[ELECTRONIC MUSIC]
So let's talk a little bit about signal flow and latency
when you're recording audio.
I like to use a metaphor of water systems,
and it's something that I picked up from another Berklee professor.
And it's a really strong metaphor to help you visualize and remember
how signal flow works in your home studio.
So first think about water falling from the sky.
That's you singing into your microphone.
That's the source.
And it runs down the mountain, through the cables,
into a reservoir, which is your audio interface.
It's going to collect all that signal, and then
send it to the filtration center, which is your computer where you're
going to do all sorts of effects processing,
and filtration of your audio signal.
And from there it flows in an underground pipe system
back through to your interface, and then out to your monitor speakers,
which is the faucet in your kitchen at home.
That process can take a different amount of time.
That journey isn't instantaneous, right?
So sometimes you can experience a little bit of latency as you're recording.
And that's actually the signal flow just taking its time
getting from microphone, interface, computer, interface, back
to your monitor speakers.
And also you have to take into consideration the different places
that it stops along the way.
You're either going to enhance or degrade that signal flow.
In order to combat latency, we're going to adjust the buffer size.
The lower that the buffer is, the harder your computer has to work,
but the faster the sound is going to get to your ears.
The higher the buffer size, the less hard your computer has to work,
but the longer that process takes.
Setting your buffer size is something you kind of
have to do, on and off, throughout the entire session
depending on what it is you're doing.
If you're recording audio, then you want to have
the buffer lower because you don't want to create latency or hear latency
as you're recording.
But if you're just playing back and you're mixing,
you might be able to set the buffer size a bit higher so
that you can hear everything seamlessly without your computer
having to work too hard.
And the way you can tell that your computer is working too hard
is that you're going to hear digital distortion.
Let's create latency and resolve latency.
I have an audio track, audio 2, I'm receiving signal
from an external interface in input 1.
And you can see, I'm already metering here.
So I'm going to arm this track, and I'm now hearing signal
coming in through Ableton.
So let's check the mic and see if we've got any latency starting off.
Check, check check.
So this sounds pretty good.
I'm hearing everything in real time, and this is the optimal condition
for recording my voice.
Now let's navigate to the Preferences menu.
And I'm on the audio tab here, and here's
where we're going to set the buffer size.
So I'm at 128 samples right now.
I'm going to choose the highest sample size, which is 2,048 samples,
and we'll see what happens when we do that.
Checking the microphone.
That's extremely latent, and it would be almost impossible to record a vocal.
So I'm going to adjust this back to 128 samples.
And we don't have any more latency.
So adjusting the buffer size is going to eliminate any latency.
And one thing I really love about Ableton
is that we're actually given a scientific cue
as to the amount of latency.
It looks like at 128 samples, we've got 8.68 milliseconds of latency.
And if I go back to that highest sample setting, I've got 95.8 milliseconds
of latency, which is considerable.
Anything above or around 13 milliseconds, a human ear
is going to consider as being latent or slightly behind.
So just balancing these settings as you move through your session
is going to be really important.