Studio A Manual

By Callum Trotter



























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
Diagram of building
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.
Control room A diagram
Control room equipment list and functions

Soundesk:
Soundcraft Sapphyre



Soundcards:
2x Motu 240mk3
Motu 1224
Motu Mixer 7s
3x Proteus/1 Emu Systems
Motu Midi Time Piece 2



Outboard equipment (racks):
TL Audio IVORY-SERIES EQ-5013 (valve equaliser)
TL Audio INDIGO-SERIES
Joemeek Photo Optical Stereo Compressor SC2.2
Frocusrite Platinum TwinTrak Pro
Masa Preamp Rectifier
Samson Powerstrip PS11
Yamaha SPX90
Roland DEP-5
Sony Delay Unit DPS-D7


MIDI keyboard:
Evolution MK-3612

Mac Pro:
Logic Pro
Logic Pro X
Melodyne
Different types of connections used

Balanced cables
Balanced cables consist of three conductors. One is the positive, one for the
negative and one for the ground. Three conductors ensure maximum protection
against hum and interference. Balanced cables work by combining two signals
that 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 one
you have complete cancellation of that audio. However if one of those signals is
a 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 the
connector. We mainly use balanced cables in the studio as they reduce
interference.







Unbalanced cables
Unbalanced cables consist of two conductors. One is audio signal and the other
one is the ground. The only problem with unbalanced cables is that they can pick
up interference from radio frequencies or electromagnetic fields causing noise
and buzzing. Unbalanced jack cables have only one black ring around the end of
the connector.













Computer connectors
USB 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 main
computer 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 much
shorter 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 console
has its own patch bay for inputting and outputting signals. For the racks and
soundcards 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 to
send signals around the studio equipment. The cable is like a jack lead in shape
but much thinner.




Explanation of the terms

Audio Interface
An audio interface is a piece of hardware that allows you to record and playback
audio from a computer. As computers became more used outside of the office
environment in the early 90s, the hardware inside the machines had to be
upgraded 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 for
speakers/ 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 playback
through your speakers or headphones. This couldnt be used for recording in a
more professional way. The architecture of the soundcards became more
powerful, 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 the
professional environment. Most soundcards are connected to a computer/Mac
via a FireWire or a USB cable.

Signal Levels
Signal levels or Line levels is the strength of an audio signal used to transmit
analogue and digital sound between audio hardware and software. The hardware
consists 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 Audio
Workstation) 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 Injection
A 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 and
microphones from electric guitars, basses, keyboards, condenser and dynamic
microphones to a mixing console through one main cable. These are used for long
distances from the instruments and microphones to the mixing console.

Patch Bay
A patch bay or patch panel is a device that allows you to connect different
types 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 many
outboard 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 Meter
A 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 clear
indication of how strong the signal is. Some outboard compression racks have
VU Meters to show how the compression is effecting the audio.

Noise Gate
A 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 in
live 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 control
more 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 of
making much more precise alterations to sound than other equalizers.
Parametric equalizers are used mainly in studio recording and live sound.





Shelf EQ
Shelf EQs are equalizers designed for low and high frequencies to be enhanced
by boost or reducing volume levels of those frequencies. Using Shelf EQs for
enhancing high frequencies like cymbals, adding a few dBs at 10 KHz. You can
edit the level, frequency and bandwidth.






Outboard Compressor
An Outboard compressor is a compressor that is hardware form rather than
software based. Outboard compressors are used a lot in studio recording and
live 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 world
of computers being very powerful software compressors are designed to cut
down the audio data by a certain percentage to simply reduce the processing
power 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.
Gain control:
This the main gain control for
microphone and line input.
The top Knob is the gain
control for microphones and
the bottom knob is the
control for line inputs.
Line input:
This is the button you click if
you are using a line input.
Phantom power:
Click this button to
activate power supply to
condenser microphones.
Phase reverse:
This button reverses the
phase 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
output from one of eight
mix busses to be switched
into the channel path,
allowing subgrouping.
Diagram/explanation of a channel strip
Channel Path
Noise gate:
The (GATE)In button
activates the noise gate in
the signal path and this is
normally placed in the
monitor path. Pressing CH
swaps the gate to the
channel path.
Flip:
The Flip button clicked
switch swaps the source
for the sidechain to the
other signal path. The
sidechain is normally fed
from the source feeding
the gate.
RNGE:
This button increases the
attenuation range of the
gate from 15 dB to 60 dB.
THR:
This button is the
THReshold control that
allows the opening of the
gate to be varied from -40
dB to +10 dB.
IN (FIL):
This button switches
places a wide bandpass
filter in the sidechain.
Having a centre frequency
variable from 70Hz to
4kHz by the FIL knob.
REL:
The RELease knob control
varies the recovery time
after gating from 0.1 to 4
seconds. If HOLD is
pressed the release time is
fixed at 0.1 seconds and
REL the HOLD time for 2
seconds.
SHUT:
The SHUT LED
illuminates to show when
the gate is shut and not in
use.
Noise Gate
This is a patented 4-band swept mid design which is arranged as two
separate sections. Both sections are normally placed in the channel path,
but can be switched into the monitor path for mixdown by pressing the
MON button.
The HF is a second
order shelving design,
giving a 15 dB cut or
boost at a fixed
turnover frequency of
12kHz.
The LF section provides
a 15 dB cut or boost at
a fixed turnover
frequency 60Hz.
The INSERT point is located before this EQ section and moves
with it when the MON button is active. The send is unbalanced
and return is balanced.
The HMF and LMF section
have a peak boost and cut
response at centre
frequencies variable
between 600Hz-12kHz
and 100Hz-2kHz. This can
be altered by using the
right-hand (black) knobs.
The entire EQ section,
HF/LF and HMF/LMF,
may be switched IN and
OUT with the EQ IN
switch.
Equaliser
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 mainly
for 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
CHANNEL FADER:
This sets the level sent to the
tape send, sub group busses or
mix. The fader has 10 dB of gain
at the top and can be cut
altogether by the CUT button.
Recording Fader
REV (FDR):
This button switches the top
fader with the bottom fader
giving you more control over the
gain.
CH PAN:
This is patented active channel
pan that determines the control
of the audio signal within the
stereo (L-R) mix and routes the
signal to any of the eight group
outputs via ROUTING
SWITCHES. Panning left feeds
the signal to odd numbered
busses and panning right to
even busses. The MIX button
routes the channel signal to the
stereo mix bus so you can use
the channel path another input.
This fader is manly used for
recording.
XFX:
The XFX button moves the
source point for the channel
fader to post the monitor fader
so that you can use the tape send
as additional sends when mixing.
BNCE:
The BNCE button when pressed
sends every channel that is in the
bounce to two output channels
ready to bounce.
PFL (AFL):
The PFL (AFL) button allows
direct monitoring of the audio
signal at either the input of the
channel fader (PFL) or the output
of the channel CUT switch (AFL).
Monitor Fader
The channel fader output normally
feeds the Tape Send socket and
signal level, this is selectable to +4
dBu or to -10 dBV by pressing the -10
button 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.
The monitor path is normally fed
from the Tape Return input, to make
use of the automatic input switching
in the tape machine. A trim control
with a range of +/-10 dB is provided
on the lower section of the input
gain control and this is determined
at the nominal centre 0 dB point.
Send (TAPE MON):
This button switches the feeds from
the monitor path of that modules
Tape Send, enabling you to hear the
signal being sent to the tape. The
PAN control positions the signal
across the stereo bus at a level
determined by the long-throw
monitor FADER. This fader has +10
dB gain at the top of its travel and is
cut by the adjacent illuminated CUT
button.
SOLO:
The monitor SOLO system is
normally Solo-in-Place (SIP).
Pressing the solo button mutes all
the other monitor paths. Pressing
the SOLO button on a BNCEd
channel gives AFL/PFL instead,
depending on the PFL/AFL status on
the master module.
PEAK LED:
This is a multi-point peak detector
that illuminates when less than 6 dB
of headroom remains at three critical
places in the signal chain.
CH ACTIVE LED:
This monitors the output of the input
pre-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.
Microphones available
Shure SM57
Shure SM 58
2x Audix F10
Audix F15
3x Neuman KM 184
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.





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.