Have you ever noticed that a loudspeaker is
the opposite of an eardrum?
Eh, probably not.
But it’s true!
See our ears work by concentrating changes
in air pressure onto a small diaphragm that
will move back and forth with the pressure
changes.
This vibration causes stimulation in the heary
bits of the ear which your brain can, assuming
you have normal hearing ability, turn into
what we perceive as sound.
A loudspeaker does the opposite--its diaphragms
(the driver cones) vibrate to create pressure
changes in the air.
This vibration gets transferred to our eardrums
so we can hear it.
We’re sticking to simple stuff today because
the rabbit hole is just too deep.
All you need to know is that things vibrate,
which causes air pressure to fluctuate, which
causes our eardrums to also vibrate, which
stimulates the brain so that we can perceive
that vibration as sound.
This channel started as a series exploring
the history of artificial sound, and it’s
been over TWO YEARS since I last touched on it at all.
Finally we’re finishing this up with the
introduction of
DIGITAL SOUND
(emphasis added with obnoxious reverb).
Since it’s been forever, let’s go over
a brief history of sound recording technologies.
The first device which could reproduce a sound
recording was the phonograph.
Thomas Edison’s invention consisted of an
artificial eardrum, which would vibrate along
with changes in sound pressure, and with the
aid of a collecting horn, the vibration is
transferred into this stylus, creating an
up-and-down motion.
This carves a groove into a wax cylinder,
and the vibrating stylus creates an imprint
of the sound wave.
The depth of that groove becomes a literal
analog of the original sound vibrations.
Then, when the stylus is run over the now
bumpy groove, the bumps cause the diaphragm
to vibrate in the same way as it did when
it first made the bumps, and the result is
that you hear the same sound as before.
Or at least, a barely passable imitation of
that sound.
(sad sounding violin music)
Commercially produced
discs and cylinders were molded from master
recordings, and wouldn’t wear down like
the original wax cylinders.
They were played back using devices like this.
This device is called a reproducer, and
for decades all phonographs were based on
simple acoustic devices like this.
For nearly a century, this is how artificial
sound recording technologies worked.
Something (like this horn) would collect sound
waves, and recreate them onto a physical analog.
Then, that physical analog could recreate
the original sound waves when played back.
While it all started with simple acoustic
devices like this phonograph, eventually improvements
were made.
The development of the electronic microphone
was perhaps the most important.
Now, sound waves cause a receiving diaphragm
to move a coil of wire around a magnet, and
a voltage is produced in the wire as the diaphragm
moves.
This time, sound waves are recreated as a
voltage coming from the microphone, and by
amplifying this voltage and sending it into
a new record cutting device which moves its
cutting stylus as a function of the voltage
it receives, a more accurate carving of the
sound wave could be made into a disc or cylinder.
This greatly improved the fidelity of the
recorded sound, even on acoustic reproduction
devices like this.
With the proliferation of radio--which I feel
I must explain is a sound transmission technology,
not sound recording.
Just so we don’t get confused too much here--
the loudspeaker became a big deal.
Loudspeakers are the opposite of microphones--instead
of producing a voltage as a reaction to a
sound pressure wave moving its diaphragm,
a loudspeaker will move its diaphragm and
create a pressure wave as a reaction to incoming
voltage.
With loudspeakers all the rage, record players
could now use a phonograph cartridge, which
acts like a microphone for records.
The movement of the stylus as the groove vibrates
it generates a voltage which can be amplified
to drive a loudspeaker.
This gets very meta very quickly.
An artificial ear turns sounds into voltage,
and a cutting stylus turns this voltage into
a groove on a record.
Then, a playback stylus playing the record
generates a voltage as the stylus vibrates.
This voltage is then amplified to drive a
loudspeaker, which causes pressure changes
in the air around the loudspeaker, which your
ears concentrate down to your eardrums, and
now your real eardrums are vibrating in roughly
the same way that the original artificial
eardrum moved in the microphone in the first
place.
Yeah.
In essence, the record becomes a way to recreate
the original pattern of voltage created by
the microphone, so that the sound can be heard
again in a different place
at a different time.
Let’s cut out the middle bit because that’s
what’s most confusing.
A microphone like this creates an electrical
signal of fluctuating intensity based on how
its diaphragm moves.
I can just amplify that signal and send it
straight into a loudspeaker, which will reproduce
the sound in real time.
Radio accomplishes this wirelessly, but the
sound isn’t recorded.
To capture the sound coming from the microphone
to be played back later, it has to be converted
into an analog of the signal.
And that’s why it’s called analog recording
technology.
No matter if it’s a record, a cassette tape,
an open reel tape, or even a wax cylinder,
the sound information is recorded “doorectly”...
Doorectly.
Doorectly?
The sound information is recorded directly
onto something, which can then be used to
recreate a copy of the original sound information.
That something is an analog of the original
sound waves.
Improvements in sound technology were for
many years simply incremental.
Wax cylinders became shellac discs.
Shellac became vinyl.
Magnetic recording wire allowed for a reusable,
electronic recording medium.
This was improved into magnetic tape, allowing
for a high fidelity, versatile recording medium
enabling multi-track recording and editing.
And to improve on the noise of magnetic tape,
different particle formulations were developed,
and noise reduction technologies matured.
But we were still just taking some signal
from a microphone, then slapping it basically
as is onto some sort of physical medium.
And that medium was never perfect.
Poorly made tape would cause signal dropouts.
Discs would be plagued by dust and scratches,
and would slowly wear down with each play.
Because the analog medium contained the sound
in its physical properties, it was inherently
prone to wear, damage, and distortion.
Which of course would wear down, damage, or
distort the sound recording itself.
If only there were some way to encode the
sound, perhaps a way to store sound logically
rather than analogously.
Maybe if the signal weren’t the sound itself,
but instead were a set of instructions on
how to recreate it, we could get lossless,
near-perfect sound reproduction.
And thus, digital sound was born.
The heart of uncompressed digital sound is
pulse-code modulation, or PCM.
PCM’s roots can be traced back to the telegraph
days, but its invention as we know it today
for sound came from British Engineer Alec
Reeves.
I feel I must compliment Mr. Reeves on his
given name, it’s excellent.
Very good.
He first devised this digital method of transmitting
and receiving voice communication in 1937,
though it required extremely complex circuitry
for the time.
However, PCM transmission was used during
World War 2 as a way to encrypt extremely
important voice conversations, such as those
between Winston Churchill
and Franklin Delano Roosevelt.
This encryption system was called SIGSALY,
“SIGSALLY”?
“SIGSALIE”?
Or Project X, X System, Ciphony 1, or Green
Hornet.
Anyway, Project Green Sally X System Hornet
1 was much more complicated than simple Pulse
Code Modulation, but PCM was a large part
of its encryption.
So how does PCM work?
It’s actually simpler than it might seem
at first.
It’s rather like a system for repeatedly
asking what the instantaneous amplitude of
a signal is many thousands of times per second,
then simply writing that down.
Let’s look at a simple sine wave.
If this were to be encoded on a vinyl record,
the groove of the record would start out straight
in the center, then move to the left as the
signal intensity reached peak, then it would
start to move to the right, keep moving, keep
moving, and then it would pull back to the
center.
When it’s played back, the movement of the
stylus as the walls of the groove wiggle it
back and forth will recreate this signal.
And audio tape does the same thing, except
the intensity isn’t recorded as a physical
movement, but as a degree of magnetization
on the tape.
But with PCM, we aren’t even trying to recreate
the wave.
Instead, we want to quantify it and play connect-the-dots.
Let’s say I want to take 20 samples of this
waveform.
OK, I’ll divide it up into 20 chunks.
Now I just need to define the detail I can
have within each sample.
Let’s put this on a scale of 0 to 15.
That's 4 bits of resolution.
Now, at each sampling point, we can take the
closest value.
This sine wave can now be represented as the
following string of numbers.
To get the sine wave back, we simply plot
those numbers on a graph.
Then, connect the dots.
Tada! A sine…
wave?
Well, a sloppy sine wave.
But that’s only because we weren’t very
specific.
We only took 20 samples, and each one could
only be one of 16 values.
But now we know the two most crucial parts
of digital sound--the sample rate and the
bit depth.
Perhaps the most common sample rate and bit
depth of digital sound is 44.1 kilohertz,
16 bits.
This means that every second, 44,100 samples
are taken, and each sample can be one of 65,536
values, or 2 to the power of 16.
And that’s how devices like this, a Tascam
DR-05, record sound.
It’s looking at the voltage coming from
the microphone, and taking precise measurements.
Every 44.1 thousandth of a second, it takes
a voltage reading, and, well, writes it down.
It’s furiously quantifying and logging the
voltage it measures with 16 bits of accuracy,
and the result is a string of numbers that
logically represent the shape of the sound
waves that exerted pressure on the microphone’s
diagram.
Pretty neat, huh?
And it can actually write down two numbers
at a time, since this has two microphones
and records in stereo.
Inside this recorder is what’s called an
analog-to-digital converter, or ADC.
The “ADC” is the actual device responsible
for creating the stream of samples.
It takes the analog signal coming from the
microphones themselves and converts it into
a stream of discrete numbers.
If you open the files it makes in audacity,
you see what looks like a waveform of the sound.
It is a waveform, but a waveform that’s
been plotted precisely on a graph.
Zoom way, way, way in on the waveform,
and eventually you can see the individual samples themselves.
And that’s all digital sound is--
it’s a huge list of numbers strung together in order.
To get these numbers back into sound we can
hear, we need to use the opposite of an analog-to-digital
converter, or “ADC”.
So, we’ll use a DAC, or Digital-to-analog
converter.
I like it when names make sense.
A DAC will read the string of numbers, and
generate an analog voltage based upon their
values.
The DAC will smooth out the choppiness of
the samples a bit to make the resulting sound
a little more natural, and now you’ve got
an analog signal to send into an amplifier
and drive a loudspeaker.
The result is a near-perfect reproduction
of the originally recorded sound.
Here’s a very crude analogy to explain the
difference between analog and digital sound.
A vinyl record’s walls generate an analog
signal by moving the stylus left and right...
as well as up and down.
It’s diagonally moved for stereo, but just
imagine for a moment that it’s just left
and right.
A record directly creates the analog signal
via the motion of the stylus.
But a digital sound source is instead sort
of like a virtual stylus riding in a virtual groove.
The sound samples are snapshots in time of
where the stylus was.
A DAC will then create an analog signal by
running a virtual stylus through this virtual
groove and placing it at exactly the correct
location--and thus generating the appropriate
voltage level--as defined by the samples.
By using a giant list of numbers to recreate
sound, instead of the physical properties
of a plastic disc, the sound can be reproduced
flawlessly and accurately with no reliance
on the record player’s cartridge properties,
the integrity of its stylus, it’s motor,
the quality of the vinyl etc.
The biggest boon of digital sound was that
it eliminated all of the little nuances that
might change how a recording sounds.
Digital sound is in a sense, absolute.
But getting digital sound into the hands of
the average consumer took a long while.
DACs and “ADCs” were expensive components,
and the amount of raw data generated by sound
recording was immense for the standards of
the time.
Although 650 megabytes, the data equivalent
of the first compact discs, is a paltry sum
of data in the 21st century, it was unimaginably
huge in the early 1970’s, when the first
commercial digital sound recording took place.
For context, the Commodore 64, released the
same year as the compact disc, has 64 kilobytes
of ram, and that was considered huge for the
time.
A compact disc held roughly ten thousands
times as much data.
64 kilobtyes of CD quality audio lasts this
long;
(clip)
That’s not super helpful.
When we continue, we’ll look at the methods
that were used to store data from digital
recordings, and we’ll discuss the rise of
the compact disc as a robust, consumer-friendly
format for digital sound reproduction and
distribution.
Thanks for watching, I hope you enjoyed the
video!
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