studio a' manual

7/22/2019 Studio A' Manual

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Studio A ManualBy Callum Trotter


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Index:

Building diagram Health and safety Control room A diagram Control room equipment list and functions Different types of connections used:

Balanced cables Unbalanced cables Computer connectors Bantam cables (Patch leads)

Explanation of the terms: Audio Interface Signal Levels Direct Injection Patch Bay VU Meter Noise Gate Parametric EQ Shelf EQ Outboard Compressor

Diagram/explanation of a channel strip Microphones available How to route signals from the live room to the desk then to the computer

via an external compressor?

Different drum microphone set ups


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Diagram of building


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Health and safety

No food or drink in control rooms and live rooms/studios.

Be careful of wires running around the control room and live rooms.

No running.

All the equipment should be tested before use. To make sure it is

safe to use.

No smoking.

No blocking fire exists with items or people. There are 3 fire exist.Nowhere the exists are.

No we're the fire extinguishers are; there is one in the meeting

room.

Red emergency lights on the ceiling flash to show that there is a

fire in control room and live rooms.


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Control room A diagram


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Control room equipment list and functions

Soundesk:Soundcraft Sapphyre

Soundcards:2x Motu 240mk3

Motu 1224

Motu Mixer 7s3x Proteus/1 Emu Systems

Motu Midi Time Piece 2

Outboard equipment (racks):TL Audio IVORY-SERIES EQ-5013 (valve equaliser)

TL Audio INDIGO-SERIESJoemeek Photo Optical Stereo Compressor SC2.2

Frocusrite Platinum TwinTrak ProMasa Preamp Rectifier

Samson Powerstrip PS11Yamaha SPX90

Roland DEP-5

Sony Delay Unit DPS-D7

MIDI keyboard:Evolution MK-3612

Mac Pro:Logic Pro

Logic Pro XMelodyne


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Different types of connections used

Balanced cablesBalanced cables consist of three conductors. One is the positive, one for thenegative and one for the ground. Three conductors ensure maximum protectionagainst hum and interference. Balanced cables work by combining two signalsthat are out of phase from each other. When two identical signals of identical

volume are combined and one is 180 degrees out of phase from the other oneyou have complete cancellation of that audio. However if one of those signals isa different volume, you dont get complete cancellation. This is the principle

that makes a balanced audio path work. You can get both balanced XLR and jack

cables. Balanced jack cables will have two black rings around the end of theconnector. We mainly use balanced cables in the studio as they reduce

interference.

Unbalanced cablesUnbalanced cables consist of two conductors. One is audio signal and the otherone is the ground. The only problem with unbalanced cables is that they can pickup interference from radio frequencies or electromagnetic fields causing noise

and buzzing. Unbalanced jack cables have only one black ring around the end ofthe connector.


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Computer connectorsUSB cables are used for connecting the MIDI keyboard to the Mac. The

interface is USB 2.0. The USB cable interface is from male pin to a female pin

for the MIDI keyboard. USB stands for Universal Serial Bus. The maincomputer connectors we use are FireWire 400. The reason why FireWire 400

has a much faster data transfer than USB 2.0. When transferring big audio

files between the Mac and soundcards this is needed.

Bantam cables (Patch leads)Bantam cables or patch leads are used for the patch bay only and they are muchshorter than most cables used in the studio. The patch bay is connected to the

mixing desk and then to the outboard racks and soundcards. The mixing consolehas its own patch bay for inputting and outputting signals. For the racks andsoundcards to receive these signals there is an outside patch bay for inputs and

outputs. The bantam cables and patch bay is the interface that allows you tosend signals around the studio equipment. The cable is like a jack lead in shapebut much thinner.


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Explanation of the terms

Audio InterfaceAn audio interface is a piece of hardware that allows you to record and playbackaudio from a computer. As computers became more used outside of the officeenvironment in the early 90s, the hardware inside the machines had to beupgraded for the task that needed more power. Audio interfaces or soundcards

had to evolve from basic circuit boards that could only handle a beep to multi-channel recording and playback. Internal soundcards only featured an output forspeakers/ headphones and an input for microphones. Most soundcards featured

a main problem known as latency. Latency is whereby if you are recording

something, there is a noticeable delay between the actual sound and its playbackthrough your speakers or headphones. This couldnt be used for recording in a

more professional way. The architecture of the soundcards became morepowerful, more inputs and outputs where added and a more powerful power

supply was needed. This wasnt going to fit into a computer so external

soundcards were introduced. The latency problem was quickly addressed for theprofessional environment. Most soundcards are connected to a computer/Mac

via a FireWire or a USB cable.

Signal LevelsSignal levels or Line levels is the strength of an audio signal used to transmit

analogue and digital sound between audio hardware and software. The hardwareconsists of speakers, games consoles, radios, CD players, DVD players, TVs,

computers, Macs, soundcards, audio amplifiers, professional outboard equipment

and mixing consoles. The software consists of DAWs (Digital AudioWorkstation) software such as Pro Tools, Logic Pro, Ableton Live, FruityLoops

Studio, Reason, Cubase and Audacity. All this software can record, edit and

playback digital audio.

Direct InjectionA Direct Injection or more commonly used phrase DI, is a device that is used in

live and recording environments to connect line level (Jack cable) and XLR

cables. DI boxes are usually used to connect various instruments andmicrophones from electric guitars, basses, keyboards, condenser and dynamic

microphones to a mixing console through one main cable. These are used for longdistances from the instruments and microphones to the mixing console.


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Patch BayA patch bay or patch panel is a device that allows you to connect differenttypes of devices together like microphones, electric guitars, keyboards and

outboard equipment such as EQ, compression and reverb racks. Patch bays are

widely used in recording and live sound. In a recording studio there is manyoutboard equipment. Patch bays make it easy to connect different devices in

different orders for different projects. All of the changes can be made at the

patch bay.

VU MeterA VU (Volume Unit) Meter is a device that displays a representation of a signal

level in audio equipment. You will find VU Meters on most audio equipment

ranging from mixing consoles, amps and outboard racks. This gives a clearindication of how strong the signal is. Some outboard compression racks have

VU Meters to show how the compression is effecting the audio.

Noise GateA noise gate is a piece of hardware or software that is used to control the

volume of an audio signal. You set the threshold of when you want the gate to

open at. If you have a snare drum reaching volume of -6 dB but can also hear

other sounds at lower volume like the overheads then putting the threshold at -6 dB will only allow that volume of sound to come through the noise gate. Putting

it down to -10 dB threshold the overheads can still be heard. Increasing it up to-4 dB the snare sound wont be heard at all. The signal is not strong enough to

pass through the noise gate. Noise gates are very important in the studio and inlive sound as this can single out audio signals to be heard more clearly.

Parametric EQ

A Parametric EQ is a more complex type of equalizer that allows you to controlmore parameters of the sound than other equalizers. It allows you to control

the level (boost or cut), the frequency and the bandwidth. The level of each

frequency band can be controlled, the centre frequency can be shifted and the

bandwidth can be widened and narrowed. Parametric equalizers are capable ofmaking much more precise alterations to sound than other equalizers.

Parametric equalizers are used mainly in studio recording and live sound.


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Shelf EQ

Shelf EQs are equalizers designed for low and high frequencies to be enhancedby boost or reducing volume levels of those frequencies. Using Shelf EQs forenhancing high frequencies like cymbals, adding a few dBs at 10 KHz. You can

edit the level, frequency and bandwidth.

Outboard CompressorAn Outboard compressor is a compressor that is hardware form rather than

software based. Outboard compressors are used a lot in studio recording andlive sound, but mainly for studio recording. Outboard compressor do exactly the

same thing as a software compressor, so reducing the higher frequencies and

increasing the lower frequencies but with a much warmer, richer sound. What

people dont realise is that sound carries a lot a data and even in todays worldof computers being very powerful software compressors are designed to cut

down the audio data by a certain percentage to simply reduce the processingpower for the computers processers. This means that you are not getting all the

audio data that has been compressed. Outboard/rack compressors dont do this

as they only have one process to contend with. Outboard compressors are

connected to the sound cards and computer though patch leads to patch bay.


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Gain control:

This the main gain control formicrophone and line input.The top Knob is the gain

control for microphones andthe bottom knob is thecontrol for line inputs.

Line input:

This is the button you click ifyou are using a line input.

hantom power:Click this button to

ctivate power supply to

ondenser microphones.

Phase reverse:

This button reverses thephase of the selected input

to compensate for conflicting

microphone positioning.

100Hz fixed frequency:

This button is a fixed frequency

at 100Hz for the input.

SUB:

The sub button allows the

utput from one of eightmix busses to be switched

nto the channel path,llowing subgrouping.

Diagram/explanation of a channel stripChannel Path


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oise gate:he (GATE)Inbutton

ctivates the noise gate in

he signal path and this is

ormally placed in theonitor path. Pressing CH

waps the gate to the

hannel path.

ip:he Flip button clickedwitch swaps the source

or the sidechain to thether signal path. The

dechain is normally fed

rom the source feedinghe ate.

NGE:

his button increases the

ttenuation range of theate from 15 dB to 60 dB.

THR:

This button is the

THReshold control thatallows the opening of the

gate to be varied from -40

dB to +10 dB.

IN(FIL):This button switchesplaces a wide bandpass

filter in the sidechain.

Having a centre frequencyvariable from 70Hz to

4kHz by the FILknob.

REL:

The RELease knob control

varies the recovery timeafter gating from 0.1 to 4

seconds. If HOLDispressed the release time isfixed at 0.1 seconds and

RELthe HOLDtime for 2

seconds.

HUT:

he SHUT LEDuminates to show when

he gate is shut and not in

se.

Noise Gate


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There are six Auxiliary

Sends. They are arranged in

two sections. Aux 1 and 2 are

intended primarily for fold

back and are normally pre-

fade and in the monitor path.

Pressing CH sources them

from the channel and POST

makes the post-fade. Aux 1

is a stereo send and follows

the channel or monitor

panpot when on pre-fade.

Aux 3-6 are intended mainlyfor effects sends. They are

permanently in the monitor

path and in post fade.

Control knobs:

These determine how much you

want to send to the Auxiliary.

Auxiliary Sends


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HANNEL FADER:

his sets the level sent to the

ape send, sub group busses orix. The fader has 10 dB of gain

t the top and can be cut

together by the CUTbutton.

Recording Fader

REV(FDR):

This button switches the top

fader with the bottom fadergiving you more control over the

gain.

CH PAN:

This is patented active channelpan that determines the control

of the audio signal within the

stereo (L-R) mix and routes thesignal to any of the eight group

outputs via ROUTING

SWITCHES. Panning left feeds

the signal to odd numberedbusses and panning right to

even busses. The MIXbutton

routes the channel signal to thestereo mix bus so you can use

the channel path another input.

This fader is manly used forrecording.

XFX:

The XFXbutton moves the

source point for the channel

fader to post the monitor faderso that you can use the tape send

as additional sends when mixing.

BNCE:

The BNCEbutton when pressed

sends every channel that is in thebounce to two output channels

ready to bounce.

FL(AFL):

he PFL(AFL) button allows

irect monitoring of the audiognal at either the input of the

hannel fader (PFL) or the output

f the channel CUT switch (AFL).


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Monitor Fader The channel fader output normallyfeeds the Tape Send socket and

signal level, this is selectable to +4

dBu or to -10 dBV by pressing the -10button on the rear panel. Although

the Tape Send can be sourced from

the group outputs corresponding to

the module position by pressing the

BUS-n button.

he monitor path is normally fed

rom the Tape Return input, to makese of the automatic input switching

n the tape machine. A trim control

ith a range of +/-10 dB is provided

n the lower section of the input

ain control and this is determined

t the nominal centre 0 dB point.Send(TAPE MON):

This button switches the feeds fromthe monitor path of that modules

Tape Send, enabling you to hear the

signal being sent to the tape. The

PANcontrol positions the signal

across the stereo bus at a level

determined by the long-throwmonitor FADER. This fader has +10

dB gain at the top of its travel and is

cut by the adjacent illuminated CUTbutton.

OLO:he monitor SOLO system is

ormally Solo-in-Place (SIP).ressing the solo button mutes all

he other monitor paths. Pressing

he SOLO button on a BNCEdhannel gives AFL/PFL instead,

epending on the PFL/AFL status on

he master module.

PEAKLED:

This is a multi-point peak detector

that illuminates when less than 6 dBof headroom remains at three critical

places in the signal chain.

CH ACTIVELED:

This monitors the output of the inputpre-amp and illuminates if the level

exceeds -20 dBu.

These are four mute buttons that

allow the monitor path to be

controlled by the master mute busses

from the master module.


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Microphones available

Shure SM57

Shure SM 58

2x Audix F10

Audix F15

3x Neuman KM 184


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How to route signals from the live room to the

desk then to the computer via an external

compressor?

First you need to record you need to record your audio. Once this is

done we are going to create a new channel in Logic and click the R to

record enable it. Your recorded audio is on channel 1 in the stereo

mix on the Monitor faders (bottom faders). We are going to send

this signal to a group output. Group output 2. We click group output

1-2 and then pan it to the right side. After this we click bounce.

Now when bounce is enabled to audio is being sent to the group

output, so dont panic that there is no audio coming out. We now

heard over to the patch bay and find group output 2 and insert the

patch lead into the Joe Meek compressor in left in. We then grab

another patch lead and insert it in Joe Meek compressor left

output and then in to the top patch bay on the racks any free input.So input 10.

On logic the new channel you created needs to have input 10 and

output whatever channel is free on the sound desk, so channel 4.

Loop your audio, out the volume fader up for channel 4 and make

sure the compressor is on and you will hear your audio going through

the compressor.


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Different drum microphone set ups

This is the startand way in setting up microphones to record a drum kit. As you

can see in the photo the overheads are placed either side of the drum kit. They

both face the snare and have to be the same length away from the snare

otherwise you can phase. You measure this just using a cable. This gives a more

wider steroe recording.

This positioning is called

the X & Y. In the photo

you can see that the

microphones are

positioned in the middle

of the drum kit. One

microphone is faced to

one side and the other

microphone to the other

side. This is a very good

way in capturing a

stereo image of the kit

with no phasing

problems.

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