Why Group Tracks?

There are numerous situations in which groups become useful. You might simply want to mute or solo a particular group of instruments — even if you’re using sub-mixes, you might want to work with small groups within the sub-mix (such as instantly muting all your overheads, room mics and ambience mics so you can check on the close-miced drums).

Perhaps you’ve got a perfect relative mix that you don’t want to disturb — for instance, the relative mix between a set of guitars — but you want to bring the whole group down in volume so you have more headroom in your master fader or sub-mix. There’s only one good way to do it quickly and without throwing your mix out at all.

As you learn about the Pro Tools grouping feature, I’m sure many more scenarios will present themselves to you.

The Groups List

With the Pro Tools window layouts as they are by default, you’ll see the Groups list in the lower left-hand corner of both the Edit window and the Mix window. The groups are listed in the main area of the window, and a small downward-pointing arrow in the upper right corner of the Groups list provides you with a menu of options. This menu allows you to create groups, modify them, and do a variety of other things.

On the left-hand side of the Groups list, where the scrollbar will be if you create enough groups to bring one into play, there’s a button with the letters A and Z in it which determines how groups are sorted — alphabetically or in the order they were created.

While the Groups list looks the same in the Edit and Mix window, each one is different in that there are three types of groups — Mix, Edit and Mix/Edit — and they only display those groups that are classified as a type relevant to the window you are in.

Mix, Edit & Mix/Edit Groups

Mix groups and Edit groups both affect different variables. The Mix/Edit Group is the most versatile in that it’ll do anything that Mix and Edit groups do together. I find myself using the Mix/Edit group more often than not, but it certainly doesn’t mean the other two are less useful and I know users who rarely make use of Mix/Edit groups. Pro Tools will select the Mix/Edit option by default.

Edit groups affect the following:

• Track view
• Track height
• Track timebase
• Audio and MIDI editing functions
• Automation editing functions

Which pretty much means anything you’d normally do in the Edit window.

Mix groups are significantly limited in Pro Tools LE when compared to Pro Tools HD — I’m not sure why you need to pay tens of thousands of dollars just to affect something as simple as input monitoring or plug-in bypass — but as most readers will be on LE systems, I’ll list only those variables LE Mix groups can affect:

• Main volume
• Automation volume
• Main mute
• Solo
• Send level
• Send mute

The last four are options you can select when you create the group, whereas the first two are inherent.

Creating a Group

To create a group, hit Cmd+G (Mac) or Ctrl+G (Win), or bring up the Groups list drop-down menu and click on New Group. The Create Group dialogue box will appear.

It’s wise to name it something other than “Group 1” — something that tells you what the group is for immediately. Get in this habit from the start as you’ll be creating plenty of groups once you realize their usefulness.

The next option to set is the group type, as discussed in the previous section. Most of the time you’ll just set up an Edit and Mix group.

Under Linked Within Groups, there are a bunch of checkboxes — Mutes, Solos, Send Mutes, Send Levels. If you tick these, the group will affect those variables. If you don’t, when you try to solo the group, only the track you’re clicking on will solo. If you want an all-round group tick them all — if the group has a more specific purpose only choose what you need.

The most important part of the dialogue box takes up the most space. The “Available:” list shows you all the tracks in your project that aren’t currently assigned to the group, while the “Currently in Group:” list shows you those that are. The arrow buttons between these lists allows you to move tracks from one side to another.

You can do things a bit more quickly by selecting the tracks you wish to group in the Edit or Arrange window first, then pressing Cmd/Ctrl+G to create the new group. Once you’re in the Create Group window, press the “Add” button at the bottom of the window to add all tracks you have selected in the Mix or Edit windows to the group.

Working with Groups

There are a few basic tasks you’ll probably need to carry out on a regular basis:

Modifying a Group

You may need to modify a group, whether that’s to correct a mistake or simply to change the function of a group once it has served its purpose. To do so, simply right click on the group in the Groups list and press Modify. This will bring up the Modify Groups window, which is identical to the Create Group window.

Duplicating a Group

If you want to duplicate a group — perhaps you need the same selection of tracks for a different purpose or the same settings for another selection of tracks, simply right click on the group in question in the Groups list and select Duplicate.

Deleting a Group

As with modifying and deleting a group, simply right click on the group in question in the Groups list (or alternatively, select a group of groups and right click if you wish to delete more than one) and click on Delete.

Enabling Groups

The only thing left to do is learn how to enable and disable groups. It’s incredibly frustrating at first — if you’re anything like I was when somebody first showed me groups, I’d forget I had a group enabled and go to change a single track’s level, and twenty track’s levels would change!

To enable or disable a group all you have to do is click on it in the Groups list. If it has a blue-grey background, it’s enabled. If the group is not highlighted by any background other than the light-grey of the Groups list itself, it is disabled.

Automation Interface

The basic automation interface can be seen in the Edit window amongst our track controls. Underneath the record, solo and mute buttons is a dropdown menu that is, by default, set to “waveform”. In waveform view, you are able to slice, splice, move and crossfade regions, making it a very useful default view. To automate data on the track, though, we’ll have to change the setting. To do so, click and hold and the list will appear. Feel free to play with these options, but for now we’ll keep waveform selected. There’s a more efficient way of accessing and modifying automation data.

There’s a right-pointing arrow to the left of each track strip. Click on this and you’ll see a new lane appear. By default, this will be set to the volume variable and show a straight line beneath your waveform regions. The first lane will have a plus and minus icon on its left — click the plus arrow to add more lanes, and the minus to remove the lane in question.

At this stage, add two more lanes and set them to mute and pan. Your track should look something like this:

Volume, Panning & Mute

Volume, panning and mute are the basic automation variables for every audio track. Now that we’ve created several automation lanes for them, we can automate without fiddly flicking between view settings on the main track, and when we’re not automating we can simply tuck the automation lanes away with the small arrow icon used to reveal them in the first place.

Automation is edited with the pencil or the smart tool (or, as a more cumbersome alternative to the smart tool, you could use the selector and grabber). With the pencil tool you simply click on the lane and draw:

Regardless of the type of pencil you select, it’s pretty much the same click-and-drag process. As you can see, I haven’t used it on the mute lane — it’s possible, but not efficient. Use the smart tool for that.

With the smart tool you select a section of the region and then hover near the top of the lane until your cursor turns into a rectangle without a line on top (sort of like a squared, elongated U). Then pull up or down, depending on whether you want to increase or decrease the value of the variable, until you’re happy. In the case of the mute variable, there are only two values — on or off — so you can only pull until it snaps into one of the two positions. In this screenshot, you can see changes to the mute lane via the smart tool:

The smart tool isn’t just useful for muting; it’s good wherever you want to accurately affect a specific section of a track, such as chorus vocals. When you want to pan your two lead vocal tracks to create a “doubled” chorus vocal while leaving the verses on the same tracks centered, the smart tool is your friend.

Automating Plug-in Settings

If you have a plug-in active on an insert you can automate just about any variable that plug-in uses to affect the sound via automation. However, unlike the basic automation variables, they aren’t listed in the drop-down by default. There’s a good reason for that: even simple plug-ins usually have a massive number of variables and if you only have the basic EQ, compressor and limiter trio on your track, you’ll still have more variables listed than is manageable.

Other programs like Logic Pro have compromised between ease of accessibility and an uncluttered list by grouping the variables of each plug-in under their own drop-down menu, but we’re not using Logic Pro, so until Digidesign implements an easier method we’ll just have to make do!

You’ll need to have access to your inserts, which will generally mean going to the Mix window as the insert list is hidden by default under Pro Tools 8. If you’re not keen on jumping between windows, there’s still a way to show the insert list in the Edit window. At the top of the track area, click on the small icon representing columns with a downturned arrow to the right of it:

Then click “Inserts A-E” to show the first five inserts on the track. If you’ve gone nuts with inserts, you can add more the same way.

Now that you can see your inserts, right click on the name of the one you want to automate and click Automation Dialog. You’ll see a window like this:

Select any variables in the left column you wish to control and click on the right-pointing arrow to move them to the right column. Variables in the right column will show up under the automation lane drop-down menu.

Notice that to save time, you can switch between inserts on a track without leaving this window. It’s a small touch, but one that prevents headaches.

Now add a new automation lane, click and hold the drop-down menu, and you’ll see the plug-in added to the list. Hover over the plug-in and the drop-down will expand to reveal the selections you made earlier. Once you’ve clicked on one of these, you can automate it:

Automating Sends

You now know almost everything required to do most jobs that need automation! Automating sends, though, is a big part of almost any mix, so let’s finish off by covering this. It’s pretty simple!

Once you create a send in the Mix window (or the Edit window if you’ve added a few other columns since I showed you how to reveal your inserts), you’ll automatically see that send in your automation lane drop-down selector. Of course, it’ll have a name like “Bus 5” which is not helpful when you’re dealing with 30 tracks and somewhere between one and five sends on each. Rename the send by right-clicking on it and pressing Rename.

Now when you check your automation lane drop-down selector you’ll see the name of the send and, like inserts, hovering over it will produce an extended drop-down with variable names inside. In this case we have “level” and “mute”. Send automation is blisteringly simple once you know anything about basic automation, but the terminology is slightly different in that instead of volume, we have level, which is the percentage of the track’s main volume being sent to the bus. It’s the same as modifying the level fader on the send itself.

Explore

Automation is a useful tool with the knowledge you’ve gained from this tutorial alone. However, there’s much more to it. We may cover that at some stage in the future, but for now, explore and find out what other great things you can do with Pro Tools 8’s automation feature set!

Cleaning Up Unused Files

Without a doubt, one of the biggest nuisances with digital multi-track recording is the massive amount of hard drive space each project takes up, especially after doing tens of takes on each instrument! I try to keep things tidy during the recording process, though you do need to be careful not to delete the takes you wanted to keep, and be sure that you won’t want the takes you’re not using later on.

Go to the Regions list to the right of the Edit window in Pro Tools. It’s probably best to widen it so you get a good view of what you’ll be deleting. Here’s a screenshot of an incredibly disorganized project’s region list:

Click on the arrow-in-a-circle button in the top right of the Regions list to open a drop-down menu. Under the Select menu, click on Unused. Your regions list will look like this:

Now go back to the drop-down menu and click on Clear. You’ll see the following dialogue box:

It’s important that you click Delete and not Remove if you want to clear hard drive space. Remove will simply take the files out of the list, leaving them in the folder, and making it really difficult to get rid of unused files later on. If you have lots of audio files to delete, it can get tedious confirming each deletion — holding the Alt button while confirming will delete all files in one fell swoop.

Backing Up Your Session

Pro Tools automatically backs up your project file, so any non-destructive changes can be reversed later on (audio files are not backed up, so destructive changes truly are destructive and final). It’s also easy to drag-and-drop a copy of your project folder from your computer to an external hard drive, or burn the session to CD. But what happens if you lost your Pro Tools CD? Or your Digidesign or M-Audio interface stopped working, meaning the draconian protection measures Digidesign makes use of will prevent Pro Tools from opening at all?

You could spend a decade sorting out your Audio Files folder, especially with editing and crossfades to consider, but there’s a much easier way to prevent this scenario from occurring. It’s not a fun process, but it’s much better than the alternatives. We’re going to save copies of each track’s audio in a way that they can easily be imported into another DAW without spending hours trying to get the timing of each audio file right.

First, we need to get each of our tracks prepared for back-up. Most tracks look something like this:

The track is split into many regions and hence audio files, which will make life hard in the event of an emergency. You’ll also note that this track doesn’t start playing for quite some time after the start of the song, which will make putting in the right spot quite difficult. The solution is to consolidate the region.

Make sure no groups are currently selected. Select all tracks (Ctrl+A on PC and Cmd+A on Mac, or Edit > Select All) and then go to Edit > Consolidate. Assuming everything has been selected between the start and end points, this will turn each track into one audio file, and all these audio files can be inserted at the start of a new project and everything will line up perfectly. Here’s what the track I showed you before looks like now:

At this stage, follow the steps in the first section of this article to clear our unused files. You’re now left with one audio file for each track, so by copying the Audio Files folder in your Pro Tools project folder, you can backup every bit of audio you’ll need to import into a new project or another DAW.

If you find the Region list still contains more files than your project has tracks, you may have excess playlists. Click on the small arrow next to the track name:

Click Delete Unused, and then select all playlists you don’t wish to keep and confirm their deletion. You’ll then need to repeat the steps required to delete unused files from the Region list.

Optionally, at the end of this process you can zip the Audio Files folder. This makes archival a little simpler, and more space efficient, than keeping copies of the folder. Be warned that it’s also slightly easier to end up with a corrupted backup this way, simply because everything is in one file. It’s much easier to retrack one instrument because a particular file in an unzipped folder was corrupted than to start recording the song from scratch in the case that your zip file goes bad.

File & Track Naming

The Audio Files folder is notoriously difficult to make sense of. Compounding this problem is the fact that most Pro Tools users don’t bother to set up naming conventions from the very start of the project. The earlier screenshots showed a project that, while it had some track names in place, wasn’t terribly organized at all. It’s terribly easy to get around the problem of disorganization.

The rule is that files are named after the tracks they’re recorded on. So, the best thing you can do is create all the tracks you know you’ll need at the very start of the project, before you’ve recorded a single take, and name them all. If you name the tracks after you’ve started recording, those files will not take on the new track name.

If you want to record a track that wasn’t anticipated at the start of the project, as will undoubtedly occur in almost every project you work on, it’s not a problem — just create a new track and name it before continuing. The habit of creating every track you think you’ll need at the start of the project just counters the tendency to get lazier as the project goes on by having everything organized from the start.

Next time, we’ll continue looking at Pro Tools project management, but we’ll cross from the drudgery of file management to organizing things so that it’s easier to mix and comp by exploring groups and playlists.

The Parametric EQ

Let’s take a look at the plug-in itself. I’m using the one that comes with Pro Tools LE, but you can use any parametric EQ in any DAW with these steps:

There are several controls you’ll be using all the time, but three you’ll be using the most. We’ll cover those three in a second – let’s get the others out of the way first:

• In: the In button turns that particular EQ control on or off. If you equalize a frequency and then decide you don’t want to keep the change, you can just turn off that EQ band until you need it, instead of having to zero out the settings.
• Shelf/Notch: the two buttons next to the EQ band name determine the shape of your EQ when it is at one end or the other of the frequency spectrum. Those in the middle are notch, meaning they affect a set range of frequencies, but these end bands can be shelved which means they are affected from the bottom of the frequency spectrum (for the LF) up to the set frequency, or the top of the spectrum for the HF.

Gain

Once you’ve pushed a band in, the first control you should tweak is the Gain control. The EQ have no effect without some gain reduction or addition, no matter what you do with the other controls. Gain determines how much of a certain frequency is added or removed. Gain is the vertical axis on the EQ graph, and the taller it is, the more of that frequency is being added:

Q

Q determines how wide or narrow the EQ band is. A setting of 0 will pretty well encompass the entire spectrum (depending on your gain) while a setting of 10 will only affect a very small range of frequencies. Here’s a Q that’s fairly average, though a little on the narrow side:

Frequency

The third control is Frequency. This determines which frequency the band affects, or in most cases where the Q determines that a range of frequencies will be affected, where the center of the frequency range is.

Subtractive EQ is the Best EQ

As we just discussed, you can either boost a frequency or attenuate it. However, just because you can boost doesn’t mean you necessarily should: it’s better to pull frequencies down. When you increase a frequency, the plug-in has to create extra sounds that weren’t there before. When you attenuate, you’re just reducing part of the existing sound, so it stays more natural and realistic.

So what do you do if you want to get a beefier bottom-end? Simple – pull down the high-end!

Here’s a picture of what subtractive EQ looks like:

And this is what additive EQ looks like:

Of course, you can boost if you want to, but it’s a good idea to try the subtractive approach first.

Cut Narrow, Boost Wide!

While we’re on the topic, there are a few best practices for cutting and boosting frequencies. When you’re cutting a frequency, it’s best to make it narrow (a higher Q) and a bit deeper, whereas if you’re boosting, it’s better for it to be wider (a low Q) but shallower (in other words, use gain sparingly).

This is not a hard and fast rule. If you’re recording at home, or have a less-than-perfect take in the studio, you may find yourself cutting wide fairly regularly. And if you just need a sound to poke through in a small range, you might introduce a narrow (though still shallow) EQ.

You can get an idea of how it’s done by looking at this image:

Low and Hi Pass Filters

I find one of the reasons I most frequently use EQ is to slap a low or hi pass filter on.

A low pass filter boosts or attenuates the high frequencies, while the high pass filter does the same for the low frequencies. Usually it’s attenuation that’s happening, though you might find a slight LPF boost on drum overheads, for example, gives things a bit more sparkle.

When I mix a session, I slap a HPF on every track other than the kick drum and bass guitar (unless I have other instruments in the session that are there to be bass instruments). Low frequencies get muddy ridiculously quickly, so it’s important to be ruthless to keep your studio sound sparkle. It’s worst in rock and metal, which is what I usually work with, so do some tests before taking my advice completely in a lighter genre.

Cutting Holes

Even when you’re not controlling low frequency build-up, you’ll need to cut holes in certain instruments. It’s important to let an instrument dominate it’s primary frequency range, so you should cut holes in your instruments that will compliment each other. For instance, the human voice is (generally) strongest in 3.5kHz, so if I found another instrument was competing with our vocalist in that range, I’d pull them down there like so:

This ties into planning your session and arrangement. Find instruments to fill up each major block along the spectrum, and then make sure they are the strongest in their ranges with the help of subtractive EQ.

Finding Problem Frequencies

Before you get to the stage in your session where you’re cutting holes to prevent instruments from fighting each other, you’ll be using EQ to remove “problem” frequencies. If you find the snare is to boxy or the guitars are to jangly, this trick will help you identify the problem and fix it.

Give one of your bands a high Q (10 is not too high in this case) and raise the gain as high as it goes.

Now you need to perform a “sweep” along the spectrum until the problem sound becomes really prominent. When you find the frequency where the problem is at it’s worst, reduce the gain and change the Q until you’ve controlled it. It’s a fairly simple trick, but you’d be surprised that so many people attempt to fix a problem with EQ before they’ve located the frequency range where it’s occurring.

Step 1

Once you’ve got your drums recorded, do a basic mix and try to get the levels where you want them. You want to get the best relative mix between the elements of the kit that you can — don’t despair too much if the kit as a whole is not as strong as you’d like it to be at first, since that’s what this tutorial will teach you to fix.

Step 2

At this point, with a relative mix done, solo each track and fire up an EQ plug-in. Listen to each sound carefully and boost and reduce frequencies as you feel necessary. If you’ve done a great job of recording a great kit in a great room with great hardware, you may not need to do this, but usually you will.

Step 3

At this point apply compression to your drum tracks to keep the peaks down and to prevent drum sounds from totally disappearing after mixing to a particularly loud section. That said, keep the compression as light as you can, because our aim is to ensure that the majority of our kit sound is a natural one with very little compression so we can pull up the parallel compression auxiliary as necessary to fortify it.

Step 4

Got to Track > New… and set up two stereo auxiliary tracks. Click and hold on the input selector and choose a free stereo bus for each. Now’s a good time to name each appropriately — the first can simply be Drums while the second might be Drums Parallel or whatever will remind you of the track’s purpose quickly.

Step 5

Set the output on each drum track to the first auxiliary track’s bus input. This is your main drum level controller. At this stage it may be wise to solo the main drum aux and check your relative mix without any processors applied to the group aux. Ensure that there are no peaks showing on it and check that you haven’t missed any level adjustments you should’ve made after applying EQ and compression.

My uncompressed drum submix sounds like this:

Download audio file (Drums-NoComp.mp3)

Step 6

With your drum aux still soloed, have a close listen and decide which drums are lacking in punch that need to be more upfront in the mix, and also have more impact and tightness. Create a send on each of these tracks to the second aux. Unsolo the first aux and solo the second so that you can get the mix right with the send faders.

An alternative method is to create a send directly from the drum auxiliary to the compression auxiliary, but I like the ability to choose which drums are given that extra oomph, especially since most of my projects have a larger number of drum tracks recorded.

Step 7

Set up a compressor on the second auxiliary. I’ve used the standard Compressor/Limiter Dyn3. Dial in the settings you’d like. You can afford to set the compressor to work considerably harder than you normally would as this will be mixed in under the main drum mix. As you can see in the screenshot, I like to work it particularly hard, especially on rock tracks. This is largely because rock tracks use more drums and more ambience microphones in order to get a big stadium feel, and lose a lot of that crucial rock impact as a result.

Get the threshold down until the gain reduction meter is showing no less than 6dB of gain reduction and probably closer to 12dB for this purpose. You could even go as far as 12 to 18dB.

Often the attack and release are brought way, way down — even as low as the Compressor/Limiter Dyn3 can go. I didn’t want too much pumping to occur for this particular track and was happy with the impact of my drums, so I’ve actually used reasonably tame settings for a parallel compression bus.

Step 8

Mix the drum mix and parallel compression mix together to taste. I prefer to use much less of the parallel compression mix than the main drum mix to retain a natural feel, with the parallel fader brought up just enough to get the tightness I need. Whether you follow my route or want heaps of compressed drums in there, do keep it tucked under the main drum mix at least a little.

The end result should sound something like this:

Download audio file (Drums-PComp.mp3)

It still retains the natural character of a rock drum kit but gives the kick, snare, hat and toms more power and also more presence in the mix. Despite the crazy compression settings we used, though, nothing sounds like it’s pumping too much. This is the effect you’re after when using parallel compression on a drum mix.

Rewire is often underestimated and in many cases goes unused. Using Rewire correctly can actually expand your sound palette, avoid unnecessary rendering and even add extra processing tools to your DAW. Let’s take a look at how Rewire works, some of its history and a few situations we can use it in.

Step 1 – Understanding Rewire

Rewire is a pretty impressive piece of software engineering, allowing MIDI, audio, transport commands and even instrument patches to be transferred between applications. Using the newest Rewire 2, an impressive 256 audio channels and 4080 midi channels, across 255 independent busses can be utilized. That’s more channels than most of us need and certainly enough to handle even the most demanding projects.

Even though it works with cutting edge software applications, Rewire technology is in no way new and is in its tenth year of existence, having been introduced in 1998. Originally developed in a joint venture by Propellerheads and Steinberg, we first saw Rewire appear in the now discontinued (but classic) Rebirth software synthesizer. Since then the protocol has become an industry standard, with just about every major DAW manufacturer adopting it as their primary inter-application streaming method.

If the concept of ‘Rewiring’ applications still seems a little daunting to you just think of it as a virtual patch bay, with ‘cables’ between your audio apps. The real beauty of the system is the diverse types of data Rewire can carry, so sync, transport, midi and audio are all taken care of.

Even in its complexity Rewire is a really elegant protocol, with most of the routing taking place in a pretty invisible fashion, so all you have to take care of are a few basic control panels.

Rewire was first developed to link Rebirth to Cubase VST

Step 2 – Compatible Applications

Rewire and its relationship with most DAWs has really matured over the years. In most cases its integration and performance is rock solid. This pedigree also shows in the long list of applications that support Rewire out of the box. The bottom line is that if you own a major DAW it is very likely to speak the Rewire language.

The list of applications supported is a little long to list here, but you can check whether or not your software is by visiting this page on the Propellerheads site. The list also gives specific details as to which features are supported in each application.

It’s worth mentioning that the slave/host system that Rewire uses at this point, as applications need to be opened in the correct order if things are going to work properly.

One application acts as the host or master, while the other is slaved to it. This means that the host’s transport controls and BPM settings will apply to the slave, so if you hit play on the host the same will happen in the slave. Also audio is streamed from the slave to the host, so if you are trying to get audio from one application to another you need to decide in advance which is the host and which is the slave.

For an application to be used as a host it must be opened first. Any application opened after that will be treated as a Rewire slave. You are not limited to one application here and can open numerous different programs to act as Rewire slaves, but there can be only one host.

The Rewire panel in Cubase 5 running Reason as a slave

Step 3 – Using Rewire to Add Instruments & Features

The main way Rewire is used is to expand the functionality of a piece of software. Let’s say you are working in Cubase 5, and your track is coming along nicely but you want to quickly audition some loops you have warped in Ableton in real-time.

Of course you could import them one by one and re-analyze them using Cubase’s system but this would take time and could slow up the creative process. Alternatively to harness the power of Ableton’s elastic audio functions and to use our previously treated loops, we could simply use Ableton as a Rewire slave.

Once set up you can easily stream audio from anywhere in Ableton to any number of channels in Cubase 5’s mixer. You now have the features of both pieces of software available in one session.

Ableton is capable of being both Rewire slave and host

Just as you may want to share features between DAWs, you may also want to use sounds or instruments to expand you current applications sonic palette.

Using Reason as a virtual rack of sound modules and synths is the perfect example of this. Reason is generally very light on your computers main CPU, so it can be a really wise move to use Reason’s sound sources if you are looking to maximize your machines potential. Rewiring Reason to Cubase or Logic (for instance) can provide excellent results.

The mixing capabilities of the larger DAWs combined with the extensive instrument list in Reason makes a great combo, and remember any of these instruments can be sent to an individual Rewire channel for further processing.

Reason as a Rewire slave

Step 4 – Rewire as Mixdown Tool

Just as some people may want to tap into an application’s instrument list whilst constructing a project, others may feel that they need extra mixing features when it comes to the mix-down and mastering stages. Rewire can really help in this area and if you’re not already using it for this purpose, it may be just the thing you’ve been looking for.

Let’s say you have completed an entire project in Reason or Ableton Live. You may feel you need extra mixing and automation capabilities, with more advanced routing options. While you may love Live’s environment for creating music, Cubase, Logic, Digital Performer or Pro Tools could give you the edge when it comes to mixing your track.

It’s simply a case of setting up Ableton Live as a Rewire Slave and sending as many independent channels as you need into your Rewire host. Of course, you don’t have to send every single channel in your project separately, you could send them as groups. This can save time and still give the required results.

Once your mix is complete, the separate ’stems’ coming into your Rewire host can be bounced for later use and both applications can be saved as usual for instant recall of your mixing session.

Cubase mixing multiple channels from Reason

The routing on the Reason hardware interface

Step 5 – Using Rewire as Vocal Processor

Another more unusual use for Rewire is as a vocal processor connection. Celemony’s excellent Melodyne software and also Yamaha’s Vocoloid can both be connected to hosts via Rewire.

Melodyne provides near perfect pitch correction and formant manipulation and although it comes in plug-in form, the stand-alone version offers a fuller feature set and a more complete interface. When used with a Rewire host the playback of any pitch corrected signals is perfectly synced with your host’s clock. This means you don’t have to render your work immediately and any adjustments can be made on the fly.

The excellent Melodyne studio can used as a Rewire slave

1. All New Interface

When first opening Pro Tools 8, the most noticeable difference is the new interface. The previous interface looks outdated in comparison. The dark gray is much easier on the eyes for those all night mixing sessions. In Pro Tools 8 the interface can be customized simply by holding command on Mac (control on PC) then clicking and dragging the tool section you want to reposition.

The waveform view in regions has been overhauled. Waveforms are now 16 bit resolution which makes them more detailed. Also the region background is transparent. This is extremely useful for making sure your transients are where you want them on the grid.

2. Universe View

An entire session can be viewed at a glance with universe view. For anyone familiar with Adobe Photoshop’s navigator panel, universe view functions similarly, allowing the user to quickly pan around a session with ease. The universe sits docked at the top of the edit window and can be resized as desired. All tracks are represented as thin bars and match the track’s color. On the right edge of the universe view there is a small scrollbar as well as an arrow button to collapse or expand the panel.

3. Track Compositing

Making a comp track is easier than ever with Pro Tools 8. Loop record a few takes of a guitar solo or vocal phrase and then switch the track view from waveform to playlists. Now all the recorded takes show up under the main track. Each take can then be played back individually and applied a one through five rating. Simply right click a region and select rating to rate. This is a useful feature when working on a session over a long period of time, making it easy to remember which takes were the best. To create a comp track use the selection tool to pick out the best parts of different takes and press promote (the little arrow to the left of the track). Now the selection has been moved up to the main track and you are are on your way to the perfect comp.

4. MIDI Editor

Previously Pro Tools was a little lacking in the MIDI department compared to Cubase and Logic. That has changed with the addition of the MIDI Editor window. MIDI tracks now default to region view in the edit window. Double clicking a MIDI region brings up the MIDI Editor which includes its own set of tools (including the smart tool and scrubber) independent from the edit window. Multiple tracks can be viewed in the editor at once by clicking the dots on the left hand side of the track names. On the right hand side, the pencil represents which track is editable. MIDI notes are now darker or lighter based on their velocity. The MIDI Editor can either be docked at the bottom of the edit window or be its own separate window.

5. Score Editor

Composers and arrangers can view, organize, and even edit all of their tracks in the score editor, another new window added in version 8. Those who like working with traditional sheet music will find this feature invaluable. This window also has its own independent tool set and offers many functions of Sibelius built-in. Using the pencil tool it is easy to add and move notes in the score editor. The grid value determines the length of the note that the pencil tool creates.

6. Mini Grand

With its simple interface, the awesome sounds of the Mini Grand will become the go to plug-in for many producers. The included presets sound great; Real Piano, Compressed Dance and Underwater Life are some of the standout presets. The dynamic response is great and the eight velocity layers per key make Mini Grand feel more like a real piano.

7. Vacuum

Vacuum is a great analog tube synth modeler. Being a mono synth, Vacuum is mainly for leads and basses. Vacuum’s controls are broken down into ten main sections. Starting from the left side there are two tube oscillators, the mixer (with drive and ring modulation), tube low and high pass filters, two envelopes, two independent channels of modulation (each with four sources feeding one of five destinations), age and amplitude. At the bottom, Vacuum also has an arpeggiator built-in. Throw on the new AIR Multi-Delay for a great full sounding synth track.

8. Xpand2

This is the second version of Digidesign’s sample playback/synthesis virtual instrument. Xpand2 contains many impressive new patches totaling 1.5gb of content. Each new patch is marked with a “+” in front of the name. The interface has been completely redesigned allowing quicker access to settings that used to be spread across multiple pages. The orchestral instrument sounds really sit well in a mix. An entire composition can be created using only Xpand2.

9. DB-33

This virtual instrument is actually two plug-ins in one. It is an emulation of a Hammond organ and a Leslie cabinet. A guitar or vocal track (or anything else) can be routed through the Leslie by adding the cabinet as an insert to any track. On the organ, selectable tone wheels and the ability to dial in a key click level result in a very satisfying Hammond replication, but as any B3 enthusiast knows, the Leslie cabinet is an inseparable part of the playing style. Look behind a real Leslie cabinet and you won’t find any controls, but all of the nuances you’d want to control are conveniently available in this plug-in. It’s easy to adjust the Leslie to fit whatever sounds you are putting through it.

10. Boom

Boom is a new drum machine virtual instrument featuring ten electronic kits and a very easy to use step sequencer. The edit mode switch lets you use buttons 1-16 to pick patches or edit them. Patches can also be edited inside the grid on the top left. One great feature which may not be immediately apparent is the little screw at the bottom of the controls for each drum. Tightening or loosening this screw tightens up or loosens the sound of the drum. For example loosen the screw to make the kick sound flabby and boomy or tighten the screw to make the kick sound hard and sharp.

11. Automation and Controller Lanes

Now automation and controller lanes can be viewed simultaneously along with the track waveform view. This makes editing automation easier because not only can multiple lanes be visible at once, but the regular track waveform view also remains visible. Click the arrow at the bottom of the track’s colored tab in the edit window to access this feature. To view additional lanes press the plus sign on the automation lane.

12. Elastic Pitch

Elastic pitch is a convenient real time pitch transposition tool. This editing tool is not meant to replace Melodyne or Autotune, but is designed for the purpose of transposing the pitch in a specific area in a region in real time.

To use elastic pitch select an elastic audio region with the hand tool, and then right click the region and select "Elastic Properties" to bring up the elastic pitch window. Whether the singer missed that high note or the guitarist didn’t quite bend high enough during the solo, elastic pitch is quick and easy way to the fix. Elastic pitch can also be used to alter and create variations of the melody.

13. Air Stereo Width

Included with Pro Tools 8 are twenty new Air plug-in effects. Stereo Width widens the stereo image of a track. To get started put the delay and width all the way up to hear the effect working, and then adjust the mode and frequency bands as desired. This effect works great with techno style synth sounds to help achieve a larger than life atmosphere.

14. Air Enhancer

The Air Enhancer acts as a harmonic exciter. Adjust the gain of the frequency being enhanced and generate extra harmonic content by turning up the depth knob. Experiment using this plug-in to enhance the highs on a synth lead from vacuum or the lows on a kick from boom.

15. Air Multi-Delay

Air Multi-Delay features 5 delays, each with its own time, pan and level knobs. Also included is the ability to send one delay into another using the “From” and “To” knobs. All of the control available makes for a lot of fun coming up with new textures.

16. Air Filter Gate

Air Filter Gate has many great patterns to choose from. The filter section provides a Low Pass, Band Pass, High Pass and Phaser Filters. At the bottom of the plug-in there is a modulation section for even more creative possibilities. This plug-in is a natural fit for dance tracks.

17. Torq LE

Torq is a great stand alone program bundled with Pro Tools. It does the beat matching for you by automatically syncing the beat and tempo of your tracks.

To get two songs in sync, first find two songs that are close in tempo (techno/trance songs are easiest). Next drag each song onto a deck on the top left and right sides of the interface. The first time you open the song Torq analyzes it to determine the tempo. After both songs are analyzed press play on Deck A. Then press the arrow on Deck B that says sync to Deck A. Now check that the crossfader is in the center and press play on Deck B. Both tracks should now be playing in perfectly in sync.

18. More Included Plug-ins

Pro Tools 8 comes bundled with additional plug-ins for free which were previously paid options.

• Maxim (a mastering peak limiter with dithering built in)
• Eleven Free – includes the two Digidesign created amps from Eleven
• D-Fi Bundle (LoFi, RectiFi, SciFi and VariFi)
• SansAmp PSA-1 (a guitar modeler)
• TL Utilities (TL InTune, TL Autopan, TL MasterMeter, TL Metro)

19. Quick Start Welcome Screen

Now every time Pro Tools is launched you are greeted by a welcome screen. From here new sessions can be created, recent or past sessions can be opened and sessions can be built from templates. This makes organizing templates much easier than previous versions where there was no place to view templates. Also many starter templates are included for different workflows. At the bottom of the window, parameters for the session can be configured.

20. Additional Features

There are even more great features and plug-ins that come with Pro Tools 8. Some plug-ins deserving honorable mention are: Air Reverb, Air Vintage Filter, and Structure Free. Digidesign has upped the audio track count in a session to 48 mono or stereo audio tracks in Pro Tools LE. The Music Production Toolkit 2 further extends track count up to 64. The plug-in limit per track has been doubled from 5 inserts to 10 inserts per track. To access these additional inserts go to "View" in the menubar and select "Inserts F-J" under "Mix Window Views".

Now a session can be opened with all plug-ins made inactive by holding shift while the session opens. This allows sessions to be opened quickly when searching to find the correct one. Finally, automatic update has been added so you no longer need to remember to check Digidesign’s site every few months to keep up to date.

Conclusion

Pro Tools 8 is a massive upgrade. With Pro Tools LE for only $149 USD, any one of the included plug-ins is worth more than the price of the upgrade. Go and grab your copy and leave 7.4 behind.

What is compression?

Compression is the process of lessening the dynamic range between the loudest and quietest parts of an audio signal. This is done by boosting the quieter signals and attenuating the louder signals. The controls you are given to set up a compressor are usually:

• Threshold – how loud the signal has to be before compression is applied.
• Ratio – how much compression is applied. For example, if the compression ratio is set for 6:1, the input signal will have to cross the threshold by 6 dB for the output level to increase by 1dB.
• Attack – how quickly the compressor starts to work.
• Release – how soon after the signal dips below the threshold the compressor stops.
• Knee – sets how the compressor reacts to signals once the threshold is passed. Hard Knee settings mean it clamps the signal straight away, and Soft Knee means the compression kicks in more gently as the signal goes further past the threshold.
• Make-Up Gain – allows you to boost the compressed signal. as compression often attenuates the signal significantly.
• Output – allows you to boost or attenuate the level of the signal output from the compressor.

Compression Types

Compressors come in various different flavors. These are used by engineers for different tasks and some sound far better in certain situations than others.

VCA Compression

Voltage Controlled Amplifier compressors use an integrated circuit to give very precise control. They are less colored and suffer from very few side effects like distortion, which make them ideal for lots of different tasks. The dBx 160 is a VCA compressor.

Opto Compression

Opto, meaning optical, describes the light sensitive circuits that control the compression amount in opto compressors. They often react more slowly than other compressor types, but this can be desirable. The famed Teletronix LA2A is an optical compressor that many producers swear by for vocals and mix bus compression. The LA2A is also a ‘leveling amplifier’ — which means it is working nearly all the time, not just when a threshold is reached.

FET Compression

Field Effect Compressors use transistors to emulate a valve sound with more reliability, but with a higher signal to noise ratio. They are popular for vocals and great for drum compression. The Urei 1176 is a FET compressor.

Valve Compression

Valve compressors work in one of the three ways described above, but use valves in the amplifier circuit to get that ‘creamy’ sound. The LA2A, which is an opto compressor, uses valves.

How Set Up a Compressor

1. Whether you’re using a hardware compressor or a plug-in, setting up works the same way. Insert the compressor on the channel you want to compress.

2. Adjust the threshold until the peaks in the signal are pushing over the threshold and triggering the compressor. Unless, of course, you really want to clamp something—like a live bass maybe—in this case it can work to make it push over the threshold all the time.

3. Set the Ratio to suit the material. Bass guitars sound good at 4:1, drums at 2:1, vocals also at 2:1 and electric guitars anywhere from 2:1 to 6:1.

4. The Ratio and Threshold work together. Adjust them together and see how they affect the output.

5. The attack and release controls shape how the compressor reacts. A fast attack would be useful for a rapper or anything that has sudden peaks early in the signal. Slower attack times suit mastering uses and buss compression.

6. The release control can really affect the sound of the compressor. Short release times cause the compressor to sound like it’s working hard, but long release times sound more natural.

7. Use the make-up gain and output control to sit the signal back into the mix without adding any unnecessary noise.

8. Setting the hard/soft knee would depend on the material. Hard knee works well for drums, bass and percussive stuff. Soft knee is more transparent and better for vocals and some guitar parts.

9. Look-ahead. Plug-in compressors often have this feature. It uses a slight time delay on the whole song to give the compressor a sneak preview of what’s coming. This allows it to catch all the peaks in the smoothest possible way. It can sometimes cause the compressor to lose its ‘character’ so don’t use it by default—only if necessary.

For every rule about setting up compressors, there’s someone who has broken the rules and made a great sounding record, so experiment.

A final word of warning—compressing on the way to your recording format, be it tape or hard disk—can’t be undone. Use compression sparingly whilst recording. Save it for the mix until you’ve got enough experience to know you’re not overdoing it.

Example Audio

Here is the same drum loop processed with the standard Apple Logic compressor with a variety of settings.

Download audio file (loopclean.mp3)

The clean drum loop with no compression applied.

Download audio file (loopcomp1.mp3)

Opto compression with a slow attack and release, 6:1 ratio and a very low threshold. This is the classic drum pumping sound.

Download audio file (loopcomp2.mp3)

This is the same compression as used above, but with a fast attack.

Download audio file (loopcomp3.mp3)

FET compression with a 2:1 ratio, medium attack and release, and the threshold just peaking.

Download audio file (loopcomp4.mp3)

Same as above, but with a 30:1 ratio. This sound is really getting crushed.

So what plug-in compressors can you use to emulate the sounds of the classics? Well, there are legit emulations of the 1176 and LA2A, and other classic compressors. But there are alternatives.

The Waves RVox sounds very similar to the Urei 1176 and works in a similar way. Hard to beat on vocals.

The PSP Vintage Warmer can sound just like an LA2A. It can also sound very much like an array of tube compressors, like the Manley and Thermionic Culture stuff.

The PSP Mixpressor2 can be made to sound very similar to things like the Fairchild compressors and also the 1176.

Step 1: Find The Frequency

Finding the right frequency to adjust is, of course, the most important thing. With time, some of the more common frequencies become second nature, but what if you’re dealing with a new sound, or just don’t have the experience to know where to start? Here is an easy way to find the right frequency every time.

What you need is a parametric EQ, or at least an EQ that allows you to control the target frequency. Boost one band all the way. If the band has a “Q” control make it quite high (Q stands for “quality factor” and it controls how much on either side of the target frequency is being affected).

Then, play the sound and slowly sweep the frequency back and forth until you find the point where the tone you are looking to focus on is loudest. Make a note of the frequency and put the EQ back to zero. You now know the frequency where your target tone occurs and can cut or boost appropriately.

In the audio samples below we have a fiddle track. The first sample is the track as is. The second sample is using this technique to isolate the croak of the box against the strings.

Download audio file (step-1a.mp3)

Example 1 – Fiddle without EQ

Download audio file (step-1b.mp3)

Example 2 – Fiddle with EQ

Step 2: Think First, Record Second

Before you hit record think about where this instrument is going to sit in the mix, and what it needs to accomplish. For example, an acoustic guitar in a two-piece band will need to be fairly rich and full. But, if an acoustic guitar is part of a ten-piece band, then any fullness will be buried and will just end up contributing to a muddy mix.

These two situations require very different tone, and so therefore should be recorded differently. Take time at the very beginning of the recording process to think about what role every instrument has in the mix and plan its tone accordingly. This should affect your choice of instrument, mic, mic placement, and what sort of room you choose to record in.

Step 3: Understand What You Need To Worry About

Don’t waste your time and energy. It’s important to understand that it’s okay if an individual instrument sounds terrible when you listen to it by itself, as long as it sounds great in the mix. Any individual track only needs to sound good on its own if you hear it on its own at some point in the song; otherwise all that matters is how it sounds in the mix.

In fact, the qualities that make an instrument sound fantastic solo, are often the ones that make it hardest to polish in a full mix.

Step 4: Don’t “Fix In The Mix”

EQ should be the last resort. That is to say, try to get your tone as perfect as possible right from the beginning. If you’ve followed step 2 then you’re half way there, but don’t fall into the “I’ll fix it in the mix” mentality. If you’re not completely happy with the tone you’re getting without an EQ then keep trying.

Mic placement can be one of the biggest factors here. Don’t be afraid to spend the time trying as many different placements as necessary to get the right tone. Keep in mind that small changes in placement can make a big difference. If you have great tone from the start, then EQing during the mixing process will be little more than massaging the sound into place.

Step 5: Cut Narrow, Boost Wide

It’s a good rule of thumb that when cutting it’s best to use a narrow (high) Q, while it is better to have a wide (low) Q when boosting. This will help keep your EQ subtle.

Step 6: Make Cutting Your First Instinct

There are two reasons why it is better to cut than to boost. The first reason is that excessive EQ boosting in a mix usually results in muddiness and loss of clarity. The second is that too much boosting can lead to phasing problems.

In a nutshell, phasing problems occur when waveforms get slightly out of alignment. The result to your tone can be drastic and is generally very undesirable—but I’ll leave the details of phase for another tutorial. Boosting should be done sparingly.

Step 7: Check Into Low-Mid Rehab

Hi, my name is Mark and I’m a recovering low-mid junky. The low-mid range is where all the fullness and body lies for many instruments. For this reason it can be tempting to give those instruments plenty of low-mids. The problem is that all those low-mids fight for room in the mix and if you aren’t careful you’ll be left with a muffled, unintelligible mess.

This problem is furthered by the fact that the low-mid range is an overlapping point for many of the instruments most common in modern music. The chart below shows the approximate range of some common instruments, including their harmonics. You’ll notice that the low mid has a lot going on. Be aware of what’s happening in the low-mid range of your songs and use the EQ appropriately. In some cases it may be necessary to change the arrangement or instrumentation of a track to avoid a low-mid mess.

Step 8: Make Room

Think of your mix as a physical space. The more you put in that space, the smaller the items need to be to fit nicely. So, the more instruments you put in your mix, the harder it will be to fit everything in.

In step 6 we talked about the pile up that occurs in the low-mids. Well, with each additional instrument in a mix, the more important it becomes to keep an eye on the areas where their tonal ranges overlap (look again at the chart from Step 7). Each instrument needs its own place to sit in the mix, so any time there is a common range you need to pick which instrument takes the forefront in that frequency range.

For instruments that have the same basic range, such as bass and kick drum or two guitars, you can use the EQ to interlock them making them both distinct.

This means that any frequency boosted on one should be cut in the other and vice-versa. In the example of a bass and kick drum, if you boost the thump of the bass (100 Hz) then cut at 100 Hz from the kick drum.

Check out the sample file I recorded for an example of some quantized audio.

Example of quantization

Step 1

To become comfortable with the beat correcting process, it is best to a start off with a guitar track because it has many large transients and peaks within the waveform. If you cannot play guitar or do not have any guitar tracks available, bass and vocal tracks will also work fine. Drums are great too, assuming they are on multiple tracks.

It will also help a good amount if your track has breaks or rests similar to the screenshot below. Start off creating a session as normal, and set up either a click track or a drum machine plug in. It is important to play along with this steady rhythm so that you will hear the effect of beat detecting during playback. Lastly, make sure that the “Tab to Transients” and “Keyboard Focus” buttons are checked. These two functions will be helpful throughout the process.

Step 2

First, we will quantize using beat detective, which is a bit more complex than elastic time. Once you have an audio track ready to be quantized, make sure the timeline is set to “Grid” mode and use the hand or selection tool to pick a part of the region you want to beat detect.

After you have the region selected, press the “E” key on your keyboard to get a zoomed in view of the waveform. You will need to closely watch the timeline marker and beat markers on the transients while beat detecting. Now go to “Event > Beat Detective” in the menu bar.

Step 3

Now we have to create the beat markers so beat detective knows where audio should be on the grid. In the first section of beat detective, check to make sure that the start and end beat matches the area of the region you selected. If it does not you will have to click the “Capture Selection” button or type in the numbers manually.

In the drop down list labeled “Contains,” select the quickest note value that occurs in the region you are quantizing. If there is even one note shorter than the “Contains” settings specify, the beat markers will not pick it up.

Step 4

At the end of the window you will see two drop down menus. Depending on what version of Pro Tools you have (M Powered, LE, or HD) you may or may not see the first drop down labeled “Normal” in the screenshot below. The first menu should be left at “Normal” if it exists, and make sure the second menu has “Enhanced Resolution” selected.

Now click “Analyze” and drag the sensitivity slider until you see beat markers placed against all of the notes in the waveform. Check that “Resolution” is set to Sub-Beats. If beat detective makes extra markers in the waveform these can be deleted. You can edit the beat markers by using the hand tool. Double clicking will create a new beat marker, and Option (Alt on a PC) clicking a beat marker will delete it. Also, by clicking and dragging you can move a beat marker, if you feel it is in the wrong place.

Getting the markers right is the most time consuming part of beat detective. Critical listening is required in order to make sure markers are in the right locations. It is sometimes helpful to play the track back at half speed to hear and to see when the timeline hits a spot in the waveform. Half speed playback can be accomplished by holding shift and pressing the spacebar.

Step 5

The next section on the left of the beat detective window called Groove Template Extraction is mainly for extracting a groove from an audio track and applying it to midi data. We do not need this for quantizing, so let’s skip to the section after that called Region Separation. All you need to do here is click the button labeled “Separate” at the bottom right hand corner.

This will spilt the region into many parts based on where the beat markers exist. Now go to the next section called Region Conform. To hear the full effect of the quantization check off “Strength” and drag the slider to 100%. “Exclude Within” and “Swing” should be left unchecked, but these settings can be adjusted later in order to get the feel you want.

After this, click “Conform” in the bottom right and you should see the regions move slightly. If they move too far you may want to go back and check your beat markers.

Step 6

The audio track has now been quantized, but we aren’t done yet. We have to make sure that there are no region gaps (unwanted silent spaces), clicks, or pops in the audio. To prevent these things from happening, go to the last section called Edit Smoothing. In this section there are only two options, Fill Gaps or Fill and Crossfade.

Fill and Crossfade yields the best results for clean audio so select that and leave the Crossfade Length at the default 5ms. After you have it selected, simply click the “Smooth” button at the bottom right. Now it’s time to listen to your fully quantized track!

If everything was done right, the track should have a very tight and possibly, ‘almost too perfect’ sound. Remember that if any part sounds off, just hit undo a few times to get back to before the regions were separated. Then you will be able to edit the beat markers again. Repeat these steps on as many tracks as you wish to correct. Use “Tab to Transients” to assist in selecting within a region, and remember to hit the E key to zoom in on your selection.

Once you go through this process a few times it becomes a very simple and relatively quick way of quantizing audio. Go on to Step 7 to learn a much easier way of quantizing!

Step 7

With Pro Tools version 7.4 it is possible to quantize tracks using elastic audio. Elastic audio will generate markers for you without any work. To enable elastic audio click the arrow button at the bottom of the track and select Polyphonic in the list (bottom arrow shown in screenshot). Wait a couple seconds for the warp markers to generate and then click the waveform button and switch it to warp view (top arrow in screenshot).

Step 8

Here comes the cool part. Use the hand tool to select a region in the warped track and go to Event > Event Operations > Quantize. The window that you see below will appear. Set the quantize grid to the note value you need and leave all the other settings alone. Click apply and just like that your track has been quantized!

Ironically elastic audio seems to have a better algorithm for automatically detecting beats. So why not use elastic audio all the time? Well, for one reason: having many warped tracks can eat up cpu cycles, though minimal in comparison to heavy plug-ins. Also, there are some tracks where elastic audio will not be accurate, and it becomes necessary to use beat detective in order to make sure tracks quantize the way you want.

More often than not though, elastic audio is a great way to quickly quantize your audio tracks.

Step 9

You have now fully quantized your track using elastic audio, but if something does not sound right you will need to know how to edit warp markers. Editing warp markers is very similar to editing beat markers in beat detective.

Use the hand tool to click and drag a marker into a new position, and option (alt on a PC) click a marker to delete it. A marker can be unlocked by double clicking with the hand tool. A new marker can be created with a click from the pencil tool. Experiment with these different options to create the effect you are looking for. Now you have learned how to quantize audio using both beat detective and elastic audio. No more recording dozens of takes until you get it right. Have fun quantizing!