Guide

to
Using
Studio 1

Studio 1 Setup
I am about to do a block diagram of
Studio 1, number each component and
then write what they are, what they do &
why theyre there.

1)
2)

3)

2)

3)
5)

4)

6)

11)

7)

10)
8)

9)
12)

1) This is the Computer Screen. This is integral and needed so that

you can see what you are working on.
2) This set of speakers are the KRK Systems Rokit 5s. These
speakers have a frequency response of 45hz 35khz and are
needed to hear what is being played.

Image Credit:
http://medias.audiofanzine.com/images/normal/krk-rokit-5-g2-521259.jpg

3) These speakers are the Adam A77xs which are near-field monitor
speakers which means they should sit on or just behind the mixing desk,
within a couple of feet of the engineer. These speakers have a frequency
response of 38 Hz - 50 kHz.

Image Credit:
http://www.adam-audio.com/files/images/speakers/gallery/A77X_gallery.jpg

4) The AMT8 is an 8 Port MIDI Interface. This interface has seven

of its MIDI i/os on the back of the interface and one on the front.
The AMT8 is designed to work with USB as well as Macs & Pcs with
serial ports.
One of the main pros of using this interface is that it will improve
the tightness and timing of your MIDI playing/recording.

Image Credit:
http://www.homerecording.be/forum/attachments/f28/8077d1300880140-amt8.jpg

5) The Focusrite Saffire Pro 40 is an Audio Interface which

upgrades the sound output of your computer and also increases
the resolution and sound quality of your recordings going into your
computer. Audio Interfaces (Soundcards) are also needed to power non-passive
speakers. This interface is for inputs 1-8 in the live room.
The Saffire Pro 40 has 20 inputs & 20 outputs, 8 mic preamps,8 analog I/O, (2x
Mic/line/inst combo XLR, 6x mic/line/combo XLR) ADAT I/O, 2x S/PDIF I/O, 2
monitor outs + 2x inserts, monitor switch, 2 separate headphone buses, MIDI
thomann in/out, 2 FireWire connections, Saffire Pro 40 Control zero latency DSP
mixer/router. Quote taken from http://www.thomann.de/gb/focusrite_saffire_pro_40.htm

Photo credit:
http://g-ecx.images-amazon.com/images/G/01/musical-instruments/detail-page/
sc_b001mzqez2-03bckfrt_lg.jpg

6)Focusrite Octopre MKII - 8-channel mic preamp with A/D converter, 8x award winning
Focusrite preamps, 8x ADAT outputs thomann (24bit/96kHz via 2x lightpipes), -10dB pads, 5x
LED input metering and direct out on each channel, internal clocking and external clocking via
BNC word clock, 8x analogue inputs (2x mic/line/instrument combi XLR & 6x mic/line combi
XLR), 8x analogue outputs (8x 1/4" balanced thomann jacks), JetPLL jitter-elimination. Quote
taken from: http://www.thomann.de/gb/focusrite_octopre_mkii.htm
This is also an Audio Interface which
upgrades the sound output of your computer and also increases
the resolution and sound quality of your recordings going into your
computer. Audio Interfaces (Soundcards) are also needed to power non-passive speakers.
This interface is for inputs 11-18 in the live room.

Photo credit: http://www.mydukkan.com/media/Focusrite.FocusriteOctoPreMkII15966.jpg

7) The Samson S-Phone is a 4 channel headphone mixer/amp which

allows you to send the audio from the computer you are working on
to multiple artists in the live room. There is an LED meter to display
and easily allow you to control the overall stereo input.
Each of the S-Phones 4 channels has three headphone outputs,
These outputs have an overall volume control.

Photo credit:
http://www.samsontech.com/site_media/legacy_docs/S-phone.jpg

8) The SPL Monitor and talkback controller model 2381 allows the
engineer to speak to the artist in the live room wearing headphones with
its integrated microphone. The idea of this device is to give DAW users
the same type of monitor-control and source-switching features you'd get
on a large mixing console, also includes a 'musician' input allowing the
performer to monitor the source being recorded directly from the output of
a preamplifier, rather than from the computer's output, so that there's no
latency while overdubbing. Quote taken from: http://www.soundonsound.com/sos/
apr05/articles/aplmodel2381.htm

Photo Credit: http://spl.info/fileadmin/

user_upload/produkte/mtc_2381/
MTC_2381_MK2_1500_l.jpg

Photo Credit: http://spl.info/fileadmin/

user_upload/produkte/mtc_2381/
mtc2381_rear1500.jpg

9) The Focusrite ISA One is a single channel mic preamp built into
a sturdy metal casing which is also highly portable. the preamp
has both balanced XLR and balanced jack inputs to accept mic or
line-level signals. The DI channel has its own gain control, a switch
to match impedance for active or passive guitar pickups
(470k(omega) or 2.4M(omega)), an unbalanced input jack and an
unbalanced 'link' jack output, which can be used to route the DI
signal to a guitar amplifier. The DI also has a separate balanced
XLR output on the back panel. Quote taken from: http://www.soundonsound.com/
sos/oct08/articles/focusriteisaone.htm

Photo credit: http://

d3se566zfvnmhf.cloudfront.net/sites/default/
files/styles/cta_scale_640/public/Image-2.png

10) The Novation Impulse 49 is a USB MIDI controller keyboard

which is used to send MIDI data to a computer.
Novation Impulse 49, USB MIDI-Keyboard with 49 Keys semi
weighted with Aftertouch, DAW- and Plug-in-Control, 8 backlit DrumPads, thomann 9 Fader 55mm, 8 Control Dial, 6 Transport-Buttons,
Pitch- and Modulation-Wheel, 2 Octave-Switches, Arpeggiator-, BeatRoll- and Clip-Launch-Buttons, LCD-Display, USB-Connection,
Expression- and Sustain-Pedal-Connection, MIDI I/O.
Quote taken from: http://www.thomann.de/gb/novation_impulse_49.htm

Photo Credit:
http://
medias.audiofanzine.com/
images/normal/novationimpulse-49-435458.jpg

11) Emagic Logic v5 & Logic Control MIDI + Audio Sequencer Hardware Control Surface. This piece of hardware is for people who
want to streamline their workflow on Logic and also a nice alternative
to using Logics on screen mixer. It has support for up to 8 channels
and with the attachment, another 8 channels.

Photo Credit:
http://
media.soundonsound.com/
sos/apr02/images/
emagiclogic1.gif

12) The Apple Power Macintosh G5 Tower is the primary computer

Used in Studio 1. This is a crucial part of the set up as you would
not be able to do anything without this.

Photo Credit:
http://upload.wikimedia.org/wikipedia/commons/c/c6/
Power_Mac_G5_hero_left.jpg

Photo Credit:
http://upload.wikimedia.org/wikipedia/commons/
7/7e/Power_Mac_G5_back_upright.jpg

Chamber
of
Reection

The track I chose to cover for Multi-track recording purposes was

Mac DeMarcos - Chamber of Reflection. I started my process by
firstly opening up Logic 9 on the G5 Power Mac in Studio 1 and making
all of the Drum tracks and choosing the correct inputs they would be
sent to in the Live Room so that I would be prepared and not have to
do this at a later time.
Kick Drum microphone - Input 1
Top of Snare microphone Input 2
Underneath the Snare microphone Input 3
Rack Tom microphone Input 4
Floor Tom microphone Input 5
Left Overhead microphone Input 6
Right Overhead microphone Input 7

Once I had the drum template ready to record, I downloaded the song
with a YouTube converter and added it to Logic on its own individual
track (I deleted this at the end when it wasnt necessary to have in my
mix anymore). There were a few reasons I did this, one was so that I had
a reference/comparison point if I lined everything up on Logic as I chose
to replicate the original drumming pattern which is pretty simple. Another
reason I temporarily added the original was so that the drummer who
drummed for me on my recording could put on a pair of headphones,
with the jack plugged into the Stage Box in the live room so he could
hear the original and practice his timing to the song and just practice
getting the drums correct in general while he was playing along.
This is only doable due to the Samson S-Phone Headphone amp and the
SPL Monitor and talkback controller model 2381 which I spoke about
earlier.

The drum microphones I used were:

Kick Drum Audix F6
The f6, which is characterized with a hypercardioid pickup pattern for isolation and feedback
control, is equipped with a LM Type A (Low Mass) diaphragm for natural, accurate sound
reproduction. frequency response of 40 Hz - 16 kHz and the ability to handle sound pressure
levels of 140 dB Quote from the Audix website - http://www.audixusa.com/docs_12/units/
FP7.shtml
Photo Credit: http://www.audixusa.com/docs_12/units/f6.shtml

Photo Credit: http://recordinghacks.com/images/graphs/audix/f6-freq.png

Top Snare Shure SM57

The SM57 is a dynamic microphone with a cardioid polar pattern which means that it hears
sounds straight ahead of it and isolates ambient sounds from behind which cause interference
and feedback. Instead of the SM57 I could have used the Audix f5 which has a hyper cardioid
polar pattern which is similar to the cardioid polar pattern but with a narrower pickup than
cardioid polar patterns and also has a greater rejection of ambient sound.
The frequency response in the SM57 is from 40 Hz - 15 kHz.

Here is the Frequency Response Chart -

Bottom Snare Shure SM58

The Shure SM58 is a widely known and used microphone with a Cardioid polar pattern which
means that it hears sounds straight ahead of it and isolates ambient sounds from behind which
cause interference and feedback. Again, instead of the SM58 I could have used an Audix f5 which
has a hyper cardioid polar pattern which is similar to the cardioid polar pattern but with a
narrower pickup than cardioid polar patterns and also has a greater rejection of ambient sound.
The frequency response is from 50 Hz - 15 kHz

Here is the frequency response chart

Tom microphones I used Audix F2

The Audix F2 is a microphone with an Hypercardioid polar pattern which is similar to the
cardioid polar pattern but with a narrower pickup than cardioid polar patterns and also has a
greater rejection of ambient sound. I used these microphones due to their frequency
response which is from 52 Hz - 15 kHz.

Frequency Response Chart

Overhead Microphones Audix F9s

The Audix F9s are the microphones I used for both of the Overheads I set up to record with.
These microphones are pencil condenser microphones. Condenser microphones typically
produce a better quality of audio recording. Condenser microphones unlike dynamic microphones
need phantom power (which is 48Volts) to record with which gets sent through the audio
interface.
The F9s have a cardioid polar pattern and a frequency response of 40 Hz - 20 kHz.

Frequency response chart

Setting up the Drums

I chose to use a Spaced Pair set-up of the microphones for my recording of the drums.
I decided to use this specialized drum microphone set up because I felt that I would get the
fullest stereo sound from the drums and cymbals that were being played in my recording.
What I could have done would have been to experiment with different specialized drum miking
techniques such as the X-Y technique or something like the Glyn Johns technique although I
dont feel like Glyn Johns technique would have been to my liking for this recording as there are
no Toms being played in my recording of Chamber of Reflection.

Here is some information about this particular stereo microphone set up from:
http://www.shure.co.uk/support_download/educational_content/microphones-basics/
stereo_microphone_techniques
One of the most popular specialized microphone techniques is stereo miking. This
use of two or more microphones to create a stereo image will often give depth and
spatial placement to an instrument or overall recording. There are a number of
different methods for stereo. Three of the most popular are the spaced pair (A/B), the
coincident or near-coincident pair (X-Y configuration), and the Mid-Side (M-S)
technique.
(Continued on next slide)

Spaced Pair Continued..

The Spaced Pair (A/B) Technique
The spaced pair (A/B) technique uses two cardioid or omni directional microphones spaced
3 - 10 feet apart from each other panned in left/right configuration to capture the stereo
image of an ensemble or instrument. Effective stereo separation is very wide. The distance
between the two microphones is dependent on the physical size of the sound source.
For instance, if two mics are placed ten feet apart to record an acoustic guitar; the guitar will
appear in the center of the stereo image. This is probably too much spacing for such a small
sound source. A closer, narrower mic placement should be used in this situation. The
drawback to A/B stereo is the potential for undesirable phase cancellation of the signals
from the microphones.
Due to the relatively large distance between the microphones and the resulting difference of
sound arrival times at the microphones, phase cancellations and summing may be
occurring. A mono reference source can be used to check for phase problems. When the
program is switched to mono and frequencies jump out or fall out of the sound, you can
assume that there is phase problem. This may be a serious problem if your recording is going
to be heard in mono as is typical in broadcast or soundtrack playback.

1) Setting up the drum microphones I started with the Kick drum

microphone was an Audix f6 which I placed just inside the drum angled
towards the middle of the skin to capture the sound of the beater against
the skin more efficiently which I thought could potentially sound
interesting for my recording. I could have experimented with microphone
placements and angled the microphone where the skin meets the metal
to get different tones. The Kick microphone was sent to Input 1.

2) The microphone I placed for the top of the snare was a Shure SM57
dynamic microphone. I placed this microphone looking over the top left
of the snare drum but facing towards the center on a stand a few inches
off the skin so that the drummer wouldnt hit the microphone. I used an
Aquarian 14 Studio Ring on the snare to help prevent resonance from
the skin of the drum. I could have also used a microphone rim clip which
clips onto the rim of the drum and you slip the microphone into the
holder. This saves you having to set up microphone stands and saves a
lot of space. The top of the snare microphone was sent to input 2.

3) The microphone I placed underneath the snare to record with was the
Shure SM58 dynamic microphone. I placed the microphone underneath
the snare drum and angled it upwards towards the snare wire so I really
captured the rattle as the skin gets hit. I used an Aquarian 14 Studio
Ring on the snare to help prevent resonance from the skin of the drum.
What I could have done was angled the microphone differently to get
different sounds and less immediate pick up of the rattle of the snare
wire. The bottom of the snare microphone was sent to input 3.

4) Although the song I recorded had no Tom action, I still set up the full arrangement of
microphones to pick up the sounds of the other drums being played to get a fuller sound. For
the Rack Tom and Floor Tom I used two Audix f2 microphones set up on microphone stands
slightly over the side of the drums and facing towards the center of the drum skin to pick up
as much sound as possible. The microphones were placed a few inches above the drum so
that they wouldnt interfere with the drummers drumming abilities. I also used Aquarian
Studio Rings on both Toms (10 for the Rack Tom & 16 for the Floor Tom) which sit just on
the skin to avoid unnecessary resonance from the drum skins. What I could have done
differently would be to use rim clips for the Tom microphones which clip directly to the rim of
the drum. I also could have angled the microphone to different parts of the skin to pick up
different tones of the skin. These drums were sent to inputs 4 (Rack Tom) and 5 (Floor Tom).

5) The microphones I used for both of the overhead recordings were Audix f9s which are
pencil condenser microphones. I set these up with two individual microphone stands in a
Spaced Pair drum microphone set up. I set the microphones up to both face the snare drum
to capture the sound in stereo. The way I made sure that there would be no lag from each
microphone was to measure they were both identical distances from the snare drum itself. I
did this by using a spare XLR cable to measure from the center of the snare to the head of
the f9. I could/should have used a measuring tape instead of an XLR cable just to reduce
that tiny risk of wear and tear. I also could have set up the overhead microphones in an
altogether different specialized miking technique like the X-Y or Glyn Johns techniques.

Once I had the drums recorded on Logic, on the bottom snare I went to
the inserts bar and chose the Gain setting, I then clicked the little box
that says Phase Invert. I did this because the top snare mic and the
bottom snare mic are facing each other, this can cause issues with
phasing. When you invert the phase on one of these recordings you can
really hear the difference. I selected all of the drum tracks and sent
them to their own bus which I suitably named Drums.

I then proceeded to use the Compression insert on the Drum bus. I had
the bus level at 0dB for all the drums. Something I could have
experimented with to get different compression on individual drums.

I chose to have the threshold at -25.0dB so that I would get softer drums as on my track I
didnt want the drums to overwhelm the rest of the mix.
I chose to have a low ratio so that the compression wasnt completely flattening my audio.
The knee was adjusted next to be set to a harder knee so that you hear the compression kick
in faster over a softer knee which lets the compression kind of sneak up on the listener.
I raised the attack of the compression so that it would instantly compress the drums and
make them all around softer. I turned the gain down slightly as the drum signal was still very
strong.

I also added some Tape Delay to the drum bus. I noticed that the original song has
some delay on the drums and where the hi-hats are played consistently throughout
the song it gives them a trippy sound, as well as the whole drum kit sounding a little
more psychedelic which is what I was aiming for. I scrolled down the Logic tape delay
pre-sets and found one which was close to producing the delay which I wanted from
the drums. I tweaked the Feedback, Wet & Dry on the tape delay to further adjust the
sound until I reached the level of delay which I desired and is blatant in the mix.

Pre-recording the Bass Guitar

Next thing I did was to DI the bass guitar. I used a Jack to Jack cable to plug the bass guitar into
the audio interface in the iMac room. I could have recorded in Studio 1 using the ISA One single
channel pre-amp to get a better signal on my recording. I made a new audio track on logic and
clicked the record ready button to make sure the signal was coming through okay. I had a look
online for the original bass tab as I liked the original bass line. I found a YouTube video which
had the bass tab in the description..
After tuning the guitar using Logics built in Tuner insert and playing along
with the track a few times to get used to it we noticed the tab was wrong on
a few parts and quickly corrected it to what we thought it should sound like.
The bass player played this riff once and I looped it through-out the whole
song as it does not deviate. I could have recorded a few takes and chose
the one that was to my liking for how I wanted the recording to sound.

Post recording the Bass Guitar

Once I had the bass recorded I added compression on the inserts bar, I used
compression as the first note that the bass player hits gives off a strong
signal and using compression I brought down that dynamic peak to coincide
with the rest of the signal. Because I looped the bass riff I could have used
Automation to bring the initial notes played that I thought the signal was too
strong on down to a suitable level and then looped it to get the Automation
all the way through the song.

This is the compression I chose to use on the Bass Guitar.

I added some reverb to the Bass Guitar using the Logic Insert AVerb.

I only applied a little reverb on the bass guitar because I wanted it to ring out
a little but not overwhelm the rest of the mix. I slightly lowered the Predelay
from the default of 20ms so that the reverb would kick in slightly quicker.
I had quite a small Room Size so that the Bass doesnt boom out but you
can still hear the reverberation quite well.

Pre-recording the Lead Guitar

To record the lead guitar for Chamber of Reflection I initially made a new audio track in Logic
and named it Lead Guitar. I used a Jack to Jack cable to DI the guitar into the ISA One single
channel pre-amp in Studio 1.
To get the guitar signal to come through to the computer to record I had to plug the pre-amp into
the audio interface using an XLR cable and change the input to Input 2.
We were lucky to find a tab for the song translated into guitar as the original song uses no lead
guitar, only synthesizer's.

We played through the riff a few times to practice as this doesnt change at all so Id only have
to get one perfect take and then I could re-use it throughout the song. With the lead guitar it
took a little longer than the bass to get recorded as I was really picky about the timing of the
hammer ons & pull offs but it was worth it as I was very pleased with the final sound.
What I could have done differently is used an amp and recorded that sound source instead of
recording with a clean tone directly into logic and then later using logics built in guitar amps to
change the sound.

Post Recording the Lead Guitar

Once I had the lead recorded I firstly copied it to the other parts of the track where it is played
and made sure the timings were right.
Next I added the compression, mainly for the hammer ons and pull offs where the signal came
through particularly strongly. I felt there wasnt too much compressing needed though as I still
wanted there to be a varied dynamic range on the lead. What I could have done here was made
the ration higher to really cut out the peaks on the highest notes.

I used an Amp Designer preset to get close to the tone I wanted which was lo-fi tremolo. I kept
the digital microphone in the center of the digital amplifier but a bit further back to pick up the
sound to the amplifier but a little more spread out than if I had the microphone closer.
I slightly raised the gain and pretty much kept the EQ centered but with a little treble boost to get
a higher frequency sound. I left the amplifiers built in Reverb off as I later added my own reverb
using the Space Designer and after some experimenting with them both on I found that these two
clashed. I chose the tremolo version of this amplifier so I kept that feature on and didnt switch it
to Vibrato as I didnt want the pitch to be changed, only the note to be reiterated. I kept the depth
& speed centered so the tremolo level was pretty relaxed and not reiterated too much for how I
wanted it to sound. I raised the presence of the amp slightly so you could hear it more in the mix.
What I could have done was experimented with logics other amp sounds more.

Next on the lead guitar I added the Space Designer reverb insert. I chose to use this reverb
insert as I thought I could get more control over the space in which the lead guitar would be
played out into. I had the pre-delay quite low in terms of milliseconds so that the reverb would
kick in sooner. I could have experimented with this to make the guitar reverb drone out for longer
after it was no longer being played. I found that the Wet & Dry in the reverb output worked well
when they were quite close to being centered. I found with too much wet on the space designer
clashed with the guitar amp and rang out sounding too harsh.

Chorus & Verse Synth

For the synths in this song I chose to almost replicate the original style but not hold the
notes for quite as long on each third chord change as I wanted to change up the synths a
little bit. I was able to get the chords played in straight off the bat from the lead guitar tab
which were:
Am
Bm
C
Am
Bm
Em
For the intro, chorus & verse synths I found the sound I thought was closest to the original
(which was Logics Blue Carpet) as I liked the pitch and sustain on the original
synthesizers. I could have experimented with synth sounds and added more notes in this
track to make a more complex song.
Initially the synth was played in with only two keys pressed down but I felt like it was
missing something so I added the third note to make it a triad which I much preferred the
sound of.

In the verse synth I added a bass note to the to the initial notes being played to bring the synth
and bass guitar closer together as there are no vocals in the mix I thought it would be more
pleasant to listen to.

I also cut the highest note in the E Minor chord as I felt the pitch in the highest note was
bringing the synth too much attention in the mix and I didnt want it at the foreground even
though there were no vocals during the verse.
The Intro & Chorus synth were together but the verse synth was on its own individual track.
Both of the synths produced the same sound though as the inserts I added were the same on
both.

The inserts I added to the synthesizers were the Amp Designer to change the original sound of
the Blue Carpet logic synth to something closer to the original synth in Chamber of Reflection
and Averb for some slight reverb on the synths to stop them coming through too harsh and to
fade out slower and also fade into each other smoother instead of just drop straight out.

As you can see, I had the Gain slightly lowered because the synths were playing more than one
note at a time it was coming through quite blurred and with the gain down even a little bit the
sound became a lot clearer. The Bass, Mids & Treble are all slightly lowered as I felt the synth
sound originally had boosted frequencies. I left the built in reverb off because I had already
added a little reverb with the AVerb insert and preferred that over the amps reverb. I had the
switch for Tremolo/Vibrato on and had it switched to Vibrato because I wanted the pulsating
change of pitch over the reiteration of the same note which would be tremolo.

I had the setting for the Depth quite deep so that the amount of pitch variation was quite a lot
but left the speed synced with the depth so that rate of the pitch variation wasnt too extreme
and unpleasant to listen to. I could have changed the vibrato to tremolo to get a reiterated note
which could have been interesting or just taken the vibrato away completely to get a more
droning note being played.
Lastly to the synth inserts I added a small amount of reverb with the help of Averb. I left the
pre-delay as default 20ms as I thought it came in just right as it was. I slightly raised the mix
percentage to add slightly more reverb to the synth mix. I did this because I wanted to try to get
the synthesizers played notes to blend into each other seamlessly. I could have added more
reverb to get a more droning, psychedelic overlay of synthesizers.

Just before each chorus and the outro for three chords at a time, I added an extra
layer of sound. I chose the Church Organ on Logic and played in Am Bm Em in time
with the verse Synth.

In the original song there is something similar to this before the chorus and outro
but a lot more noticeable and sharper. I chose to keep my replication a lot more
subtle because I wanted to create a more relaxed version of the song.
I used the velocity tool to make the MIDI messages come through extremely soft as
the church organ has naturally quite a strong output with even a soft velocity.
What I could have done was raised the velocity and made this synth more
noticeable in the mix.

I added Averb reverb to the Church Organ. I lowered the pre-delay by 2ms as the default is set to
20ms. I did this so that the reverb comes in a little sooner than default.
I raised the reflectivity of the reverb to 75% to get a bouncier sound from the church organ. I
could have raised this more to get more reflectivity on the sound which would make the reverb
more prominent.
The room size I set to 154 was because with this church organ I thought it sounded better with
more of a booming output with the reverb than a smaller room size which would make the reverb
sharper but less effective.
I raised the mix percentage to 65% to get more reverb into the mix. I could have raised this even
higher to get more reverb but would have risked the sound being overwhelming and being too
blatant in the mix which I didnt want.

I used a Tremolo insert on the church organ to continuously reiterate the chords which I played
in. I chose to do this to give the church organ a slight bit more of a presence in the mix even
though the overall gain of the church organ was quite low as your ears would pick up the
frequencies from the organ due to the panning.
I did use the Slow Panning preset on this Tremolo insert as I found the effect suitable for the
sound that I wanted in my mix.
I could have experimented with the tremolo settings to get an interesting panning effect and
potentially bring this organ more to the foreground whenever it comes in on the track.

Equalisation
In sound recording and reproduction, equalization is the process commonly used to alter
the frequency response of an audio system using linear filters. Most hi-fi equipment uses
relatively simple filters to make bass and treble adjustments. Graphic and parametric
equalizers have much more flexibility in tailoring the frequency content of an audio signal.
An equalizer is the circuit or equipment used to achieve equalization. Since equalizers,
"adjust the amplitude of audio signals at particular frequencies," they are, "in other words,
frequency-specific volume knobs.
Equalizers are used in recording studios, broadcast studios, and live sound reinforcement
to correct the response of microphones, instrument pick-ups, loudspeakers, and hall
acoustics. Equalization may also be used to eliminate unwanted sounds, make certain
instruments or voices more prominent, enhance particular aspects of an instrument's
tone, or combat feedback (howling) in a public address system. Equalizers are also used
in music production to adjust the timbre of individual instruments by adjusting their
frequency content and to fit individual instruments within the overall frequency spectrum
of the mix.
The most common equalizers in music production are parametric, semi-parametric,
graphic, peak, and program equalizers. Graphic equalizers are often included in
consumer audio equipment and software which plays music on home computers.
Parametric equalizers require more expertise than graphic equalizers, and they can
provide more specific compensation or alteration around a chosen frequency. This may be
used in order to remove (or to create) a resonance, for instance.

Quote from: http://en.wikipedia.org/wiki/Equalization_(audio)

When I equalised my mix I started with the Bass Guitar and Kick/Bass drum as I knew
they would both be at similar frequencies. I opened the channel EQ on both of these
channels and used the Analyzer to show me where their frequencies were coming in at.
I noticed that in my mix, the bass guitar was a lot more prominent than the Kick drum so
I decided to give the kick drum the precedent over the bass in terms of frequencies so on
the 50hz range I boosted the Kick drum by +6.0dB and attenuated the Bass by -6.0dB.
I feel like this made the kick drum a lot more noticeable in the mix and lowered the bass
just enough that they fit together.
What I could have done would be to do the opposite and have a really bass guitar heavy
mix which could have been sonically interesting and made a unique remake of this song.

Next with the EQ after listening to the individual drum tracks I noticed that there was
a resonance being emitted by the bottom snare recording at about 510Hz, so what I
did was to open the Channel EQ and try to pin-point where this frequency was by
creating a point and boosting the dB to find the frequency. What I did once I thought
Id found it was to then attenuate the dB and cut that frequency out of the mix
altogether for that channel.

Next I checked where the lead guitar was coming in on the EQ, this seemed to be about the
500Hz mark so because I attenuated the 500Hz from the bottom of the snare I was able to
raise the lead guitars frequencies to make it more prominent in the mix which is what I
wanted. A very noticeable chorus riff.
What I could have done was added another guitar track to accompany this one and then have
to think carefully about which one to give the prominence to in the mix.

The last thing I did to the EQ was to analyze the Rack Tom & Top Snare frequencies together.
Although the rack Tom wasnt actually played throughout the song, the microphone placed over it
still picked up the other drums being played.
The Rack Tom & Top Snare microphones were both picking up the snare being played more than
anything else but I thought that the Snare sounded too sharp for what I wanted it to be in the mix.
What I did was to attenuate the snare frequency at about 200Hz by -6.0dB and lift the frequency
coming from the Rack Tom recording at 200Hz by +6.0dB and I found that this really softened
the snare as the Rack Tom microphone was distanced away enough for it to give less peaking
frequencies to the mix.
What I could have done would have been to heighten the snare frequencies to get a snappier,
sharper attack sound from them.

What is Audio Compression?

Dynamic range compression (DRC) or simply compression, reduces the
volume of loud sounds or amplifies quiet sounds by narrowing or
"compressing" an audio signal's dynamic range.
Audio compression reduces loud sounds which are above a certain
threshold while quiet sounds remain unaffected. The dedicated
electronic hardware unit or audio software used to apply compression is
called a compressor.
Quote from: http://en.wikipedia.org/wiki/Dynamic_range_compression

Compression is all about controlling the peaks and troughs (dynamics)

that occur in your mix when, for instance, the quieter vocal moments
are drowned out by the guitarist during recording and/or playback. In a
nutshell, it squashes the loudest peaks and boosts the quieter troughs,
meaning you can up the overall track volume to get that extra punch.
Quote from: http://www.thewhippinpost.co.uk/mixing-music/compression-audio-mixing.htm

Compression
Logic has multiple circuits for the built in Compression insert.
VCA: Uses a Voltage Controlled Amplifier. Known for their fast gainreduction
abilities, examples include SSL's famous bus compressor and the Dbx 160.
FET: Uses Field Effect Transistors. Compressors based on these designs have a
'valvey' sound, but are also capable of pretty fast response times. Examples include
the Universal Audio/UREI 1176.
Opto: Uses a lamp and photoresistor. By their nature, optical compressors react
quite slowly to transients, which can be a good thing in some cases! Examples
include Teletronix's LA2A and the Joe Meek/Ted Fletcher designs.
Platinum: This is Logic Pro's original compressor 'model' and it can still be useful in
some situations, as it has a fairly transparent quality.
ClassA_R & ClassA_U: Quite what these emulations are based on is anyone's guess,
but the names suggest variable 'mu' devices combined with Class-A amplification,
similar to devices from Manley Labs..

Quote from: http://www.soundonsound.com/sos/jun11/articles/

logic-tech-0611.htm

Compression
Threshold - how loud the signal has to be before compression is applied.
Ratio - how much compression is applied. For example, if the compression ratio is set for
6:1, the input signal will have to cross the threshold by 6 dB for the output level to
increase by 1dB.
Attack - how quickly the compressor starts to work.
Release - how soon after the signal dips below the threshold the compressor stops.
Knee - sets how the compressor reacts to signals once the threshold is passed. Hard
Knee settings mean it clamps the signal straight away, and Soft Knee means the
compression kicks in more gently as the signal goes further past the threshold.
Make-Up Gain - allows you to boost the compressed signal. as compression often
attenuates the signal significantly.
Output - allows you to boost or attenuate the level of the signal output from the
compressor.

Quote from: http://music.tutsplus.com/tutorials/the-beginners-guideto-compression--audio-953

Additional Notes & Thoughts

Photos on location taken with a Cannon 1000D
Continue Chamber of Reflection, add vocals, change bass line a little