually so you can hear the
difference a room mic makes to the sound of a drum kit. Keep this in
mind when recording, as a few room mics might come in handy for
various mixing tricks down the line.

As you can hear in the audio
file below, there is a radical increase in sound whether we are using
the room mic for ambience or not.

Download audio file (oh3.mp3)

Some recording and mixing
engineers choose to use their room microphones for reverb, choosing
not to use digital reverb engines because they prefer the sound of
their recording room. Whatever you choose to do, it’s always good
practice to have one or two mics as ambient room microphones. We’ll
come back to this microphone next time, as I have a separate function
for this microphone that has nothing to do with reverb.

Until Next Time

Stay tuned for the next
installment in the the guide to effective drum mixing. We’ll be
continuing to look at what comes next, namely equalization and
compression. We’ll be learning to equalize correctly, enhancing the
fundamental sound of the specific drums as well as cutting out
unwanted or unattractive frequencies. We’ll be looking at some
examples of compression, the New York Compression trick as well as a
few other mixing tips you can use to spice up your drum sound.

The name send/return came from the practice of sending a sound out from an amp, into an effect processor, and then back into the amp again before finally playing back through the speaker. Although the effected signal never really “returns” to the audio source (the audio track) in digital methods, the name has stuck.

Step 1

Fire up Logic. I’ll be using Logic Pro 9, but this should work just fine in most other versions from the not-to-distant past. Create a new empty project, or you can open an existing one if you want to experiment with a work-in-progress.

Create a new audio track by navigating to the Track > New… menu option.

Once the track is ready, record or import some audio onto it.

Alternatively, you can pick an existing track with audio already in it, if you’re using an existing project.

Step 2

Open the Mixer using the tabs at the bottom of the Arrange screen, and just above the Transport bar. We’re going to create what most audio workstations call an auxiliary track. The auxiliary track will receive audio from other tracks and send just the effected portion to the output of the track (usually the master or a submix).

Go to the Options menu — there’s one at the top of your screen and one in the Mixer, so make sure you use the Mixer’s Options menu. Select Create New Auxiliary Channel Strips. Generally speaking you should make it a stereo track (often things like delay plug-ins will take mono sounds and make them stereo).

You should double-click the track label in the Mixer and rename it, so you don’t forget what it’s for. Complex projects with multiple send/return effects can get confusing!

Step 3

The next step is to tell Logic which “pipe” to feed into the aux track. Busses are a lot like pipes that siphon a portion of audio from one track through to another. We can connect our outgoing sends and auxiliary inputs to different pipes, or busses.

You set the input bus by going to the I/O section of the auxiliary’s channel strip, clicking and holding on the lozenge that will generally by default say “Input 1-2”, and selecting a bus from the Bus sub-menu. In this example, I’ll choose bus 1.

It’s a good idea to jot down a basic list of the busses you’ll need to use regularly to route effects and keep it on your desk or on the screen while you’re working on a project. With multiple audio tracks to add effects to, it can be a pain to scroll back and forth trying to remember which one to use all the time.

Now that we’ve selected an input bus, we know that every time we send audio down that path, it’ll be received and effected by our auxiliary track.

Step 4

Our aux track currently does absolutely nothing except receive audio! Let’s change that by adding an effect. You probably know how to do this, but for the few who don’t, create an insert on the auxiliary track — you can do this by clicking and holding on one of the empty Insert lozenges in the channel strip, and selecting the desired effect. This will also bring up the Insert settings, so you can configure it right away. You’ll no doubt come back to fine tune this once you’re listening to audio come through, but dial in the rough settings you’re after now.

Your aux strip should look like this by now:

Step 5

Now it’s time to head back to your audio track. You should have some audio on it so you can check both the levels of effected audio and whether you’ve configured the effect to your liking.

Unlike the aux track, which has one fixed input, the audio track can send audio off to multiple busses. It will still go from the audio track to the master output — we’re just sending a certain level of the audio signal to be effected. If you’re looking to fully saturate some audio, then you need to place the insert directly on the audio track.

Below the Inserts section of the audio track’s channel strip is the Sends section. Click on one of the empty lozenges here and hold it until the menu comes up. Navigate to the Bus submenu and select the same bus number that you set the auxiliary track to. If you followed my advice and labeled the aux track, Logic will kindly append that label to the bus number your aux track is using.

To set the level of audio sent to the bus, click and drag on the circle next to the send. You don’t need to send a whole lot — around -15dB is a good starting point. At 0dB you’re sending just as much audio to the aux as you’re sending to the master outputs. Subtlety is generally best here.

Always ensure that you set the levels using the sends. Leave the aux track’s fader at 0dB at all times. It might seem more difficult to micromanage a whole bunch of send levels but by trying to control the volume of effects in your project with the aux fader you’ll only introduce mixing balance problems.

You’re now done creating your first send/return effect! The beauty of this system is that you can send multiple audio sources to the same effect processor, and you can repeat step 5 on as many tracks as you like. Your effects should be more cohesive, and save a heck of a lot of computer processing power since you don’t have to run inserts on every channel strip!

Jim Pavett is a Grammy-nominated engineer working out of his
studio in Arizona. His studio, Allusion Studios, has been around for a
long time and is one of the most prominent recording studios in
southern Arizona. He’s worked with diverse artists such as Marty
Friedman of Megadeath, Alicia Keys, and Beyond Words. His method and
attitude towards digital recordings is what makes his recording
crystal clear and pristine. Not only stuck to the studio as he has
done multiple on location recordings, the more prominent location
being the Great Wall of China. I had the opportunity of interviewing
him on a variety of subjects concerning audio recording, mixing and
the modern state of the studio business.

Photo by Joe Ancona Productions

Tell us a little bit about how it all got started for you in
the audio industry.

I was a young kid in my teens, and
always had a passion for really high-end gear. So at the age of 13-14
I bought the second generation 4-track cassette deck ever made – the
Tascam 244 I believe. A few months later I got an SPX-90 and that
kinda kicked it all off. From there I literally started the name of
Allusion Studios, which is the same name I hold today. And here I am.

How do you record in your facility?

It’s multi-room with plenty of ISO
booths. A little bit different than anybody else. Instead of having a
huge room and using Gobos to isolate everything, I have the guitar
player, bass player and drummer all in the same room, but the walls
are wired to where their amplifiers are in separate rooms. So you
have full isolation, but you get the full visual and feeling of the
band playing all at once but getting the sound quality of overdubs.

Since the beginning, you’ve always been in digital. So would you classify yourself as a modern recording engineer, as
opposed to somebody who’s striving for that vintage sound?

Yes and no. What has allowed me to
sift through all the fads and all the misinformation is my education
in electrical engineering – years of doing analog and digital research
through my degree. I’d read interviews in magazines, whether it’s Mix
or EQ or whatever. In these magazines you’d notice that someone was
being endorsed by some company and they’re just doing whatever the
marketing ploy was to try to sell gear, and none of it was really
true.

So when digital first came out, let’s say the Tascam DA-88 – a 16 bit
machine – sounded phenomenal. It was a great machine but people were
like, “Oh well, digital sounds harsh and it’s this and that.”

Well….no, it’s actually 99.9% exactly what you’re putting into it,
and if you’re compensating your high-end EQ all the time, because
you’re used to analog not giving it to you back and you’re doing the
same thing by sticking a digital recorder in there, then yeah, it’s
going to sound harsh. You know, if you’re talking 12 bit or 8 bit,
yeah, that was horrible sounding. (laughs)

There was that whole
analog-digital war and as far as I’m concerned I have so much low
end, and digital for me has always been amazing. Back in the
day everything rolled off at 40Hz so there’s actually a lot more
low-end management that you have to do when you’re recording now.

Does digital sound different? Yes. But
with a little bit of compression and if you want some saturation,
great! You can create that stuff and still have an amazing sounding
recording in digital.

I never thought that digital was a subpar
technology. The reality of that over the years, which is why I
created my educational series, is that it’s not the technology that
sounds worse, it’s that the parts that are coming out are no longer
being held within a professional environment. It’s by people that
don’t have degrees as audio engineers or any kind of engineering
background at all, they just throw up some speakers in a room and
call themselves a studio.

But that’s not how you make great records. It’s the knowledge. So I decided to create a full educational series
to kind of combat that and help educate the public on how to be
better.

Tell us a little bit about this educational series.

Well, for years I’ve always wanted to do some kind of educational product.
And it’s something that I would not be able to complete on my own
since I’m kind of a practical perfectionist. I don’t go too crazy, but
I like stuff to be done right, put my best foot forward.

But I like
to teach from a different way. People might think that recording is
learning a particular software program. The manual of the software
program teaches you how to use that program – which doesn’t teach
you anything about the art and science of recording. So, with my
understanding of the technology and the laws of physics, I decided to
take that information and really start teaching people the physics of
sound, because, it’s not really about the equipment.

You could give
me a (Shure SM) 57 and a cassette four-track and I’m going to do
wonders with it because I not only have all the knowledge, but I have
over 20+ years of experience. You could give a brand new student my
half a million dollar facility here and it’s going to sound like
crap (laughs) because they don’t know what they’re doing (laughs). So
it’s not really about the equipment, it’s about the knowledge, your
expertise and what you do with it.

So the first two DVDs in the series presents the physics of sound, and
it’s kind of a general knowledge about everything in the studio –
“What you should know.” It’s not too math-oriented, but we do
show all the diagrams and frequency charts and such.

And as we go
through this, the 3rd DVD
is going to be all about acoustics. People need to understand that
everything that produces sound is coupled to it’s environment.
Whether that’s an instrument, or whether that’s your speakers. I
could buy a 60 thousand dollar pair of speakers, if I put them in a
bathroom or gymnasium it’s going to sound horrible. So people need to
understand that buying these expensive speakers does not mean you
will have the best mix in the world. Especially when they’re mixing
in a harsh environment.

So we’re going to go through multiple phases
of different scenarios of The Songwriter, The Low-Budget Studio and
The Professional Studio, figuring out how all these different
scenarios might be acoustically treated.

And then we’ll have DVDs that will be, let’s say, one full, maybe dual DVD
on just drum miking techniques. Going through all the different
scenarios and actually give you live examples of how microphones
work, listening to the difference between this position, that
position etc. What happens when you have three of the same exact
microphones one inch away from each other, what do they sound like?
What do they sound like one foot away from each other? So even in the
first two DVDs, we’ve actually done some of that stuff already, just
in the microphone techniques section. Testing and giving you sounds
that you could realistically hear. And we did it here in my actual
studio so we were able to control what we were doing and making sure
it was the real deal.

Photo by Joe Ancona Productions

For drum miking samples using
different microphones, check out Jim’s microphone sample page at Pure Wave Audio.

What’s your mixing system like?

The current system is a mixerless Nuendo/Wavelab system that has really
high-end, discrete mic pres and Lynx converters. We’re using
Millenia, Neve and SPL for most of our mic pre-amps. About half are
tube, half are solid-state. We also have Universal Audio, the 610 for
some vintage qualities.

Compression wise we’re using the Distressor
and 1176, and the Neve channel-box has some compression built-in too.
Inside the box we’re using the Duende SSL plugins that have all the
amazing EQ’s and the bus compressors built into it. We also have the
Quad UAD2 card in there, with almost every plugin available, and also
the Waves platinum plugins.

So you’re doing all of your mixing inside the box?

Yeah, you know one of the things about mixing in the box is depending on
your platform. Different platforms whether it’s Pro-Tools, Nuendo,
perform differently. And Nuendo seems to be the most amazing platform
that doesn’t have any limitations. It’s a full 32 bit floating point
system. So you’re not losing anything when you’re doing calculations
in the box. Other systems, I can’t say that, they’re fixed point systems.

And because of that, if you want the best and you’re
doing more than 16 tracks of audio, you really have to mix outside
the box. Whether that’s mixing in the box and then using an analog
summing device, like an SSL or a SPL, or if you want some flavor you
want to get a Neve summing box. But the SPL and SSL mixing boxes are
really pristine, high-end boxes. With Nuendo and my system, I don’t
feel a need for that. I’ve actually done tests with Nuendo and
pristine summing boxes. And we will just say, I am not using a
summing box.

Let’s talk a little bit about recording. As a drummer, do you find that you
concentrate more on the foundation of the song, as a
producer/engineer? Do you build it like a house?

Yeah, you know one of the things I found out through working with many
songs over the years is that you have to have that solid foundation
of the rhythm section. Your drums are kind of like your stereo
palette that everything is then applied on top of. And if your drum
sound is just not that great, and let’s say you have the most amazing
guitar sounds and vocal sounds, usually when a listener listens to
that they’ll say, “Well this is a horrible recording, something
just doesn’t sound right.” But if you have the most amazing drum
sounds and you have really crappy sounding guitars usually the
perspective of the listener will be “Wow, that’s a cool production
idea that they did to the guitars.”

(Laugh)

They don’t actually think it’s a bad recording, it’s just a choice.

Yeah, so I did a test once, with one of my friends who had this amazing
$20,000 home stereo system. And he’s like, “Find
the best CDs in your CD collection and I’m going to find the ten best
CDs in my CD collection. Come on over and we’ll do a listening test.”

When I started finding these albums, they didn’t have everything they
were supposed to, and what was missing was the low end. And what I’ve
noticed is that the low end is what makes the band sound either
close, or far away from you and weak.

What I thought were going to be
the CDs I was going to bring over to his house did not make the cut.
Hearing them on my normal home stereo was like, “No, that’s not it.”
The production is amazing, but it’s not the final sound I was looking
for.

So I finally grabbed ten CDs, went over to his place and started
creating this scale that was like, “What
row are we sitting in?” I
feel like I’m in the 400th row – not good enough. Then we pop in a CD and be like, “I’m sitting in the
5th row, this is amazing.” You’re right there, it’s in your face. And it
doesn’t mean it’s dry, it just means that it’s…

Close?

It’s close, and full – and bam it’s right there. And that’s what I like.
That’s my personal preference. I can do anything for a client, but
that’s my personal preference. If I was going to produce something I
want it to sound like the 5th row.

So we did come up with a good, ten CDs that were just amazing
sounding. And they were different, some were more lower, some were a
little brighter, some were a little more organic, but they all had
that “right there” sound. And a lot of that had to do with the low end
of the album. And that low end is very tied into what the drums sound
like. That was kind of like a huge realization about 5-6 years ago
that really started changing a lot of thing on how I start mixing and
what was important.

Do you want to name drop any of those recordings you were talking about?

Yeah, “Cryptic Writings” by Megadeath has a really well produced sound and
has an amazing bass guitar sound. Avril Lavigne’s album “Let Go” I
consider to be the perfect mix, mixed by Chris Lord Alge. Bonnie
Raitt’s “Luck of the Draw” is a good one, as well as James Taylor’s
“Hourglass”. I use two of my own for reference as well, Amber
Norgaard’s “Rising” and Ozlo’s self titled album “Ozlo”. I keep all of
them on file on my hard drive, so I can always compare if I need to.

Since drums and low end are so important, what’s your take on the bass
player?

Now, when I produce a band, the one thing that I tell people is that the
bass player is the most important part of the band. Whoever you
choose as a studio musician to come in, or if it’s the band itself,
that bass part is the most important. And people are like, “What
are you talking about? It’s the songwriter/rhythm and the vocals.”
Well yeah, in the final mix, it’s all about the vocal and whatever
the lead instrument is, coming to the audience. But that bass part
makes that song.

That song can be a heavy rock song, it can be a
country song and it can be a jazz song just because the bass player
changes what he’s playing underneath that songwriter’s chord and
vocal. So unless you get the bass figured out, that song could fall
apart and just go in any direction it shouldn’t be going. And that
bass player has to be the most solid thing.

If you have an amazing
player with an amazing instrument and that low end comes through,
you get that close produced sound. You get the right sound for the
song in note selection and it ties in with the drummer better because
he’s aware of what’s going on rhythmically. That’s what changes the
productions out there.

So when recording, do you use a lot of EQ and compression when going in?

Well, over the years, depending on what equipment I have for use, things
have changed. Obviously there are people out there like Al Schmitt,
that don’t use EQ whether they’re mixing nor tracking. You know, if
he’s working on a big orchestra thing and the violins aren’t just
popping out then he changes the microphone to a different microphone
that has more presence to it. I totally applaud that philosophy. You
need to capture right, you need to put it in the system right. That
being said, should you be scared to touch an EQ? Absolutely not. It
you need to radically do something, then you need to radically do
something.

Are you using any special brands of microphones or are you just using the
normal standards, the 414s, the U87s…?

Over the years I started out small but slowly upgraded. I used a U87 to
find my benchmarks and I found some mics that really sounded as good
back when I did have a budget, and those microphones I still have
today in my mic collection. But you know, for the drums I’ve been
using a lot of Audix. They’re dynamics and condensers and they’re
pencil condensers. D2, D4, D6, i5, I don’t have any (Shure SM) 57s
anymore. All my 57s are now i5s, just because it’s a better
microphone design. Sounds very similar, but better. SCX-25s, SCX-1s,
then I have a U67 which is usually one of my main vocal mics. So I
have the number 1 tube vintage mic that a lot of people use.

But some of the new manufacturers have some amazing technology, like
SE-Gemini II. If I had a choice between my U67 or the SE Gemini II
I’m gonna grab the Gemini. Because
I don’t have to EQ that one. But on the U67 I always like to
just….add a hair of EQ. I also have the SE Titan, which is a
Titanium diaphragm which has just a little extra brilliance to the
microphone. The new Neve SE series RnR-1 ribbon mic – it’s the first
ribbon mic to go all the way up to 25 Khz. And it’s just a work of
art when you look at this thing, so I’ve been using that thing on
everything lately.

I also have a Royer 122 that I use, AT4050s, great all around mics. I
just acquired recently for the first time ever C-414 TL-2s. Haven’t
been using them much, but I’m getting them in there once in a while.

The microphone that I’ve been using a little bit, and I want to
explore more is the SE4400 A microphone. It’s kind of like the C-414
in the sense that it has all these different controls, all these
different patterns, roll offs and pads. But the mounting system is
just amazing because it’s shock-mounted and you can have it within
the cradle, but you can also flip it around to where the cradles is
here and the microphone is out by itself. It’s still shock-mounted
but it’s not in the way of the cradle so you can get it really close
to tom drums and not have that shock-mount in the way. So that’s kind
of the next thing I want to dig into, and start experimenting with
more condensers as drum mics.

People are always very interested in kick drum sounds, and you have a very
interesting approach to miking up your kickdrum.

I have a May mic system built into my kick drum, so there’s a clip
inside of it. What I’m using is a (Audix) D6 and it’s about ¾
back from the beater head into the drum. It’s pointing a little bit
towards the beater head but it’s only about 2-3 inches off the actual
shell itself.

And I have a XLR jack mounted to the shell of my DW
drum. But what this allows me to do is actually close the front head
of the drum, so I get the full resonance of the drum. Now, the D6 has
a V-curve kind of EQ already. It has a lot of low end, and a lot of
high end. So now when you close this drum up, here’s what you get – you get more low end resonance, you get the boom of the drum coming
through, and with the D6 in there it’s able to get all that, so you
don’t need double mics.

You don’t need one mic inside on the beater,
you don’t need one mic outside the drum which not only is going to
bleed from all your cymbals. But you don’t have to put any absorption
to block all that. So for me, it’s the ultimate system. I get
everything I’ve ever wanted, and it’s built in. All I have to do is,
literally plug an XLR cable into my drum and it’s done. I get the
same sound at gigs too.

When mixing, what do you start doing?

I edit all the tracks, get rid of all
the noise, do whatever kind of fade-ins needed, and deal with
problems. After that, I start working on the drum sound.

These days,
when you’re working in the box, your scratch mix can be the starting
point of your mix. If it’s one of those things that was written in
the studio, you probably have a pretty good scratch mix already going
on that you wanna work off of. But I normally start from scratch and
I try to get the drum mix as clean as possible. I want every nuance
of every instrument to come through and also try to get rid of every
nuance that I don’t like; tom ringing for instance.

Once I have a
really good mix on the drums, I will keep going back and tweaking
things as more instruments are put into the mix. A lot of times when
you’re soloing things, you’ll actually give it more low end and
fullness than it really needs to be in the final mix. So I try to
keep that in mind.

So you don’t end up with a very
muddy or boomy drumkit in the end if you…

If you’re EQing the drumset by itself, after a while you kind of know not to go there (adding more low end)
because you know what works by experience. You wait until you have
more things in the mix going on.

So I usually start with the drums,
and whether you’re using room mics on the drumkit, or a plug-in
reverb, the reverb is going to tell the story of the song. That’s
your ambiance that the rest of the band might be in. For me that’s a
big issue of where the song is going.

So along with the drums I’ll
also be looking at the vocals. If I personally tracked the song I
already have an idea of where to go. But I usually start with the
drums and start building. I spend a lot of time on the bass. For me,
the low end is the most important, and the hardest. It’s kind of that
thing you have to wrestle with a little bit and really get dialed in
correctly. Otherwise it’ll be too subby or too obnoxious. You won’t
hear all the notes. So that’s where I start, it’s the whole
foundation of the song.

So when you’ve got the drum reverb
going, would you say you try to fit the rest of the instruments like
they were inside that same room?

Ah, sometimes yes, sometimes no. If
you’re going for that sound, then yeah, that’s the reverb I’ll be
using on the guitars and other rhythms. I’ll probably be using less
reverb on the guitars and more delays and interesting little nuances
though. But it could be possible that there’s no reverb on the rest
of the instruments and there’s only delays, the drums only having
reverb or it’s just the room mics going. Then it’s all about the
vocals, the vocal reverb and trail delays. It just really depends on
what the song is and where it’s going.

What’s your take on compression? Do
you slam things, or do you like to let them breathe?

When it comes to actual tracking with
compression I use minimal compression to add a little extra body and
keep things in check. But I don’t slam stuff ever, just minimal low
ratios. Bass guitar, vocals, acoustic and lead guitar. Around 2dB on
2:1 ratio for control. I’m not opposed to putting things to a higher
ratio, but I usually don’t. I’m looking for the nuances of the
instrument and sometimes compression will rip all that out of the
instrument.

A lot of people do the drum compression thing but I
actually don’t. I play with it every once in a while, usually only
in the mix, never on the way in. I am very well known for my drum
sounds. I love them too, they’re alive and real. But you have to have
a good drummer that plays consistently.

The drummer that hits the
kick drum really light and then on every first beat of the measure
cracks the hell out of it is going to be very problematic. You get
some super high end and then you get it super dull for the rest of
the song. That’s stuff you have to pull out a lot of tools to try to
fix.

If you’re going to use compression it’s to tailor the sound, not
to fix the sound. So I don’t really do the whole stereo room mic
squash. I play around with it but it’s not a necessity for me.

So you’re not a big fan of the New York Compression Trick?

When it’s done right it’s great. One of
the problem I see in today’s home studios is everybody thinks they
have to use compression on everything. They don’t even understand
what compression is so they use presets that has no reference to what
they’re working with. And they got 24 channels of compression going
on and they’re wondering why their mix sounds like a wet dog. The reality is that you just destroyed all of your audio with something
that you don’t know how to use.

I use subtle stuff. A lot of times I
really watch the detection circuit, whether I’m rolling off into the
detection circuit or not. Then when it comes to mixing if there’s
something that needs to be taken care of, if it needs to be brighter
I’ll use 1176 or if needs a warmer sound I’ll use a LA2A. I’ve been
really getting into subtleties of gear. But I’m never really digging
in more than 2-3dB.

So when you have your levels and
pans set and got a rough mix going, what kind of effects are you
using to take your mix from good to great?

My philosophy on effects is whatever
you need that song to be what it is. But when it comes to production
value , what I think is brilliant production is when you have tons of
effects going on but to the average ear it’s not noticeable. And
that’s brilliant.

So super short delays and subtle chorusing?

All sorts of crazy
stuff. Maybe you might have a different reverb in a chorus or you
have this one delay that comes in at this little section on the
vocal. The subtleties that makes the listener know that the mix is
breathing, but they don’t know why. For some reason this mix sounds
better than this other mix that just sounds static throughout.

When I have the freedom to do production and really go at it, I’ll do things
like that. It’s never to purposefully do random stuff, only if it
calls for it and adds to the song. That’s the key, Does it add? Does
it take you somewhere else? Because if it doesn’t there’s no point in
having it in there, it just clutters everything up. I’ll spend my time
really working on subtle things.

But a lot of it is just delay,
chorus, a little bit of reverb, not too much reverb. I might be using
a doubler effect or something that changes stuff up. It’s more about
moving the effects around in real-time than having all these
different parts. The same goes for arrangement, whether it’s
background vocals or when stuff comes into the song. It’s the same
thing. If it’s always on at all times it’s kinda boring, but if it
pops in here or if that little background part is moving or panning
across. Answer and call responses going all over the soundscape gives
the listeners ear a treat.

Photo by Joe Ancona Productions

What’s your work method when mastering?

90% of the time it’s in the box. If it
comes in digital format I’ll leave it there. I do have some tube
units to warm stuff up if I needed to, but with the amount of plugins
I have, I can do most of the job in the box anyways. I try to stay
away from multiple A/D D/A conversion. It’s been a frustrating
process over the years. We were given a format that had 80+dB of
dynamic range. And today it’s just a volume war where there’s no
dynamic range at all. People forgot they have a volume knob. That’s
what the power amp is for, not what the CD is for.

There’s not one mastering conference
you’ll go to where you’ll hear a mastering engineer say, “Oh that’s
so great. I love squashing the hell out of everyone’s music. It’s so
pleasing to my ear.” I was at the last mastering convention at AES
recently, and we had the ten top mastering engineers, Bob Ludwig,
Doug Sax etc.. They start talking about this new Metallica album that
is just destroyed!

Here they are thinking, well maybe this is the
turning point. We’re never going to see this again, we’ve finally
gone too far! and now we can start going back in the other direction.
And sure enough, Metallica says, “You know, we’re going to leave it
the way it is. We’re going to go down in history as the loudest CD
ever”.

The reality is, louder is not better. If it turns on like a
light switch for five minutes and then turns off, it’s so ear
fatiguing, it’s so distorted and annoying to the human ear. Nobody
wants to listen to that over and over. You have 80 dB of dynamic
range, but now you have 2dB of dynamic range on this Metallica album,
if even. I like the band but I’m not buying that CD, cause it’s just
a waste of talent far as I am concerned.

So people like Bob Katz have tried to
put a standard in place. Somewhere between -12dB average RMS levels
of the music, and that’s not the peaks. The peaks will be hitting
-0.1dB, they won’t be hitting 0dB. If you want that super hot level
you can put it to -9 but anything more than that you’re just
destroying everything. Multiple compression techniques, EQ and
limiting are going help get you up there but a lot of home studios I
hear what they’re mastering and it’s so distorted and clipped it’s
actually crunchy sounding. And then they put it to MP3.

Sometimes trying to find a good sounding CD is difficult these days. We are now
turning that point in generations that kids are growing up today not
knowing what a full fidelity CD sounds like. For two reasons, one is
that they’ve never bought a CD, just downloaded it. And the second
one is that CDs are being produced in bands homes now and not
professional studios.

Since so much music is done in home
studios today, do you think big recording studios are a league away
from home studios or do you think people can get away with good
recordings in home studio basement.

There is truth to both in what you’re
saying. If you’re recording an orchestra, yes, it’s all about the
acoustics of that room. You need a huge acoustically treated room to
take care of that. So yes, there’s definitely need for a big huge,
well equipped studio. That has to exist.

Now if you’re going to do
let’s say, Sara Bareilles, she plays piano and she’s a great
vocalist, and she’s doing pop music. Do you want to record the piano
and drums at home? Not necessarily, you want that in a stellar
recording space so you can get those room mics and this great sound.
So that all makes a huge difference in the recording. A stellar,
acoustically sound facility is what it’s about. It’s about the
physics of sound. And that’s what people need to learn. It’s not
about the recording equipment. It’s about the knowledge of the
engineer.

So if you have an experienced engineer
or producer, whether that’s you or somebody else, there’s no reason
to go renting a $1500-2000 a day studio to do vocal tracks when you
could have a nice isolated booth at your house with great recording
equipment. If that’s what you’re going to be doing all the time, then
hey, all the power to you. So yeah, is it a necessity for the big
places? No, but there’s definitely a reason you want to be there and
do certain tracks. And most of it is the engineer or the acoustics of
the facility. Most great engineers work in great facilities so it all
goes hand in hand.

What do you think is the most
important thing in the recording chain?

The thing to focus on is always the
weakest link. Is there a difference between a recording interface
all-in-one unit and separate discrete converters? Yeah. Is it subtle?
It could be, depending on what brand it is and the quality of the
jitter clock etc. Today we have some amazing sounding Firewire
clocks. Back in the day things used to be more about name brands,
like, “Oh I have this UA4A compressor and I have this Sony,” where today I think it’s more about what sounds good and what does
the job for you.

There’s so much stuff and so many manufacturers today…

And the technology has gotten better.
We got to keep in mind that the technology does not sound worse
today. It’s way way better than it used to be 10-15 years ago. The
difference is who’s controlling it. I have Presonus gear that I use
for live recordings and it’s amazing sounding because of it’s clock.
In the studio I have Lynx, and I have SSL. That’s my benchmark. Does
that mean I can’t record a Number One hit on gear that only cost $500?
Absolutely not. I can record a hit, but part of that is who am I
working with, how good is the song, what are my connections and
experience as an engineer. That’s the whole formula.

You mentioned you do on-site
recording, what’s you biggest on-site recording to date?

My biggest onsite recording was a
situation where not only did I have to record it, but I had to build
the system and fly it out there. I built a redundant 48 track (96
total), multitrack digital recorder system. Flew it out to China for
the first international artist concert on the Great Wall. It
consisted of Alicia Keys, Cyndi Lauper, Boyz II Men, Doyle Bramhall
II, Nelly McKay and Silvia Tosun.

Pretty big names.

Very huge names. It was crazy because
we had this 40 foot turntable as a stage. It was on some kind of
mechanism where you flip a switch and within five minutes it would turn
a full 180°. So they split the turntable with a little “Mini
Great Wall”.

Cyndi Lauper was performing while the Alicia Keys
band was setting up on the stage behind. They had 48 channel snakes
on either side that were both split. Those 96 split again to 192
channels coming into my room. I had to process all of those channels
down to two redundant 48 track recorder system. And it’s all live so
once you go, there ain’t no turning back. (laughs)

Pretty stressful, but probably pretty rewarding in the end.

Yeah it was really intense. You really
get so pumped and psyched. Once you hear the director say “AND
WE’RE ROLLING!”(laughs) it was just like you were running a
marathon. Your adrenaline is going crazy. It was awesome.

So that’s the other side of
recording. On-site recording is different than being able to take
your time in the recording studio.

It’s a whole different beast and what
you really have to watch out for in live recording is making sure
you’re never clipping. I just did another recording recently with the
band Calexico, and they were filming for this special documentary
DVD. They had over 44 microphones live on stage and we were recording
56 channels worth of audio. There were mariachis with trumpets and
they’re doing trumpet solos into the vocal mic instead of the trumpet
mic. And then when they hit that high note note you suddenly watch
your meters go “WEEEEEH.” (laughs)

And it’s just random, putting
out fires every second. Just trying to make sure everything is not
clipping. Sometimes the live thing is very stressful, because you’re
trying to do the best job you can and keep everybody in line, but 44
mics on stage for just a rock band?! They had two percussionist, the
drummer, stand up and regular bass Two keyboard players, vibraphone
and other random instruments. Two guitar players, slide, pedal steel.
Bunch of vocal mics. You name it, it was there. It was intense.

Has your approach to engineering
changed since you started 25 years ago, not only because of
digital advance but your thoughts and mentality?

I’m sure my approach has changed over
the years. In fact I notice that when there’s something I want to
achieve – let’s say the ultimate drum sound – I spend years
achieving it. Then finally I one day: “This is the ultimate drum
sound! This is the holy grail.” Then all of a sudden on the next
project I go, well now what? And then I go in a whole different
direction.

So there are things that change and sometimes you have a
perspective on something that all of a sudden radically goes in a
180° direction. One of those things for me, happened to be ribbon
mics. I never understood why anybody would want to use ribbon mics.
And then one day the lightbulb went on and now it’s like one of my
favorite microphones. Not for every application, but for a lot of
applications that were always problematic for me. Now it’s like the
holy grail of microphones to take care of specific situation. How did
I live without it for all these years?

So over time, things have changed.
There’s times where I don’t want to use EQ on anything. I recorded a
24 track New York Hardcore band and the only EQ I used was maybe 1dB
on the snare drum and the rest was flat on the way in and flat on the
way out and it’s an amazing sounding album. I don’t know how but it
just worked. I’m proud of those accomplishments, but I don’t need to
do that all the time. Heading in different directions keeps me
engaged.

So would you recommend recording
enthusiasts constantly try to push themselves in different directions
and not get stuck in a comfortable way of recording.

Yeah. And especially from a production and a engineering standpoint. Once I
was talking to agents about representing me as a producer and
engineer. And they gave me some samples of other people that they
worked with. I popped in the CD and listened to the whole thing. That
was really interesting. All the songs sounded very consistent and
everything sounded good. It wasn’t until the secretary came up to me
and said, “No no, those were 10 different bands on that CD.” And I
was like “What?!?” I didn’t even realize it was 10 different bands
because the guy made every band sound exactly the same and that
doesn’t work for me. I mean, how did I not notice it wasn’t the same
lead vocalist? That’s how exact it sounded. That’s scary to me.

You don’t want to pigeonhole yourself. It’s ok to say that you want to do
one thing and do it well, it’s a great business plan. Specialize,
don’t be everything to everybody. But don’t paint yourself into a
corner. Because it’s going to cause you not to have better
opportunities in the future. In this aspect, knowledge of your
recording medium is essential. In the end, knowledge is power, and
education is the key.

For more of Jim’s work you can find him at the Allusion Studios website, his portfolio page or his online audio store, Pure Wave Audio.

Some vocalist pronounce their S’s more than others. For us recording enthusiasts, this can pose a problem. S sounds have more energy and can be annoying to listen to because they tend to pop out at you, out of context to the rest of the vocal. We want our vocal to sound smooth and clear, and not jump out at you every time the vocalist has a word that starts with an S.

Many audio programs come with so-called de-essers. De-esser are
compressors that only compress the S’s, evening out the energy and
smoothing out the pronunciation of the S’s. De-essers work great but
can be confusing. Logic comes with a de-esser that I have to admit I
have no idea how to use. I have never gotten to grips with that
particular de-esser, and in the end I gave up using it and just created
my own, using some simple routing tricks.

In the following steps, I’m going to walk you through creating a
simple de-esser using a compressor, a side-chain and a little
routing. Since these steps are fairly simple and don’t especially
belong to Logic in a sense, this same technique can be used in any
DAW. Creating a simple de-esser like this can be a real life saver
when the time comes to mix your track.

Step 1 – Duplicate Your Vocal Track

This is the vocal melody we’ll be
processing. It doesn’t have a lot of sibilance, but enough to work
with for now. It even seems to have a little lisp at the end, but
we’ll be ignoring that as well.

Download audio file (vocal.mp3)

We need to duplicate the vocal track we
need to process with our make-shift de-esser. Let’s create a new
track below the main vocal and name it “DE-ESSER”.

Now let’s copy the vocal track above to
this new de-esser track. Grab the vocal track, hold SHIFT and ALT and
drag it down into the new track. That should copy our track and by
holding down the shift key we don’t nudge it or otherwise compromise
it’s location.

We are going to process this vocal
track, pinpointing the S sounds of our vocalist so we can
compress only those sounds later.

Step 2 – EQ your vocal track

Let’s find our offending S sounds in the frequency spectrum and
insert an EQ on the DE-ESSER track and filter out the unwanted
frequencies. Sibilance is found in the range around 7Khz so we’ll be
filtering out the low end up to 6.5 and filtering out the high end
from 7.5Khz or so.

Now take a bell curve and put its Q setting to the highest
number possible so it becomes a big notch boost. By boosting all the
way to the top and sweeping around the 7Khz range we can find where
the S sounds pop out most prominently. When we find that we tighten
up the filters around it, only letting that S sound pass through.

There is another way of pinpointing the sibilance using only the
filters. If you put the Q of the low pass filter, for example, at its
highest (100), it creates a spike where it starts to filter. That way
we end up with almost the same result, but maybe even more exact.

What we end up with is this. No vocal melody to speak of, only
sibilance.

Download audio file (sibilance.mp3)

We don’t actually want to listen to this horribly EQ’d vocal track
so we’ll put the output of this particular track to NONE. Now we can
process our vocal using this EQed track without compromising the
sound of our mix, a ghost track if you will.

Step 3 – Compress Your S’s Using the Sidechain

Most compressors have a side-chain setting. By using the
side-chain we are giving a different track the ability to control our
compressor. In this case it comes in handy because we want the
super-pronounced S’s we did in the last step to control when the
compressor starts pumping.

Lets insert a compresor on the main vocal track.

Now let’s side-chain our vocal compressor to the de-esser track.

Now everytime an S comes through in the singing we should be
seeing the compressor react.

Step 4 – Tweak Your Compressor for a Subtle De-essing Effect

We need to tweak our compressor so our de-essing isn’t an audible
effect. We don’t want anyone to notice the compressor working. We
want it to subtly cut out the S’s and leave everything else intact.
Because we’re only channeling
the compressor to that certain sibilance frequency, we can have it work pretty hard without it crushing the rest of the
audio. As said before, it’s side-chained to the frequency area around
7 Khz, so we’ll push the compressor pretty hard in order to get rid
of the sibilance.

Set the Threshold to around -20 dB so it acts even when the vocal
is singing softly and the ratio pretty high so it cuts off all the
sibilance abrubtly. The way the ratio works is that for every signal
that goes over the threshold it gets cut in relation to the ratio
set. So by having a high ratio setting, like the 8:1 I have below,
for every signal that goes over the threshold it gets divided by
eight, and lets only one part get through.

I set both the attack and release fairly fast because we don’t
want the compresor to linger too much, as it’s sole job is to cut out
as much sibilance as possible. This way, as soon as an S sound comes
in the compressor attacks it immediately but is fast to go back to
it’s original position. Lastly, be sure to set the compressor setting
to peak rather than RMS because we want the compressor to be working
on the peaks of the signal but not the average RMS value of the
vocal.

Now, listen to the vocal, in context with the rest of the track.
We’ve eliminated a lot of the sibilance the vocal had before and now
have a more level, controlled track to work with. Now the vocal sits
better in the track, without our unwanted S’s jumping out at every
possibility.

Download audio file (vocalfinal.mp3)

We have a great vocal track, without the sibilance. Using these
methods you can create a de-esser on any DAW, not only Logic.
De-essers are just simple frequency specific compressors, so by
pointing our compressor in the correct direction, or frequency, we
get the desired result we want.

Conclusion

Because you EQed everything but the S sounds of the vocals, and got
rid of the sibilance by subtly compressing them, you end up with a
cleaner and smoother track. Now you can relax, knowing that your problematic vocalist
is not going to give you problems at the mixdown stage. Never mind his
over pronounced S’s when tracking, you can safely record him knowing
you can effectively dampen those sounds later on.

Although this tutorial has been completed in Record, similar results can be achieved in Reason as the MClass mastering suite is available in both programs. Obviously Reasons lack of audio playback may mean a small workaround but we’ll take a look at this as we progress.

Step 1 – Your Finished Mix

There are a couple of ways you can attack mastering your project in Record. You can either export your entire project into a stereo file and bring this into a new ‘mastering’ project, or the project can be mastered on the fly.

The second approach simply involves strapping your mastering processors across your active project and either tweaking them as you go, or spending time at the end of the track perfecting the sound. Although this can be a useful approach for quickly previewing the project with some processing applied, I tend to prefer starting a new project and importing the dry mix as a stereo file.

The unmastered project.

This method may seem like its a little more work but it separates the writing and mastering processes as two clearly defined parts of your workflow. It also keeps the mastering project clear and uncluttered, you will also find that your CPU headroom is greatly increased using this technique. For the record I tend to use this approach regardless of the DAW I am using to write or master in.

So first up let’s get our project exported. Ensure that there are no mastering processors active on the master output. It’s also a good move to disable Record’s master buss compressor here as we will use it in the mastering process. Remember the less processing your apply to your master now the more freedom you will have to change your mind at a later stage. This untreated file will also be perfect if you need to submit your track for mastering at another studio.

If you wanted to attempt mastering in Reason using this method, which isn’t an ideal scenario, you can load your unmastered tracks into a sampler and trigger them with one continuous midi note. To be totally honest if you are a Reason user it is much easier to master your track on the fly and have the mastering devices loaded throughout your project.

The unmastered audio:

Download audio file (1.mp3)
.

Step 2 – Setting Up Your Mastering Project

First of all start with a blank Record project, This is achieved by either deleting any devices present or using the blank template option in the preferences. Now simply add an audio track, this is the track that will contain our unmastered mix.

The new blank project.

You can now import the unmastered stereo file onto the new track we just created. With the channel and master fader at 0db, the track should play back at exactly the same level as it did on export. In fact this will be a mirror image of your file in every respect. If it’s not then something has probably gone wrong at export and you should go back and check your settings.

The imported unmastered track.

With everything playing back correctly we are ready to start applying some processing to our file. Before we get into the next stage this an ideal point at which to save your project so that any changes are not lost to the abyss!

Step 3 – Buss Compression

Buss compression is a really important part of a achieving a great final master and luckily Record is very well equipped for this process. Records mixer is actually modelled on an SSL series 9000 console and faithful to the original it features it’s famous master buss compressor.

Record’s master buss compressor is a pretty simple device and if you have even basic experience with compressors then the controls should be immediately familiar. There is a ratio, threshold attack and release controls here so nothing overly complex to get your head round.

Record’s master buss compressor.

As with most mastering treatments we are looking at broad, wide brushstrokes. In mixing you may use pretty extreme compression settings to achieve creative effects or to fix problem audio files but in mastering we should be thinking about subtle settings that enhance rather than ‘fix’. This means using a certain amount of restraint and remember that mastering is all about adding that last 5-10% to a mix. Even great mastering won’t make a bad track good.

With all this serious stuff in mind you can try to apply around 3 or 4 db of compression to your master. You will best off using low ratios and reasonably slow attack and release times. This will ensure that you don’t generate any unwanted pumping artefacts.

The settings used for the demo track.

You will notice that the buss compressor’s controls move in ‘notches’ as opposed to a smooth continuos motion like other processors. This is common in professional mastering processors and allows better recall in settings, you will also find the points available are very musical and often work perfectly in the mix.

The general idea of using a compressor across your whole mix is that you create a finished track with a more even dynamic signature. (i.e. with less difference between loud and quiet sections). Buss compression is also famous for acting as a sort of ‘audio glue’, creating a cohesion between instruments which may sound slightly disconnected.

If you are very new to the concept of buss compression you could try reading a tutorial I have already written on the subject.

As many of you will already be aware there is another compressor in Record and Reason that is up to the job of performing master buss compression. The MClass compressor is perfect for the task and has another clear, easy to use interface.

The Mclass compressor.

To use the Mclass compressor or any other mastering device we have to use them as master inserts. The next step will look at how we can set up our own custom mastering chains using this method.

The audio with some basic buss compression:

Download audio file (3.mp3)
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Step 4 – Inserting Mastering Processors

Of course it is likely that we will need more than just a buss compressor to get a polished sound, so let’s look at how we add more mastering processors to our project.

The master section of Record is really just a Combinator in disguise, it is capable of housing as many devices (or inserts) as you like. In this case we want to build a mastering chain so we will be using two or three of the Mclass devices here.

Record’s master insert section.

There are a few ways to get the processors into the master insert section. An easy way is to use preset mastering chains, this is achieved by using the disc symbol in the master insert section on the main mixer. The browser will pop up and show you a decent list of preset chains for specific mastering scenarios.

Although using pre sets is quick and convenient I think its always best to understand whats going on under the hood. With this in mind lets insert our own devices.

On the master channel in the rack you should see a button called ’show inserts’, click this and a new open space should appear at the bottom of the Combinator. By simply right clicking when in this blank space you can create new devices and they will automatically be routed to the master output. Using this method we’ll add one device at a time.

Inserting processors in the master section.

Step 5 – Equalisation

So as we already have Record’s buss compressor taking care of our mixes dynamics but if we want to brighten things a touch or clean up our bottom end we will need some EQ, the Mclass equaliser is perfect for this. Again the focus here is on quality not complexity, so no one should really have a problem with the controls.

The Mclass Equaliser.

The first thing to do is engage the low cut filter. This gets rid of anything below 30hz, reducing rogue subsonics and allowing more signal to pass into our subsequent processors in the chain and ultimately this will allow a high perceived volume in our master.

Now whether you are adding some upper mids to your mix or cutting a bit of low mid you should keep things subtle. As I mentioned earlier broad wide strokes are the name of the game, so low low Q values and no more than about 3 or 4db of gain on any one frequency. This will avoid coloration and stop your mix sounding hyped.

Bottom line is if you are having to add or remove huge amounts of any one frequency, you need to go back to the mix to see why!

A bit of Mild EQ:

Download audio file (5.mp3)
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Step 6 – Stereo Enhancement

There’s nothing like a nice wide master, it adds interest and depth to your sound and can give the track a great sheen. Saying all that, this really is one area where you have to tread carefully and employ the right amount of control.

Using the Mclass stereo imager device we can add just the right amount of width in just the right places. It’s really just a case of choosing the right crossover frequency here. The device employs two separate bands of processing, which have independent width controls.

The Mclass stereo imager.

With your upper band chosen (it helps to use the solo buttons to isolate the correct instruments) you can start to add some width. The lower band should really be left as is, or you can even make it slightly more mono. This will ensure your all important low frequency parts, such as your bass and kick are left untouched.

Some light stereo expansion:

Download audio file (6.mp3)
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Step 7 – Limiting and Exporting the Final Master

“Louder, make it louder” … well louder it is. The Mclass maximiser is a limiter dedicated to art of loudness. It’s really just a typical brick wall limiter with a really cool soft saturation circuit clamped on for good measure.

The Mclass Maximizer.

By simply setting your desired output ceiling you can drive the input to create a denser and louder signal. Try using long attack and release times to avoid any pumping effects. If in doubt opt for the auto release feature. This really is a simple processor so you should be making super loud mixes in no time … not that this is necessarily a good thing, but I’m sure many of you will do it anyway.

In all seriousness try not to go too much over around 3-4db of gain reduction here. You will know if its too loud as you will start to bleed from the ears or pass out. Try to keep some of the dynamic response of your tune, as music is supposed to be a dynamic medium!

The beauty of Record’s mixer and audio support is that you can import multiple version of your mix, maybe containing different takes etc and quickly audition them. Parallel compression and multiple instances of the same track in your mastering project is also easily attainable … but this is perhaps another tutorial !

The final master settings.

The finished mastered audio:

Download audio file (7.mp3)
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