Guide to Using Studio 1

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.

Studio 1 Setup

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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 Drum-Pads, thomann 9 Fader 55mm, 8 Control Dial, 6 Transport-Buttons, Pitch- and Modulation-Wheel, 2 Octave-Switches, Arpeggiator-, Beat-Roll- 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/novation-impulse-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-guide-to-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