Better Latent? Never!

Jun 1, 2002 12:00 PM, By David Battino

OVERCOMING LATENCY

Fortunately, musicians don't have to keep track of all these little steps — playing 21.5 ms ahead of the beat so their overdubs will sync with previously recorded tracks, for example. “All modern driver architectures have a mechanism to report the latency to the host application,” explains Emagic's Gerhard Lengeling, “so the application can simply move the recorded data position on the timeline” to compensate during overdubs.

Many newer audio interfaces feature zero-latency monitoring, which routes a copy of the incoming signal directly to the interface's output (see Fig. 2). Because the copy you're monitoring doesn't pass through the computer, you hear it instantly, but you won't be able to hear any software-based effects processing. If you do want to record or monitor with effects, however, you can use the old-school solution: simply route the input signal through an outboard hardware mixer and effects, then monitor the mixer's output rather than the computer's.

Similarly, you can use an audio interface such as the Korg OASYS PCI (see Fig. 3), which has onboard digital signal processing (DSP) chips. This design puts the outboard effects on the audio card itself for instant aural gratification. The ultimate (albeit the most expensive) approach is to use a complete DSP-based recording system such as Digidesign Pro Tools. These systems minimize latency by doing all their processing on custom hardware, using the host computer solely for graphics and disk access. (You still get the 1.5 ms A/D and D/A conversion delays.)

DRIVE TO SUCCEED

There are ways to improve latency without hauling out the heavy artillery, though. One of the best is to update your drivers, the tiny pieces of code that handle communication between the music software and the audio interface. Under Windows, this generally means substituting ASIO or WDM drivers for the standard, comatose MME drivers. (Note that your interface and software must support the new driver type.) The default Sound Manager drivers on the Mac provide reasonable latency performance (about 11 ms), but Macs can also benefit from an ASIO upgrade. Most audio interfaces support a variety of drivers; the sidebar “Too Little, Too Latent” offers tips on testing them in your setup. For background on ASIO, WDM, and other driver standards, see “Desktop Musician: Musical Protocols,” in the December 2001 issue of EM and also online.

As mentioned, a faster computer can reduce latency by doing an equivalent amount of processing in the space of a smaller buffer. But you can ease the load on your current CPU by turning off extensions and background tasks. Some people even replace graphics cards that handle interrupts poorly, delaying the computer's response to audio interrupts.

Upgrading your operating system may help, too. “Possibly OS X will be the audio platform for its good real-time behavior,” says Adam Castillo of interface manufacturer M-Audio, which has achieved latencies of less than 1.5 ms on the new Mac OS. The company has also had success under Windows 2000 and XP, clocking 1.5 ms latencies using its Delta cards, Cakewalk Sonar, and WDM drivers. A recent study at the Peabody Institute found that the Linux operating system outperformed both OS X and Windows in some configurations. You can read the study at http://gigue.peabody.jhu.edu/~mdboom/latency-icmc2001.pdf.

Emagic's Clint Ward puts it succinctly: “Latency really occurs when someone is trying to use the native audio system on a slow CPU with no RAM and an old audio I/O device with terrible drivers.”

WORKING LATENT

Whoever said “It's never too latent” probably wasn't a musician. But although some amount of latency is endemic to computer music systems, it's not necessarily the show-stopper that it's often made out to be. Every day, live bands crank out killer performances while their members are standing 15 feet or more apart. Although we don't normally think of it in such terms, this separation creates a “latency” of 15 ms, which is ten times the amount easily obtainable with today's sound cards and software. When you consider that early computer musicians had to wait hours to hear a single note, we've got it pretty easy.

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David Battino is the editor of EM's 2002 Desktop Music Production Guide. He would like to thank Adam Castillo of M-Audio, Greg Ondo of Steinberg, Dug Wright of BIAS, and Clint Ward and Gerhard Lengeling of Emagic for their contributions to this article.

TOO LITTLE, TOO LATENT

When you're confronted with arcane control panels for juggling buffer size and CPU overhead, it's easy to forget that the ultimate goal is to make music. While most experts I spoke with described complex scenarios involving oscilloscopes, auxiliary computers, and other lab-coat apparatus, Clint Ward of Emagic shared this uniquely musical way to optimize any new audio device or driver:

1. Install the driver and start your test with its default buffer settings.
2. Launch a virtual instrument such as the Emagic EXS24 software sampler and load up a drum kit. (A swelling string sound will not work for this test.)
3. Play the drum kit in real time and make a judgment on the feel.
4. Change the driver's buffer settings to the lowest possible value.
5. Play the drum kit again and make a judgment on the feel and the stability of the driver at that setting.
6. Record track after track until the driver or the CPU falls down.

Ward remarks, “In this test you will either experience problems immediately (at the lowest latency settings) or you will eventually hit a computational wall. It will shed immense light on the maturity of the driver, the audio device, and [whether] you need a new CPU. If the driver doesn't fall down until you reach the computational roadblock, you have a potential winner. Increase the buffer size until you perceive latency, then ease off one notch. I feel you give up way too much CPU at the lowest setting when a less greedy one will still feel fine.”

As Brian Smithers noted in “Desktop Musician: Musical Protocols,” in the December 2001 issue of EM, another trick is to reduce the size of the buffer while recording soft-synth parts, then restore it to a safe level during mixdown. That technique provides fast response when you need it most — during live performance.