As we explored last week, sound is

pressure variations in the air and that's

continually variable.
It's not discrete steps, but constant.
The computer can't understand that kind
of information.
Only thing the computer can deal with is
strings of numbers.
Things represented in 1 and 0s, what we
call binary information.
So, there's a process to go from the
continually variable sound into the
stream of ones and zeros.
And that process is called a sampling
process which we talked briefly about
before.
But I'd like to go into a little more
depth here.
And before we even get to sampling I want
to talk about binary information.
You're going to find that so much of what
you deal with and so many of the numbers
you see when dealing with music
Production is really based on this kind
of information.
So, binary information is based primarily
on the bit.
And a bit is a single, kind of, memory
location, and everything comes down to
bits.
In a single bit, is a 1 or a 0 and that's
all you have.
And every number is collections of those
ones and zeros.
So the number of bits, determines the
maximum number of states, or the biggest
number that you can represent.
If we have a single bit, we can represent
two things, on or off.
Maybe heads or tails on a coin.
or on or off of a switch or a number from
0 to 1.
If we want to represent larger numbers,
we have to start collecting bits into
words.
And a word is a, a collection of bits.
And very often, in the computer, we make
a standardized kind of collection.
You'll have MIDI data, for instance,
uses, commonly uses seven bit words.
Or if we're dealing with digital audio,
we use 16 bit words.
So it's important to know kind of how
many values you can represent with a
specific number of bits.
Or with a specific word length, how many
bits using a, in a particular word.
If I have a one bit word it's going to be
two values.
If I have a two bit word we can actually

represent four values.

And it's really just all the permutations
of ones and zeroes with that many bits.
So if I have a two bit word, we have
values of zero, zero, zero, one, one zero
or one one, so we can represent four
things.
So if I wanted to represent the seasons
of the year, I could use a two bit word
and get Spring, Summer, Fall, and Winter
no problem out of that.
But if I wanted to represent something
larger, I would need a longer word
length.
Now, it's good to know how to kind of
know with numbers how many bits, what's
the largest number you can represent.
And it's always two to the power of the
word length is going to give you the
number of, numbers, the value that you
can represent.
So take it on yourself, let's do a little
quiz here.
What can you get, how many values can you
represent, if you have a 2 bit word?
What about a four bit word?
Seven bit?
Eight bit?
16 bit and 24?
You're going to see that every time you
add a bit, you actually double the number
of values.
And it gets very large very quickly.
Now when we're representing sound in
digital audio.
You're going to find that there's a
couple standards, CD standard is a 16 bit
word.
And when we start measuring sound to
create, to, to create digital audio,
we're going to find that we make many,
many measurements very fast.
And each one of those measurements has a
specific word length.
it's also known as the bit depth, but I
like the term word length better because
words have length.
bits don't really have depth, do they?
So CD standard, is 16 bits.
So that means every we measure sound,
we're using a 16 bit word.
And that really is a great number.
We can represent, I mean everything you
hear on a CD, is done in that quality but
in the studio we tend to use a higher
word length.
And maybe 24 bit would be a great setting
for when you're recording.
And what that gives you is a wider
dynamic range, so we're going to find

that the two,two,two really important

parameters in digital audio.
What we're talking about right now is
word length is related to amplitude and
the one we're about to talk about is the
sampling rate and that will be related to
frequency.
So, word length, the longer the word
length, so if I have 24 bit word length I
have a wider dynamic range.
And that is going to really, it's not
going to be that perceptible, but what it
does allow you to do is not record at
such a high value.
And we noticed that when we're recording
we had to set our input gain very
carefully.
And we wanted to get as loud as possible
without clipping without ever distorting.
Well if we're recording at 24bit, we can
actually record a little bit quieter and
still get a good recording.
And that's the benefit because you're not
going to be so close to distorting as
you're recording.
So I really recommend when you are
recording, record at 24 bit and you might
want to take a moment to go into you DAW
or into your audio interface preferences.
Or even look for the switch on the
outside of your audio interface to see
just how you can convert between or how
you can set the interface with the DAW to
work in 24 bit mode.
It's an important characteristic, an
important thing you want to decide before
you start.
whenever you're working with these
digital audio principles, you want to
make sure you set them once, and then use
them throughout our project.
There can be small issues if you do
change these settings in the middle of a
project.
So you want to create kind of a standard
for yourself.
So, the word length is going to control
your dynamic range, how you know, and we
also call it the resolution of the
recording that you're going to have in
your DAW.
And that's related to amplitude.
The other digital audio property we're
going to be dealing with is the sampling
rate.
So like we said before, when we're
converting from analog to digital, we're
making many, many measurements per
second.
And each one of those measurements has a

specific word length, which we just

talked about.
But how often we do the measurements is
known as the sampling rate.
And that has to happen very fast and on a
perfect clock, right, over and over very
carefully.
And the sampling rate has to happen
actually, we, we have to measure over
40,000 times per second to be able to
accurately represent the continuously
variable signals in the air as a digital
representation.
So it has to happen very fast and very
accurately.
And there are many settings we can have
here and there are different settings for
that sampling rate.
But I'd like to just talk about why we
would choose a specific sampling rate and
the fact is, the higher the sampling rate
the higher frequency that can be
represented accurately in the digital
domain.
And this frequency that, that can be
represented accurately is known as the
Nyquist frequency, but really is just
half your sampling rate.
So if I look at a sampling rate of say
44,100 hertz we can accurately represent
Half of that in the digital domain and
that will be 22,050 hertz.
If we think back to the human hearing
range, we said that the highest thing a
human can hear and that's even kind of an
extreme is 20,000 hertz.
So the CD standard sampling rate of
44,100 hertz can accurately represent
everything we can hear as human beings.
Which is why we chose it for this CD.
Now there are some benefits to go to a
little higher sampling rate.
In working with video, and working with
with videographers you find that they
have a standard setting of 48,000 hertz.
Now I don't think the, the difference in
sound is very audible, I don't think you
can really hear the difference very much,
but it does make it easier to work with a
wide variety of people.
So I would suggest you actually use the
higher settings than we have in a CD.
And that I would say a 24 bit recording
or 24 bit word length is a good idea, and
I would switch over to 48000 hertz sample
rate.
I'd like to take a moment to demonstrate
some of these digital audio principles.
I have here an audio file that was
recorded at 48,000 hertz.

So the sampling frequency was 48,000

hertz.
Its a 4 second WAV file, and what I have
recorded in it is a sine wave playing a
500 hertz tone.
Let's hear the wave form.
[SOUND].
If I zoom way in, you'll see it has a
sine wave shape, a very smooth kind of
shape.
In the sonogram display we have a single
line at 500 hertz, and in the spectrum
analyzer, we had a single peak at 500
hertz.
A sine wave is a special type of wave
form because it's energy at a single
frequency.
We said earlier that most musical sounds
are energy at a fundamental frequency and
then they have partials or harmonics
above that.
And we saw that with the sonogram and
spectral analysis of my voice earlier.
A sign wave is special because its energy
at a single frequency; you can think of
it as only a fundamental tone.
Now, we said earlier, that this is
recorded at a 48,000 hertz sampling rate.
And if I zoom way in we'll actually see I
have the sine wave.
And if I zoom further in we can see the
individual points that are those
individual measurements of digital audio.
So, when I play back this sine wave at
48,000 hertz, it's going to play each one
of those at a 48,000th of a second.
So we hear one sample and then a 48,000th
of a second later, we hear the next
sample.
Then a 48,000th of a second later, we
hear the next sample.
What happens if I play this back at a
different sampling rate?
Well, let me try it.
The, the sampling rate originally is
forty-eight thousand hertz.
I'll play it back at 96,000 Hertz.
Now, let's hear it.
[SOUND].
Do you notice what's different?
Now we see a peek at 1k, 1,000 hertz.
And we have our fundamental frequency
line here at 1,000 hertz.
We also see that the wave form is now
half the length.
It was four seconds before, and now it's
at two seconds.
So playing something back at a different
sampling rate, is like speeding up or
slowing down a, a record in a turn table.

In that if I speed it up twice as fast,

from 48,000 to 96,000.
And if I speed it up twice as fast we're
going to have half the length, and it's
going to be double the frequency, or an
octave higher.
Now, I point this out because sometimes
you get errors like this.
So this is the original 500 hertz tone.
[SOUND] .
Playing back at a 48000 hertz sample
rate.
What if I was to change this a little bit
to put it at 44100?
[SOUND] .
It's a little bit out of tune.
And it is a little bit longer.
So sometimes, when working with digital
audio, you'll have a sample rate
mismatch, where something's played back
at the wrong sample rate.
And it sounds as if a record was played a
little faster or a little slower.
Or a lot faster or a lot slower,
depending how big the mismatch is.
So, something to be aware of, and to
watch out for.
And also something that you might want to
play with creatively at some point if you
want to speed something up or slow
something down drastically.
Now the next thing I would like to point
out is just how good we are at hearing
individual samples.
I'm going to zoom in around two seconds
here.
Quite a bit.
To the point where we see the individual
samples.
Now rarely do you need to adjust the
level of an individual sample but
sometimes for corrrective purposes if
there's a digitial glich.
you do want to go and edit individual
samples and this DAW has the ability to
do that.
So I will grab and just move one of these
samples in both left and right, just a
little bit.
So you see, I'm just changing a single
sample.
Remember there are 48 thousand of these
per second right, so it's a very very
tiny slice of time actually a 40th
thousandth of a second.
And what I find amazing is how obvious it
is when we hear that.
That little mismatch, that tiny little
moment, is clearly audible.
Let's hear.

[SOUND].
It's kind of hard to hear in this, see in
this spectrum slate, but there's a, a
thin blue line there.
Right here's the thin blue line, that's
that click.
And I sure heard it.
Let's hear it again.
[SOUND].
That high frequency click, that was that
one sample out.
So, even a single sample can have a
dramatic impact on your audio and the
quality of your audio.
It's really nice to know what these kind
of digital glitches are so you can
identify when you hear a problem.
But it's also really important to know
just how specific and how perfect that
audio has to be.
It has to come out exactly in time,
48,000 times per second, and those
samples have to be perfect every time.
If you think about it, that's actually a
lot of computational power that needs to
happen in the computer.
To make sure each one of those samples
goes out perfectly 48,000 times per
second.
Because even if just one of them is
wrong, it's going to be an audible click
for the listener.