Sound Specialists Franklin OH

Latency, the time between hitting a note and hearing it, occurs when you plug your guitars into computers and process the sound with amp simulators. This article is about the measures you could take to solve the problem. If you are annoyed by this problem, this article is right for you.

University of Dayton
Dayton, OH
 
Jazz Arts Studio
211 South Main Street
Dayton, OH
 
Mid. School Of Music
(513) 705-9258
1221 1ST Ave Ste G
Middletown, OH

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Buddy Rogers Music
(513) 829-8333
4771 Dixie Hwy
Fairfield, OH

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Capital University
(614) 236-6183
Columbus OH
Columbus, OH

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Jazz Arts Studio
(937) 223-0100
Dayton OH
Dayton, OH

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University of Dayton (University of Dayton Department of Music)
(937) 229-3936
300 College Park
Dayton, OH
 
West Chester Academy Of Music
(513) 829-2345
8374 Princeton Glendale Rd
West Chester, OH
Hours
Monday through Friday 9 a.m. to 9 p.m.Saturday 9 a.m. to 4 p.m.Birthday Parties ages 1 to 4 by appoi

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Cedarville University
(937) 766-7728
Cedarville OH
Cedarville, OH

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Youngstown State University
1 University Plaza
Youngstown, OH
 
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Lowering Latency

WHEN YOU PLUG YOUR GUITAR INTO A COMPUTER and process the sound with amp simulators, you’ll experience latency—the time between hitting a note and hearing it. Actually, you experience it with amps, too. Sound travels at about one foot per millisecond, so if your amp is five feet away, there’s about a 5ms delay before you hear the note you just played.

The main reason for latency is the computer’s need for a sample buffer. Audio is a continuous process—you can’t have any interruptions. But computers do a zillion things at once—check the keyboard, send signals to the monitor, arbitrate priorities, etc.—so the computer stores several milliseconds of audio samples in a memory buffer. If the computer has to rush off to some other task, there’s some audio in reserve, so audio can continue streaming. If the buffer runs out of samples, your audio will crackle, pop, or die altogether. Not good. But increasing buffer size means more latency. So, you have to find the sweet spot between minimum latency and maximum reliability.

Finding the Sweet Spot

The spec for latency is given either in milliseconds or samples. Milliseconds are more convenient, but most interfaces use samples. Does it matter? No, because all you care about is finding the sweet spot. So…

• Set your audio interface hardware buffer for 512 samples.

• Verify that you can play audio without it crapping out.

• Lower the audio interface latency a little bit at a time—256 samples, then 192, then 128, and so on—until the audio performance deteriorates.

• Now you know how low you can’t go. Increase the buffer size until the audio is reliable again.

Latency-Decreasing Tactics

• For lowest latency on a Mac, use Core Audio interfaces. For Windows, use ASIO interfaces. (Note: Windows programs typically let you choose various interface protocols such as DirectX, MME, and WDM, but choose ASIO—never “emulated” ASIO—for the lowest latency.)

• Faster computers let you get away with lower latencies.

• A program doesn’t have lower latency just because it says it does—use your ears! Some programs list only input latency, but there’s output latency, as well.

• The more your computer has to do, the harder it is to get low latency. Running an amp sim in stand-alone mode, where the computer devotes all its resources to the sim, gives the lowest latency. Run the sim as a plug-in in a DAW that’s running a zillion tracks, plug-ins, and virtual instruments, and you’ll need to increase the sample buffer size. Savvy recordists cut guitar parts through sims early on to ensure low latency. Then, after the guitar parts are done, they increase the latency to handle a progressively more complex project.

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