the Basics

fuquam

New member
New to recording and even newer to mixing. I understand the concept of mixing each instrument differently so they stand out on thier own. Problem is I have this 31 band EQ in front of me and not sure what to do for drums vs guitar vs vocals vs bass . . . . Can someone explain to me the basics of EQing each track. My mixes right now sound a little muttled together. I'm using nice equipment and mics. I just need a basic lesson on mixing. Any takers? Attached screen shots would be great. I know if differs from recording to recording and there are many factors that go into a mix.
 
Graphic eqs are more commonly used for PAs.

What's you set-up like? We need more details on your gear, I think.
 
Eq

I'm using a MOTU MK firewire unit and am currently using Garageband but also have Logic as software. The 31 band EQ is in the software. Guitar is mic'd with a Sure 57. Bass is direct in. Drums are mic'd with Nady drum mics (5 total). Vocals recorded with a condensor mic.
 
The "art" of EQ
by Aaron Trumm

EQ can be used in a variety of situations, from live sound to recording to tape to mixing down. Mainly, it should be used to enhance signals that have some problem. The golden rule of EQ is less is more. If something seems fine without it, I avoid EQing it at all. Then, if I do use it, I try to remain subtle. My personal golden rule is nearly never EQ signals going to tape (as in a multitracking situation). I always try to get the original sound on tape, then I can mess with it later. Putting EQ (or any other effect) on tape usually just leads to trouble. The other rule (the silver rule :) ) is cutting is almost always better than boosting, especially when fixing problems. For example if a guitar sounds too thin, first try cutting high frequencies and boosting the gain a bit, instead of boosting the lows. The more clutter you can remove from a mix, the better. A better example is I very often cut a bit of high away from hats. Another example is, many times you may not hear something well in a mix...You might try cutting some frequencies in a different track that seems to be interfering, rather than boosting in the track you want to bring out. With these basic rules in mind, I'll tell you my rules when I enter a mixdown session:
3. Rule Of Opposites: Usually, tracks with high sounds, (a high guitar, hats) need cutting in high frequencies and boosting in lower, and vice-versa. This is really only a starting guide, not a rule. Also, sounds that interfere with eachother can be separated in a mix by EQing them in opposite directions.
4. Bass usually needs a boost in the mid range somewhere and sometimes the high. This way it can cut through and be heard on smaller speakers.
5. Kick drums usually need that same mid and/or high boost on a subtle level so they too can cut through on smaller speakers. For hip-hop, kick needs a low end boost, but NOT TOO MUCH.
6. Snare drums always sound warmer with a boost in the low-mid range and some cut of the highs. An annoying CRACK can be softened with this high cut. Sometimes I boost the lows in snares to make them even fatter. But it really depends on the snare sound. The rule of opposites usually applies here. Snare sounds that were thin to begin with I usually warm up a bit, and heafty snare sounds I might thin out a bit.
7. Hats almost never need any EQ if they're recorded clean. Usually an EQing for my hat tracks is to cut highs to get rid of an annoying hiss.
8. Guitars are simaler to snares for me. A thin original guitar might need boosting in mids and lows (depending on what the desired sound is, and what else is present in the mix) or a heafty guitar might need to be thinned out a little by cutting lows and low-mids.
9. Vocals usually like to have a boost in the mids or high-mids, but it depends on the voice. Vocals nearly always get lost amongst guitars...a good way to deal with this is the rule of opposites. Boost mids in the vocals and cut them in the guitar, or something similar. Vocals can also have annoying hiss or sibilance, and sometimes cutting high frequencies can help that.
10. Strings, and more specifically good string patches from a synth, usually need little EQ. If they are merely a support player, I may thin them out a tiny bit, or if they are meant to be present, I may thicken them in the mids a little (or sometimes the opposite...this stuff is highly subjective). But they usually work well left alone. Really clean piano or keyboard synth patches are the same way.
11. I like to leave reverb returns alone, but if the reverb becomes annoying and noisy, cutting some high can soften it up a bit...same with strings.
12. Extreme EQ setting create sounds of their own. Experiment. But for a non-novel track, be subtle.
13. AC hum from a track can almost always be fixed by cutting 60 Hz all the way off. (Sometimes this can take away from bass or kick sounds, but I believe that most frequencies audible in a song are above 60 Hz).
14. Play with EQ settings thoroughly to find appropriate settings.
15. I don't mix horns too often, but when I do, I like to leave them alone. Clean horn tracks usually seem fine to me.
16. NEVER EVER EVER force yourself to EQ a track that sounds fine, just because you think you should use the full capabilities of the studio. NEVER NEVER NEVER!
If anyone out there has rules they use for their mixes, especially for instruments I don't mention or use much, send 'em along. :)
If you have questions, or have noticed I have left something out, or misspelled, or mis-explained, or (god forbid! hehe) I'm wrong, mail me at Σφάλμα! Δεν έχει οριστεί σελιδοδείκτης.or Manny Rettinger at Σφάλμα! Δεν έχει οριστεί σελιδοδείκτης.. Σφάλμα! Δεν έχει οριστεί σελιδοδείκτης.

A Basic Guide for EQing

by Devin Devore of Σφάλμα! Δεν έχει οριστεί σελιδοδείκτης.
Some History
Dating as far back as the 1930's, the equaliser is the oldest and probably the most extensively used signal processing device availible to the recording or sound reinforcement engineer. Today there are many types of equilisers availible, and these vary greatly in sophistication, from the simple bass and treble tone control of the fifties to advanced equipment like the modern multi-band graphic equaliser and the more complex parametric types. Basically, an equaliser consists of a number of electronic filters which allow frequency response of a sound system or signal chain to be altered. Over the past half century, equalisers design has grown increasingly sophisticated. Designs began with the basic 'shelving filter', but have since evolved to meet the requirements of today's audio industry.
Understanding EQ and its Effects on Signals
There are two areas of equalisation that I want to cover. Those two areas are vocals and music. I'd like to discuss the different effects of frequencies within audio signals. What do certain frequencies do for sound and how we understand those sounds. Why are some sound harsh? Why do things sound muddy? Why can't I understand the vocals? I'll try and answer all of these question and hopefully bring some light to the voo-doo world of EQ.
Vocals
Roughly speaking, the speech spectrum may be divided into three main frequency bands corresponding to the speech components known as fundamentals, vowels, and consonants.
Speech fundamentals occur over a fairly limited range between about 125Hz and 250Hz. The fundamental region is important in that it allows us to tell who is speaking, and its clear transmission is therefore essential as far as voice quality is concerned.
Vowels essentially contain the maximum energy and power of the voice, occurring over the range of 350Hz to 2000Hz. Consonants occuring over the range of 1500Hz to 4000Hz contain little energy but are essential to intelligibility.
For example, the frequency range from 63 to 500Hz carries 60% of the power of the voice and yet contributes only 5% to the intelligibility. The 500Hz to 1KHz region produces 35% of the intelligibility, while the range from 1 to 8KHz produces just 5% of the power but 60% of the intelligibilty.
By rolling off the low frequencies and accentuating the range from 1 to 5KHz, the intelligibility and clarity can be improved.
Here are some of the effect EQ can have in regards to intelligibilty. Boosting the low frequencies from 100 to 250Hz makes a vocal boomy or chesty. A cut in the 150 to 500Hz area will make it boxy, hollow, or tubelike. Dips around 500 to 1Khz produce hardness, while peaks about 1 and 3Khz produce a hard metallic nasal quality. Dips around 2 to 5KHz reduce intelligibilty and make vocals woolly and lifeless. Peaks in the 4 to 10KHz produce sibilance and a gritty quality.
Effects of Equalisation on Vocals
For the best control over any audio signal, fully parametric EQ's are the best way to go.
80 to 125
160 to 250
315 to 500 Sense of power in some outstanding bass singers.
Voice fundamentals
Important to voice quality
630 to 1K Important for a natural sound. Too much boost in the
315 to 1K range produces a honky, telephone-like quality.
1.25 to 4K
5 to 8K Accentuation of vocals
Important to vocal intelligibility. Too much boost between 2 and 4KHz
can mask certain vocal sounds such as 'm', 'b', 'v'. Too much boost between
1 and 4KHz can produce 'listening fatigue'. Vocals can be highlighted at the 3KHz
area and at the same time dipping the instruments at the same frequency.
Accentuation of vocals.
The range from 1.25 to 8K governs the clarity of vocals.
5 to16K Too much in this area can cause sibilance.

Instruments
Miking instruments is an art ... and equalisers can often times be used to help an engineer get the sound he is looking for. Many instruments have complex sounds with radiating patterns that make it almost impossible to capture when close miking. An equaliser can compensate for these imbalances by accenting some frequencies and rolling off others. The goal is to capture the sounds as natural as possible and use equalisers to strighten out any non-linear qualities to the tones.
Clarity of many instruments can be improved by boosting their harmonics. In fact, the ear in many cases actually fills in hard-to-hear fundamental notes of sounds, provided the harmonics are clear. Drums are one instrument that can be effectively lifted and cleaned up simply by rolling off the bass giving way to more harmonic tones.
Here are a few ideas on what different frequencies do to sounds and their effects on our ears.
31Hz to 50Hz These frequencies give music a sense of power. If over emphasised they can make things muddy and dull. Will also cloudy up some harmonic content.
80Hz to 125Hz Too much in this area produces excessive 'boom'.
160Hz to 250Hz This is the problem area of a lot of mixes. To much of this area can take away from the power of a mix but is still needed for warmth. 160Hz is a pet-peeve frequency of mine. Also, the fundemental of bass guitar and other bass instruments sit here.
300Hz to 500Hz Fundamentals of string and percussion instruments.
400Hz to 1K Fundamentals and harmonics of strings, keyboards and percussion. This is probably the most important area when trying to control or shape to a natural sound. The 'voice' of an instrument is in the mids.
To much in this area can make instruments sound horn-like.
800Hz to 4K This is a good range to accentuate instruments or warm them up. Too much in this area can produce 'listening fatigue'. Boosts in the 1K to 2K range can make instruments sound tinny.
4K to 10K Accentuation of percussion, cymbals, and snare drum.
Playing with 5K makes the overall sound more distant or transparent.
8K to 20K This area is often what defines the quality of a recording or mix. This area can also help define depth and 'air' to mix. Too much can take away from the natural sense of a mix by becoming shrill and brittle.
Here are a few other pin point frequencies to start with for different instruments. In a live sound situation, I might event pre set the console's eq to these frequencies to help save time once the sound check is under way. These aren't the answers to everything... just a place to start at.
Kick Drum:
Besides the usual cuts in the 200Hz to 400 area, some tighter Q cuts at 160Hz, 800Hz and 1.3k may help. The point of these cuts makes for space for the fundamental tones of a bass guitar or stand up. I have also found a high pass filter at 50Hz will help tighten up the kick along with giving your compressor a signal it can deal with musically. 5K to 7K for snap.
Snare Drum:
The snare drum is an instrument that can really be clouded by having too much low end. Frequencies under about 150Hz are really un-usable for modern mixing styles. I would suggest a high pass filter in this case. Most snares are out front enough so a few cuts might be all that is needed. I like to start with 400Hz, 800Hz, and some 1.3K. This are just frequencies to play with. Doesn't mean you will use all. If the snare is too transparent in the mix but I like the level it is at, a cut at 5K can give it a little more distance and that might mean a little boost at 10K to brighten it up.
High Hats:
High hats have very little low end information. I high pass at 200Hz can clean up a lot of un-usable mud in regards to mic bleed. The mid tones are the most important to a high hat. This will mean the 400Hz to 1K area but I've found the 600Hz to 800Hz area to be the most effective. To brighten up high hats, a shelving filter at 12.5K does nicely.
Toms and Floor Toms:
Again, the focus here is control. Most toms could use a cut in the 300Hz to 800Hz area. And there is nothing real usable under 100Hz for a tom... unless you are going for a special effect. Too much low end cloud up harmonics and the natural tones of the instrument. Think color not big low end.
Over Heads:
In my opinion, drum over heads are the most important mics on a drum kit. They are the ones that really define the sound of the drums. That also give the kit some ambience and space. These mics usually need a cut in the 400Hz area and can use a good rolling off at about 150Hz. Again, they are not used for power.... these mics 'are' the color of your drum sound. Roll off anything that will mask harmonic content or make your drums sound dull. Cuts at 800Hz can bring more focus to these mics and a little boost of a shelving filter at 12.5K can bring some air to the tones as well.
Bass Guitar:
Bass guitar puts out all the frequencies that you really don't want on every other instrument. The clearity of bass is defined a lot at 800Hz. Too much low end can mask the clearity of a bass line. I've heard other say that the best way to shape the bass tone is to roll off everything below 150Hz, mold the mids into the tone you are looking for, then slowly roll the low end back in until the power and body is there you are looking for. If the bass isn't defined enough, there is probably too much low end and not enough mid range clearity. Think of sounds in a linear fashion, like on a graph. If there is too much bass and no clearity, you would see a bump in the low end masking the top end. The use of EQ can fix those abnormalities.
Guitar/piano/ etc.:
These instruments all have fundamentals in the mid range. Rolling off low end that is not needed or usable is a good idea. Even if you feel you can't really hear the low end, it still is doing something to the mix. Low end on these instruments give what I call support. The tone is in the mids. 400Hz and 800Hz are usually a point of interest as are the upper mids or 1K to 5K. Anything above that just adds brightness. Remember to look at perspective though. Is a kick brighter than a vocal? Is a piano bright than a vocal? Is a cymbal brighter than a vocal?
In Closing
Equalisers are one of the most over looked and mis-used pieces of gear in the audio industry. By understanding equalisers better, an engineer can control and get the results he or she is looking for. The key to EQ'ing is knowing how to get the results you are looking for. Also, knowing if its a mic character or mic placement problem. EQ can't fix everything. It can only change what signal its working with. Equalisers are also a lot more effective taking away things in the signal than replacing what was never there.

Reverb
Reverb is an important studio tool. It can be used to add realistic ambience to a sound that was recorded in a dead, dry room, or to electronic or synth sounds. About everything we hear has some reverb to it, so when we hear an untreated sound, it sounds uncomfortable, and unnatural.
Back in the sixties and seventies before there was digital reverb, studios used plate reverb. They would hang a thin piece of metal inside some frame work, and vibrate it using a voice-coil assembly. Then they would mic the metal plate with contact mics and feed that back into the mixer. The only problem with this method was that it sounded metallic and bright. After so many years of hearing this, people were used to it, and the new digital reverbs sounded strange to them. Now, digital reverb units repeat little fragments of the sound wave thousands of times to recreate reverberation. Most reverb units have hall sounds, room sounds, and, of course, plate sounds which are great for drums.
Basic Rules for Using Reverb
• The effect sounds the best when used in sparingly. Don't swamp tracks in it. Use the least possible to get the desired effect. The best engineers know when they have used too much.
• Sounds with a lot of bass, such as the kick drum or bass guitar are best left with little or no reverb. If you do use it, keep it short and bright, or cut the low frequencies on the reverb return. Otherwise you'll have a big mess before you know it.
• Obviously the more reverb you use, the farther away a sound will seem. This can be used to push certain things back in the mix such as backing vocals, but once again, don't load it on.
• Many times your effects unit will allow you to use many different types of reverb in one mix. Theres nothing wrong with using a couple of different reverb styles all within the same mix, it will just sound more interesting to the ear.
Useful Settings
• Drums
Style:
Bright Plates, nonlinear
Length:
Between 1.1 and 2.5 seconds
Pre-delay:
Around 20 milliseconds
• Vocals
Style:
Plate or short hall
Length:
Between 2 and 3 seconds
Pre-delay:
Between 20 and 60 milliseconds
• Piano
Style
Hall or concert hall
Length:
Between 2 and 4 seconds
Pre-delay:
Between 5 and 50 milliseconds
• Electric Guitar
Style:
Room or Plate
Length:
Between 1.5 and 3.5 seconds
Pre-delay
Between 20 and 50 milliseconds
• Strings
Style
Plate or Bright hall
Length:
Between 1 and 2.5 seconds
Pre-delay
Between 20 and 80 milliseconds

10 Steps to a better Mix.

By: Howard Mangrum
The following is a ten step procedure for the mixing of a song. These steps can be varied in any way necessary to accommodate the themes or concepts of the song or materials to be mixed. Please be aware that the detail of each step can change depending on your equipment and the song. Sometime the song may not be fully developed and attempting to mixdown will make this evident, one of the reasons why even the professionals do rough mixes. Final mixes are best approached when your ears are fresh, not at the end of an all day tracking session. After mixing various projects you will develop your own procedure and you can feel free to throw this out, I mean store this for further reference along with all of the other bad song ideas that your friends have come up with.
1. Normalize & Mute
Normalize each track by panning to the center, take the EQ section out or verify all settings are zeroed (this may be in the 12 o'clock position), and turning down (off) all Aux Sends so that there are no effects. Pull all faders down (some people mute each channel, then un-mute them individually as they proceed).
(This is a good place to play a reference CD, to help ensure the monitoring system is performing properly and you have a good referenced starting point for your mix.)
Review any notes taken during the tracking process and your preproduction notes. Setup the signal routing scheme, configure patch bays. Compressors and noise gate can be patched in and normalized so they have little or no effect (put device into bypass if possible, set noise gates to a low threshold, etc.). It should be possible to assign outboard effects (reverbs) to the various tracks at this point based on the song concept and basic ideas of the sonic landscape.
 
Continued


(It is perfectly acceptable to determine the concepts and sonic landscape as you progress through these mixing steps, this is art and there are no rules, just guidelines or opinion.)
2. Loop play
Set the tape deck to play the song in loop-mode if possible. This allows the following steps to be completed in a continuous procession.
3. Critique & EQ
Critique each track individually. Start by soloing (un-muting or only bringing up one fader) each track to ensure proper gain setting by observing your level indicators. Setup noise gates and compressors if necessary. Perform your first rough EQ, do this EQ as fast as possible, don't spend more than a few minutes per track
(the point of dimensioning returns is close at hand during this first pass, the perceived frequency distribution will shift/change, as all tracks are mix together)
(Periodically switch the EQ section out and back in to help ensure you are making improvements.)
(General approach to EQ; if you have a parametric with tunable 'Q' use it to fix frequency problems, get rid of the bad sound. Use shelving EQ's to do gross adjustments to the sound, since they effect a large range of frequencies.)
(It is usually better to cut, so work to cut the bad and if you have remaining control use this to enhance.)
(Adjustment to the EQ/frequency content can make dramatic changes to the gain structure of your signals/sounds, be sure to keep an eye on this and make adjustment accordingly)
4. First Mix
Bring up each track to start building your mix. The order should mimic the priority of each track, this depends on the style of the music and your personal tastes.
(It is standard practice to start with the foundation, such things as drums & bass)
(This is were you start to build your sonic landscape, you determine which sound should be out front and which sound should be in the background.)
(If necessary this is a good place to draw a sketch of your stage setup of the band, to help visualize your sonic landscape)
5. Re-EQ
Re-EQ tracks where necessary. Listen for too much sound (muddy) in each frequency range, where, you have instruments or sounds that are in the same basic frequency range and may conflict or mask each other.
6. Pan
Pan tracks/sounds to complete the setting of your sound stage.
(This step is done in conjunction with re-EQ, step 5. The overlapping frequencies maybe less offending after panning)
(Periodically monitor your constructed sonic landscape in mono to ensure that phase cancellation and sound masking are not going to cause you any problems, your mix should stand-up in mono as well as stereo, with only the basic imaging shifting.)
7. Effects
Setup the reverb and other effects. When applying reverb, keep your sonic landscape in mind. You are setting the outer-boundaries of your sonic landscape at this point.
(It is easy to over use reverb and other effects, generally turn-up the effect to a point where they become dominate then back them off to they just meld into the background)
(Be sure to keep a written record of which effects are used where and the programs of the effect units, with any special settings and/or signal routings that have been employed)
8. Balance Mix
Listen to the mix and ensure you can hear each sound and the over-all balance between each sound is correctly portioned.
(This is a good point to perform a rough mix to tape and playing it on a secondary monitoring system to help gain a second prospective and ensure the main monitoring system is not leading you down the wrong path. Studio monitors can reduce the perceived impact of various settings and the amount of such things as reverb.)
9. Map Moves
Map out any move that maybe necessary, such as:
• level changes
• muting of tracks
• panning
• effect changes
(map the move to a tape counter and/or a smpte time readout, keep a written list of these moves)
(learn to perform the moves on-the-beat, tap your foot and count)
10. Practice Mix
Practice the mix, learn to play the console/mix like an instrument. When you are confident with your mix start recording it to your mixdown deck.
(It is usually good to perform a few mixes, like any performance each will be different and one will usually be preferable.)
(It is common at this point for you to realize that you have not determined how the song should start or end, map these moves out as above. Be sure to allow for some pre-roll & post roll time)
(Performing a mix should be similar to performing on an instrument where moves and other events happen on the beats of the song, your mixing moves should have rhythm to them.)

Roger Nichols Recording Guide
Setup:
• How many people (musicians) will be in the recording room and how will they be arranged
• What Instruments will they be playing and what special requirements need to be met
• How big is the room (or rooms). If sharing an isolation room, consider grouping of instruments for least adverse leakage.
• Isolation between instruments should be considered. Is some of what is being recorded going to be replaced (stand in vocalist, not the real solo, etc.) Determine how best to isolate the instruments (baffles, Tube Traps, blankets, foam, plywood).
Cables:
• You can never have too many cables or adapters. All cables must have previously been ascertained to be in proper working order. Cables that have been previously suspect and checked to find nothing wrong should be labeled as such until they have successfully worked in a session. (this is in case of a cable problem, the first cable to check would be a previously faulty cable.) Anticipate problems as much as possible.
Microphones:
• Choice of microphones. What mics are available for the session? What mics are specifically requested by the client? Are there notes from previous sessions with the same musicians that pointed out a unique requirement or a mic that worked rather well in a particular situation.
• Impedance must be matched. (Lo, Hi, Inline Xformer) Thin sounding microphones (reduced low frequency response) usually means that the impedance is not matched properly. Connections must be matched. (XLR, 1/4", DIN, Teuschel). Polarity must be matched (Pin 2 hot - Pin 3 hot?)
• Phantom requirements must be ascertained. If mics are split to multiple consoles, such as live performances, only one console should provide phantom power. Make sure that mic splitter will pass phantom (some will not). If console does not provide the proper phantom voltage, use external pass through phantom power modules. If the microphone has it's own power supply, make sure that phantom is turned off to that mic, otherwise distortion and noise may result. (Guaranteed in some instances)
• Can't have phantom on with unbalanced microphone.
• Try direct box for synths and electric instruments. Try different direct boxes (they are like microphones and have a coloration of their own). Active (phantom powered) direct boxes may not have ground isolation capabilities and may cause ground loops (which result in a buzz or hum). Some consoles will let you pluginstruments in directly. (check for impedance matching.)
• Pickups on acoustic instrument can be added in with the microphone sound.
• Mic patterns must be chosen properly for the job. Understand proximity effect in microphones.
• Microphone placement may not be the same each time you record in a similar situation. It may depend on the individual player or instrument.
• Listen for reflections off of music stands (use foam or towel) when recording vocals. Listen for extraneous noises from squeeky chairs or rattling instruments.
Speakers and monitoring:
• What do they sound like. Have you heard these speakers before in a different environment? Does the control room color the sound of the speakers so that you must compensate for that difference?
• Placement of speakers in control room may effect the way they sound (experiment with different placements)
• Try not to use speakers (instead of headphones) for monitoring in the studio during recording. The leakage will hamper an otherwise good recording. If this must be done, there are methods whereby two speakers are fed a MONO signal and placed out of phase. The microphone is then placed in the phase null between the speakers. Extreme caution should be taken when employing this method.
• Use distribution system for multiple headphones, don't just parallel a bunch of headphones from the console or cassette machine headphone outputs.
• "More Me" headphone distribution systems for individual mixes to each musician can help the recording process immensly.
Console:
• Trim vs. volume control. Don't clip the input. In some situations it may be desirable to set all faders to "zero" and establish the initial recording level with the input trims. This provides an instant graphical representation if microphone levels change drastically during the recording (the fader is pulled down or up from it's reference)
• Check the sound through the console, select the correctrouting path, with inserts disabled.
• With no EQ, listen. Does it sound good, is it muffled, scratchy, far away,or boomy?
• Is the mic facing the wrong way, (this happens often) or are you listening to wrong input.
• Impedance mismatch between the console input and the source. (line, mic, instrument)
• Bad cord, connection, patch bay, patched in wrong hole, patchbay normal not broken- mic going too many places at once
• Balance vs. unbalanced - pin2 vs. pin 3 (unbal pin 3 at one end + unbal pin 2 = SHORT)
• Bad instrument - change try another
• Bad playing technique or position - try something else, face Mecca.
• Move the mic a little - start with close micing, then move the mic away
• Acoustic guitars, pianos, Bass, Standup bass, Drums
• Go in the room and listen to the instrument with a finger in one ear.
• Ask the player - chances are he has recorded this instrument before and has some idea as a starting place.
• If he says "This is what I do all the time and it always sounded good before" then there is probably something else wrong.
Tape Machine:
• Machine on input. Monitor through the machine (good idea in case you are overloading the machine input) make sure that whole signal path is working right. (what you see on the meter may not be what you think is going there.
• Listen to output of machine with no music playing. Listen for hums, crackles or buzzes. If the meter is reading something, then there is probably a hum or other noise that you didn't notice.
• What kind of metering? Digital metering is the most accurate. Peak meters second best Analog VU meters, depend on what music is playing (click, hi hat, organ, etc.) Percussive instruments should indicate lower on the VU meter for proper recording level.
• Don't forget to make sure that you are using the correct tape for the machine. Bias, tape stiffness & head wrap.
• Noise reduction dbX Dolby A, B, C, S, SR
• Don't use noise reduction on the SMPTE track. Make the SMPTE track one of the edge tracks. (cross-talk). Don't use noise reduction on digital recordings.
• Autolocate is a nice feature. Also autopunch, cycling etc.
• Don't forget to clean the machine. Digital machines need cleaning too. Follow manufacturers guidelines.
Recording:
• Start the machine in plenty of time before the song begins. Allows machine to get up to speed. Allows plenty of SMPTE for future lockups.
• Let the machine keep running a little while after the take. In case you want to add something at the end, or cross-fade into the next tune, or?
• Make sure you have enough tape for the take you are about to record. If what you are recording is longer than a reel of tape, plan a break in the music for changing tape or get a second machine for A/B rolling. (the second machine is placed into record before the first machine runs out of tape.
Overdubs:
• Must be able to monitor output of machine
• Good headphone mix.
• Try not to use speakers to monitor during overdubs
• Test punch in capabilities of machine Punch during sustained playing & punch right on beat. Play back. See if there is glitch and see if there is any delay. If delay, modify punch technique accordingly.
• If big glitch, don't punch during sustains or punch on back beat or someplace that will mask punch.
Effects - EQ:
• Equalizers - change the tonal characteristics of the audio. They have at least bass and treble controls. Most desirable is four band sweepable parametric EQ.
• Graphic equalizer. Usually 5 to 31 frequency bands, each fixed in frequency. Usually with slide pots to show a graphic representation of the frequency curve.
• Peak vs. Shelving EQ.
• Tuning EQ by EAR
• Use EQ to:
• Compensate for low listening levels
• Make the blend between different instruments more pleasing
• Compensate for bad frequency response in some device
• Reduce noise
• Special effects like telephone voice
• Reduce apparent leakage between instruments
Effects - Compressors & Limiters:
• Compressors keep levels more constant by automatically detecting level changes above a set level and riding the gain.
• Use compressors on individual instruments, not mix. It will be less audible.
• Attack time settings determine the "punchiness" of the instrument. Peaks get through before the compressor actually clamps down. Faster attack will make for a smoother sound.
• Limiters are faster than compressors and are there to LIMIT the amount of signal passing. These are usually there to protect equipment such as radio transmitters or speakers from overloading.
Effects - Noise Gate:
• Noise gates work like a soft on-off switch. As the level of the sound gets below a set point, the signal is turned off, blocking any residual noise that may creep through. If not set properly they can be worse than the noise.
Effects - Delays & Echoes:
• A delay by itself has no effect but to delay the signal. When the delay is heard mixed with the original signal, we have a sometimes more interesting sound.
• Echo & reverb units control the amount of feedback sent to the input of the delay as well as the number of taps off of the delay line. These signals mix together to form artificial reverberation as found in different size enclosed spaces.
• Doubling (recording the same instrument playing the same part twice) can be simulated by using a delay of from 9 to 30 milliseconds. This fattens up vocals and instruments and can make it sound like there was more than one instrument playing the same part.
• Short delays can also add fake ambiance to a recording that was too dead sounding.
• Chorusing is caused by modulating the delay time. This modulation causes a change in pitch as well as a change in the delay time. This produces a wavy effect in the sound.
• Flanging effects are created by using a delay of 10 to 20 milliseconds and changing the delay amount slowly between those two parameters. The delayed signal mixes with the original signal and some of the frequencies are out of phase with each other and cancel or augment each other. A change in delay time changes the frequency that is affected.
Harmonizers - Octave dividers - Aural Exciters:
• Harmonizers are used for pitch shifting effects. They can be used to fix bad notes in some cases, or to add harmonies in other cases.
• Octave dividers add an additional tone one to two octaves below the original signal. This can fatten up an otherwise wimpy bass or kick drum.
• Aural Exciters work by adding slight distortion and phase shift to the signal. This can brighten up an otherwise dull sounding instrument. They usually work the best if there is a rich overtone sequence present in the original sound. The work great on snare drums, background vocals, and string pads.
Combining tracks:
• On analog machines do as little as possible. If you have to combine vocals - record a bunch, combine to one track - record next bunch - combine to next track. At all costs avoid bouncing to adjacent tracks (feedback). Watch out for track next to SMPTE.
• Combination digital and analog machine. Record vocals on ADAT & combine to one track on analog deck. Same quality as one original recording on analog machine. Adjacent tracks not a worry on digital machines.
Comping tracks:
• Recording multiple tracks and combining to make one master track. If you can do it on the digital deck, it is better. If you must do it on analog deck, try not to do it multiple times.
• If the way you work is to try 4 takes, then comp, try 4 more then comp again. Don't use the comp track as a component and bounce to new track, try to take any pieces and comp them into the existing comp track. That way comp track will never be more than one generation down. (Make a safety track if you have trouble punching in tight spots.
Mixing:
• Clean up tracks. Erase unwanted material (with the supervision of the producer). Mixing will be easier
• Make a cue sheet reminding you when to make what moves
• Levels. Different DAT machines use different reference level.
• What does reference level mean? analog vs. digital.
• In analog recording, "Zero" is a level reference at which there is 3% harmonic distortion. Above this level there will be more distortion but a better signal to noise ratio. Audio contains peaks which may be above this zero reference by as much as 20dB. Analog tape compresses this information and records it with more harmonic distortion, but for the small instance that the peak lasts, this may not be a problem. If recordings are made at a lower level, the distortion figures are lower, but the signal is dropping into the noise floor of the tape.
• In digital recording, "Zero" is the level above which no additional information can be recorded. This results in hard clipping of the sound. Anything above "Zero" is not recorded. A reference level of 18dB below "Zero" allows room for peaks in the audio to be recorded without clipping. Because the noise floor is so low in digital (98dB below "Zero") having a reference at -18dB does not really effect the quality of the recording.
• Echo. Don't use too much of a good thing. Use just enough to provide the ambience or effect necessary.
• Effects. If you have empty tracks available on the multi track tape, record the effects to free up equipment for something else, or to save time in re-mixing.
• Limiters (use on record and playback - different ratios) Effects on vocals should be kept to a minimum.
• Panning and stereo placement should be determined by the final destination of the mix (TV, video game, Surround Sound, CD, CD Rom). Keep in mind the center buildup phenomenon. Avoid placing something all the way to one side. (keep in mind stereo listening and being able to hear from opposite side of the room)
Mix machines:
• Analog 2 track
• Revox, Tascam. Otari, Ampex, Sony, Studer
• 30 ips vs. 15 ips vs. 7 1/2 ips
• 1/2 inch vs. 1/4 inch
• Dolby vs. Non Dolby
• dbX and other noise reduction.
• Center track time code
• Cleaning of machines
• DAT
• 44.1kHz vs. 48kHz
• Emphasis on or off
• External or built in converters
• Type of DAT tape Computer backup DAT tape, Apogee, HHB.
• Don't use 3hour tapes unless machine is designed for it
• Cleaning of machines. Use DAT cleaning tapes properly.
• Input pause wears the heads.
• CD-R
• Marantz, Carver, Yamaha, Studer, Phillips, Micromega.
• Cassette
• No comment
• Mixing back to two tracks of multi-track
• Multitrack 48kHz or 44.1kHz? Stuck with whatever multi is.
• Digital or analog.
• Updating mix without remixing
Sample Rate Conversion:
• To get from one sample rate to the other or VSO final mix?
• Alesis AI-1
• Roland SRC-2
• N-Vision
• Z-Sys
• Analog out - in
Editing:
• To change the arrangement of the song
• To assemble all of the tunes in order for distribution or going to mastering
• Razor Blade editing (try to keep blood to a minimum). Razor blade editing can be performed on reel to reel digital tapes under certain circumstances. Special precautions need to be adhered to and a bad edit may not be reparable.
• Hard disk editing Akai, Sound Tools, Sonic Solutions, Turtle Beach Roland, SADiE, RADAR, etc.
• Optical disk editors AKAI, Sony PCM 9000
• DAT editors Sony, Otari, Fostex Music editing vs. assembly editing
• Pause editing DAT is a NO-NO unless plenty of time between cuts.
• Editing for vinyl records (South America etc.)
• Editing for cassette master
Pre-Mastering:
• Assemble in the correct order with proper spacing
• Don't do pause edits on DAT machines unless 5 seconds around edit
• (If that is the only way, let mastering do it)
• Consistent levels (If you don't do it, Mastering will have to)
• EQ All of the selections should have similar tonal quality
• When you are done, Make a Digital Copy. Don't send your only tape
• Some plants can accept CD-R as master. It must not be Multi-Session
• All Plants accept Sony 1630
• Some plant will accept DAT( not if they have to edit)
• Include accurate timing sheet (where you want each cut to start
• Make them send you a ref (plant or mastering facility)
• If everything done (eq, levels, editing) copy DAT to 1630
• PQ codes on tape? or PQ time sheet. Music @ 3:00 into tape
• If you can afford it, good idea to let mastering facility EQ and level correct your tape. You want your product to be competitive with everyone else so it has to sound as good. Third party reference is good.
• Think about breaks for cassette. Second side should be shortest
Labels:
• Multi track labeling of boxes and track sheets.
• DAT labels & J cards
• Cassette
• CD labels.
Keeping notes:
• Keeping good notes. Which mic on which instrument.
• Which sequence was used to print to tape
• What was the tempo
• Which SMPTE interface was used to drive sequencer
• What kind of direct box was used through what preamp?
• Was instrument delayed? if so, which delay and by how much?
• What kind of tape was used and what was the machine set up for
• What was the reference level for recording.
• What reference tape was used to set up machine
• What reference tape and levels were used for Mix?
• What effect units were used and what were the settings?
• Limiter settings for vocals or whatever?
• If you printed alternate mixes, what were the differences?
• Were they printed at different levels or different VSO settings?
 
Dont know who wrote this

Pro's guide: Compression We guide you through an advanced lesson in compression...
Compression has got to be the most common processing effect applied during recording and mixing, but it's also perhaps the most difficult technique for anyone new to the game to get their head around. The fact is, most processing techniques produce an immediately audible effect, but compression, in many cases, is an altogether more subtle creature, which definitely does something to the sound, but in a way that might not always jump out and grab you. So how do you find out what compression does and how and when to use it? Simple... just sit back and read our pro's guide.
Compression basics
At a fundamental level, the function of a compressor is to decrease the dynamic range of a signal, that is to reduce the difference between the quietest and the loudest sounds, the end result being to effectively make the loud sounds quieter and the quiet sounds louder. The point of compressing a sound is to boost its average loudness and to give it a more consistent level with less fluctuations, which translates to more presence, punch or prominence in a mix.
Take a vocal, for example, where the performance may be inconsistent, with some phrases or words sounding quieter than others. This wouldn't sit terribly well in a mix and would get swamped in places by other instruments. If you compress the vocal, the quieter phrases will be brought up, the louder ones tamed, and the whole thing will sit better and be consistently audible throughout the song. And it's not just individual sounds that can benefit from compression; compressing a whole mix can make it sound louder and perhaps more exciting, and can help it to stand out when played on the radio.
Can you hear it
Although all compressors are basically designed to do a similar job, the various models, makes and designs can all sound different from each other. Some units, usually dedicated compressors, are quite subtle and transparent in action, whereas others often have additional functions alongside their compressing abilities. Some compressors can add a little distortion to the sound, while others are perceived to fatten up sounds and add a little warmth.
And then there is the question of whether or not compression should be audible. While there are some musicians and engineers who think compressors shouldn't be heard to be working, other people make the most of the pumping, sucking or breathing effects that can be coaxed from a compressor by setting a too short release time. However they are used or abused, compressors are an indis-pensable part of modern music production.
Know your knees
A compressor works by effecting the level of a signal once it rises above a threshold level, set by the user. The amount of effect or 'attenuation' is known as gain reduction, and is directly dependent on the compression ratio; the higher the ratio, the heavier the compression. A 2:1 ratio means that when the input signal exceeds the threshold by, say, 2dB, the resultant output signal will only increase by 1dB.
Compression at really high ratios of 20:1 and above is known as limiting. The highest ratio is infinity:1, known as hard limiting or brick wall limiting, where the signal can get no louder once it has reached the threshold, or its limit, in other words. While some compressors have separate controls to set both ratio and threshold, others simply have a single control to turn up the compression and all the mathematics will be done internally without you noticing.
The point where the input signal hits the threshold and starts to get compressed is known as the 'knee'. A compressor will incorporate either hard or soft knee operation, although some compressors incorporate a choice of both modes. With a hard knee, compression kicks in once the input signal exceeds the threshold. Soft knee operation sees a more gradual onset of compression as the input signal approaches the threshold, resulting in a gentler and less obtrusive effect.
Compressor controls
Envelope control, in the form of attack and release knobs, is a feature of some compressors, although some have automatic envelope control. Attack is the time it takes for compression to commence after the threshold has been reached and release is the time it takes for compression to stop after the signal has fallen below the threshold.

The final parameter on a compressor is the gain control, known as make-up gain. Because a compressor reduces the dynamic range of a signal by turning down the loud sounds, the overall peak signal is reduced. The make-up gain is used to turn up the overall output of the device so the peak signal level of the compressed sound is turned back up.
One other feature worth a mention is the sidechain. This is basically the sensing circuit that detects the incoming input signal. In some compressors it can be accessed via rear panel connectors allowing an external input to affect the compressor's operation; more on this later.
Practical uses
Compression can be used on individual instruments when recording or mixing, and it can be applied overall to a final stereo mix. There is little difference between the way compression is used in top professional studios and our home studios; the choice of compressors may be restricted by cost (check out some compressor options in our Buying guide box on the right), but the techniques and applications apply universally.
Vocals are always a prime candidate for compression. The human voice has a wide dynamic range, 'from a whisper to a scream' as the old saying goes, and compression is used to effectively reduce this range so a consistently high level of vocal can be recorded and so the vocal stays consistent throughout a track.
The usual method with vocals is to compress them while recording and again during the mixing process, if more compression is necessary. The important thing to remember about compression is once a sound is recorded with compression, the compression becomes an integral part of that sound and cannot be removed later... you are stuck with it.
The advice is to err on the side of caution. Don't pile on the compression while recording unless you're absolutely sure that's the sound you want; you can always experiment with more compression at the mix stage. For vocals, try a ratio of 4:1 while recording and, perhaps, up to about 5dB of gain reduction on the loudest bits. The actual amount of gain reduction needed will vary from singer to singer; let your ears be the judge.
Bass guitar is another instrument that benefits from compression. It gives it a more even and, perhaps, fatter sound; try ratios up to about 10:1. Compression can also be used to smooth out the sound of both acoustic and electric guitars, although distorted or overdriven electric guitars played through valve guitar amps are naturally compressed anyway. With cleaner guitars the compressor's attack control can be adjusted to a slower setting to allow the transient or front end of the note through before the compression kicks in for a slightly different sound.
The same technique works well on percussive instruments. If a more defined attack is needed, roll back the attack time slightly to check the effect and keep a relatively short release time so the compressor can recover between beats. Compression also works on more electronic sounds, but be aware the sounds in some synths and samplers may already be compressed.
Mental experiments
While all the above examples involve the whole sound being compressed, there is an alternative technique that can lead to some interesting sounds. The idea is to split a sound so it comes up two desk channels and to compress just one of those channels. Mixing the two sounds - compressed and uncompressed - makes the sound more exciting and dynamic, without compromising the original sound. This technique works well with drums, whether it be a single snare or a submix of several drums. Try using heavy compression and tweaking the attack and release controls so the compressed sound is pumping away rhythmically, then gradually mix the compressed sound in with the uncompressed one to get a larger than life drum sound.

As previously mentioned, external access to the sidechain allows the compressor's action to be influenced by whatever is connected. Equalisers are commonly connected in this manner for a type of frequency-conscious compression. The compressor will respond more vigorously to any frequencies emphasised by the EQ. In this way a compressor can be used as a de-esser to clamp down on sibilant sounds by boosting up the EQ for those frequencies where sibilance occurs (usually between 4kHz and 9kHz).
Another use of the sidechain is for 'ducking', whereby one signal can control the gain of another. This technique can be used to tighten up the sound of a rhythm section or, more specifically, the way a kick drum and a bass sound mesh together. If the bass is being compressed, a split signal from the kick drum can be fed into the sidechain input. This means every time the kick drum plays, the bass is compressed a bit more and reduced in volume for the duration of the kick drum hit, resulting in a tighter rhythm and smoother bass end. Setting the release control is crucial in this instance to bring the sound of the bass back up at exactly the right time.
Finally, compressing a whole stereo mix can give a professional sheen, and can make it sound louder, tighter and punchier. Use a stereo compressor or a dual mono one in Link mode, so there are no undue shifts in the stereo image, and use quite a low ratio. Alternatively, multi-band compressors allow you to apply differing amounts of compression to different frequency bands of the track, so you can, say, tighten up the bottom end in relation to the top.
The one thing to bear in mind is it can be easy to overdo things. If your track is being mastered for release it may be wise to leave the job to an experienced studio engineer for optimum results.

Step-by-step: compress a live vocal
1.Plug the compressor across the insert point of the desk channel through which the vocal is being recorded, or, if you are using a separate preamp, connect the compressor between the preamp output and the recorders input. If you are using a compressor which allows you to set a ratio like this dbx 160x, use a lower ratio like 3:1 or 4:1.
2.Gradually turn the Threshold knob until the compressor is working on the signal. The amount of compression or gain reduction is usually indicated on a meter (in this case it's a red LED ladder) and you can use this as a guide to set the gain reduction to be between 3dB and 5dB on the loudest notes.
3.As compression effectively decreases the signal level, compressors have an Output or Make-up Gain knob to compensate for the level drop. Turn the knob until you have sufficient signal to feed the next piece of equipment in the chain.
Step-by-step: compress a drum loop
1.Connect the compressor across the insert point of the desk channel that is handling the drum loop, or plug it in line with the output of your sampler. For this example we'll be using a Drawmer LX20 compressor.
2.The LX20 is a dual mono unit where the two channels can be linked. In this case the drum loop is mono so take the machine out of Link mode and just the left channel is used. The expander is a rudimentary gating device that we don't need so should be turned off.
3.Some compressors have separate ratio and threshold controls, whereas others, like this unit, combine them into a single compression knob. Turn the knob up until the signal is heavily compressed, with a large amount of gain reduction indicated on the meters.
4.The pumping or breathing effect is dependent on the attack and release controls. Set the attack to the fastest position and release to the slowest. Now move the release control through its range to its fastest position and note how the sound changes. You'll hear rhythmic compression effects at fast release settings
5.Now, with the release control on a fast setting, move the attack control for slower attack times and note how the sound changes. The relationship between the two controls will give many sound variations, and it may be possible to get the compression turning on and off in time with the track.
6.Once you have got a sound that you like, use the Output or Make-Up gain knob to set the output level to match the next piece of equipment.
 
20 TIPS ON... MIXING
PAUL WHITE delivers a crash course in instant mixing.
The vocals sound great, the drums are really kicking and the guitars are exceptional, but put it all together and what have you got? A mess! Sound familiar? Until you've gained plenty of experience in mixing music, the process can seem very frustrating. There are probably as many correct ways to tackle a mix as there are successful engineers and producers. Even so, I've taken 20 tips that I've found to be helpful over the years and presented them below in the form of a checklist. These are not immutable rules, just general guidelines that can be broken any time you feel you can get away with it. Have fun!
Put the mixer into neutral (EQ flat, aux sends down, routing to Left/Right only and so on), before you start work and pull down the faders on any channels not in use. Make sure all unused aux sends are set to zero and that unused mixer channels are unrouted as well as muted, as this will further reduce the level of background noise. If you don't do this, you may find effects on tracks that don't need effects, or unwanted tracks creeping into a bounce due to a routing button being left down. You should also have a track sheet for your recording from which you can label the mixer channels. The time-honoured way to do this is to use masking tape and felt pen, so that you can peel the whole lot off when the job is finished.
Optimise the gain settings not only for the multitrack returns, but also for all effects sends and returns and for your external effect units. Also ensure that your master recorder is being driven as hard as possible, without overloading on signal peaks. These simple measures can significantly improve the clarity of your mix. If your recording is going to be digitally edited, leave any fade-outs until the edit stage, and don't try to chop off the noise that precedes or follows the mix -- you may need this when setting up a digital denoiser that requires a bare noise 'fingerprint' for calibration purposes.
Subgroup logical sections of your mix, such as the drum kit or the backing vocals, so that you can control the overall level of the subgrouped elements from a single fader or stereo pair of faders. This allows you to control the mix using fewer faders, and fewer fingers! Be aware that any channels subgrouped this way must also have their effects routed to the same groups(s), otherwise the effects level won't change as you adjust the group fader.
Where level adjustments need to be made, mark the fader settings with a chinagraph wax pencil and, if necessary, take note of the tape counter or timecode locations at which the level changes occur. This way you can solicit help from other musicians in the studio if the mix gets too busy. If you're lucky and are using mix automation, listen to the whole mix through without watching the levels, so that you can concentrate on the balance of the instruments.
Don't assume that your ears always tell you the truth. Rest them before mixing and constantly refer to commercial recordings played over your monitor system, so that you have some form of reference to aim for. This is particularly important if you use harmonic enhancers, as your ears can grow used to the effects of over-enhancement very quickly.
Don't overdo the effects, especially reverb, as this can clutter your recording and take away the contrast that is needed to give your mix punch. As a rule, the drier the sound, the more up-front it will sound, while heavily reverbed sounds tend to move into the background. If you need strong reverb on lead vocals, try to add some pre-delay to the reverb effect and adjust both the vocal level and reverb level so that the vocal sits comfortably over the backing.
Don't pan bass sounds such as kick drums or bass instruments to the sides of the stereo soundstage, as these high energy sounds need to be shared equally between the two stereo speakers for best results. As a rule, very bassy sounds contain little or no directional information anyway, although bass sounds that also contain a lot of harmonics can sound more directional.
Leave any final EQ and effect adjustments until the full mix is playing. If you work on any single instrument in isolation, it's likely to sound different when everything else is added. If you can avoid using any heavy EQ, the result is more likely to sound more natural.
Try not to have too many instruments competing for the same part of the audio spectrum. The mid-range is particularly vulnerable, so try to choose the best sounds at source. You can improve the separation when mixing by using EQ to narrow the spectrum of the sound you're working with. Try rolling off some low end and occasionally taking out any excessive top end. This is sometimes known as spectral mixing, where each sound or instrument is given its own space in the audio spectrum. A good example of this is the acoustic guitar which, in a rock mix, can muddle the low mid. If you roll off the low end, you still get plenty of definition, but the mix will seem far cleaner. Sidechain filters on noise gates (set to Key Listen mode) are often very good tools for trimming the high and low ends of sounds without unduly changing the section you want to keep. Don't over EQ sounds as they're likely to sound unnatural, especially when boosting. As a rule, good external equalisers will sound better than your console channel EQ when you're trying to make significant tonal changes. If you can confine your EQ to gentle shelving cut or boost rather than using heavy sweep mid, you're less likely to end up with nasal, harsh or phasey sounds. If possible, fix problems by using EQ cut rather than boost. The human hearing system is less sensitive to EQ cut than it is to boost. This is especially true if you are using a low-cost equaliser or the EQ in your desk.
Compress the vocals to make them sit nicely in the mix. Few vocalists can sing at a sufficiently even level to be mixed successfully without compression. Soft-knee compressors tend to be the least obtrusive, but if you want the compression to add warmth and excitement to your sound, try an opto-compressor or a hard-knee model with a higher ratio setting than you'd normally use. Be aware that compression raises the background noise (for every 1dB of gain reduction, the background noise in quiet passages will come up by 1dB), and heavy compression can also exaggerate vocal sibilance.
From time to time, check your mix balance by listening from outside the studio/bedroom door. This tends to show up level imbalances more clearly than when listening from directly in front of the monitors. Nobody is quite sure why, but it works.
Don't monitor too loudly. It may make the music seem more exciting (initially), but the end user is unlikely to listen at the same high level. High monitoring levels also tend temporarily to shift your hearing perspective and can lead to permanent hearing damage. It's fine to check the mix loudly for short periods, but most of the time, it's useful to try and mix at the level you think the music will eventually be played. (Forget I said this if you're mixing dance music for nightclubs!)
Check your mixes on headphones as well as speakers. Headphones show up small distortions and clicks that you may never hear over loudspeakers. However, don't rely solely on headphones for mixing, for they represent the stereo image differently to loudspeakers and are notoriously unpredictable at low frequencies.
Don't vary the level of the drums and bass unnecessarily during a mix, as the rhythm section is traditionally the constant backdrop against which other sounds move. Natural dynamics within rhythm instrument parts is OK, but don't keep moving the faders on these sounds.
In a busy mix, try 'ducking' mid-range instruments such as overdrive guitars and synth pads under the control of the vocals, so that whenever the vocals are present, the conflicting sounds fall in level by two or three dBs. Just a little ducking can significantly improve the clarity of a mix. Use a fairly fast attack time for the ducker (which may be either a compressor or a noise gate that has ducking facilities), and set the release time by ear. Shorter release times will cause more obvious gain-pumping, but in rock mixes, this can add welcome energy and excitement.
If you are recording a primarily MIDI-based track, try not to look at your sequencer display while mixing; the visual stimulus interferes with your ability to make subjective judgements based only on the sound. If necessary, close your eyes. Watching your sequencer progress through the arrange page can also give you a false impression of how well the arrangement is working, which is why some composers prefer hardware sequencers.
If a close-miked sound seems unnaturally lifeless, but you don't want to add any obvious reverb, try an ambience or early reflection setting to induce a sense of space. The shorter the reverb time, the easier it is to move the treated sound to the front of your mix.
Listen to your finished mix again the day after you've finished it, as your perception is likely to change after resting your ears overnight. Also check the master recording on as many different sound systems as you can, to ensure it sounds fine on all of them. Even then, save all your mix information and track sheets, including effects settings, as you never know when you might want to try to improve on the 'final mix'!
 
EQ: How and when to use it
EQ: How and when to use it
Published in SOS March 1995
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Technique : Effects/Processing

Equalisation is one of the most powerful tools in your sonic toolkit and can be your greatest enemy or your greatest ally in the battle for the perfect sound. DAVID MELLOR gives advice on how and when best to use it.

The next time you make a recording, as an experiment set all the EQ controls of your mixing console to their centre positions and leave them there until you have finished the final mix. Don't be satisfied with anything less than perfection, and don't give yourself the excuse that you can't get a good sound because you were not able to use the EQ.
EQ is a very powerful and effective item in your sonic toolkit, not unlike a circular saw in fact! But you wouldn't use your Bosch or Black & Decker for a fine carving, would you? No, you would use basic hand tools and, most importantly, your skill and judgement. As a recordist, it is your own abilities which are going to be most important to the degree of success of your recording, and you should always use the appropriate tool for the appropriate situation.
It is always best to ensure that you get as good a sound as possible from the microphone, synth or sampler coming into the mixing console. If you start off with good sounds, then a good result is almost inevitable. It is becoming increasingly popular to use microphones for recording, even when DI (direct injection) is possible, because of the wider variation of tonal qualities available. Even small variations in microphone position make vast differences to the sound picked up. It is a sign of an expert recording engineer that he or she will listen carefully to the sound from the mic and adjust its position and angle, and even try out several microphones, rather than pretend that it is always possible to get it right first time.

"Graphics are great for EQing an entire mix so that you can shape the sound as a whole, even after you have processed the individual elements."

Once you have built up your skills in this area, then you can think about using EQ. I could spout all sorts of proverbs about the things you can't make silk purses out of and the things you ought not try to polish, and these proverbs apply especially to EQ. You should always aim to use EQ to improve an already wonderful sound. If the sound isn't good without EQ, then you will never end up with anything but second best. The only time you should ever use EQ to 'save' a sound is when you have been given a tape to work on that was recorded by a lazy engineer.
Just as there is an art to creating a brilliant sound, there is an art to bringing that sound to perfection, and also blending several sounds together to make the perfect mix. Van Gogh didn't learn to paint overnight, and no-one is born with the inbuilt ability to EQ. It's a skill that is learned by experience and a good deal of careful listening.
WHAT IS EQ?
As a first step (although I know 99% of you have used EQ already!), let's see what EQ is and what it does. Then I'll move on to looking at the machinery and techniques.
Figure 1 shows one of the parameters you would expect any item of sound equipment to aspire to -- a flat frequency response. This, or at least a very close approximation, will be the frequency response of your mixing console with the EQ controls set to their centre positions, or with the EQ buttons switched off. Here, the balance of frequencies of the original signal is preserved in correct proportion at the output. In other words it is just as trebly, tinny, harsh, nasal, honky, bassy or boomy as it was when it left the microphone; or just as perfect perhaps.
Notice that the frequency response indicates what the EQ does to the sound. A cymbal will naturally have strong high frequencies, for example, and that emphasis towards HF will be preserved by a flat EQ setting. Likewise, a flat EQ will reproduce perfectly the boomy bottom end of an undamped bass drum.
If Figure 1 shows a flat response between about 20Hz and 20kHz, Figure 2 and Figure 3 show two of the curves you might expect to get from a mixing console EQ. Oddly enough, measuring the EQ and plotting the curve is something that only 0.001% of recording and sound engineers ever get around to doing at any stage in their creative careers, and only 0.0001% have their own equipment to do it to any reasonable accuracy. Even if it's hardly ever done, except on the test bench, it's a useful concept which you can carry around in your head without ever bringing to the forefront of your mind. So if a producer ever says to you, "Let's have a little more presence in the vocal", your subconscious mind will retrieve the bell-shaped curve of Figure 2 from your memory bank while your conscious mind adjusts the controls and judges the sound.
In Figure 2 we are adding an EQ boost, and there are three parameters that we would like to be able to control (if the EQ has knobs for all three). First and foremost is the frequency: this boost could be centred on any frequency according to the instrument and according to which characteristics you want to accentuate. Second is the gain, which is the degree of boost and can be measured in decibels (dB) at the centre frequency. Some mixing consoles even calibrate this control in dB, and a good thing too! You might like to have a range of up to 12 or 15dB as a maximum. Gain can also be negative, producing an EQ cut, which would be written as a gain of -6dB (or whatever) at the centre frequency, so the curve would dip downwards. EQ cut, by the way, is a vastly underused resource on many consoles, but more on this later...
The third parameter is Q, which is only occasionally offered on mixing console EQ. As well as being the star of the last ever episode of Star Trek: The Next Generation (or so my crystal ball informs me), Q is a measure of the width of the bell-shaped curve -- the bandwidth as some might say. A low Q -- 0.3 is low -- will allow the EQ to cover a wide range of frequencies, while a higher Q -- 5 is high -- will allow you to home in on a particular feature of the sound.
The bell-shaped curve of Figure 2 is often referred to as 'peaking' EQ, and applies to all mid frequency range EQ sections and a good proportion of high and low frequency EQ sections too. Figure 3 shows a 'shelving' EQ, where the boost (or cut) extends from the chosen EQ frequency all the way to the extreme end of the range. I have shown a low frequency shelving EQ in boost mode, but it could have been a high frequency cut with a similarly shaped but differently orientated curve. It isn't possible to say which type of curve is better, for it depends on what you want to achieve, but some consoles have a button to allow you to choose.
OUTBOARD EQ
Mixing console EQ is getting better and better, particularly in the low-to-mid price range. There was a stage where I was sure that the designers were inventing their EQs with the aid of a pointed finger and a pocket calculator rather than a keen pair of ears and advice from practising recording engineers, but this is no longer true of most console EQs. Nevertheless, no matter how good the EQ on your mixing console, there will come a time when you need to use an external or 'outboard' unit. This might be because you need a facility not available from your console EQ, or you might prefer to use an EQ unit for some subtle characteristic sheen it gives to the overall sound.

"You should always aim to use EQ to improve an already wonderful sound. If the sound isn't good without EQ, then you will never end up with anything but second best."

Outboard EQs come in two basic flavours: graphic and parametric. A good graphic equaliser typically has 30 or so slider controls for frequency bands nominally covering a third of an octave each. You would use two for stereo. The basic idea of a graphic is that as you set the slider controls to achieve the sound you want, the levels of the sliders 'draw' the EQ curve, as if you had measured and plotted it the long way. Unfortunately, graphic equalisers are somewhat economical with the truth and only give a rough idea of the actual curve. This is because each band does not cover only a third of an octave; its effects are felt most there but the slider will actually affect frequencies belonging to two or three bands either side of it to a distinctly noticeable extent.
Whatever deficiencies graphic equalisers may have under the Trades Descriptions Act, they are still very useful tools to have around. Mixing consoles can handle basic EQ tasks better and more quickly, but there are certain applications where graphic EQs have the edge. More on this shortly.
The alternative to a graphic outboard EQ unit is the parametric EQ. This is so called because it offers control over all three EQ parameters I mentioned earlier -- frequency, Q, and gain. A good parametric EQ unit may offer five bands, which cover the entire frequency range, or you might find three fully parametric bands with dedicated low and high frequency bands too.
USING EQ
Successful equalisation requires good equipment and a thoughtful approach from the engineer. Experienced engineers EQ by instinct and their fingers operate the controls as fluently as a jazz pianist tickles the ivories. But this fluency doesn't come automatically, it can only be won by experience. Anyone can grab the low frequency knob and wind up the bass to the maximum, but if you are serious about your recording then you will realise that it isn't just yourself you have to please; you have to consider what other listeners like and what systems they may be playing the recording on.
There is also a good technical reason why you should think before adding a lot of bass: for a given level of input, any small or medium size loudspeaker will produce much more sound at mid frequencies than at low, and if you boost the low frequencies too much then the overall level the speaker can achieve without significant distortion is less -- sometimes much less. It's a matter of compromise: the more bass you add, the lower the overall level can be. This also applies to other frequencies in the mixing console itself.
Adding EQ adds level, and it is very easy to boost the signal so much in the EQ section of the console that you run into clipping and distortion. Since the fader comes after the EQ, lowering the fader will do nothing to solve this. The answer is to reduce the gain, to allow the signal a little more headroom if necessary. One further technical point: changing the EQ of a signal nearly always changes the level, so each time you adjust the EQ you will have to consider moving the fader to compensate. It's something that will come automatically after a time, but newcomers to recording often concentrate more on the change in the sound itself and don't notice that it has suddenly become more or less prominent in the mix.
Enough of the technical stuff, recording is an artistic occupation so let's consider the subjective facets of EQ. If we consider individual sounds first, let's assume that the signal coming from the microphone is already as perfect as can be, being the result of careful positioning and angling. Each instrument has certain bands of frequencies that are strong and some that are weaker. The human voice, for example, is very strong around the 3 to 4kHz region, no matter whether male or female, or what note is being sung. When using EQ, you will be considering which characteristics of the sound you want to accentuate, or which you want to reduce. One way to consider this might be to imagine an instrument which was an 'average' of all real instruments, where the characteristics of normal instruments were smoothed out into something that had a neutral sound. When EQing a real instrument, you will either want to exaggerate its individual characteristics and make it more distinctive, or reduce its individuality and make it more like this hypothetical 'average' instrument.
This is quite simple to do, and we can make use of the standard sweep mid range control that is found on most mixing consoles, with controls for frequency and gain. A fully parametric equaliser with a Q control can offer even more precision.
First set the gain control to a medium amount of boost -- the three o'clock position of the knob is usually okay. Now sweep the frequency control up and down to the limits of its range and listen for the frequencies at which the effect is strongest. These are the frequencies in which the instrument is rich. Boosting the instrument's strong frequencies will enhance its individual characteristics and, for example, make a clarinet even more dissimilar to an oboe or any other instrument. In effect, you are making the clarinet even more clarinet-like.
When you have found the instrument's strongest frequency band, set the amount of boost according to taste and always compare what you are doing with the flat setting. If you have EQ sections to spare, you may be able to cut down on frequencies which don't enhance the sound of the instrument. Some instruments which are not known as bassy instruments nevertheless have a high low frequency content; cymbals for instance. On many occasions it will be well worth cutting down on frequencies which you don't consider to be any use to the instrument, freeing up a space in the frequency spectrum for another instrument to use.
Enhancing the sounds of individual instruments in this way is useful, but watch out when mixing that you are not boosting the same frequencies on each instrument. It is a trap for the unwary to boost every instrument at around 3kHz to help it cut through at a frequency where the ears are very sensitive. This will produce a mix that is very tiring to listen to.
The opposite of the enhancement technique is where you lessen the individuality of each instrument and make it more like our hypothetical 'average' instrument. To do this, find the instrument's strong frequencies with the mid EQ set to boost as before, but then cut these frequencies, by as much as you feel appropriate. This won't make the instrument sound better in isolation, but it will help it blend in with the other instruments in the mix.
Many aspiring engineers do not appreciate how useful EQ cut can be, but the expert will skilfully share the frequency spectrum among all the instruments so that each has its own space and doesn't have to fight with the others for attention. Using EQ in this way can result in a powerful and full sound from a small number of tracks.
Mixing consoles differ in the usefulness of their high and low frequency EQs, and it is often necessary to bring in an outboard EQ that can do the job better. I would say that it is the purpose of the low frequency control to add 'weight' to the sound without making it 'boomy'. These are subjective terms I know, but I think we can all appreciate the difference between a sound which is firm and solid in the bottom end, and one which has plenty of bass but gives the impression of being out of control. In the other direction, the low frequency control should cut low frequencies that are not contributing anything useful to the sound, while retaining the depth and body of the low mid. At the high frequency end, you should be able to cut any 'fizz' from the sound while still leaving it clear and incisive, and you should be able to make the sound brighter without the extreme top becoming aggressive. If you can't achieve all this with your console's EQ, you may have to spend a thousand pounds or more on an outboard unit that can.
When you have explored all the possibilities your console's EQ can afford and you have visited your local hire company for outboard units that perform the same function only better, you'll be keen to get your hands on a graphic equaliser. This is a rather different animal which appears at first to offer the ultimate in flexibility: just raise or lower the frequency bands you are interested in for quick and precise control. Unfortunately, you will find that precision is lacking because each individual band alters frequencies over quite a wide range on either side of its nominal centre frequency.
This doesn't mean that graphics are useless -- far from it. Graphics are great for EQing an entire mix so that you can shape the sound as a whole, even after you have processed the individual elements. If you know your way around, you can do this by taking a couple of outputs from the mixing console back into two channels and using the console's EQ again, but you'll only be applying more of the same, and doing it the graphic way really is much more satisfying. Graphics are also great for adding bite to a sound: just raise one or two sliders somewhere in the upper frequency region and you will make the sound more cutting without lifting the whole of the high frequency range. Experiment at your leisure.
PROBLEM SOLVING
If you are working on a tape made by another engineer who isn't quite as fastidious as you, then you may find yourself faced with problems that EQ can help rectify. Unwanted sounds have a knack of finding their way onto recordings, particularly live recordings. If you have a 50Hz mains hum, for example, then a graphic will be able to help at only a little loss to the musical sounds on the recording. You can also use a parametric equaliser set to a high Q to home in on the unwanted frequency. Some equalisers have special notch filters to cope with precisely these situations. 50Hz hum may be removed to a reasonable extent, but the buzz caused by lighting dimmers may be impossible to get rid of. If the buzz isn't too harsh then you can try cutting the 50Hz fundamental and its harmonics at 100Hz, 150Hz, 200Hz etc. I can't promise anything, but it may make the recording just listenable.
Apart from hum or dimmer noise, if a recording is too noisy then very often the noise is most noticeable at high frequencies. Here you can use your EQ to strike the best compromise between cutting as much of the offending component of the noise as possible while still retaining some brightness in the sound. You may be able to apply a little boost at high mid frequencies, although the result will remain a compromise.
Even if the recording has no hum, buzz or noise, it may previously have been over-EQ'd. It is quite difficult to ameliorate the results of over-zealous EQing, particularly if some frequencies have been cut to a large extent. Trying to boost these frequencies back up again may result in an unacceptable amount of noise becoming apparent. Once again, compromise is necessary, although if you were dealing with one instrument from a multitrack mix you may be able to patch in a noise gate to help in this instance.
THE FUTURE OF EQ
There is no doubt that the designers of EQ both in mixing consoles and outboard units are going to pay far more attention to the sound of the EQ rather than the technical specs. Some manufacturers have started to drop the conventional 'low', 'mid' and 'high' labels and describe their controls with names such as 'bottom', 'sheen' and 'glow'. I don't think this is a bad idea, since it will focus our energies less on the technicalities and more on the sound the EQ produces. I wouldn't be at all surprised to see EQ being combined -- not just in series within the same box, but fully integrated -- with compression or distortion far more often than it has been up to now.
Whatever the future may offer, EQ will always be one of the most powerful tools in your recording toolkit, so make the most of it.

EQ WITH BOLDNESS
When adjusting the amount of EQ to apply (ie. the EQ gain), it's tempting to adjust it very carefully and change the setting in small increments. The problems with this method are: (a) that if the EQ setting isn't right then it is wrong and thus needs total reconsideration; (b) that the ear quickly grows used to changes in the frequency balance of a sound.
It may not always be appropriate, but the next time you want to change the EQ level of a sound, grab the control firmly, twist it all the way up and all the way down and quickly settle on a new position which will hopefully be just right.

EQ TERMINOLOGY
CUTOFF FREQUENCY The frequency at which a high or low frequency EQ section starts to take effect. Also referred to as turnover frequency.
SLOPE The rate at which a high or low frequency EQ section reduces the level above or below the cutoff frequency. Usually 6, 12, 18 or 24dB/octave.
PASS BAND The frequency range that is allowed through.
STOP BAND The frequency range that is attenuated.
FILTER An EQ section of the following types:
HIGH PASS FILTER A filter section that reduces low frequencies.
LOW PASS FILTER A filter section that reduces high frequencies.
BAND PASS FILTER A filter section that reduces both high and low frequencies.
NOTCH FILTER A filter that cuts out a very narrow range of frequencies.
GAIN The amount of boost or cut applied by the equaliser.
Q How broad or narrow the range of frequencies that is affected.
SWEEP MID A middle frequency EQ section with controls for frequency and gain.
PARAMETRIC EQ An EQ section with controls for frequency, gain and Q.
GRAPHIC EQ An equaliser with a number of slider controls set on octave or third octave frequency centres.
BELL An EQ with a peak in its response.
SHELF A high or low frequency EQ where the response extends from the set or selected frequency to the highest or lowest frequency in the audio range.
HF High frequencies
LF Low frequencies
MID Midrange frequencies
TREBLE Hi-fi enthusiasts' word for HF.
EQ OFF BUTTON The sign of a good mixing console!

EQ HINTS & TIPS
• If your mix sounds 'muddy', boost the main frequency range of each of the principal instruments. Boost 'decorative' sounds even more and pull the faders right down.
• If you can't get your tracks to blend together in the mix, cut the main frequency range of the principal instruments.
• To make vocals stand out in the mix, boost at around 3kHz.
• For extra clarity, cut the bass element of instruments which are not meant to be bass instruments.
• Adding EQ boost often adds noise. Listen carefully to arrive at the best compromise.
• Changing the EQ changes the level. Always consider re-adjusting the level after you EQ.
• If you add a lot of EQ boost, you may run into clipping and distortion. Reduce the channel's gain to eliminate this.
• If you use EQ to reduce feedback in live work, take care not to take too much level out over too wide a range of important frequencies, particularly the vocal 'presence' range around 3kHz.
• If your mixing console has an EQ Off button, use it frequently to check that you really are improving the sound.
 
Compression and limiting have been covered in SOS before, but like the brown mould that you blitz every few months in the bathroom only to watch gradually return, questions on the subject steadily build up again, mere months after we explain the basic principles in an article such as this one!
On the one hand, musicians are encouraged to give an enthusiastic and dynamic performance, while on the other, their levels must be controlled to some extent, if we are to create musically acceptable mixes. One tool that is vital in helping us to do this is the compressor, but before looking at how they work, I'd like to outline the types of problems they are designed to solve.
While the faders on a mixer can be used to set the overall balance of the voices and instruments that make up a piece of music, short-term changes such as the occasional loud guitar note or exuberant vocal scream are less easy to deal with manually. When I first started recording, compressors were too expensive for home use, so we had no alternative but to 'ride' the faders. Once you've used a compressor to control your levels, however, you come to appreciate that there are certain things it can do that the human engineer is just too slow to manage. For example, unless you've played the track through and memorised exactly where the loud and quiet spots are, you'll always respond too late, because you can't start to move the fader until you hear that something is wrong. A compressor, on the other hand, will be aware of a level problem virtually as soon as it happens. Fortunately, good compressors are now relatively inexpensive, and next to reverb, a compressor is probably the most important studio processor to own -- at least for those who work with vocals or a lot of acoustic instruments.
For the benefit of those who are still a little unsure as to what a compressor does, it simply reduces the difference between the loudest and quietest parts of a piece of music by automatically turning down the gain when the signal gets past a predetermined level. In this respect, it does a similar job to the human hand on the fader -- but it reacts much faster and with greater precision, allowing it to bring excessive level deviations under control almost instantaneously. Unlike the human operator though, the compressor has no feel or intuition; it simply does what you set it up to do, which makes it very important that you understand what all the variable parameters do and how they affect the final sound.
In order to react quickly enough, the compressor dispenses with the human ear and instead monitors the signal level by electronic means. A part of the circuit known as the 'side chain' follows the envelope of the signal, usually at the compressor's output, and uses this to generate a control signal which is fed into the gain control circuit. When the output signal rises past an acceptable level, a control signal is generated and the gain is turned down. Figure 1 (p.116) shows a simplified block diagram of a typical compressor circuit.
SETTINGS AND CHARACTERISTICS
Threshold: With manual gain riding, the level above which the signal becomes unacceptably loud is determined by the engineer's discretion: if it sounds too loud to him, he turns it down. In the case of a compressor, we have to 'tell' it when to intervene, and this level is known as the Threshold. In a conventional compressor, the Threshold is varied via a knob calibrated in dBs, and a gain reduction meter is usually included so we can see how much the gain is being modified. If the signal level falls short of the threshold, no processing takes place and the gain reduction meter reads 0dB. Signals exceeding the Threshold are reduced in level, and the amount of reduction is shown on the meter. This means the signal peaks are no longer as loud as they were, so in order to compensate, a further stage of 'make-up' gain is added after compression, to restore or 'make up' any lost gain.
Ratio: When the input signal exceeds the Threshold set by the operator, gain reduction is applied, but the actual amount of gain reduction depends on the 'Ratio' setting. You will see the Ratio expressed in the form 4:1 or similar, and the range of a typical Ratio control is variable from 1:1 (no gain reduction all) to infinity:1, which means that the output level is never allowed to rise above the Threshold setting. This latter condition is known as limiting, because the Threshold, in effect, sets a limit which the signal is not allowed to exceed. Ratio is based on dBs, so if a compression ratio of 3:1 is set, an input signal exceeding the Threshold by 3dB will cause only a 1dB increase in level at the output. In practice, most compressors have sufficient Ratio range to allow them to function as both compressors and limiters, which is why they are sometimes known by both names. The relationship between Threshold and Ratio is shown in Figure 2, but if you're not comfortable with dBs or graphs, all you need to remember is that the larger the Ratio, the more gain reduction is applied to any signal exceeding the Threshold.
Hard Knee: This is not a control or parameter, but rather a characteristic of certain designs of compressor. With a conventional compressor, nothing happens until the signal reaches the Threshold, but as soon as it does, the full quota of gain reduction is thrown at it, as determined by the Ratio control setting. This is known as hard-knee compression, because a graph of input gain against output gain will show a clear change in slope (a sharp angle) at the Threshold level, as is evident from Figure 2.
Soft Knee: Other types of compressor utilise a soft knee characteristic, where the gain reduction is brought in progressively over a range of 10dB or so. What happens is that when the signal comes within 10dB or so of the Threshold set by the user, the compressor starts to apply gain reduction, but with a very low Ratio setting, so there's very little effect. As the input level increases, the compression Ratio is automatically increased until at the Threshold level, the Ratio has increased to the amount set by the user on the Ratio control. This results in a gentler degree of control for signals that are hovering around the Threshold point, and the practical outcome is that the signal sounds less obviously processed. This attribute makes soft-knee models popular for processing complete mixes or other sounds that need subtle control. Hard knee compression can sometimes be heard working, and if a lot of gain reduction is being applied, they can sound quite heavy-handed. In some situations, it can make for an interesting sound -- take Phil Collins' or Kate Bush's vocal sounds, for example. The dotted curve on the graph in Figure 2 (p.118) shows a typical soft-knee characteristic.
Attack: The attack time is how long a compressor takes to pull the gain down, once the input signal has reached or exceeded the Threshold level. With a fast attack setting, the signal is controlled almost immediately, whereas a slower attack time will allow the start of a transient or percussive sound to pass through unchanged, before the compressor gets its act together and does something about it. Creating a deliberate overshoot by setting an attack time of several milliseconds is a much-used way of enhancing the percussive characteristics of instruments such as guitars or drums. For most musical uses, an initial attack setting of between 1 and 20 mS is typical. However, when treating sound such as vocals, a fast attack time generally gives the best results, because it brings the level under control very quickly, producing a more natural sound.
Release: The Release sets how long it takes for the compressor's gain to come back up to normal once the input signal has fallen back below the Threshold. If the release time is too fast, the signal level may 'pump' -- in other words, you can hear the level of the signal going up and down. This is usually a bad thing, but again, it has its creative uses, especially in rock music. If the release time is too long, the gain may not have recovered by the time the next 'above Threshold' sound occurs. A good starting point for the release time is between 0.2 and 0.6 seconds.
Auto Attack/Release: Some models of compressor have an Auto mode, which adjusts the attack and release characteristics during operation to suit the dynamics of the music being processed. In the case of complex mixes or vocals where the dynamics are constantly changing, the Auto mode may do a better job than fixed manual settings.
Peak/RMS operation: Every compressor uses a circuit known as a side chain, and the side chain's job in life is to measure how big the signal is, so that it knows when it needs compressing. This information is then used to control the gain circuit, which may be based around a Voltage-controlled Amplifier (VCA), a Field Effect Transistor (FET) or even a valve. The compressor will behave differently, depending on whether the side chain responds to average signal levels or to absolute signal peaks.
An RMS level detector works rather like the human ear, which pays less attention to short-duration, loud sounds than to longer sounds of the same level. Though RMS offers the closest approximation to the way in which our ears respond to sound, many American engineers prefer to work with Peak, possibly because it provides a greater degree of control. And though RMS provides a very natural-sounding dynamic control, short signal peaks will get through unnoticed, even if a fast attack time is set, which means the engineer has less control over the absolute peak signal levels. This can be a problem when making digital recordings, as clipping is to be avoided at all costs. The difference between Peak and RMS sensing tends to show up most on music that contains percussive sounds, where the Peak type of compressor will more accurately track the peak levels of the individual drum beats.
Another way to look at it is to say that the greater the difference between a signal's peak and average level, the more apparent the difference between RMS and peak compression/limiting will be. On a sustained pad sound with no peaks, there should be no appreciable difference. Peak sensing can sometimes sound over-controlled, unless the amount of compression used is slight. It's really down to personal choice, and all judgements should be based on listening tests.
Hold Time: A compressor's side chain follows the envelope of the signal being fed into it, but if the attack and release times are set to their fastest positions, it is likely that the compressor will attempt to respond not to the envelope of the input signal but to individual cycles of the input waveform. This is particularly significant when the input signal is from a bass instrument, as the individual cycles are relatively long, compared to higher frequencies. If compression of the individual waveform cycles is allowed to occur, very bad distortion is audible, as the waveform itself gets reshaped by the compression process.
We could simply increase the release time of the compressor so that it becomes too slow to react to individual cycles, but sometimes it's useful to be able to set a very fast release time. A better option is to use the Hold time control, if you have one. Hold introduces a slight delay before the release phase is initiated, which prevents the envelope shaper from going into release mode until the Hold time has elapsed. If the Hold time is set longer than the duration of a single cycle of the lowest audible frequency, the compressor will be forced to wait long enough for the next cycle to come along, thus avoiding distortion. A Hold time of 50ms will prevent this distortion mechanism causing problems down to 20Hz. If your compressor doesn't have a separate Hold time control, it may still have a built-in, preset amount of Hold time. A 50ms hold time isn't going to adversely affect any other aspect of the compressor's operation, and leaves the user with one less control to worry about.
Stereo Link: When processing stereo signals, it is important that both channels are treated equally, for the stereo image will wander if one channel receives more compression than the other. For example, if a loud sound occurs only in the left channel, then the left channel gain will be reduced, and everything else present in the left channel will also be turned down in the mix. This will result in an apparent movement towards the right channel, which is not undergoing so much gain reduction.
The Stereo Link switch of a dual-channel compressor simply forces both channels to work together, based either on an average of the two input signals, or whichever is the highest in level at any one time. Of course, both channels must be set up exactly the same for this to work properly, but that's taken care of by the compressor. When the two channels are switched to stereo, one set of controls usually becomes the master for both channels -- though some manufacturers opt for averaging the two channel's control settings, or for reacting to whichever channel's controls are set to the highest value.
ALL IN THE EAR
You may have noticed, or at least read about, the fact that different makes of compressor sound different. But if all they're really doing is changing level, shouldn't they all sound exactly the same? As we've already learned, part of the reason is related to the shape of the attack and release curves of the compressor, and of course peak sensing will produce different results to RMS, but at least as important is the way in which a compressor distorts the signal. Technically perhaps, the best compressor is one that doesn't add any distortion, but most engineers seem to like the 'warm' sound of the older valve designs which, on paper, are blighted by high distortion levels. The truth is that low levels of distortion have a profound effect on the way in which we perceive sound, which is the principle on which aural exciters work. A very small amount of even-harmonic distortion can tighten up bass sounds, while making the top end seem brighter and cleaner.
The best-sounding contemporary compressor designs include valve models with a degree of distortion built in, while others use FETs, which mimic the behaviour of valve circuits. As digital recorders and mixers are introduced into the signal chain, more people are becoming interested in equipment that can put the warmth back into what they perceive as an over-clinical sound.
USING COMPRESSORS
One problem newcomers to recording seem to have is deciding where in their system to patch the compressor. A compressor is a processor rather than an effect, so it should be used via an insert point or be patched in-line with a line-level signal (for more on patching effects and processors, see my article 'The Ins and Outs of Patching' in SOS March '95). If you have a system without insert points and you want to compress a mic input, you may be able to use your foldback (pre-fade send) in an unconventional way to get around the problem, as shown in Figure 3. Here's how to do it:
Plug the mic into a mixer channel, set the mic gain level as normal, but turn the channel fader completely down. Turn the pre-fade aux send control to around three-quarters up, and do the same with the pre-fade master control, if there is one. Turn the pre-fade send fully down on all the other channels. Now you can take your mic signal (now boosted to line level), from the pre-fade send output, feed it into the compressor and bring it back into another channel of the mixer -- this time into the line input. And there you have it: your compressed mic signal.
Most engineers will normally add some compression to vocals while recording, and then add more if necessary while mixing. Working this way makes good use of the tape's dynamic range, while helping to prevent signal peaks from overloading the tape machine. It is best to use rather less compression than might ultimately be needed while recording, so that a little more can be added at the mixing stage if required. If too much compression is added at the beginning, there's little you can do to get rid of it afterwards. Similarly, if you have a compressor with a gate built-in, it might be better to leave this off when recording, and only use it while mixing. This will prevent a good take from being wrecked by an inappropriate gate setting.
A further benefit of gating during the mix is that the gate will remove any tape hiss, along with the original recorded noise. If a gate is allowed to close too rapidly, it can chop off the ends of wanted sounds that have long decays, especially those with long reverb tails, so most gates (and expanders) fitted to compressors have either a switchable long/short release time, or a proper variable-release time control.
SIDE EFFECTS
Most of the sound energy in a typical piece of music occupies the low end of the audio spectrum, which is why your VU meters always seem to respond to the bass drum and bass guitar. High-frequency sounds tend to be much lower in level and so rarely need compressing, but even so, high-frequency sounds in the mix are still brought down in level whenever the compressor reacts to loud bass sounds. For example, a quiet hi-hat occurring at the same time as a loud bass drum beat will be reduced in level.
One technique to reduce the severity of this effect is to set a slightly longer attack time on the compressor, to allow the attack of the hi-hat to get through before the gain reduction occurs. This is only a partial solution, and if heavy compression is applied to a full mix, the overall sound can become dull, as the high-frequency detail is reduced in level.
Going back to the subjective effect of subtle harmonic distortion for a moment, some compressor designs make use of harmonic distortion or dynamic equalisation to provide an increase in high-frequency level whenever heavy compression is taking place. This helps offset the dulling of high-frequency detail, and can make a great subjective difference, but it isn't a perfect solution.
More elaborate compressors have been designed which split the signal into two or more frequency bands and compress these separately. This neatly avoids the bass end causing the high end to be needlessly compressed, but it can introduce other problems related to phase, unless the design is extremely well thought-out.
DE-ESSING
Another side chain-related process is the de-essing of sibilant vocal sounds. Sibilance is sometimes evident when people pronounce the letters 's' or 't', and is really a high-pitched whistling caused by air passing around the teeth. If a parametric equaliser is inserted into the side-chain signal path of a compressor and tuned to boost the offending frequency, the compressor will apply more gain reduction when sibilance is present than at other times.
Most sibilance occurs in the 5 to 10kHz region of the audio spectrum, so if the equaliser is tuned to this frequency range and set to give around 10dB of boost, then in the selected frequency range, compression will occur 10dB before it does in the rest of the audio spectrum. The equaliser should be set up by listening to the equaliser output, and then tuning the frequency control until the sibilant part of the input signal is strongest. Figure 4 shows how a compressor and equaliser may be used as a de-esser. Some compressors have a built-in sweep equaliser, to allow them to double as de-essers without the need for an external parametric equaliser.
GENERAL GUIDELINES
For some general advice on compression settings, take a look at the 'Useful Compressor Settings' box elsewhere in this article. I should stress that these are just to get you started -- the ideal settings vary from compressor to compressor, which is why I come up with slightly different figures every time I write on the subject. The more gain reduction is used, the higher the level of background noise, so never use more gain reduction than is necessary.
 
continued
Virtually all recorded pop music has a deliberately restricted dynamic range, to make it sound loud and powerful when played over the radio. The more a signal is compressed, the higher its average energy level. In addition to compressing the individual tracks during recording or mixing, the engineer may well have applied further compression to the overall mix. This can be very effective, but don't choke the life out of a mix by over-compressing it either.
When it comes to individual tracks, it is pretty much routine to compress vocals, bass guitars, acoustic guitars and occasionally electric guitars, though overdriven guitar sounds tend to be self-compressing anyway! The most important of these to get right is the lead vocal, because even modest dips in level can make the lyrics difficult to hear over the backing.
Sequenced instruments are less likely to need compression, because you can control the dynamics by manipulating the MIDI data in the sequencer. My own rule is to avoid compression (or any other form of treatment) unless it's absolutely necessary. Even with vocals, if somebody gives me a perfectly controlled vocal take, I wouldn't want to compress it just because compressing vocals is the done thing. Compression is a very valuable studio tool, but like all tools, it is just a means to an end -- not an end in itself.
"Next to reverb, a compressor is probably the most important studio processor to own.""Virtually all recorded pop music has a deliberately restricted dynamic range, to make it sound loud and powerful when played over the radio.""Technically perhaps, the best compressor is one that doesn't add any distortion, but most engineers seem to like the 'warm' sound of the older valve designs."

DUCKING YOUR RESPONSIBILITY
In addition to their more conventional applications, compressors may also be used to enable one signal to control the level of another. This is known as ducking, and is frequently used to allow the level of background music to be controlled by the level of a voice-over. When the voice-over comes in, the level of the background music drops, but whenever there is a pause in the speech, the background music is restored to its former level, at a rate set by the compressor's release control.
To try ducking, you'll need a compressor with a side chain access socket. This allows an external signal to control the compressor action rather than the compressor's input signal. When an external signal is patched in to the side chain, its dynamics will control the gain reduction of whatever signal is passing through the compressor at the time. Let's assume that a piece of background music is being played through the normal compressor input, but that the side chain input is being fed with a voice signal from a mixer send or direct channel output. The diagram in this box shows how this is set up in practice. When the voice exceeds the threshold set by the user, the compressor will apply gain reduction to the music signal, and when the voice pauses, the gain will return to normal at whatever rate is set by the release control.
Ducking is often used in broadcast, to allow DJs to interrupt and spoil perfectly good pieces of music. Exactly how much the music will be turned down depends on both the threshold and ratio settings, and some experimentation will be necessary. The attack time should normally be set fairly fast, but the release time should be long enough to stop the music surging back in too abruptly. A release time of a second or so is a good starting point.
Even though ducking is possible with a compressor as described, it is even easier to achieve using a gate equipped with a dedicated ducking facility, such as the Drawmer DS201. If you have one of these gates, then I suggest you take the easy way out and use it. The technique is not confined to radio voiceovers: it can also be used creatively when mixing music. Perhaps the most useful application is to force backing instruments such as rhythm guitars or pad keyboard parts to drop in level by a dB or two when vocals are present, or when someone is taking a solo. When mixing, a change in level of as little as 1dB can make all the difference between a solo sitting properly in the mix, and either getting swamped or being over-loud.
Ducking can also be used in a similar way to push down the level of effects such as reverb or delay, so that they only come up to their full level during pauses or breaks. This is a useful technique to prevent mixes becoming messy or cluttered.

UNWELCOME GUESTS
Every time we apply say 5dB of gain reduction to a signal by compressing it, the peak level is reduced by 5dB, but the low level sounds remain unchanged. If we now use the Make up Gain control to bring the peaks back up to their previous level, we have to apply 5dB of gain. This means the quieter signals will also be 5dB louder than before. The outcome is that any noise present during the quieter parts of the input signal is also amplified by 5dB.
Obtrusive noise during pauses can be gated out using a gate or expander before the compressor, though many compressors come fitted with their own, built-in expanders or gates for this very purpose. However, the gating action can only mute pauses -- you're still stuck with any noise that is audible above wanted parts of the signal.

USEFUL COMPRESSOR SETTINGS
SOURCE ATTACK RELEASE RATIO HARD/SOFT GAIN RED
Vocal Fast 0.5s/Auto 2:1 - 8:1 Soft 3 - 8dB
Rock vocal Fast 0.3s 4:1 - 10:1 Hard 5 - 15dB
Acc guitar 5 - 10ms 0.5s/Auto 5 - 10:1 Soft/Hard 5 - 12dB
Elec guitar 2 - 5ms 0.5s/Auto 8:1 Hard 5 - 15dB
Kick and snare 1 - 5ms 0.2s/Auto 5 - 10:1 Hard 5 - 15 dB
Bass 2 - 10ms 0.5s/Auto 4 - 12:1 Hard 5 - 15dB
Brass 1 - 5ms 0.3s/Auto 6 - 15:1 Hard 8 - 15dB
Mixes Fast 0.4s/Auto 2 - 6:1 Soft 2 - 10dB (Stereo Link On)
General Fast 0.5s/Auto 5:1 Soft 10dB



To the best of my knowledge, compressors were first developed as a means of keeping the levels of location movie sound under control, shortly after the industry decided that talking pictures had earning potential. They were soon adopted by the music recording industry as a means of keeping the vocal excesses of untrained pop singers under control, but along the way the benign side-effects of heavy compression became a production trademark. Indeed, compression is as much a part of modern music-making as digital reverb.
Though the use of compression is not always as well understood as it could be, the fundamental workings of these devices are pretty straightforward. Essentially, a compressor is a processor designed to reduce the dynamic range of an audio signal by applying gain reduction when the input exceeds a certain level. In other words, when the sound gets too loud, the compressor turns it down. In the context of pop music, this is a useful way of applying automatic level control to singers who may not be able to restrain themselves on louder notes. In addition, vocalists find some phrases and words easier to sing than others, and the outcome is usually a performance that fluctuates in level by a considerable margin from phrase to phrase -- and even from word to word. You may have experienced this in your own demos made without compression, where some sung words and phrases tend to be obtrusive while others get almost completely lost beneath the backing -- and, aside from the considerations of vocal intelligibility, unplanned changes in level make a recording uncomfortable to listen to. Furthermore, because pop music tends to have a fairly restricted dynamic range compared with, say, classical music, a degree of routine compression can make the vocal sit more comfortably at the correct level in a mix. Though vocals are the most obvious candidates for compression, most acoustic instruments work better in a pop context when their dynamic range is deliberately restricted. The same is true of electric guitars and basses.
One aspect of compression that causes confusion is whether it makes loud sounds quieter or quiet sounds louder. The mechanism of compression means that loud sounds are reduced in level, but most compressors have an output level control that allows any gain lost by compression to be restored or made up for. If you apply enough make-up gain to bring the signal peak levels back to where they were before compression, the quieter signals will be louder than before, so you can think of compression as both a way to make loud sounds quieter and to make quiet sounds louder. Figure 1 should help to explain this, as it shows an uncompressed signal, a compressed signal, then the same compressed signal brought up to the same peak level as the original. As quieter sounds can, in effect, be increased in level, compression has the effect of boosting the average signal level which, in turn, means that the average energy level is higher. This often results in a more powerful or punchy sound, even though the peak level is unchanged.
COMPRESSOR ACTION
A typical compressor comprises a gain control element, such as a VCA, and a photocell and diode arrangement, or an FET gain cell in series with the input signal. A second part of the circuit, known as the side-chain, monitors the input signal to establish its loudness or level. The signal level is continually compared with a threshold set by the user, and when the signal reaches or exceeds the threshold, a control signal is sent to the gain element to reduce the level of the signal. Though this might sound a little complicated in engineering terms, it's almost the exact equivalent of listening to a recorded track over monitors and pulling the fader down when you feel it's getting too loud. Indeed, manually controlling levels in the way I've just described is known as gain riding, and a compressor is simply an automatic gain rider. The problem with doing the job manually is that, unless you've played the track through and memorised exactly where the loud and quiet spots are, you'll always respond too late to changes in level, because you can't start to move the fader until you hear the start of the offending loud or quiet sound. Add to that the reaction time of a typical human being and you can see why you'll always be chasing the problem rather than curing it!
Before I explain how to set up a compressor, it probably makes sense to run through the various controls you're likely to encounter.
CONTROLS & TYPES
All the compressors I've ever used worked via a threshold system of one kind or another. With the simplest form of compressor, life is very black and white -- if the signal is below the threshold set by the user, nothing happens to it, but as soon as it reaches the threshold, it is turned down by a specific amount. In the case of what's known as a 'hard-knee' compressor, the threshold level is well defined, but in a so-called 'soft-knee' compressor, the gain reduction is introduced more gradually.
• RATIO: The ratio control is very important, because in most compressors this determines the severity of the gain reduction to be applied once the signal reaches the threshold. The higher the ratio, the more gain reduction is applied and the stronger the compression effect. If the ratio is made high enough, the signal level can, in effect, be prevented from ever getting past the threshold, and this situation is known as limiting. Though a limiter requires a theoretical compression ratio of infinity:1, any ratio above around 10 is so close to true limiting that it is usually referred to as such. Because most compressors have enough ratio range to allow them to be used as limiters, they are often termed compressor/limiters.
Ratio is defined as the number of dB by which the input level needs to increase to cause a corresponding 1dB rise in output level. If, for example, a compression ratio of 5:1 is set, an input signal exceeding the threshold by 5dB will cause only a 1dB increase in output level, as shown in Figure 2.
• HARD KNEE: As touched upon earlier, a conventional compressor has no effect on signals that are below the threshold, but as soon as they reach the threshold, gain reduction is applied at the ratio set by the user. This is known as hard-knee or hard-ratio compression because the onset of compression is sudden and occurs as soon as the threshold level is reached.
• SOFT KNEE: Because hard-knee compressor can sometimes sound a little abrupt or heavy-handed, the soft-knee compressor was developed. With this type of compression, gain reduction starts a few dBs below the threshold, but at a very low ratio. As the signal gets close to the threshold, the ratio increases, until at the threshold the ratio is that set by the user. Usually the ratio increases over a range of 10dB or so before the threshold is reached. This type of compression isn't quite as positive as hard-knee compression, but in some applications it can sound smoother and more musical. Figure 3 shows the characteristics of a soft-knee compressor.
Soft-knee compression is often used when the compression needs to be 'invisible', such as when you're keeping a mix level under control, whereas hard-knee compression is used in situations where it doesn't matter if you can hear the compressor working. Indeed, the audible side-effects of hard compression are often used as production devices to make vocals or specific instruments stand out in a mix.
TIME CONSTANTS
Earlier, I compared compression to the manual process of pulling a fader up and down. Just like the human engineer who does this, a compressor side-chain has a finite reaction time. It may be a lot faster than a human, but it's still true that a conventional compressor can't start to pull the signal level down until it has reached the threshold, and if that signal happens to be a snare drum with a near-instantaneous rise time, the compressor has to work incredibly fast to prevent the sound from shooting past the threshold level. In fact we don't always want to prevent the signal from overshooting as, in cases where brief peak overshoots aren't critical, the subjective result can actually be better than 'perfect' compression. For this reason, 'attack' and 'release' controls are provided to determine how quickly the gain is pulled down once the threshold is reached, and how long the gain takes to rise back to normal once the signal falls back below the threshold. Creating a deliberate overshoot by setting an attack time of several milliseconds is an effective way of emphasising the percussive nature of drums. Too short a release time can result in level 'pumping', while if the release time is too long quieter sounds following a loud beat may be reduced in level even further.
Setting the best attack and release values for a given type of material can take a certain amount of skill and experience, and if the programme material is constantly changing in dynamics, no one setting is going to be quite right -- which is why programme-dependent attack and release time were developed. An Auto function continually adapts the attack and release characteristics to the material being processed, by monitoring not only the input level but also the rise and fall times of signal peaks. Such systems can be very effective, especially on complex mixes or vocals.
If you were to set a very fast attack and a very fast release time, in addition to level pumping you might also end up with audible distortion, due to the fact that the compressor would be trying to work on individual cycles of the input signal rather than on its overall envelope. This phenomenon is particularly noticeable when the input signal is from a bass instrument, as the individual cycles are long enough to allow the compressor to respond. To get around the problem, it is necessary to increase either the compressor's release time or its hold time. Hold time is a short delay that prevents the compressor from going into its release cycle until a certain time has elapsed. All you need is a hold time longer than the wavelength of the lowest audio frequency and the problem is cured. Few compressors nowadays seem to include a variable hold control, but many have a fixed hold time built in, which ensures that the problem will never arise. If distortion does become audible at fast attack and release settings, and you don't have a hold control, you must increase the release time until the distortion stops.
SIDE-CHAIN SENSING
To continue comparing the compressor side-chain to the human hearing system... the compressor will always go by the average level of the sound rather than by the peak level, because the human hearing system tends to average out sounds in such a way that short, high-intensity peaks might actually sound less loud than a continuous sound at a lower level. That's one reason why the old-style VU level meter became so popular -- the sloppy response offered by VU meters is pretty similar to the way we humans perceive sound levels.
For a compressor to respond to averaged signal levels, it needs what is known as RMS level-detection circuitry. Such a system will invariably let short peaks slip by, and though this doesn't matter so much in the case of analogue recordings (where brief level excesses translate to brief increases in distortion) there are situations, such as when recording digitally, where peaks need to be better controlled. For that reason, some compressors are fitted with peak level detectors, which respond to signal peaks, no matter how short. In addition to keeping a better check on peak levels, these compressors can work better on drum sounds, where average signal levels bear very little relationship to what the signal is actually doing. Some compressors use RMS sensing, some use peak sensing, and some use a system that is somewhere between the two. Others give you the option to switch between one type and the other. Always try both settings if you're lucky enough to have a compressor that offers both. As a general rule, peak detection works best with percussive sounds.
STEREO LINKING
When you're compressing stereo signals it's necessary to ensure that both channels are subjected to exactly the same amount of gain reduction, otherwise the stereo image will drift from side to side whenever the signal in one channel is louder than that in the other. For example, if a loud sound occurs only in the left channel, the left channel level will be pulled back, and as a result the mix will appear to swing towards the right channel, where less gain reduction has been applied. The Stereo Link switch of a dual-channel compressor usually sums the side-chain inputs together, then controls both channels from the same side-chain. It may be necessary to set up both channel controls in the same way (the control settings are usually averaged, in this case), or you may find that one channel becomes inoperative and the other channel's controls affect both channels.
USING COMPRESSORS
 
continued

A compressor should be patched into a mixer via an insert point, or connected in-line between one piece of equipment and another. Compressors should not normally be used with aux sends. It is common practice to add some compression to a signal while recording and then apply more at the mixing stage, should further control be necessary. This approach makes good use of the recording medium's dynamic range and, to some extent, protects against unexpected signal peaks. However, it's usually better to apply a conservative amount of compression during recording, which means you won't get as much protection against peaks as if you were hard limiting. Having said that, if you apply too much compression there's no easy way to undo it afterwards. Likewise, if the compressor has a built-in expander or gate, this might be better left switched off during recording, as a gate which has been set up badly can completely ruin an otherwise perfect take. Furthermore, if you save the gating until you mix, any noise inherent in the recording medium itself will also be gated out. If the gate settings are wrong, you simply reset the gate, then roll the recording again.
SIDE-EFFECTS
Perhaps the most common shortcoming of conventional compressors is the unwanted modulation of high-frequency sounds, due to large amounts of gain reduction brought on by high-intensity bass sounds. In most music, especially electrically assisted music, the majority of the sound energy emanates from the bass end of the spectrum, obvious examples being the kick drum, bass synth, and bass guitar. Any high-frequency sounds that occur at the same time as high-energy bass sounds will obviously be compressed along with the bass, and it's quite common to hear hi-hats and other bright sounds being pulled down unnecessarily. One way to get around this is to use a multi-band compressor that applies different amounts of gain reduction to different sections of the spectrum. In practice, though, these are costly and rarely sound natural. A more pragmatic solution is to set a slightly longer attack time, to allow the attack of the hi-hat, for example, to pass through the gain-control element before any gain reduction takes place. How successful this is depends very much on the design of the individual compressor and on how much gain reduction is being applied.
It's surprising how much the sound quality of different compressors differs depending on their design and on the type of gain-reduction elements used. Tube and FET compressors tend to introduce a little even harmonic distortion, which has the effect of brightening up the sound, whereas compressors based on photocells tend to sound quite gentle. Even VCA-based compressors can vary greatly -- unsophisticated designs often dull the sound or appear to cloud the mid and high-end detail, whereas a really good VCA compressor can sound almost perfectly transparent. The main artistic differences tend to occur when the compressor is being driven hard, which is why certain models are valued for the effects they create rather than for their integrity.
SUMMARY
There are almost as many different compressor characters as there are compressors, but there are a few basic rules that can be applied to setting them up.
• LIMITING: If you want to use a compressor as a limiter, mainly to control excessive peaks, you need to set the threshold fairly high and use a high ratio. The signal will then be unprocessed most of the time, but when a peak does occur, it will be controlled very firmly. A fast attack and release time is best, though if the sound appears to pump you'll need to lengthen the release time until the pumping is acceptable.
• THICKENING: There are times when you want to use a compressor just to thicken up a sound, and in this instance it's probably fair to say that you want to bring up the level of low sounds. To do this, set a much lower ratio -- perhaps as little as 2:1, or even less, but set the threshold quite low so that you still get between 6 and 12dB of gain reduction showing on the meters. A longer release time may give a smoother sound, but every sound is different, so let your ears decide.
• SETTING UP: Once you've decided whether you want to thicken or limit, setting up a compressor is quite easy and you don't really have to think about the threshold level much at all. Once you've set the ratio, adjust the threshold control so that around 6-12dB of gain reduction shows on peaks and you'll have a good starting point. You can then adjust for more gain reduction if you want audible pumping, or back off the threshold for less gain reduction if you want to be subtle. Always adjust the release time to be as short as possible without pumping -- start out at between a quarter and a half of a second -- and start with a fast attack. For percussive sounds, lengthen the attack while listening to the result -- you should set it just long enough to give the sound a good transient kick, and to avoid obvious gain modulation of high-frequency sounds. Smoother sounds such as vocals can be dealt with using a faster attack setting or, better still, an auto setting if you have one.


DUCKERS
You may know that compressors with side-chain inputs can be used to make one signal control the gain of another -- most of us are familiar with this technique through DJs using duckers to enable them to talk all the way through our favourite records. Personally, I prefer to use a gate with a ducking facility to create this effect, as I find it more predictable in operation and easier to set up, but you can use a compressor by feeding the signal you want to control into the main input, and the signal doing the controlling into the side-chain input. If, for example, music is fed into the main input and a DJ's voice is fed into the side-chain input, whenever the voice level exceeds the threshold, gain reduction occurs at the ratio set by the user. Figure 4 shows how ducking is achieved.
Ducking DJs (other than literally) isn't very inspiring, but you can use the effect quite creatively in a mixing situation by forcing parts of the backing track to drop in level to make a solo or vocal more audible. It's probably not a great idea to duck the whole backing track, but keyboard pad sounds or rhythm guitars could be usefully dropped in level by a dB or two for the sake of a clearer mix. If too much gain reduction is used, the gain pumping will become noticeable -- but many '60s hits pumped like mad, and they sounded great. Part of using effects is knowing how to make them sound good by abusing them creatively.
Ducking can also be used to control the level of effects such as delay or reverb -- indeed, many effects units now include the facility to do this automatically.

COMPRESSION AND NOISE
For every dB of compression applied, the signal-to-noise ratio is worsened by 1dB, assuming that the make-up gain is set so that the maximum levels of the compressed and uncompressed sounds are the same. This isn't because compressors are noisy, but because the quieter parts of the original signal, plus any noise it may contain, will be raised in level by compression. It is possible to use a gate to keep noise levels down, but care should be taken to minimise the noise at source first. If noise is a problem, it's essential to use as little compression (gain reduction) as you can get away with.

DE-ESSING
Some vocalists are more sibilant than others, and what starts off as a mildly irritating trait can become magnified out of all proportion by the time you've used your best capacitor mic, added a touch of 6kHz EQ boost for that extra sizzle, compressed the signal, and added a bright reverb. Fortunately, sibilance tends to occur in the 4-8kHz part of the spectrum, making it fairly easy to identify.
There are some units around which do an excellent job of reducing sibilance. These are known as de-essers, and the best ones act like compressors, but they only cut the section of the audio spectrum where sibilance occurs. However, it is quite possible to use a conventional compressor as a de-esser, providing it has a side-chain input and you have a spare equaliser. If the equaliser is patched into the side-chain signal path of a compressor, and set so that sibilant sounds are emphasised, the result is a compressor that responds more vigorously to sibilant sounds than to ordinary vocal frequencies. For example, if the equaliser is set to give around 10dB of boost only to sibilant frequencies, compression of sibilant sounds will occur 10dB before it does in the rest of the spectrum. Figure 5 shows how a compressor and equaliser may be used for de-essing. The shortcoming of this simple approach is that when sibilance is detected the level of the whole vocal is dropped, not just the level of the sibilant part of the spectrum. For this reason, you need to set a fairly fast attack and release time for the compressor, and settle for only a moderate amount of improvement, otherwise the voice will sound 'lispy' every time the compressor operates.
Published in SOS April 1997 Thursday 24th March 2005
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They are are a couple of articles Ive collected from the net over the years.


Remember that there is no need to eq and compress everything. If you concentrate on tracking your instruments to absolute perfection, then the song will mix itself.
 
Holy Crap! Thanks for a lot of great info! I had to quit reading about halfway through the second post, as I'm at work and need to get back to work :D, but I'll be coming back to read the rest for sure. Thanks, Pingu :)
 
Hey Pingu,
I am new to mixing/recording, and to this forum.
I just wanted to say that was an incredible post.
Here have some "Rep Points" !!!!
I saved it in MS Word, so I can refer to it. :D
Thank you so much
Gary
 
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The things you find looking for something else

thnx pingu that was so informative and to think I was just doing a search on noise gates... found this thread at work and was so into it got into trouble for goofing off and being unproductive you are awsome :D
 
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