Tempus? Fugit!

The dread evil of latency lurks and skulks through our vale, dragging time to a standstill and marking a greater darkness. Defeat it? We shall try. We shall try.
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“It don’t mean a thing, if it ain’t got that swing. . . .”

So sang one of the great masters. And this maxim is never truer than when you’re recording music. And it ain’t going to swing — or even be in rhythm — if what the performers are hearing as they record doesn’t relate in time to what’s actually being stored on the hard disk platters.

We’re talking about one of the great Tolkien-grade evils of the digital recording age: latency. Unfortunately, we don’t have noble Hobbits to undertake a quest to defeat this evil. But on the plus side, it won’t require a great wizard to vanquish this foe in your studio, either.

Here’s the deal: When you’re recording to most computer-based DAWs, the sources that you are recording are sent through an audio interface, into the computer, recorded to hard disk, sent back through the audio interface, and routed out to headphones or speakers for monitoring.

Easy enough.

The problem is that, as fast as today’s CPUs are, routing through the interface/computer/interface chain takes time. A small amount of time in real world terms, but time nonetheless. In many cases, enough time that there will be an audible rhythmic offset in the signal compared to previously recorded sounds that are playing from the DAW. Swing as hard as you want; if your overdubbed tracks are always playing several milliseconds late, the final result is going to sound like orc crap.


Perhaps the easiest way to deal with audio interface latency is to avoid the problem entirely. A variety of audio interfaces now include the ability to mix live incoming signals with audio coming from the computer, then send that mixed signal back out for the performers to monitor without passing the live signal through the computer; the incoming signals are simultaneously routed through the interface to the computer to be recorded to your DAW software. Many of MOTU’s interfaces, two examples being the 896HD and the Traveler, feature “CueMix DSP,” which essentially puts an 8-bus mixer into the interface, allowing you to monitor “live” inputs with no latency. MOTU isn’t alone in providing interfaces with zero-latency hardware monitoring features: PreSonus, M-Audio, TASCAM, Aardvark, E-mu, Digidesign, Edirol, Lexicon, and other manufacturers offer such capabilities on their boxes, too.


If your audio interface doesn’t offer latency-free monitoring, and purchasing a new one isn’t in the budget, there’s still a solution for avoiding the latency issue: use an external hardware mixer. You don’t have to have a massive room-filling Neve console to turn this trick; depending on how many sources you need to record simultaneously and monitor from your DAW, you could get by with as few as two or three channels. Any mixer that’s set up to handle multitrack recording will work for this application. But manufacturers have stepped up, and you can get by without a multi-bus mixer. Small-format mixers, such as the Soundcraft Compact 4 we reviewed back in the January ’05 issue, are designed specifically to defeat interface latency right out of the box. The Compact 4 mixer has dedicated recording outs for feeding live inputs to your soundcard or DAW, and dedicated “playback” inputs, which would be fed from your soundcard or DAW outputs. The signal from your DAW is blended with the live inputs for monitoring using a cleverly labeled “Mix” control.

But even if your mixer doesn’t come from the factory with connections labeled for connection to your computer, you can still probably figure out a way to make it work. The key is finding a separate output from the main outs for your live input signal to use to feed your DAW; some mixers have direct outs on each channel, others have insert jacks that can be pressed into service, and as a last resort, you could use an aux or effects send. Here’s how it works:

1. Route the signal you want to record into a channel or channels of the mixer.

2. Now we want to find an output to carry that signal to your DAW. You don’t want to use the mixer’s main outputs, since we’ll be using those to carry the blended mix of our live input and the DAW’s output. Check the jackfield of your mixer, looking for a direct output that corresponds to the input you’re using. If you spot one, you’re golden; simply connect it to your DAW input.

If there’s no direct out for the channel, look for an insert jack. An insert jack provides a send (output) and return (input) for the channel where a processor, such as a compressor, can be connected. We’ll use the “send” portion of the insert to serve as a direct out for our channel.

But wait: Most inserts have both send and return connections carried on a single TRS jack. How do we get a separate send connection? There’s a tricky solution: on many mixers (Mackies being a prime example), there are two “clicks” as you push a ¼" connector into the insert jack. If you push a cable in so it “clicks” just once, you can use the jack as a send (direct out), allowing you to feed a DAW input from the channel. Check your mixer’s owner’s manual or check with the manufacturer to determine if this will work for your particular board.

Can’t find a direct out or an insert on your mixer channel? All is not yet lost! If your mixer has an aux send that can feed external devices, you can still get by. Connect the aux send output to your DAW input. Turn up the aux send on the channel you’re recording, and turn up the master aux send level as well.

3. Connect the outputs of your DAW to input channels on the mixer.

Now, when you record, the live signal feeding into the mixer will be routed out to your soundcard or DAW through the direct out/insert send/aux send we set up in step 2. That channel’s output will also feed the mixer’s main outs, where it can be monitored. The outputs from your DAW will also feed through to the mixer’s main outs, and can be blended with the live input so you can monitor the whole works.

One caveat: Depending on whether you are using a direct out, insert send, or aux send, moving the fader on your live input channel(s) may also change the level being recorded to your DAW. Set the gain staging on this channel for the best recording level, then use the faders on the channels being fed from your DAW to adjust the blend in the monitors to taste.


There are other tools you can use to combat input/output latency. Among the most important are settings within your DAW software for reducing the buffer size used by the program. In general, smaller buffer sizes equal smaller latency, in other words, shorter delays.

Almost all software allows you to reduce buffer settings for better latency performance. Some audio drivers/protocols also allow you to fine tune settings to reduce or nearly remove latency from the picture. Where you’ll find this setting depends on your software; it may be in the preferences, under a menu that deals with tweaky audio system settings, or hiding in another menu. Consult your owner’s manual or contact your software’s manufacturer for detailed instructions on where to access this setting, and how to set it. (See also Tech Bench, where Todd Tatnall gets into how to change buffer settings in several programs.)

One word of warning: Reducing the buffer size your software is using generally increases the load that’s placed on your computer’s central processor. In fact, with some software-based methods for reducing latency you may limit the number of plug-ins you can use — if you can use any at all. It may require some trial and error to find the ideal buffer settings for your system. And you may find that you set the buffers to one value when tracking and another when mixing — when you’re mixing and editing, input/output latency is far less of an issue.

The good news is that changes you make to buffer settings can be easily restored — you’re not permanently altering anything. Just be sure to note the default buffer settings so that you can restore them if your computer or software starts acting up. And please, experiment with buffer settings on your own time, and always make some test recordings with material you don’t care about before committing to fooling with buffers on a major session. Nothing kills an artist’s creativity more than tech problems in the studio.


The digital age has provided us with amazing new tools, capable of almost miraculous recording feats. And with all that power have come a few difficulties to surmount, including latency. But there’s no reason your tracks have to suffer from delay-induced arrhythmia. Whether you use an interface capable of no-latency monitoring, an external mixer, or optimized buffer settings, your tracks will arrive on time, in the groove, and in the pocket — swingin’.