The Air moves backwards and forwards like that across a room. I would say and
Moves in some kind of wave pattern across a room
and it will hit your
Eardrum and vibrate something which turns it into some kind of electrical signal which your brain decodes and you go
oh that's that note or that's that note when we come to get that sound into a computer the
Simplest way would be to have a microphone which basically turns sound into some kind of voltage
Badly Drawn wave like that what I'm [interested] is is how?
Once that voltage gets down the cable
You can put it into something like [a] sound card. Here's one that I've [got] sitting up here
I've actually got a few lying around and they convert
What would be an analog signal something coming through a guitar lead or a microphone cable?
And they'll turn into something that a computer can use which is basically ones and zeros a digital signal
there are two numbers that you associate with
[Wav] file one of which is called the bit depth, and then you've got another thing which is called the sample frequency
So basically these are the two parameters you
use to
Convert an analog signal to a digital signal
I'll start with the sample frequency [the] one I'll write down second
And that is basically the [number] of times per second that the computer will stab in and go
What is this level here, so if we say?
Yeah, there's a number of different levels. You know one going up to whatever this
Axis [here] is time
And it will go across here and and stab in there and go that's the level at that point
And that's the level at that point
That's the level at that point and it will record these as numbers
So you might get like 5 6 8 so the number of slices per second that it does?
is
the sample frequency and obviously the more of those you put in per second in theory the better quality the
[Audio] is going to be now for a big wave like this you don't need a particularly high sample frequency
Because what you're getting there [are] tiny little details you could probably stab in two or three times in each cycle
But obviously if you start to get a wave that's much more
Like that if you only sample there
And then the next sample you took with a you're going to miss all of that data there
It's just going to end up cutting off what you consider the treble [the] high frequencies
So the higher notes would just be cut off I'm talking about physical high frequency data
So I'm talking about high notes like like screeching notes
you know if you like run your
You have two bits of glass scraping against each other it produces a really high
Frequency which hurts your ears that is a real top and note and something. That's like a
Real load Sonic boom sort of note. That's a real low frequency because it
Physically has a longer wavelength and the [high-end] frequencies have a really short wavelength now. We only hear
up to
around
22,000 hertz so that we call that 22 k but there's the two main industry standards that use our
44.1 K and
Forty-eight K and when I say k that's thousand cycles per second. I'm going to put in the kilohertz there, so the reason that
44.1 was initially chosen because it's a strange number that isn't it 44.1. I mean where does [that] Confer basically the highest that they?
Measured at the time. I mean, I don't know the exact science on this
I'll probably be corrected, but the highest that people thought people could hear was twenty two
Thousand and
Fifty Hertz and to get the full cycle of a
22050 hurt wave you need to double the sampling frequency they picked forty four
Thousand one Hundred Hertz as the sample frequency
to be able to recreate anything that we can hear as human beings for various reasons that got changed for video so CDS and
Audio Mp3s, and other such things generally come out other both CDS
definitely doing mp3s most of the time come out at forty four point one and if you put a DVd or a blu-ray in your
Player you will be listening to audio at 48 K so
computers and other devices you know digital high fiims other stuff like that need to be readily available to convert between
Those now there's other nowadays who use 96k some microphones can't hear as high as the human ear
so maybe they're not picking anything up above 16 K so sampling a
44.1 [or] 48 K is well above what you need to pick up the sound from that particular?
Microphone now the second thing is the bit depth and the bit depth is the one that to me does make quite
an important difference
[so] when you get a final product [on] [a] cD or you get it on an or
DVd or a blu-ray or something like that most of the audio is at 16 bits which basically means that between the top level of
The plus here and the bottom level of the minus there the levels that you can have the different
Volume [levels] that you can have you can only have at 16 bits
[65,536] you'd think that's enough different levels, but realistically. It's kind of half that
Because you're dealing with a wave here. It comes down below the center line, and it comes back above
so really you're getting half of that and half of that, so you get half of
65,536 above and half of that below with the zero line in the middle so you [probably] think that
[Thirty-two], and a bit thousand looking to do the exact that's 32 and a bit thousand levels would be enough and for
a final piece of finished music that
That is kind of mastered and ready to go out it. Probably is which is why that's why we get you know
That's the CD standard. That's the DVD standard, but when you're actually dealing with the individual instruments you might end up with a drummer
Who's playing really loudly and hitting away really loudly?
And then right at the end
he's just doing a little cymbal swell [and] there's the
tiniest little amount of volume going on and you [just] want that volume to stay right to the end till the cymbal stops ringing and
unfortunately when you're dealing with those raw
Files that haven't been mixed, and there's not a lot [of] other things going on if you record it [16-bit]
You can really hear it being grainy because it's all very well saying when you're dealing with a wave
That's like that you've got loads and loads and loads and loads and loads
I mean
You could have 32 and a bit thousand levels there to deal with if the way that you're dealing with suddenly becomes only that big
You're only dealing with
that amount of the space
So this is maybe the the drummer hitting I don't know a symbol really hard there at the beginning and then it dies out
Yeah, and ten seconds after you hit the symbol which it's died out and it's become really quiet
And you want to pick up that noise as as
Perfectly as possible without it becoming distorted and fuzzy because what ends up [happening]?
Is you end up creating a wave and you get all these points and then when the computer puts it out?
It puts them as points, and then it's got draw lines between these points, and if you haven't got enough levels you end up with
Effectively square waves which don't sound particularly nice they said?
You get quite a buzzy [sound] to them
so if you hear right on the end of a fade out of a track
And you turn it up really loud in your hi-fi you might be able to hear the song starting to get a bit
A bit dizzy and a bit fizzy at the top end and that's effectively what's happening here
so when you're doing recording if you're trying to record a drum kit, and then a guitar and then a vocal and
And going through and actually layering up as a song you generally want to be recording at 24 bits
so for a 24 bit signal we end up with
16,777,216 different levels
Which is why on a professional level you record everything at 24-bit you do your mixing and Generally within the audio software it
Passes it from one place [to] another actually at 32 bits
You've got even more headroom to move so head rooms like space. Yes, space because
Let's say you've got a wave there
And then you've got another wave here and you add the two together in a piece of software
It's going to produce a wave that's a lot bigger now
If that is heating the top [and] the bottom there, and that's hitting the top of the bottom there
What does happen?
What ends up happening is they get squared off you'd add them together and that might be bigger like that
And then that would [just] be cut off the top that we cut off the bottom because these become No-go
areas for the software because you can't get a number that's bigger than
the amount of bits so you end up with a way that's like that and then it squares off and then it comes [down] and
it's As off and then it goes back [up] [and] it squares off
And it sounds distorted so so [shall] we give one example of how that might sound I'm going to get you to say hello
Hello
Right so that's how you horrendous in my ears that's called digital clipping when you use a digital audio workstation
[you're] summing together Wav files constantly because you might have a bass drum mic a snare drum mic a hi-hat mic
Two or three Tom mics two overheads two room mics a bass guitar two guitars vocals whatever, so you could have twenty four
Thirty six seventy fifty odd tracks of information and with backing vocals
And you're summing them all together and the idea being is if you've been a good engineer
You will have recorded them with enough level that
You're not keeping the level like that because you don't want the level really small because it will sound Grainy as we previously discussed
But you want to send them in with enough headroom as well, so that they don't clip
But when you sum all these things together if you've got things that are only a little bit off the top
Suddenly you're going to produce a signal which is far in excess of what?
24-bit is capable of doing so internally within the programs
they are passing the Wav files at
[32-bit] to make up for the fact that you're summing these all together and then the idea being is
Some piece of software at the end will show you that it's clipping and you put in a plug in and turn the whole
Volume down on the whole thing to get it back, so your final output
[ends] up still sitting within the boundaries of what 24-bit is now?
Realistically if you go the tiniest amount over what it is and you get the tiniest amount of digital clipping
You know if you were recording a drum kit on one snare hit go slightly over
You're not going to stop the whole recording it
But if you were recording a guitar and all the way through [the] whole thing was [just] squaring everything off it might sound pretty nasty
So you have to make these kind of fundamental decisions?
What is going to be your sample frequency and what is going to be your bit rate and generally for most applications?
[you'd] be using 24 bits and 48 k or possibly 44.1
You end up with all [sorts] of annoying?
Artifacts when you're converting from one thing to another you're either down scaling or up scaling you're [either] throwing away data
or you're adding in Data, and you're having to artificially make the data so converting between
48 Km 44.1 Vice versa, it's not the bulk stuff
You hear it's the the symbols and the real higher in frequencies the treble stuff?
That's where you're going to lose the detail by doing that but there's great algorithms out there that do all that stuff
I don't that's run it through a software click the button it does it?
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This and your is what would normally be your compass heading
So it's your your your magnetic compass heading the phone can detect all of those aspects the trick is then how we pass information?
to the plane itself