MALIK ARSHAQ
A SECOND TOO LATE
You've programmed some amazing “amazing” music on your
DAW. You're happy with how it sounds but it could use some
live instruments. You pull out your guitar, hook up your
interface, get the plug-ins in place and hit record. For some
reason, you can't seem to be able to play on time at all. Wait a
minute! We've all been at that point where the annoying delay
due to the latency ruins the vibe of your creative process.
Let's tackle this issue from its root cause. Now, this is a question that's gonna have multiple answers
based on who you're asking. There might be people who tell
you that anything less than 10ms should do just fine and
there might be people who tell you that anything more 5ms
is absolutely unacceptable. Personally, I'd suggest that you
try different latency timings and try recording, whether it's
software instruments or live instruments, and then coming to
a conclusion on what's manageable.
Why does this happen? To understand this, we'll first have to
get our heads around the concept of buffer size. Simply put,
while recording or playing back audio, a stream of data is
being transferred between the soundcard of your interface
and the hard drive of your computer. The buffer size controls
the rate of the data transfer. This means that the higher the
buffer size, the longer the amount time required to process
that chunk of data, i.e higher latency. Your first idea of a
solution would be to turn down the buffer size way down.
Problem solved, right? It should also be noted that plug-in effects can also add their
own processing latency, particularly plugins that look ahead
in the waveform. Plus, most interfaces nowadays have a
“direct” monitoring option that allows you to listen to the
input signal from the interface directly before it goes through
the processing in your DAW. Basically, what this does is that
it lets you monitor your inputs in real-time while recording
without latency. And if you're someone that's chasing “zero
latency”, the Thunderbolt interfaces are currently the fastest
available. Latency times are cut-down to less than 1ms as
it is almost as if you’ll be able to connect directly to the
motherboard of the computer.
Wrong. Setting the buffer size too low would result in the
data being fed too slow to keep up with the rate of processing,
resulting in crackles, glitches, pops and overall bad audio
quality. So, the key is to set your buffer size as low as possible
in order to keep the latency to a minimum, but at the same
time high enough to avoid distortions and pops in the audio.
The question that arises is “So, what is the acceptable
amount of latency?”
40
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Score Magazine
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