introduction to digital sound design: week two study notes

7/30/2019 Introduction to digital sound design: Week Two Study Notes

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Week 2

We've talked about how sound works, with the sound pressure waves that move the air molecules

back and forth through a compression, and then rarefaction where the sound molecules hit our ear

drums. The ear drums capture that energy and then move it into the cochlea within the ear, where

we have fluid that vibrates the basilar membrane which starts to resonate with the various ratios of

vibrations which then sends electrical signals to our brain, and we hear sound.

So let's talk about how we're going to capture that same physical energy into an electronic device to

convert that in-electrical energy into an analogue of electrical energy. This electrical energy will then

allow us to send it to through cables, into a mixer, and eventually we'll convert that into a digital

representation, which would be how the computer records that same information.

Microphones

The way that we convert physical energy into electrical energy is

through some kind of transduction. Most of the transducers that

we are familiar with are called microphones.

How do microphones work?

There are many different design microphones but I want to talk

and focus on two primarily today.

Dynamic Microphone

The dynamic microphone is the most common and durable type

of microphone that you can find.

It works under the following principle:

A dynamic microphone has membrane covering

the top of the microphone; usually this is going

to be polyvinylchloride or some other kind of a

thin material. This membrane, or diaphragm,

will move according to the air pressure. The

sound pressure waves that hit it will start tomove that diaphragm. Connected to the

diaphragm is a copper coil, surrounded by a

magnet, which has a positive and negative

charge on each side.

As the diaphragm begins to move, that coil of copper moves as well, and as the copper moves closer

within that magnetic field, it will then send a positive and negative energy electrical charge out of

the cable.

What we've got is a conversion of the electrical energy or the acoustic energy, into an electrical

signal, which will then convert this into patterns of alternating current.



In electricity there are two types of current: direct and alternating. Sound cannot be represented in

direct current. We have to have something that alternates, to be an analogue for the alternations of

compression energy in the air wave formed.

This microphone principle is also the principle used in speakers, it is just the inverse - you have a

cone which is moving, and you have a moving coil around a magnet, and then as the electrical

energy moves that magnet and coil, it moves the diaphragm, which moves the air molecules again,

and we hear the sound.

Condenser Microphone

The condenser microphone doesn’t have a magnet to

give you the polarized electrical patterns. You have

two plates at the top of the microphone, and one of

them is going to be charged to a positive electrical

energy (plates are often made of platinum or gold).The other one will be negative, and the difference

between those two is called capacitance. As the air

pressure pushes down on the top plate, it pushes it

closer to the, the bottom plate which that then creates a capacitance difference. Over time that

distance is then captured and sent out to our cable, and we get both the positive and negative

electrical energy going out to the alternating current again.

Although this microphone is more delicate and fragile, it records a much higher quality sound. But

the trick with this mic is pre-charging the plates. It can be

done in various ways:

One method is to actually have a battery on board

with either a nine volt battery or another kind of NiCd

battery.

Another method is called “Phantom Power”

Phantom power was first developed by the German

microphone company called Neumann, back in 1966. They

were working with condenser technology and along with the

Norwegian Broadcasting Company; they came up with a

standard that is 48 volt phantom power.

Phantom power is used to charge the 2 plates by plugging the microphone cable into a phantom

power source, so that the charge can be sent through the cable to the mic. This is usually an external

power source, but most mixers come with a phantom power source built in anyway, that can be

used if you are connecting a condenser microphone to your mixer.



Variables of microphones:

There are important things to consider when deciding on which microphone to use.

The technology of dynamic versus condenser.

Dynamic you'll find more in sound reinforcement, and condenser you'll find more in studio recording

applications, but you can find both in, in either.

Polar Pattern

The polar pattern is the microphone’s sensitivity towards the direction of sound coming into the

microphone. There are a few different types of polar patterns:

Omnidirectional

microphone – sound can

be received from all 360degrees ate an equal

sensitivity.

Boundary microphone -

a boundary microphone

is an omnidirectional

microphone that can be

placed, literally at a

boundary, but also has

very good reach

capabilities.

Cardioid pattern – a cardioid will give you sound in front of the microphone and some of the

side energy coming in.

Hyper cardioid or super cardioid - what those do is narrow the energy field more as you get

in front of the microphone. Usually the hyper and super cardioids will allow some energy

directly behind to come into that.

Specialised figures:

Figure 8 - A figure eight is a sort of a double cardioid. What it does is allow for sound energy,

to come in the front of the microphone and also equal energy to come in behind the

microphone. It is a very powerful figure used in recording.



Shotgun microphone – this is often used in video or film applications and is basically a very

narrow cardioid mic, and only allows receiving from a very small range.

Depending on what you're recording or your sound application, you would choose one of those

cardioid patterns. In order to have a versatile microphone and the more expensive microphones will

give you this shifting.



Microphones part 2

Another aspect of microphones that's important for us to understand is the frequency response of

microphones. We talked about the two designs, the dynamic and the condenser microphones as

being important. One for primarily used for sound reinforcement - the dynamic, and then the

condenser, primary found in studio recordings. The reason that those are done that way is not only

the design but also the amount of money you pay to get a good frequency response.

Remember we talked early on about the human ear can hear frequencies from about twenty hertz,

twenty vibrations per second up to around eighteen to 20,000 hertz. So when we have a

microphone we need to have something that can pick up the energy of that same frequency range.

A really good microphone, usually costing a little bit more money, will give you a frequency response

of at least twenty up to 20,000 hertz. Now, as you come down in price range, and design, you will

find microphones which have a narrower range than that.

If you don't have a microphone that can represent the sound energy you won’t hear any direct

difference of the kind of sounds it can record other than a slight timbral change. Timbral is the result

of all the vibrations of frequencies that are going above the fundamental, so all of those, up to

20,000 hertz affect the colour of the sound that we hear.

Some expansive mic’s mays also give you the option of changing the sensitivity and range at which it

records. If you're recording in a live setting, many times in halls you will have air conditioning sound

that will go through the vents. Usually those are around 75 up to 150 hertz. They're very low

frequencies. A microphone may allow you to reduce the sensitivity of that range so you don't pick up

the sound of the air conditioning etc.

Cable Systems and Impedance

Impedance is the amount of resistance the electrical energy faces, leaving the microphone.

Balanced cable systems

A balanced cable system can be identified as having a 3 pin connecter, connected to what is

standardly called a microphone cable, or XLR cable. (This kind of microphone cable is going to be

used in most of your applications for recording and sound reinforcement.)

Each microphone cable applies a resistance to the electrical energy leaving the mic. This resistance

is usually 600 ohm. If the symbol “lo z” or “lo – z” is on the mic, it means that it is a low impedance

microphone, and has a smaller resistance.

The good thing about low Z impedance microphones is that it will travel long distances because

there's very little resistance. The down side of that is that, if you are traveling a long distance with

very little resistance or ohms, in the cabling, you can easily pick up other radio frequencies that are

in the space or in the air.

So, along with the positive and negative pin on the mic, you have to have this third pin that acts as

shield that keeps out other signals.



Unbalanced cabling systems

An unbalanced cabling system consists of a 2 pin connector, connected to a standard microphone

cable. This specific type of microphone is called a high impedance microphone. These are used only

in recording studios; you are not ever going to be running it in a sound reinforcement situation.

The high impedance generally runs about 50,000 ohms, or 50 kilo-ohms, giving you a lot more

resistance that's built into the signal. This means that it's going to be very protective of any kind of

outside interference coming into it, but, you can only run it for a short distance.



Mixer Basics

Now that we understand how sound energy is converted into electrical energy through the

microphones, through various kinds of transduction systems, then it moves through a cable,

with various kinds of connectors and then what's going to happen is all those microphonesare going to go into something called a mixer.

The mixer is where we're going to adjust the levels, we can change some of the

amplification of certain frequency ranges, and we can basically add effects to the sound at

that point. The mixer is where everything sort of happens in the piece.

Components of a mixer:

1) Input channels



There are multiple channels of information, coming from microphones, guitars, synthesizers

etc. all plugged in the top row, of inputs. You can see here that we have the XLR Connector

for the microphones. And then we see a place for the quarter inch phone plugs. Then here

we see the RCA plugs for a CD player or an iPod or anything else that we would plug in here.

2) Stereo device input channels

In channels five -six, seven - eight, nine - ten, eleven - twelve, you will see we only have one

output volume, down here. This is because these are designed so that we can plug in any

kind of corded phone cord that already has a stereo output on it. E.g. a stereo system, a CD

player, an iPod, a synthesizer, anything that is generating stereo information.

3) Output channels

Usually on the back of a mixer, you will see, this one has a two channel output. So

everything that plugs into the top of the microphone we would then be mixing into a stereo,

a left and a right channel output. And here again you see this is another XLR output so we

can send this to our recording device if we want to record to the computer; record, send it

to speakers for, sound reinforcement.

So the purpose of the output channels would be taking the final mixed signal and into the

output.

4) Power Switch and Power Input

Also at the back of the mixer.

5) Phantom Power Switch

Most mixers today will have a phantom power source on them. If you have a condenser

microphone plugged into one of these four microphone inputs, you will need to turn on the

phantom power for, for those microphones. Phantom power only goes through these first

four channels of inputs. In case you have a condenser microphone that needs to be charged

up once you're done.

6) Pre-amplifier

The electrical energy leaving the mic is usually very low - so the first thing you have to do is

to run it through a preamplifier. The preamplifier is going to raise that signal up high enough

that we can then actually manipulate it through the mixer.

The very first knob we see at the top here, determines how much energy of Pre-

amplification we apply to the signal coming in. You want to get the correct balance between

pre-amplification and the overall amplitude. If the preamplifier is too high, the preamp will



over ride and the sound will distort. If it is too low, not enough energy will be sent through,

and you will end up turning the output level up high, which will also distort the sound.

7) Equaliser(s)

These blue knobs on the mixer above are effecting what is called equalization. Equalizationallows us to divide the frequency spectrum. (20 – 20 000 hertz) An equalizer allows us to

amplify or attenuate various ranges of that spectrum. The simplest would be on a device

that has tone controls - a base knob and a treble knob. It allows you to then take a very wide

band of frequency spectrum, and manipulate the upper and lower partials, changing the

timbre of the sound.

Above, we have a three pole equalizer: In this one we have a low frequency knob which is

set at a mid-range of about 80 hertz. Above that, we see a mid-range, which is 2500 hertz,

and then the upper one going to be about 12,000 hertz.

Some mixers give you the ability to change the mid-range of each equaliser pin, making the

frequency range it changes higher or lower.

Types of equalisers:

Parametric equalizer Graphic equalizer

Equalisers are critical when doing sound reinforcement, because in sound reinforcement

you're going to have to be dealing with what's called feedback. Feedback is the returning or

bouncing of sound. This will be different in each situation that you're working in; because

they will all have a different acoustic mapping.

So, when a sound engineer comes in to set up a sound system, they have to be checking the

room to see where the frequencies that get over-amplified in that particular room are.

Every room is different, so they will analyse, tale a spectrum analyser and will run what is

called pink noise. A pink noise generator which sends all frequencies into the sound system,

and then they measure the output to see, are there places where certain frequencies are

getting amplified because of the acoustics of the room. What the sound engineer would do

is measure those points and then on the parametric equalizer they would specify those

frequency points and then reduce those down so you end up with a flat response.



8) Pan

Pan is the next knob under the equalizer, and is very important in a stereo. If a microphone

coming in and I want to send it to the left channel, than I take my pan knob and move it to

the left and it would only come out the left channel. If I move it to the right, then it only

could at the right.

So, why you would be doing this in sound reinforcement? For example: Let say you've got

musicians on stage, and you've got speakers on either side of the stage. Let’s say the

guitarist a little to the right side of the stage. Then when you create a sonic image, you

would probably move the panning’s slightly to the right so that when the audience is

hearing that guitar player, the sound is actually coming from speaker more from the right,

making it actually looks like the sound is coming directly from the musician.



Mixer Basics part 2

9) Auxiliary sends (effects sends)

On our mixer, there are two extra knobs above the equalizer – these are the auxiliary control units.

These units send the sound to the auxiliary output control area. This auxiliary output is divided into:the stereo, auxiliary return and auxiliary send; number one and two.

What you can do is:

We can then take an instrument plugged into channel one, and send that information to axillary

output 1 or 2. I control the gain and that it will get mixed into auxiliary send channel one – in the

auxiliary control area. From there, I can then send that to somewhere else - a separate equalizer, a

monitor, speaker, another mixer, flange, reverb, delays, compressors, expanders etc.

And then coming off of the hardware, you would bring the output of that back into the auxiliary

returns and then it would get mixed in with the output signal.

For sound reinforcement purposes, the auxiliary send will have multiple purposes: you may also go

to a reverb unit or some other kind of effects. But primarily, your auxiliary sends are going to be

sending this particular information to monitors on stage.

The normal sound reinforcement configuration

You have your stage speakers, facing the audience. Usually they're on the sides of the performers,

not behind the performers, because you're going to get a potential feedback. The musicians also

need to be able to hear each other on stage, so often you will see another set of speakers; it's awhole other configuration of speakers that are pointing back toward the musicians, so that they can

hear each other on stage. They don't need to be hearing what the audience is hearing, but they need

to be able to hear, specifically what they need to, to be able to play together and that's called the

monitor system.

Today, more often than not, you'll find wireless monitors, which are being plugged into a little

earphone in their ear, and you don't have to have these big banks of speakers down on the floor. But

it's the same principle, where that auxiliary send one, that keyboard is being sent to the vocalist's

monitor through our auxiliary send channel, so that they can hear that stronger and the bass and all

the other instruments will be reduced.

So that's why, our auxiliary sends are critical, for being able to have a group play together.

I mentioned in sound reinforcement, there are three different systems often in play:

So when a sound engineer is mixing the sound reinforcement, you're doing the stage speakers,

which are trying to balance and make sure you don't get feedback. You're trying to get the right

sound mixing throughout the whole hall. Then you're trying to get these multiple monitors, all

having the right level of mix, going back to the musicians so they can play together. And many times,

you'll have a third level of amplification, which are your guitar amplifiers.



Often you will see a guitarist who prefers to have their particular Fender amp, or whatever design

they want, and often the sound engineer has to put a microphone in front of that amplifier, because

they can't just plug directly into the mixer, because they want that particular quality of that amplifier

sound into the sound of the guitar. So then you have to transfer the sound of that amplifier, that

Fender guitar amplifier, record it, and mix it correctly into that. So in some ways the way we do

sound reinforcement is a very sophisticated and sort of multiple problems can emerge on that; and

so a good sound engineer sort of understands all those three different systems and how they have

to operate together, and so the mixing console is where all of this is going to happen.

10) Potentiometer

Down at the bottom of each of these channels, you'll always have the same kind of information, a

potentiometer. Larger mixers will have a slide fader that does the same thing. It amplifies or

attenuates the electrical signal - what we call the volume control or the final gain.

So remember the pre-amp gain does the same thing, it controls how much signal gets pre-amplified,and then the potentiometer is going to determine the final mixing of that, how much of that goes to

the outputs on the back.

In professional recording studios, usually you will have a specialist who will come in and usually,

called the tonmeister - someone who's a specialist in knowing how to record sound and put the

exact kind of effects processing on it and do the exact kind of mixing it, to make it sound really great

for commercial purposes. So when you hear an artist that you hear on your downloaded sound file,

whether it's Coldplay or Lady Gaga, they usually will have one sound artist and tonmeister who

comes in to do the final mix - because the exact mixing level, how much sound goes to a gate, how

much goes to a reverb or delay, all of that information is a very subtle art form, and a tonmeister issomeone who specializes in creating a particular kind of quality of sound that, that each artist will try

to have, a world of sound that is unique to them.

11) Master output configuration

This is where we finalise all the information and send it to the next destination. We can see the

master level for auxiliary one and two and have a master control for how much gain goes out to

auxiliary one and two overall.

There's a control room output. So if we were in a recording studio, you may want to send your

master outputs into a computer to record the sound, but you may need to hear the sound in the

studio at the same time, so you would have a control room out so we can mix the signal to the

control room, and then you would have another volume control for the control room, but then your

master output volume would be controlling how much goes to the master outputs, which goes to

your mixer and to your recording device. There are multiple ways of sending the information to

wherever it needs to go, for people to be able hear it or monitor it, and then to the recordings areas

separately.

This is a 12-channel input and a 2-channel output. So when you look at a mixer, often you will see

different configurations. This is a very simple mixer but often you will find 24, 48, 72 channels of

input and so you'll see these big long mixers with lots of fader inputs, they're basically having the

same kind of functions there, but most of your more expensive mixers will have a different kind



output. So the output could be 2, 4, 8, 16, and 24. So if you want sixteen just for channels of output

that means you can assign everything to 16 different outputs that can go into sixteen different

channels in your computer program, or it can go to sixteen different speakers, or all sorts of reasons.

The larger mixers, you're probably going to have different combinations.

You'll have what's called, for example, if you see something that's a 24 by 8 by 2; if you see a mixture

that has that underneath, that means that there are 24 inputs, and then you can send that to either

eight channels and two channel outputs. So you can mix it into eight channels separately, or you can

also combine that into a two channel stereo output.



Audio File Formats

Let's talk a little bit about the different kind of ways that we can record audio and the different kind

of formats that you'll find on different computers. This sometimes can be confusing as to people, as

to what kind of format, how much memory that's taking up, and the quality of that.

Three Major Groups of Audio Files

1) Uncompressed audio formats, such as WAV, AIFF, AU, or Raw Header-less PCM

This is the main type of file used in recording studios – an uncompressed file format; which means

that when you record your sampling rate of, say audio's going to be at 44 100 or 44.1, thousand

samples per second (you usually would go even higher to 96 and up in professional studio’s), you're

recording that you're getting a lot of data that's representing the sound file into the computer, and

for professional level you don't really want to have that compressed. You want to keep the file and

as much information about the sound as you possibly can; so that as you work with it and edit it andmix it and so forth, you're not diminishing any of the potential sound.

The uncompressed audio formats that you will find are referred to as the WAV, or the AIFF or the AU

or sometimes the raw header-less P C M files. Those are the files that, that are going to be the

largest size, because of the amount information in them.

2) Lossless compression

This is a type of compression, which stores data in less space by eliminating unnecessary data; no

data is permanently removed. The audio data is just packed to save more space; 50% data reduction.

This is in some ways the most preferable of the compression formats, because you're not actually

losing any of the, the original data, you're just reorganizing it into a kind of tinier space, but you

don't get that much saving of memory off of that.



3) Lossy compression

These are compression algorithms which take the data and try to figure out which of the information

that has been sampled and stored is not necessarily information that's going to be heard, and it

takes out the data that is not really going to be affecting the sound to any great degree. There is a

(slightly audible) data reduction of 80 – 90%.You probably will find on your mobile devices and most

of the MP3's etc. is found in this lossy compression style.



So those are, the main different kind of audio formats you will find, just remember that most of

your computer sequencing programs that we'll be working with will be recording into an

uncompressed format. Once they're in either in a WAV, or AFF file, we can then usually export them

out of our computer into some kind of compressed format that can be played on your mobile

devices.



MIDI

MIDI stands for Musical Instrument Digital Interface. This was an important development back in the

early 80's when manufacturers discovered that they could send some information from keyboards to

tone modules to also several electronic devices to where they could communicate.

The Development of MIDI

Sequential Circuits (a company which made popular synthesizers back in the late 70's and

early 80's) engineers, Dave Smith and Chet Wood, devised a universal synthesizer interface

in 1981.

By January 1983, they were able to demonstrate a MIDI connection, digital interface

connection, between the sequential circuits Prophet 600 analogue synthesizer and a Roland

JP-6.

So these were different manufacturers but they both had that capability on them so the informationcould go from one synthesizer into the other one and trigger a device from that.

On 19 August 1983 MIDI specification was published.

And almost all the manufacturers of electronic instruments started to put MIDI capability on their

devices. Meaning that those devices could connect and communicate with each other.

The MIDI commend set includes note-on, note-off, key velocity, pitch bends and other

methods of controlling a synthesizer.

The MIDI commands don't send full audio information, it only send a little bit of computer data anddirects us toward the basic performance information. So what you will find a MIDI stream of

computer data is not the full audio information, but the simple note on and note off.

The MIDI protocol uses an eight bit serial transmission, with one start bit, and one stop bit.

It's running at 31.25 kilo baud sampling rate (Kbs data rate), and is asynchronous (sends data

in 1 direction, 1 character at a time

So a computer number, let's say we have a middle C that we're trying to send. A middle C, in the

MIDI stream data, would be a number 60.

Connections are made through a 5-pin midi cable / DIN plug, of which 3 pins are used.

You will not find midi connectors on a computer. If you want to have midi go into the computer you

have to have some kind of external interface, usually a separate box and midi interface that you can

buy, that would allow you to plug your MIDI connectors, cables from your synthesizers and other

devices into that MIDI interface box. From there you would usually convert it to a USB or Fire Wire,

or some other kind of connector into the computer.



MIDI controllers

The most important part of midi is: what kind of controller do we have? Because, midi, here again, is

just simple performance information. So, manufacturers started developing different kind of ways of

triggering that midi information. Obviously, the most common type would be a keyboard like this.

Where we're just sending simple data, but there are all sorts of other manufacturers that started

developing different kind of ways of capturing performance data that can be triggering MIDI

performance information.

One of the most popular was different kind of wind controllers.

Yamaha, the WX7 - There's a pressure sensor at the mouthpiece, so you're not actually playing the

notes, but the velocity sensitive trigger and pressed keys with the correct fingering of a clarinet or

saxophone form MIDI note data.

At the same time, Casio developed an instrument, similar to the Yamaha - it has pressure

information coming in from the breath and then it has pitch information, note information coming in

from the keys; but it also had a little speaker on board, so you actually can hear the sounds, like a

quasi-saxophone. But there is a MIDI connector out on this, which allowed you to go into the

computer, and it would function just as easily as these would to, to be triggering the MIDI data.

Other controllers

Now there are other kinds of companies that have developed media controller information that are

no modelled on traditional Western musical instruments. Obviously trying to get the Western

musician to be able to operate and to be able to control the sounds is one part of the market. That,

that when MIDI came out, they wanted to capture as many musicians who were trained on

instruments, to be able to use that training into triggering electronic instruments. But, there were

other experimental devices as well.

There's a company called Infusion Systems that has made a number of different kinds of sensors that

the body can trigger that send MIDI data e.g. I-Cube Touch sensor - Just a little pad sending velocity

information or continuous controller information and pitch. Then I can assign this to whatever mini

data structure I want.



MIDI part 2

MIDI messages

Channel Voice Messages

Channel voice messages are used to send musical performance information. The messages in this

category are Note On, Note Off, Polyphonic Key Pressure, Channel Pressure, Pitch Bend Change,

Program Change and the Control Change messages.

Note On/ Note Off/ Velocity

In MIDI systems, the activation and the release of a note are considered two separate events.

Note On: When I press a note and send the velocity through the instrument, the note is turned on.

Note Off: When I release and stop blowing, the instrument sends another MIDI note, the same pitch

number but with a zero velocity.

After Touch

Some MIDI keyboard instruments have the ability to sense how much pressure is being applied to

the keys, while they are depressed. Some also have the ability to, only when a note is sustained add

certain effects to it e.g. vibrato

Pitch Bend

The Pitch Bend message is unusually sent from a keyboard instrument, in response to changes in the

pitch bend wheel. Pitch Bend information if used to modify the pitch of sounds played on a

particular channel.

Program Change

The Program Change or Patch change command is used to change the sound of MIDI synthesizer box

or device. Each individual sound is loaded in and assigned to a particular program number.

Command therefor sends a program changed sound number through MIDI device, that would shift

that in that particular those

Bank Select

Controller number zero is defined as the bank select. The bank select function is used in some

synthesizers, on conjunction with the MIDI Program Change message, to expand the number of

different instrument sounds.



Synthesizer and MIDI Terminology

Polyphony

The polyphony of a sound generator refers to its ability to play more than one note at a time.

Polyphony is generally measured or specified as a number of notes or voices.

Sounds

The different sounds that a synthesizer or sound generator can produce are sometimes called

“patches” or “programs”.

Multi-timbral Mode

A synthesizer or sound generator is said to be multi-timbral if it is capable of producing two or more

different instrument sounds simultaneously. With enough notes of polyphony, and “parts” (multi-

timbral) a single synthesizer can produce the sound of an entire orchestra.

General MIDI (GM) system

The general MIDI specification includes the definition of a general MIDI sound set (a patch map), a

general MIDI percussion map (mapping of percussion sounds to note numbers), and a set of general

MIDI performance capabilities (number of voices, types of MIDI controls recognized etc.). A MIDI

sequence which has been generated for use on a general MIDI instrument, should play correctly on

any General MIDI synthesizer or sound generator.

MIDI hardware controllers

Keyboards

Wind controllers

Drums

Strings

Body sensors

Environmental sensors

Video and touch sensors

Mobile technologies

Software converters

Alternatives



Digital and Audio effects

Many times when we talk about audio effects we'll be talking about them in both a physical mixture

and the mixtures in the software.

Auxiliary sends / Effects sends

Wet / Dry mix

When we looked at The Mackey analogue mixer earlier, we saw that there were auxiliary sends

which allows us to send the input coming into that channel to an output channel. We have the same

thing built into most of your multi-track recorders and software programs. We will have auxiliary

send where we would; we would do an insert effect.

We would send it to a particular software or plugin which would have that particular effect

processing, which would then be routed back into, to the mix. So when we talk about

At this point we have two different signals: wet and dry. The dry signal is the non-processed sound

or the original sound recording. And the wet sound would be the processed sound.

When routing the sound back into your mix, you can then choose if you only want the wet sound, or

a mixture of the wet and dry sounds. This mixture is very common as it creates a more natural

sound.

Insert effect

An effect that is inserted into the signal path of an audio channel, thereby affecting the entire signal.

Send effect – return

These differ from insert effects in that they are not inserted directly into a channel but rather exist

as a “stand alone” unit.

Post-fader / pre-fader

Controlling how much processed sound is going into the mix, and at the same time, how much of the

mix is going into the effects processer.

Digital and audio effects categories

1) Dynamic Range Processing

Dynamic range is the difference between the quietest and loudest part of the signal.

It is usually used as an insert effect.

Automatically altering the amplitude of audio in such a way that its dynamic range is changed.

The dynamic range can be compressed or expanded.

Compression – narrowing the dynamic range by reducing its amplitude after it reaches a threshold

value. Often less than 3:1 ratio.

Limiter – a compressor with a very high ratio (greater than 10:1)



Expander – input is boosted faster than normal

Noise gate – low amplitude is mapped straight to zero, removing noise bellow a threshold

Sidechain trigger - triggers the effect

Ducking - reduce the level of one audio signal when another is present. Ducking to reduce music

under voice: 1) insert compressor (insert effect) onto the music track. 2) Create a send on the voice

track that sends part of the track to a Bus. This bus is then chosen in the music track’s compressor as

the sidechain trigger.

Gateing – commonly used for drums

2) Filtering

LPF - low pass filter

HPF – high pass filterBPF - band pass filter

Notch – band reject filter

Wah-wah – BPF with time varying centre frequencies

3) Equalization

EQ – typically consists of several filters of varying types; under one interface, with knobs or sliders to

control the parameters of the filters.

Peaking filters – similar to BPF, except it only affects a very narrow range of frequencies.

Shelving filters – amplifies or attenuates frequencies, above or below a cut-off frequency. (Simple

treble/bass controls on a home or car stereo)

Graphic EQ – splits the spectrum into a number of discreet bands with individual sliders.

Parametric EQ – uses one or more filters (LPH, HPF, shelving, notch or peak). Divides the spectrum

into 1-8 bands, but contains a variable centre frequency for precise control.

4) Time based effects

A memory space that stores incoming audio for some period of time and then combines it with the

original dry signal.

Stereo delay – left and right channels have different delay times.

Echo – a delay of more than 50ms

Chorus – a delay of 10 – 50ms to obtain a richer or fatter sound: modulate delay with LFO and mix

with dry signal.

Flanging – a delay of less than 10ms, modulated with a LFO, resulting in shifting peaks/notches as

the spacing between them periodically rises and lowers.



Phasing – similar to flanging in producing shifting notches in the spectrum, but uses several all-pass

filters. APFs pass all the frequencies but change the phases of the partials, resulting in notches in the

spectrum when combined with the dry signal and modulated with the LFO.

Reverberation – best used as a send effect

Has three parts:

1) Direct sound

2) Early echo’s – source of sound bouncing off walls and ceiling / high frequency energy is

absorbed.

3) Dense echo’s - later multiple bounces of the source sound off various surfaces – quieter and

darker that early echo’s.

Reverberation plug-ins

Two types:

1) Artificial reverb – uses filters and delays to mimic reflected sounds

2) Convolution (or sampling) reverb – utilizes impulse response recordings from actual spaces /

an impulse response recording was made by making a sharp sound (an impulse) from a stage

of a performance hall and then recording the sound. The mathematical technique of

convolution then results in a sound with the reverb of the hall where the sound was

recorded.

5) Other

Tremolo – “human rate” amplitude modulation

Distortion – nonlinear effect based on wave shaping functions

Time stretching and pitch changing

Sample granulation – time-domain buffer effects for breaking down signals and reconstructing them

in new combinations.

Spatialization – auditory spatial cueing

Sound morphing – extracting parameters of an analysis-resynthesize model

introduction to digital sound design: week two study notes

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