Basic Recording Tips

7/30/2019 Basic Recording Tips

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Compiled and Edited by Don Robertson

General Recording Tips

1. When initially recording your tracks, always print the hottest (loudest) signal possible to thetrack, but avoid distorting the signal. This will allow you to maximize the signal-to-noise ratioof the signal. You want the signal to be loud enough to mask any noise in the system but youdon't want it to be so loud that the signal distorts or clips. This is especially important when

recording in the digital format, analog recorders will be little more forgiving. Remember DigitalDistortion Bad.

2. Don't immediately reach for the EQ knob and don't over do it with the Reverb. These aretwo of the biggest mistakes. Rather than fiddling with EQ (equalization) if you don't like theway something sounds, try changing the source. If you are miking a guitar for example, trymoving the microphone around to alternate positions relative to the acoustic guitar (or amp, ifit is an electric guitar). Small adjustments can make huge differences in the sound.

3. Use multiple monitoring methods when mixing down and mastering your songs. This willallow you to reduce the coloration effects of your studio room. When you mix down or master

your songs, listen to the mixes on a wide variety of transducers (your headphones, the closefield monitors, your living room stereo, your car stereo, a cheap boom box, etc.). This willallow you to get the best overall mix that works in most situations.

4. Check your mix in mono(not just stereo). Make sure that elements of the mix don't simplydisappear due to cancellation.

Drum MachinesA Good TrickBuy, beg, borrow, steal or if you have too, rent a Hi Hat and a Snare. Mic them up and playalong with the machine thru the sequence. If you have never played drums this might take a

while, but the mix of live hats and a live snare mixed with the drum machine really helps ingetting a more real sound.


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Studio Monitors

A Little History LessonIn the early days of studio recording, large monitor speakers were almost exclusively used.Unfortunately, they also required high powered amplifiers and expensive acoustic treatments(often poorly installed) to the entire control room. Still, a well constructed big monitoringsystem really is impressive to listen to, something not overlooked by studio owners whowanted to impress that high paying client.

Eventually engineers and producers learned that this was not always the best way toaccurately record and mix records because the average listener was listening to music throughtheir inexpensive home stereos, radios and cassette decks. Also, large monitor systems andthe cost for the required control room acoustic treatments were going through the roof. This isparticularly true for the budget limits of smaller project and home studios which started to

grow in numbers. A new way of accurate monitoring was needed; near and mid-fieldmonitoring.

Near and Mid-Field monitors

By their definition these speakers are intended for close in monitoring. The idea was toimprove the direct acoustic path between the speaker and the listener by making it shorter,giving less opportunity for the always present indirect (reflected) sounds to get in the mix (nopun intended). With near-field monitoring, the surrounding acoustic environment becomes amuch less significant. The same holds true for mid-field monitoring except there is a little moredistance placed between the speakers and the listing position. This result can be a largersound field along with something closer to that large monitor sound.

A good set of monitors properly located in a reasonably non-reverberating room and poweredby a 100 to 200 watt amplifier will yield surprisingly accurate results at budget cost. Even thebiggest studios use smaller speakers to augment their large monitoring systems. Today nearand mid-field monitors have become proven tools in the recording business, and thats what itis, a business.

Monitor PlacementWhile near and mid-field monitors are more forgiving of the surrounding room acoustics it isalways prudent to optimize the listening environment when ever possible. Be aware of theeffect that the size of the listening room can have on low frequency response. The generalguide line is that smaller rooms will have more bottom end, but still be aware of placement ofyour monitors in a large room. If you find that your monitors are to bass heavy or the reverseto light on the bass try moving them around within your listening room.

Placement Tips:1. Try to keep the back of your monitors at least 6 away from the wall.2. Try to avoid locating your monitors near reflective surfaces.The best way to deal with this is to place your monitors out in the room away from reflectivewalls and windows.


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Monitor SpacingConsideration should also be given to the physical spacing between the monitors and thelisting position. The general rule is: The distance between the monitors is equal to the distanceto the listener. In other words, the listener and the two monitors are at the three corners of a

triangle having equal sides.

Monitor Placement TipTrying to get the just the right stereo monitor set up? Get a hold of a small compact (makeup)type mirror, get your distances down (equilateral triangle placement at mixing position). Sitexactly where you would sit as you mix, have someone hold the compact mirror in front of the"tweeter" location on each monitor and "rotate" the monitors till you can see your image(face) in each one, one identical image to the other. You are now guaranteed that eachmonitor is pointed at your head (ears) correctly and you should have a proper "sweet spot".

Note: If monitors are placed closer than three feet, the sound from each speaker becomesdistinguishable separately, which is not what you want.


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Microphones Explained

BASIC TYPES

The most commonly microphones used in audio, "Dynamic" and "Condenser.

DYNAMIC MICROPHONESDynamic microphones are commonly found in PA applications due to its general ruggednessand simplicity of use (no need for phantom power or batteries). It works rather like a speakerin that there is a diaphragm attached to a coil of hair-thin insulated wire flexibly suspended ina magnetic field. Sound waves set the diaphragm and coil in motion vibrating back and forthwhich causes the coil to cut lines of magnetic force, thus a small amount of voltage is inducedin the coil.The voltage varies in polarity with the frequency of the sound waves and in strength with theamplitude or size of the waves (the louder the sound, the bigger the waves and the farther the

coil moves hence cutting more lines of magnetic force and generating more voltage). Thisvoltage travels down the mic cable to the mixer where it is amplified and sent to the speaker.For what it's worth, a speaker works exactly the same way only in reverse - it reacts to theamplified signal by vibrating back and forth to create sound. In fact, dynamic microphones andspeakers are almost interchangeable. Believe it or not, you can connect a raw speaker, awoofer for example, to the line input on a mixer and hook the mic up to the amplifier outputs.Talk into the speaker and sound will come out of the mic. It won't work very well and you maypromptly fry the mic, but this backwards PA will actually function (briefly).Dynamic mics are best for close-up use whether for vocals, instruments or instrumentamplifiers. Certain models are also preferred for bass drum and others for brass instruments.

CONDENSER MICROPHONESCondenser microphones offer high sensitivity and smooth frequency response. They operateon a small amount of DC voltage either from a built-in battery or a "phantom" power supplyunit, or from the mixer if it has phantom power built in. This is deposited as positive andnegative charges on two thin metal plates with a small airspace or other resistive materialbetween them. This forms the diaphragm cartridge.Sound waves cause the top plate to vibrate which alternately compresses and de-compressesthe resistance. It acts as a dielectric and a signal voltage is produced that varies in polarityand amplitude with the frequency and amplitude of the sound waves. This travels down thecable to the mixer and is amplified. It is worth noting that the phantom voltage will not harm

most dynamic microphones if they are connected to a mixer which has this feature built in -nor will the sound be affected.Condenser mic technology is ideal for virtually all applications with the possible exception ofbass drum. Certain models are designed to pick up sounds at a distance or groups of people,choirs for example. Other condenser mics are first choice for acoustic instruments, especiallyguitar, banjo, mandolin, violin, upright bass, piano or anything with strings. They are alsopreferred for overhead coverage of drum sets. At one time it was thought that condenser micswere too fragile for PA applications, however they have greatly improved over the years inthat regard with many models now designed for this kind of work which virtually equaldynamic mics for road-worthy-ness.


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PICKUP PATTERNS Directionality or Polar Response

Most microphones are capable of picking up sounds over a wide area however they don't pickall sounds with equal sensitivity. The all-important midrange and high frequencys approaching

from outside a microphones pickup pattern will be detected at far lower sound pressure levelsthose which are approaching from within the pattern and will get drowned out by them.Pickup patterns can be imagined as invisible balloons, each with a particular shape dependingon the microphone's design. These shapes are what you see listed as "polar patterns" in micliterature. Although the polar plot diagram is flat-looking, in reality mics pick up sounds comingfrom above and below as well as the front and sides and even the back

Cardioid Polar Pattern

CardioidThe sound that is picked up is from a more narrow area directly in front of the mic capsule.This is what is called a "Directional Microphone Pattern" It will reject sounds that are frombehind the capsule as well as sounds that come from the side. This is a good pattern for vocalmicrophones. Microphones that use this pattern are the famous Shure SM57/58.The various jellyfish like line patterns are the different frequencies and the different pick-uppatterns that those frequencies make. If you listen closely, not only will you hear theamplitude (volume) drop off as you go around the microphone but also the timbre (tone) of

the voice changes. That's because it's hard to make a cardioid microphone that affects theamplitude without affecting the timbre. This is call "off axis coloration".

Hypercardiod

Hypercardioid

The sound that is picked up is similar to cardioid but with a tighter area of front sensitivity anda tiny lobe of rear sensitivity.


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Omni Polar Pattern

OmnidirectionalThe Omni Polar Pattern (Omni-Directional) will pick up sounds from all around the room orarea. This is what is called a "Non-Directional" microphone pattern. It is used for large choirsand orchestras as well as drum overheads and pianos. This isn't a good pattern for live gigsbut many people do use them for live. You must be careful on where you place the mics asyou will get feedback.

Half-omnidirectional or Hemispherical

Half-omnidirectional or hemispherical: Picks up equally over a 180

spherical angle. This is the pickup pattern ofPZM (pressure zone microphone)

Bi-directional

Bi-directional

It is not very difficult to produce a pickup pattern that accepts sound striking the front or rearof the diaphragm, but does not respond to sound from the sides. This is the way anydiaphragm will behave if sound can strike the front and back equally. The rejection ofundesired sound is the best achievable with any design, but the fact that the mic accepts

sound from both ends makes it difficult to use in many situations. Most often it is placed abovean instrument. Frequency response is just as good as an omni, at least for sounds that are nottoo close to the microphone.


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MicrophonesApplications

Motor City Vocal Recording Standard- Get a large diaphragm condenser microphone (U-47 or 67 at the time)- Place the microphone at eyebrow level. 6 to 8 inches away from the lips, pointing at the lips- USE A POP FILTER

Compressors The Big Squeeze

Of all the processes used in modern music production, compression is perhaps the leastunderstood. One reason is compressions sonic results are often subtle and thus hard to hearespecially for budding engineers. Another hurdle is presented by the various and differingcompressor control parameters; those, too, are typically subtle in their individual sonic effects,and they work together interactively, further complicating the stew. Then theres the confusionthat lies in the bewildering array of product types and models the engineer must choose frombefore even reaching for a control knob. For example, for a given application, should youselect a VCA-based compressor or one controlled by an opto-electrical element? Solid-state ortube designed compressor or a hybrid of the two. Then there are the analog and digitalcompressors. A hardware compressor or one that is software based? And so on.With so many variables, its no wonder compressors and compression still remains a mysteryfor many users. Yet, if you want to master the arts of recording and mixing, learningcompressions intricacies is imperative. After all, the production processes for most of todayspopular music formswith the notable exceptions of classical and some jazzrely heavily on

compression. Simply put, if youre not compressing properly, youre not getting the bestsounds possible.This article will try to help you guide through the maze of compressor options and explainpractical compression applications in plain English. We will start with the basics ofcompression, citing examples of various production techniques and the theories behind them.This article will also tell you which features to look for in a compressor and why theyreimportant.

Improvement PlanCompression falls under the broader category of dynamics processing. The term dynamicsrefers to changes in loudness level, so dynamic range is the difference between the softest

and loudest sounds that a source produces, or that a track contains. A dynamics processorspurpose is simply to increase or decrease a signals dynamic range, which alters how the levelsfluctuate within that range. Types of dynamics processors include gates, expanders, limiters,levelers, and compressors.

A compressor is a type of dynamics processor that squeezes a signals dynamic rangethatis, it reduces the difference in volume, or level, between the loudest and softest parts of aperformance. The process of reducing volume is called gain reduction. Properly applied, gainreduction makes a performance sound more consistent from beginning to end. For thatreason, compression is a great remedy for a performance in which the levels fluctuate toowidely.


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By reducing dynamic range, a compressor also allows for the processed signals overall level tobe raisedthat is, become hotterresulting in increased loudness without pushing thesignals loudest parts into distortion.Bringing up the overall level has the additional benefit of making lower-level sounds louderthan they were before compression. The result is that subtle nuances such as mouth sounds

and ghosted notesas well as burps, string buzzes, and snare rattlesare louder, clearer, andeasier to hear.Of course, you may not want to make burps, string buzzes, and other incidental performancesounds more audible. Therefore, apply compression only when musically appropriatewhenthe end result will sound better than what you started with.

You can always add compression after a track is recorded (during mix down), but sometimes itis desirable to use compression during the recording process. That approach has severalpotential benefits. For one, a compressor makes it easier to capture usable tracks whenrecording an instrument with a wide dynamic range. Moreover, solving level-fluctuationproblems during tracking frees you from having to solve them at mix down. That, in turn,leaves more time and brain powernot to mention gearfor focusing on the mixs creativeaspects.For those recording to any digital medium, using a compressor during tracking ensures thatsounds are encoded at a higher level. Because more bits are used, better bit resolution results.Furthermore, by putting a lid on peaks, the compressor also helps avoid digital clipping onextra loud notes. For those recording to analog tape, compressing during tracking allows thesignal level to be raised higher above the noise floor, which results in an improved signal-to-noise ratio.

Tricks of the TradeIn addition to problem solvingsmoothing out rough performances, improving digital

resolution and signal-to-noise ratio, avoiding digital clipping, and the likeyou can also employcompressors in numerous creative applications. For example, a compressor can dramaticallychange the envelope of a sound in much the same way an envelope generator works in asynthesizer. That and other compression tricks can give a vicious attack to a lackluster snaredrum, add crunchy edge and sustain to a mild-mannered electric guitar, make a lead vocalsound so urgent that listeners will dial 911, or pump up an entire mix until the band soundslike its exploding out of the speakers.In simplest terms, think of a compressor as an automatic volume controller. Indeed, beforecompressors were invented, engineers typically had to ride gain on a channel to maintainconsistent volume levels. (Then again, many engineers still ride gain, even when usingcompressors.) However, a compressor controls levels with a speed and accuracy that is

impossible to achieve manuallysort of like a magic genie adjusting the tracks fader withlightning-fast reflexes. The compressors control settings determine when and how much thatfader moves.Depending on how its controls are set, a compressor reduces either transient peakstheshort-lived, attack portions of a soundor the average-level portions of the sound, andsometimes both. Examples of transient peaks include the stick strike on a drum head andguitar-string plucks.


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A sounds average-level portions include a snare drum shells ringing and the sustaining of aguitar note after it is plucked. Certain instrumentsa wood block, for instanceproducemostly transients and very little sustain. Others, such as vocals and organs, typically producemild transients that barely peak above their average levels.The number of controls on compressors varies greatly, depending on design, cost, and other

factors. Units that employ voltage-control amplifiers (VCA), for example, typically have at leastfive controls: threshold, ratio, attack time, release time, and output level. Full-featured VCAmodels may offer more than twice that many controls, whereas some expensive opto-electricalcompressors may provide only two control knobs.Note: Units with fewer controls are not necessarily less capable; rather, they typically provideautomatic control of parameters such as attack and release time, or they gang twoparameters (threshold and ratio, for example) on to one knob.

High Five the five controls common to most VCA-based compressors

Threshold is the level at which compression kicks in and starts to reduce the signals level, or

gain; the threshold control lets you set that level. With threshold at 0 dB, for example, allsignals at or above 0 dB get compressed, while those that fall below 0 dB are unaffected.Therefore, to control peaks, set the threshold to a level below the level of the peaks but abovethe average level of the signal. That way, peaks that exceed the threshold get attenuatedwhile the average levels pass unaffected through the unit. Clearly, a proper threshold settingis critical to a compressors performance: if the threshold is set too high, the unit will notprocess any of the signals; if the threshold is set too low, the unit will react tothat is,attenuateevery portion of the signal.

Ratio expresses the difference between signal increases (volume) at the compressors inputand increases at its output; the number on the left refers to input and the right to output.

Therefore, the ratio control determines how much the signal will be attenuated once itexceeds the threshold. For example, a 2:1 ratio will let a signal increase in level only 1 dB forevery 2 dB it exceeds the threshold. Likewise, if the signal exceeds the threshold by 6 dB at a2:1 ratio, the compressor attenuates the signal by 3 dB, a net gain increase of only 3 dB. Inthat case, the compressors gain-reduction meter (if it has one) will show 3 dB of gainreduction.Typically, different instruments and performances call for different compression ratios. Forexample, to compress a ballads near-perfect vocal track, a mild 2:1 ratio would probablysuffice; at that ratio, and with the appropriate threshold dialed in, the compressor tightens upthe performance enough to ensure quiet phrases are not lost in the mix and higher levels arenot overbearing. At the other extreme, a bass guitar track that alternates between mellowfinger-pad technique and aggressive pop n slap can easily have a huge dynamic range.To yield consistent levels from that type of performance, a higher ratio such as 10:1 may be inorder.Note: Threshold and ratio work together to affect a signals output level. The lower the ratio,the less control the compressor has on the signal; the lower the threshold, the lower the signallevel subject to compression. The relationship between the two controls affords flexibility andsonic variation. There are, two different-sounding ways to get the same amount of gainreduction out of a compressorlow threshold and low ratio or high threshold and high ratio.


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Attacktime is how long it takesmeasured in milliseconds (ms) or microseconds ()for thecompressor to kick in once the signal exceeds the threshold. A slow attack time lets inherentlyfast transient signals pass threshold before compressing the rest of the signal; a fast attackcatches transients, but may diminish high-frequency content.Something worth noting is that manufacturers sometimes measure attack times differently.

Some specify attack time as the time it takes for the compressor to react after the threshold isexceeded, and others specify attack time as how long it takes for the compressor to reach,say, 67 or 90 percent of the maximum gain-reduction level it will ultimately achieve.Fortunately, the exact definition is of little importance, as typically attack time is set by ear.Depending on what kind of effect youre going for, simply decrease the attack time until unrulypeaks are tamed or increase it until average levels are lowered and desirable peaks getthrough unscathed. If youre having trouble hearing your settings effect, watching adownstream peak-level meter (that is, one that monitors the levels after the processthecompressors output-level meter, for example) will let you visually confirm what portion of thesound is attenuated.

Release time is how longmeasured in seconds or hundredths of a secondit takes for thecompressor to return the signal to unity gain (its unprocessed state) after the signal falls backbelow threshold. That is, once the release time passes, the compressor lets the signal passthrough unaffected. In general, slower release times result in a more natural sound.In general, set fast attack and release times when you want the compressor to do its job andget out of the way quicklyfor instance, when you want to put a lid on transient guitar plucksbut allow the ringing notes to pass through unaffected. Conversely, a moderate attack timecoupled with a long release is perfect for those Santanatype guitar solos in which you wantnotes to sustain forever. At two seconds or longer, the extended release time causes thecompressor to slowly restore compressed levels to their original (higher) gain, just as thesustained notes start to naturally die off, which counteracts the decay and makes the tails ofthe notes louder.

Output Level a compressors last control stage. That control is also known as make-up gainbecause it is used to make up for the gain reduction caused by the compressor. The usualapproach is to increase the processed signals output level so it matches the unprocessedsignals level.That creates unity gain between the two signals, which makes it easier to compare them usingthe bypass switch and ensures appropriate levels when recording or mixing.

Path Not TakenA compressor can degrade or ruin an audio signal as well as enhance it; therefore, one of themost useful features on any compressor is the bypass switch, which lets you compareprocessed and unprocessed signals. After using the output-level control to balance the levelsof the processed and unprocessed signalsa critical step because louder signals soundbrighter and fulleryou can judge whether your control settings actually improve the sound byswitching the compressor in and out of the circuit.Fortunately, most compressors provide a bypass. Typically, this is a switch that disables thecompressor circuitry; ideally, it also disables the input- and output-level controls. A hardwirebypass is usually the best design because it routes the input directly to the output andbypasses all compressor circuitry, such as input and output amplifiers and gain-controldevices.


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You Dont Need a InputMany compressors also provide an input-level control, but those are often superfluousif notundesirable. For one thing, a compressor with a wide-ranging threshold control can handlealmost any input level. So an input-level control is necessary only if the threshold range is toohigh or too low to act on the input signal as is.

For example, if the thresholds highest setting is +2 dBV and youre feeding the compressor+12 dBV levels, youll compress most or all of the time unless you somehow lower the level atthe compressors input. Thats one instance in which an input-level control comes in handy.Conversely, if the threshold ranges minimum setting doesnt go down very low, thecompressor may not kick in when fed low levels. In that case, an input-level control isnecessary to boost the input to an appropriate level.The reason an input-level control can be thought of as undesirable is it adds yet anotheramplification/attenuation stage to the circuitry, thus degrading signal quality.

Knee-jerk ReactionIn addition to the controls and parameters already discussed, several more-subtle parametersand design features often figure prominently into a compressors performance or sound. Onesuch parameter is the knee, which is related to the compressors threshold control. The kneedetermines how quickly and smoothly the compressor will transition from no action to the fullratio of gain reduction set on the unit once the signal passes threshold. Generally, acompressors knee is hard or soft, though some units provide switch able hard- and soft-kneecompression.

Hard-knee Compression The unit processes the audio signal at the selected ratio once theinput signal passes the threshold. Although useful for applications such as peak limiting andde-essing, a hard knee can sound abrupt, especially with higher ratios.

Soft-knee Compressor (sometimes called overeasy) A compressor set to soft-kneecompression, begins to compress as the signal approaches the threshold level and graduallyincreases the ratio until the signal attains threshold, at which point it equals the selected ratiovalue. The gentler, logarithmic increase of soft-knee processing tends to sound moretransparent (less noticeable) than hard-knee compression, and thus is usually preferable formost vocals and instruments.In addition to manual controls for attack and release times, some compressors offer anautomatic mode, called auto mode that does some of the tweaking for you. That is oftenreferred to as program-dependent or adaptive processing. In auto mode, the compressorsdetector circuitry analyzes the program content (the audio-input signal) and dynamically

adjusts the attack and release times accordingly.Auto modes main benefit is it precludes the need to tweak attack and release settings onperformances in which the dynamics change radically. It also lets you set up quickly yet stillget good results when the pressure is on. The downside is you lose some control over thesound. For example, you may like those peaks when the guitarist picks harderin which caseyou probably would not want to use auto mode.Some compressors offer a semiautomatic mode of operation. As the name suggests,semiautomatic mode lets the attack and release settings exert some influence on the adaptiveprocessing.


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Opto-electrical compressors may or may not offer an auto mode; however, even without one,these units provide something similar to automatic processing in that attack and releasetimesmanually set or not change based on program content.That is due to the inherent nature of opto-electrical compressors, which in general are slowerand less exacting than VCA-based designs. Because the attack and release controls on optical

compressors provide only approximate response times, many manufacturers simply put fastand slow on either side of the knob, rather than hash marks indicating exact times.

Double DutyMost dual-channel compressors offer stereo linking, a feature that lets you run two channelsfor example, stereo acoustic guitar or even an entire mixthrough the compressor and haveeach channel be attenuated the same amount. That keeps one sides level from dipping morethan the other, which would throw the stereo image out of whack.True stereo linking works by having the channel that exhibits the most gain reductiondetermine the gain reduction for the other channel. Another form of linking establishes amaster/slave relationship between the two channels in which one side (typically the left) is the

predetermined master and the other follows its attenuation pattern.It is commonly said that compression becomes limiting at ratios of 10:1 and higher, but that isnot the entire story. Actually, the detector circuits in compressors and true limiters differ bydesign. A compressors detector circuit is usually designed to detect RMS, or average, levelsrather than transient peaks. Therefore, transient peaks almost always overshoot acompressors threshold level, no matter how high the ratio and how fast the attack time is set.

A true peak limiter, on the other hand, employs a detector circuit that responds to peakenergy levels and thus reacts faster.Whereas all true compressors use RMS-sensing detector circuits, detectors for different modelscan differ substantially in their reaction times. That means two different compressors set to

the same attack, release, threshold, and ratio values may nevertheless respond quitedifferently to the same signal. (That is one of the many reasons it is difficult to recommendspecific control settings for compressing various instruments.)

Chain, Chain, ChainEvery compressor has a side-chain detector circuit that sees when the threshold has beenexceeded and tells the compressors gain-control element or amplifier to attenuate the signal.The side-chain is not in the audio path; its merely a control that tells the compressor when toattenuate the signal. The circuits for threshold, ratio, attack, and release are also found in theside-chain.Full-featured compressors typically provide side-chain inserts on their rear panels. Think of a

side-chain insert as an effects loop that patches into a compressor directly before the detector;like the rest of the side-chain, it is not in the audio path, so its effect isnt directly heard. Side-chain inserts therefore let you process the compressors input signal before it reaches thedetector. That permits de-essing and other frequency-conscious applications.


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De-essingTo de-ess a vocal, first patch the send and receive from the compressors insert into anequalizers input and output, respectively. Next, boost the equalizers high frequencies and cutits lows and mids. That causes the compressors detector to hear the vocal as having excessivehighs. Whenever the whistling sound of sibilance raises its ugly head, the sensitized detector

circuit hears it much louder than it really is, causing the circuit to vigorously reduce gain in theaudio path. With attack time set to around 50 and release time between 50 and 60 ms, thecompressor can be made to quickly attenuate the sibilance and get out so the rest of the vocalis left unchanged. Of course, the compressors threshold must also be set properlyabove thevocals average levelsfor that to work.

You can also use a side-chain insert to make the detector react to a signal entirely unrelatedto the audio-input signal. The classic example here is ducking: a side-chain application inwhich an announcers voice is set to trigger a music beds attenuation. To set up this type ofducker, play stereo music tracks through a dual-channel compressor and patch the voice-overtrack (or channel) into the side-chain inserts receive jack. Next, set the compressor threshold

low enough that it responds to every vocal utterance. When the announcer speaks, thedetector hears the voice and instructs the compressor to lower the music bed

Freq ShowThe misconception that split-band compression is the same as frequency-consciouscompression is common. A split-band compressor splits the audio signal into two or morefrequency bands so each band can be processed by its own independent compressor circuitry(each with its own controls). That lets you compress, for example, a guitars bass frequenciesdifferently from the highs.

A compressor that offersor is set up to providefrequency-conscious compression is still afull-band device acting on the entire signal. The difference between it and normal compression

is simply that the detector is set to be called into action by the prevalence of specific, user-selected frequencies.

Ones and ZerosOne advantage of digital compressor is most of them offer the look ahead circuitry. Becausethe compression algorithm is in software, the compressor can analyze what it is about toprocess and place the attack time right at the onset ofor even beforethe sound, resultingin a zero attack time. However, while a super-quick (or zero) attack time is great for catchingtransients, it doesnt always sound the best. Therefore, use such power judiciously; the crackof a snare drum without any attack just doesnt sound right.In addition, digital compressors usually offer incremental control of every parameter

imaginable, as well as the ability to store settings for later recall. Perhaps the biggest benefitof working with digital compressors is the ability to stay in the digital domain. If youreworking with a digital audio workstation or digital mixer, there are strong arguments for notre-entering the analog circuits. Most importantly, by staying in the digital domain, you avoidthe signal degradation and distortion caused by multiple conversions.If youre considering buying a hardware digital compressor, make sure it has great-sounding

A/D and D/A converters. Its also helpful if the software is upgradeable through user-installable EPROM, CD-ROM, or some other user-friendly method. In addition, you should insiston a box with a word clock input. Without word clock inputs, you will be limited to using onlyone digital compressor at a time.


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Fresh SqueezedClearly, its important to choose the right compressor for the job at hand. With compressors itis not so much the design but the execution of the design that makes a compressor good orbad for a specific application. Be wary of any generalizations about compressors. For example,that opto-electrical compressors provide transparent and natural-sounding compressionas if

that were a given. But the fact is, some optos do and some dont. As always in audio, its thesound that counts, not the propaganda.

Starting from Scratch These general rules will get you started and prevent most processingmistakes. Once you have some experience, you can tweak settings for more extremeprocessing. Just remember: the rules are meant to be broken!First, make sure the compressor is switched on and set to soft-knee mode. If processing amono track with a dual-channel unit, make sure the stereo link or slave switch is turned off.

Also, disable or bypass any other special functions such as tube-saturation circuitry, expansion,and so forth.Next, set the compressors ratio to its minimum value, usually 1:1, and the threshold to itshighest value. Those settings render the compressor inactive but still in the signal path.Now, set up the compressor for unity gain throughput. Most units have hash markstypicallylabeled 0 dBscreened around the input and output control knobs. If your unit provides thosereference marks, set both knobs at 0 dB for unity gain. If no marks are provided, youll eitherneed to call the manufacturer to find the unity gain for each knob or use a tone generator inconjunction with the units input and output meters to determine unity settings. If thecompressor has no input meter, youll have to rely on the manufacturers word.To determine unity with a tone generator (the one in your console will do), feed a 1 kHz toneto the compressors input and set the input-control knob so the compressors input meterreads the same level as the tone generators output. Then switch the compressors meters to

show output levels and adjust the compressors output control knob for the same reading. Itsnot a bad idea to mark unity gain settings for future reference.

At this point, the compressor is set so that what goes in comes out unchanged in level. Yourenow ready to make ballpark settings for processing the signal.Set the attack and release time controls to an average value, usually close to the twelveoclock position, and the ratio to roughly 2:1 or 3:1. Those mild settings reduce the risk thatyou will over compress the signal.Switch the compressors meters to show gain reduction and lower the threshold untilapproximately 4 to 6 dB of gain reduction is attained on peaks. It is most important here thatthe lowest signal levels do not exceed the threshold and trigger the compressor. In otherwords, make sure the gain-reduction meters do not kick in during soft passages.

Once youve set the threshold, its time to start varying the ratio, attack and release time andbegin listening to the results. If you want more compression, increase the ratio; if you wantless, reduce it. Use fast attack and release times for compressing only the peaks. Use slowattack and release times to make a signal sound denser. Most importantly, let your ears be theguide.

After finding settings that provide the results you want, adjust the output control to make upthe gain that was lost to gain reduction. Of course, you can add more or less than that amountif you wishjust make sure youre paying attention to proper gain staging with regard to anydownstream gear. That is, dont boost the compressors output if doing so requires you tolower the input on the next device below its unity gain setting.


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Dialing in Hot SoundsAsk any experienced engineer for suggestions on compressor settings for various instrumentsand youre likely to be met with a blank stare. He or she is not trying to be evasive. The bestsettings depend on a host of variables: the type of compressor used, the units detector-circuitry response, the amount of peak versus average energy in the track you want to

process, the dynamics of the performance, what kind of envelope shape or sound, theoutboard gears noise floor, and on and on. Use your ears is a tired phrase, but ears are stillthe best tools to contextually evaluate the sound of dynamics processing.Just the same, here are a few ballpark settings for getting great sounds from some of thecompressors mentioned in this article. As always, use your ears to make additionaladjustments if the initial sound is not to your liking.

The Exciting CompressorNote: Motown developed a mixing method that allowed presence, bite and intensity on leadvocals. Even when the vocals were mixed at an even level with the music, you could hear

every word clearly. This is a great the technique.

The Pre-Motown MixIn the 1950's and early 1960's records were generally mixed with the vocal far louder than themusic. The vocal had a very natural sound to it except the there was a lot of reverb applied tothe vocals. The artist that really had this sound was Frank Sinatra. Back then, when listeningto Old-Blue-Eye's records, you heard the music way in the background. This sound, however,wasn't exclusive to Frank. Even the "Rock & Roll" records of the time, like Elvis & Ricky Nelsonhad the vocals way out front.

The Motown Mix

Berry Gordie had a "better" idea. Motown was selling "excitement." The thinking was that therhythm of the music is what made the record exciting and what the kids danced to. Thereactually was a lot of melody and important lyrics in these old records - but rhythm was thekey. Actually Motown started a revolution in mixing and most modern rock (and even pop)releases are mixed in this style, even today.Regarding reverb, another Motown innovation was to have more reverb on the music than onthe vocal. There were three custom built reverberation chambers at Motown - all used duringa mix - unheard of in those days. Again today a typical control room today has 4-8 (or more)effects devices for reverb (and other effects).

The 1970's "Exciter"In the 70's a processing device by Aphex called the "Aural Exciter", started gaining popularity.The exciter took any instrument and generated a high-frequency signal component that couldbe added into the mix and would add "excitement" to the sound. A lot of people wereimpressed with this device (and clone devices that followed) especially to make the vocalsound brighter. Here is how to use compressor for an exciter.


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The Motown 1960's Exciting CompressorWith the Motown mix approach there were problems. If you wanted the lyrics to be heard youhad to use a lot of compression on the vocal so that the softer words could still be heard overthe higher-level music. In addition you boosted the "presence range" (around 5 kHz) with anequalizer. The only problem with this is that it took the life & natural dynamics out of the

vocal.An engineer by the name of Lawrence Horn came up with this brilliant idea. He took the vocaland split the signal so that it went to 2 different channels. Before the vocal signal went to thesecond channel, it went through a compressor. Now he had two channels of the vocal - onecompressed and one uncompressed. On the uncompressed vocal he added very little with theequalizer and he added the reverb. On the compressed channel, he squeezed the S*#t out ofit and added a ton of high-frequency equalization. What he would do is bring up the "natural"channel to full level to get the basic natural sound on the vocal. On the other compressed andequalized channel, he brought this up just enough to add excitement and presence to thevocal sound.

The result was nothing less than amazing. In the mix the vocal sounded very natural andbright. None of the music ever "stepped on" the vocal and you could hear each and everysyllable in the lyrics. The vocal never got lost.

Using the Exciting Compressor.I don't know if anyone at Aphex knew anything about this technique - BUT - the purpose oftheir product and the older Motown technique seen basically the same. As you try thistechnique out you will find it works for other instruments as well. Often the frequency of EQneeds to be changed for the instrument. The vocal works well with tons of 5kHz to 8 kHzadded to the "exciting compressor;" guitars work better with 3 kHz - 5 kHz and bass guitarswork better with 800 hZ to 1.5 kHz.

For analog recording or working with an analog console, splitting the vocal into two consolechannels is easily done with a Y-chord or similar function at the patch bay. For digitalconsoles, it's a little harder; usually the best results are obtained by actually having two vocaltracks recorded on the tape.

COMPRESSOR FX & TIPSFairly common one this, you probably know it, but if you don't it's pretty simple. Getting asnappy emphasis in this way works with most sounds, but obviously has greater effect themore hard the attack of the sound. For Example set-up your compressor for a snare drum.Simply set a slower attack somewhere in the region of 1-5 ms. This allows the initial fastattack of the snare to bust through before the compressor kicks in to crush the sound...Thislittle technique also works great on bass synth sounds with a fast attack...


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COMPRESSOR SETTINGS

Sound Attack Release Ratio Hard/Soft Gainreduction

Vocal fast 0.5 sec 2:1 - 8:1 soft -3-8 dbLoud vocal fast 0.3 sec 4:1-10:1 hard 5-15 db

Acoustic G 5-10 ms 0.5 sec 5:1-10:1 soft/hard 5-15 db

Electric G 2-5 ms 0.5 sec 8:1-10:1 hard 5-15 db

K & Snare 1-3 ms 0.2 sec 5:1-10:1 hard 5-15 db

Bass 1-10 ms 0.5 sec 4:1-12:1 hard 5-15 db

Mix fast 0.4 sec 2:1 - 6:1 soft 2-10 db

General fast 0.5 sec 5:1 soft 2-10 db

AUDIO LIMITING Controlling Peaks

In digital recording there are extreme peaks that can cause the overall average level to below. If you are mixing down to analog tape, many of these peaks have been "rounded off" bythe tape. You can control these peaks with the LIMITING function of most compressors. Thisis accomplished by setting the ratio very high (10:1 or more). According to Ben Blau of RID:

"To achieve this, engineers often seek to use very fast attack and release times with a high

ratio and a hard knee. This will very quickly reduce the gain on the audio peaks, which areoften not noticeable to the ear. This is quite common in mastering, since it allows mixes to berecorded much louder on digital media, such as CDs without going into digital clipping. Inother words, -6dB of peak gain reduction will allow a song to be recorded twice as loud toyour ears on a CD!"

Overall Mix Compression

Similar settings can be used for the overall mix to get its apparent (to the ear) level up. Usesmaller ratios (up to 2:1) and longer release times. Compressors with an "overeasy" feature

or "soft knee" work the best.


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Guitar Amps Miking Speakers

MicrophoneOne of the best single microphones for recording electric guitar is the Shure SM57. I recently

read an article on Producer/Guitar Player Pete Anderson. He was asked what microphone hepreferred or recording electric guitar. He suggested trying to find an older Shure SM57s theones with Unidyne III wrapped around the head.

Single Microphone PlacementUse a Shure SM 57 microphone; put it up in front of the speaker. Just kidding there is a littlemore to it.

General PlacementThe most commonly used placements are; One, mic the center of the speaker, two, the sweetspot (see figure 1). Then there is number three, splitting the difference. Depending on the

cabinet/speaker being used it is best to try all three.Start dialing in a sound by moving the mic back and forth. Try positioning the microphonenearer and farther from the speaker. If the cabinet has no grille, you might start with the micas close to the speaker as you can get, which will give you a nice proximity boost in the lowend then try pulling back a bit to let the sound develop.

Axis PlacementTheres (at least) one more variable you can experiment with: the orientation of themicrophone on- or off-axis. Many engineers will immediately go for on-axis (with themicrophone aimed 90 degrees, or straight into the speaker) to get the best overall frequencyresponse. However, theres an infinite range of off-axis positions, with the microphone turnedat a slight angle to the speaker. (Actually, given that the speaker is cone-shaped, unless

youve accounted for the slope in the speaker surface, you may be slightly off-axis by justpointing the microphone straight into the cabinet.)Turning the microphone slightly or a great deal off-axis does two things. First, thefrequency response of the mic will vary depending on the on-/off-axis positioning check outthe polar pattern chart for any directional mic to see what I mean. Youll see that the high-frequency response, in particular, changes as you turn the microphone off-axis.Second, youre changing what the front full-range response part of the mic is seeing. Ifyou rotate off-axis toward the center of the speaker, the front of the mic will see more of thatas the main source (and again, take into account the slope of the cone of the speaker, andwhere it is hitting the mic to determine just how far off-axis you really are).


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EXPERIMENTTry re-amplifying the guitar signal into the amp. That is, record a dry guitar signal using adirect box into my recorder. Then use an interface box there is a variety to chose from tosend that direct recorded track into an amp/speaker. This provides you with the exact sameguitar performance each time. If you dont have a interface box, try to play (or have your

guitar player play) as close to the same thing as possible with each pass. You want to hear thesonic difference that changes in mic position make, not variations in the guitar performance.Record a take with the mic aimed straight on. Angle the mic a little bit off-axis dont changeanything else. Record another take. Angle the mic a bit more, record another take. Continuethrough a good range of mic rotation. When youre finished, youll have a session where youcan A/B among the various mic angles. Make sure you take good notes, so when you listenback you know what youre listening to.Now put your ears to work. Which position sounds best to you? Keep in mind that whatsounds best in isolation may not be what works best in the context of a song. And maybe thetone you like best is with the mic straight on if thats the case, wonderful!

Add a Room MicrophoneIf you're going for a more open and ambient sound try incorporating another microphone,preferably a wide diaphragm condenser. Move the microphone back four to eight feetdepending on your needs. In this case the microphone can be pointed in the general directionof the speaker. Again start dialing in a sound by moving the microphone back and forth toachieve that sound you are looking for.

THATS STILL NOT ALLOne final tip: The idea of turning a mic off-axis isnt limited to guitar speakers. Any time yourepositioning a mic on a source, how you orient it on- or off-axis will make a difference in the

tone. Learn what your microphones sound like as you turn them, and youll find the expandedcolor palette to be useful in many situations.


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Top 10 Guitar Recording Mistakes

1. Forgetting to check for mono compatibility. You love your cherished, vintage Axe BlasterFlanger with its super wide stereo spread. Ah, but the way they get that stereo spread is byflipping the phase 180 on one of the output channels. This may sound great live, but when

the signal gets re-combined in mono, portions of it (maybe even all of it) will disappear. Ouch.This can also happen with stereo microphones on a single sound source, so always check whata track sounds like in mono before you sign off.

2. Stringing along with dead strings. Yes, change your strings before that important recordingsession and no, adding compression to increase sustain is not a suitable substitute. With newstrings, your axe will sound brighter, notes will sustain longer, and tuning will be moreconsistent. Dont just boil them go ahead and splurge, spend the $2-$4, and re-string.

3. Using automatic double tracking instead of playing the part twice. Its that popular presetin your multi-effects: Automatic Double Tracking, where the processor copies your signal,

delays it a bit, detunes the copy to humanize it, and then recombines it with the straightsignal. Although ADT is a valid effect in its own right if you want a sort of more focusedversion of chorusing, nothing substitutes for doubling a part by actually playing it twice.Furthermore, when you record each part on a different channel, you can spread the stereoimage one track more right, the other more left for a bigger, more enveloping sound.

4. Mixing direct and miked signals without compensating for delay. Heres the deal: Soundtravels at about one foot per millisecond, while electrons move at 186,000 miles per second.So the miked signal arrives at your mixer at the speed of sound, while the direct signal arrivesat the speed of light. If the mic is one foot away from your speaker, zoom in on the tracks andshift the miked signal ahead in time by about a millisecond until they line up. Youll hear a

much fuller, punchier tone. This is particularly important with bass.

5. Falling into a mic rut. You found a condenser mic that sounds great on acoustic guitars,and have a favorite dynamic mic for amps. And youve used them forever. But maybe youneed to experiment. Why be normal?

6. Not orienting an electric guitar for minimum noise. Pickups are appropriately named,because they pick up a lot more than strings like buzzes, electrical hash, dimmer noise, andthe like. The good news is that the pickup is directional, and changing the guitars position canmake it less prone to picking up garbage. Dont use your ears; look at the meters, because thelevels will be really low. If the noise is hitting at 45dB, it may not be that obvious, but it will

be if you start adding effects like compression. Try moving the guitar position, and you may beable to get that noise down to 55 or even 60 dB.

7. Turning up your amp too high. We all know that you need to turn an amp up to a certainpoint to get a good tone. But dont go past that point widely known as 11. Why? Aside fromthe possibility of overloading your mic, things in the room will have more of a tendency torattle, and poor room acoustics may be overemphasized. As Johnny Cochran once said, Onceyou get your tone, leave it alone.


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8. Forgetting to bring a spare set of tubes. Tubes fail, tubes go soft, and they sometimes do itat in opportune moments. . nuff said. Remember, if one tube of a matched set fails, youneed to replace them both. Its a good idea not to trust the tubes you buy, but to try them outimmediately in your amp to make sure they actually work. Once youre satisfied theyre okay,pull them out and save them for when theyre needed.

9. Not paying attention to tuning. This doesnt just mean tuning up before the session; we allknow thats a good idea. But have you adjusted bridge intonation lately? Just changing stringscan be enough to throw the intonation out of whack. You may not notice that theres anyproblem until you start recording, and everyones listening to your guitar under the audioequivalent of a microscope. In my experience, few things can destroy a session faster thanhaving to adjust intonation on a guitar with dead strings (mistake #2), because it will be nextto impossible to get it in tune.Tempers will fray, harsh words may be exchanged. And while youre at it, leave a tuner in-lineat all times, or use the tuner in a piece of software (e.g., Native Instruments Guitar Rig andCakewalk Sonar both have built-in guitar tuners). Its better to take 30 seconds to check

tuning before recording a part than having to re-record the part because the tuning was off.

10. Using a stompbox with an AC adapter. or for that matter, with batteries. If you record witha stompbox that can use batteries or AC, try both and see which sounds better. With some oldstomp boxes, the AC adapter might add some noise or buzz that batteries will eliminate.Conversely, if the batteries arent super-fresh, the lower voltage may degrade tone. Moral ofthe story: When you show up at the session, bring both the AC adapter and a fresh set ofbatteries.

Of course, there are plenty of other mistakes that guitar players make in the studio, fromsnorting cocaine to bringing in annoying people who arent a part of the band. But if youre

working with an engineer, one of the biggest mistakes is not letting the session evolveaccording to the engineers working style. Your job is to play a great part; the engineers is torecord. Dont worry too much about any fine points that should be reserved for the mix (not

fix it in the mix, but perfect it in the mix). Give the engineer a lot of space, and dont try todo two jobs at once. If youre really concerned that the recording isnt right, then record a drypart so you can re-amp later if necessary.


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Recording Acoustic Guitar

While the acoustic guitar remains one of the simplest instruments by design, it also remainsone of the hardest to get a great sound on in the studio. Here are three basic rules torecording acoustic guitar.Rule 1 - Use condenser microphones. They will always sound better than dynamicmicrophones for acoustic guitars. There are several condenser microphones that are currentlyon the market. Some start as low as $125.00 that can sound great for acoustics.Rule 2 - Strings: New strings will always sound better than old. Heavier gauge strings soundBigger. Lighter gauge strings sound brighter.Rule 3 - A limiter/compressor will almost always help you get a better sound.

Note: Any standard round-hole acoustic guitar has a danger zone: The sound hole pumps outa solid column of boomy low end. Avoid placing a mic in that column, or even aiming onedirectly at the sound hole, and youll get better results.

STRUMMEDHow you mike a strummed steel-string guitar depends largely on how the track will be used.These types of tracks can range from gentle strumming accompanying a vocal to hard-drivingstrumming in a rhythm section.

Many engineers start with the standard position: a large-diaphragm cardioid condenser micplaced 1224" away, at guitar neck level, slightly to the players left (assuming a right-handedplayer), and aimed at the point where the neck meets the guitars body. If the strummedguitar will be solo, you may want to add a second mic slightly off to the right of the player.

Aim the second mic at or behind the guitars bridge sometimes you might even aim slightlyin front of the bridge for a brighter sound with more pick attack. If youre in a decent room,

consider adding a spaced pair of distant microphones to add depth to the sound.

For a hard-strummed part, you could use a condenser mic in the standard position aimed atthe neck/body joint, but also experiment using dynamic microphones. Try using the standardShure SM-57 or similar models. You wont get as much detail or bottom, but youll hear a full,midrange sound with a lot of attack and drive. This can be perfect for rock styles where fidelityis less important than punch. In general, one mic is fine for this style; you want the guitar tosit in the track and drive it. The subtleties of room or fancy stereo miking will be lost or clutterup the track.

FINGER PICKED

For a finger picked guitar, especially one that will be used solo, you want to capture all thedetail of the performance, with good dynamics, solid midrange presence, and full bottom end.For this a large-diaphragm condenser microphones work best, although good results can behad using small-diaphragm condensers. A finger picked guitar can be quiet and delicate; lookfor a clean microphone and a preamp with plenty of gain.

The standard acoustic mic position mentioned previously is generally a good starting point.In some cases, with a little adjustment of the mic position, this may be all you need, especiallyif the guitar will be in a mix with other instruments. Try augmenting that mic with a stereo pairpulled back to get some room sound, and give sonic depth and space.


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CLASSICAL GUITARTo record a true classical guitar performance (as opposed to a nylon-string guitar played inanother style, such as jazz or pop), purity is your priority. With a classical recording,documenting the performance is usually the goal. So, choose microphones and preamps thatare clean and uncolored. You don not want high-end hype or too much midrange presence

boost. Often classical recordings will have the microphones set farther away from theinstrument/player than recordings for other styles. This means that the room will play a bigpart in the sound try to record in a good one.

Classical guitars are miked in mono and stereo with microphone(s) placed as far as five or sixfeet away, pretty much directly in front of the instrument. The idea is to capture what theaudience would hear. The classical guitar is a low-volume, delicate instrument. Place yourmicrophones too far back, and youll have too much room. You may also be forced to use somuch gain that mic or preamp noise become an issue.

Try recording classical guitar with a spaced stereo pair of large diaphragm condenser

microphones in cardioid or omni pattern placed back about three feet from the instrument,and spaced about three feet apart. The result is bigger and more present than many

traditional classical recordings. Consider adding a mid/side position from three or four feetback this will add control over the stereo width of the final tracks, and are set up for mid/sidedecoding.NOTE: Use a stereo miking technique if the guitar will be solo; if its a duet with anotherinstrument, try using one mic for a tighter sound.

That SoundCountry/Pop

For that Eagles "Lyin' Eyesstrummed sound. Place the microphone about 6 to 8 inches fromthe guitar's sound hole, but angle the microphone toward the area where the fret board andthe sound hole meet. If you point the microphone directly into the sound hole, it will be veryfull probably much too full, and very boomy. Use a compressor/limiter to knock down anypeaks (3:1 ratio), and set the threshold a little lower to give it a slightly "squashed" or tightersound. Set the threshold higher to just limit the peaks and give a more open sound. You mayneed to EQ out some bottom end Boom. If so, try rolling off some bottom (100Hz), or cuttinga couple of db at 300Hz. To add some "silk" on the top end, try something in the 8-10K range,but be careful, too much will add noise to the track. Positioning the mic so it angles toward thepick will give more attack-less sweetness.

Eric ClaptonFor his classical/gut-string guitar sound. Use a condenser microphone and place it about teninches away from the guitar, about 3 to 4 inches up the neck, but aim it at the players pickingfingers. This angle will reduce boominess by virtue of the microphones cardioid polar patternproducing a natural roll off when it's aimed off-axis, while simultaneously delivering the attackof the fingers. Try and say that three times in a row! The added distance will pick up some ofthe guitar body's resonance. A compressor is a must for this case because of unexpectedpeaks. A 4:1 ratio is a good place to start, but set the threshold fairly high so that the most ofthe guitar's natural dynamics are left in tact.


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Some More Approaches to Recording Acoustic Guitar

Use this especially for tracks that need a big sound, or if the guitar is detuned and you needdeep bottom end.Place one mic in front of the guitar, a bit further back than the standard position. Position asecond mic to the players right, and slightly in front, so it forms an equilateral triangle with

the players right ear and the front microphone. Experiment with the right microphonesposition; try it at knee level, looking up toward the guitar body behind the bridge, or at earlevel looking down at the guitar body behind the bridge. The meat of the sound will comefrom the front mic, but placed correctly (move it around, youll know when you hit the rightspot) the right-hand mic will fill out the bottom end with tight, full, round lows. Use thisapproach with steel- and nylon-string guitars; for nylon, pull the right-hand mic back, or turn itdown in the mix a bit.


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Here are a few other approaches to try:

Stereo small-diaphragm microphones, 1824" back, the same distance apart, with themicrophones pointing a few degrees in toward the guitar body. Mid-side from a few feet back.

For those into overkill: one large-diaphragm condenser mic centered on the guitar, twospaced small diaphragm condensers a few feet back and a few feet apart. Two roommicrophones, pulled back enough to capture a balance of guitar and room. For addedversatility, track a pickup on the guitar as well. Use the large-diaphragm with some of thepickup as the main sound, use the spaced small diaphragms to give the sound width andrichness, and then mix in the room microphones for spaciousness and depth. This will alsowork well for surround applications. Warning: Getting all those tracks in-phase may take somework.

Positioning the MicrophoneTry getting down on your knees and position your ear as if it were the microphone while the

guitar is being played. Move your ear around to find "sweet spots". Here is the Da factor -Please don't try this with electric guitar! If you have help on the session, have them move themicrophones around while you listen on the studio monitors.

Nashville Acoustic TuningThe Low E A D G strings are replaced with thinner strings tuned one octave higher thannormal, such as the extra (illusion) string on a 12 string guitar. This formula is used a lot by

the Rolling Stones. It is a great trick when you want to record more than one acoustic.

Mixing Acoustics

When mixing acoustics guitars for rock or alternative tracks, you will usually have an electricguitar or two in the track as well. Try to pan the acoustic and electric across from each other.Send one full left, and the other full right. You'll quickly discover that the electric willoverpower the acoustic and the most effective way to even them out is to compress theacoustic a little bit more than what you may have already done going to tape so you can bringthe acoustics level up high enough to compete with the electric.

Another simple but effective trick is to have the acoustic and electric guitars play parts thatcounter each other rhythmically (giving them each their own space), and have them each playin a different octave. That will give you a full sounding track that remains open and airy at thesame time. You can also make an acoustic guitar sound bigger or more rock-like by panningthe original to one side and a delayed signal (short delays are best) of the same guitar to the

other side. That effect can be taken one step further by using the pitch change option on yourdelay to "de-tune" one of the guitars just a pinch (one cent is a good place to start). The delaywill provide the brain with the psycho acoustic information it needs to perceive the guitar asbigger, while the pitch change will make it appear "fatter."


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Recording Bass GuitarRecording a Bass Guitar track without a groove no matter how well recorded is Useless.

BASS-IC INSTINCTSThe central message is this: your goal when recording bass should be to get as clean and full-

frequency a sound as you can. Keep the signal path as short as possible.Get your tone from the instrument and the components you're using rather than with EQ.

You're better off saving effects and heavy compression for the mix, where you can messaround with the track to your heart's desire. That way, you have the option of going back toyour original sound if you want to.

Keeping The Bass Guitar

In its first octave, the bass guitar generates fundamental frequencies between 40 Hz and 80Hz. Thus you could say that the fundamental notes that the bass puts out are between 40 and100 Hz. The instrument also puts out harmonics between 200 and 400 hertz, two octaves up.

If you take an equalizer and dip, using a shelving curve at 100 Hz and reduce the first octaveof the bass guitar, all of the harmonics of the instrument become accented. You can also"replace" the energy lost in the bass by accenting 300 Hz with a boost of about 5dB. Once youdo this you will find the instrument sounds like this:

1. It has adequate lows and body at loud listening levels. If you cannot saythis, reduce the first octave roll off to 1 or 2 dB.

2. The bass will have a more even sound as it plays different notes, oftenmaking a compressor unnecessary to even out the bass line.

3. The bass guitar part will be very distinguishable at low listening levels.

2.5 Methods to record a Bass Guitar Track.

Method 1You will need a (DI) Direct Box to change the high impedance signal of the bass guitar into alow impedance signal that your console's microphone preamp needs to see. The DI will alsohave a thru plug that you can use to go to an amplifier.The thru is a dry signal and is the same as if it came directly out of the bass. Add gain to your

signal by using the mixing consoles microphone preamp's gain control. Use compression whenrecording bass. Adjust EQ as necessary for the sound and style of the music. Generally thewarmth area of the EQ spectrum is down in the 100-200 Hz area. Watch your recording levelsand be careful not to overload.


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Method 2Start off with the bass's volume at 10 and the compressor set to 0 dB with the ratio at 1:1.Have the bass player strum some loud three-note chords, and set the preamp gain so thatyou're just overloading your recorder. That way, when the bassist plays normal lines, you willhave plenty of headroom. If your bass player is planning to play loud, strummed chords in the

song, turn the preamp down far enough to let those loud phrases pass through it withoutoverloading the recorder.Next, have the bass player play along with the track (or the band, if you're recording live).Make sure the drums especially the kick are loud and clear in the player's mix. Turn downanything that won't help him or her lock tight to the groove.If at that point you feel that the bass is uneven, kick in the compressor. Try a 2:1 ratio tostart. Keep the compressor attack slow enough to let you hear the attack of each note. Keepthe release fast enough so that each note is not affected by the note before it. Be careful: ifthe release is too fast, the compressor will chatter and distort as long notes sustain. (Ifyou're using two compressors as part of a DI-and-amp setup, start by setting the compressors

similarly, then fine-tune them to taste.)You should shoot for 3 to 4 dB of gain reduction. Remember, you can always compress moreat mix time. You will probably have to increase the output gain of the compressor slightly tocompensate for gain reductions.

Then there is the Bass POD and other direct recording devices. If you have access, givethem a try. Many good recording have been made using these handy little units.


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Recording Vocals

Unless you have the proper isolation necessary to keep all the other live tracks out of the vocalrecording. Meaning: Not At The Same Time As Everything Else. However, recording a scratchvocal during the tracking process can often help the musicians concentrate on the song at

hand, and allow them to get the cues necessary to record a great track. In this case you canhave the vocalist in the control room or in another adjoining room listening on headphonesand singing into any available microphone.Pick an isolated room or corner away from any noise source. (Furnace ducts, etc.) Put down acarpet and cover your music stand with a towel or rug to avoid reflections. If the area is openand too live, rig some diffusion by putting up a curtain, blanket or a rug on opposing walls.Before the session, if you have the resources try a number of microphones. Make sure thearea is well lit. Check the headphone mix yourself on the same headphones the singer(s) willbe using. Don't take for granted that the mix sounds the same in the studio as it does in theControl Room. Keep the session moving, avoid re takes because of engineering mistakes.

Positioning the microphone is crucial in getting a clean sound without any plosives. A popscreen will be needed in addition to precise positioning. Plosives are the Ps and Ts that rufflethe diaphragm of the microphone, causing unwanted low end information to get onto yourrecording. Some repositioning by the singer to the microphone may be necessary to achievethe tonal balance you are looking for.

Recording Background Vocals

There are probably as many background vocal techniques as there are background vocalists.But there are many recording techniques that you may not have tried before. Here are some

other options for recording and miking background vocals.Probably the most widely used background vocal recording technique is having the vocalist(s)go out and sing the background parts into the same mic that the lead vocalist just finishedsinging into. Its probably the most common because its the easiest. But its not always thebest option.

LAYERINGThis is another variation on the one-voice/one-mic easy method outlined above. Change upthe recording path. Instead of just using the lead vocal mic and signal path, use somethingelse. Set up a different mic, (through a different preamp if you can) so theres not so muchlayering of the same characteristic sound. A bright lead vocalist might sound great for one

track, but three or four stacks of that same brightness could easily be overwhelming.When recording a group background vocal, you can use the same one-mic technique, but ifyoure using a cardioid mic, the outside singers may lose definition since theyre off-center onthe mic and there is typically less presence as one moves off-axis. Try using the mic in awider, hypo-cardioid (as opposed to hyper-cardioid) pattern or better yet, try switching themic to the omni directional pattern. This delivers excellent results because most microphoneshave flatter frequency response in omni and sound much more real.


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Two things to watch for when using an omni directional pattern; both related to the room.First, make sure any boundaries (walls, windows) are far enough away so as not to createcomb filtering due to reflections. (rule of thumb is 3:1, but try to be even farther than that.)Right in front of the control room window, where the vocalists frequently sing, is notorious forreflections into the back of the mic. Second, watch out for the sound of the room. While it may

be flattering on the first and second pass, by the time you layer five or six tracks, theroominess may overwhelm the direct vocal sound.

TWO OF A KINDWhen using two singers, you can have them sing into one mic, or give each vocalist their ownmic. Two singers into a single mic is most common, but if the mic is directional, yourecompromising the presence of each singer since they cant both be on-axis on the mic at once.If they balance well, the best method of getting absolute presence is to have themsimultaneously sing into opposite sides of a figure-8 patterned mic. Each singer can get asclose as they want, but without the low-end build-up that using two cardioids can give you.This works very well as long as you have two singers who can balance themselves, or if

theyre singing in unison.

The balance between the two voices is decided by distance from the mic and volume of thesingers. Make adjustments in the volume by having the singers move closer or farther fromthe mic. Make sure to put any reflective surfaces to the sides of the mic, which have thegreatest rejection. Ribbon micphones are quite good for this, especially if you have a less-than-wonderful sounding recording space, because theyll pick up less room than a condenser.

THREES A CROWDWhen recording a background trio, its fairly common to have them gather around and singinto the lead singers cardioid mic. In this case, only one of the three singers is truly on-mic

and the other two are just filling in on the sides. The presence difference can be shocking.(Try it: Listen through headphones as you sing or speak into a directional mic while movingfrom on-axis to off-axis you may be surprised at how much sonic difference being off-axismakes.)For stacking background vocals with a group of three, try this: Get two variable-patternmatching microphones and set them up as an M/S pair. Put the mid (M) mic where you wouldnormally put the mic in front of the three singers. Then put the side mic (S) above or below it.The best way to do this is to get the vocalists in tight around the microphones (within 12" 24") to maximize the stereo image. Try positioning the singers at 9:00, 11:30 (off-center leftin front of M), and 3:00. Then pan the stereo outputs from the M/S pair hard left and hard

right. Youll hear one singer left, one near-left, and one right.On a second pass, flip the pans and youll hear the opposite: one singer right, one near-right,and one left. The main advantage to this, over a single mic panned hard left and right, is thatyou dont end up with mono left, center, right. You end up with a stereo background vocalgroup that surrounds the lead vocalist without occupying the same space. Put the singer at11:30 whose part is farthest away from the lead singers melody, as their part will crowd thelead the least. By moving the singers around the microphones, you can fill the space from leftto right and still leave room in the middle for the lead vocalist.


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Recording Drums

Always check your drums in mono. If anything in the kit seems to disappear, then something'sout of phase. Be systematic in tracking down the culprit. If you follow this prescription closelyand then, and only then, start to experiment with slight modifications of positions, level and

eq, you'll find yourself getting a drum sound that just might sound professional. Individualdrummers have drastically different levels of "feel," and feel is very important to the sound,sometimes more important than the drums themselves or anything you can do at the board.The farther the mic is out it is from the head, the roomier the sound, but the more potentialyou have for phase problems.

Equalization for Drums12.5 kHz - Air (transients) Region Above6.3-8 kHz - Snare Crack Peak3.1 to 2.5 kHz Midrange Harshness (strongest)800 to 1.2 Hz Adds Papery Tone500 Hz Prominent Head Ring315 to 400 Hz Adds Woody Character to Drum125 & 250 Hz / 200 Hz strongest - The Meat (fundamental)80 Hz Lowest Useful Overtone

Snare DrumFor the snare drum, the always a safe and highly effective choice is the venerable Shure SM57microphone. Set the mic up at a 45 to 60 degree angle with the capsule about an inch or twoabove the head and about two inches from the side, again pointing at approximately a 45-degree angle into the middle.

Hear is an interesting tip. Try pointing the microphone at the drummer's crotch, not that it's aparticularly good sounding part of most drummers anatomy, but because it's away from thehi-hat and any potential leakage problems. You could also place a second microphone belowthe bottom head. This will really add to the sound. If you do this, you try reversing the phaseon the bottom microphone.Fig.1 is a good representation of a snare drum microphone placement. About 2-3 inches offthe head and pointed at the drummers, well you know. Then there is the standard in from theside position. This technique does help keep the microphone out of the drummers way andvice versa.


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Kick (Bass) DrumThe microphone should be placed about half way in to the drum itself and pointing at thebeater. Here is a tip. Bring the mic in from the right side of the drum and angle it at the beateryou will be avoiding leakage from the snare drum. You can experiment with the depth of themic, but always keep the mic pointed at the drummer's shinbone on the leg that controls the

hi-hat and in line with the beater.

Options:A) To speed up the decay, you can put a blanket or pillow inside the drum, resting on thebottom, and touching the front head. Another method is to place the mic inside the bassdrum, right on top of a pillow (if a pillow is used).B) Throw a sandbag in the drum to weigh it down. Let the sandbag touch the head (that the

beater hits) just enough to dampen out any obnoxious overtones, but not the good, naturalsounding ones.C) If there isn't enough attack; you can place a second mic on the drummer's side pointing atthe point where the beater hits. This mic doesn't have to have great low frequency response,

its purpose is to get more "slap".The procedure is about the same as on snare drums.

Keeping the Kick Drum

Often the Kick Drum is still heard at low listening levels due to the attack of the instrument.Sometimes the foot has a "cardboard" type quality to the sound which can be reduced with a300 to 400 Hz dip using the equalizer. Use the amount of dip that makes the drum sound thebest - usual amounts vary between 3 dB and 9 dB. When you reduce this frequency on thefoot drum, you also tend to get better distinction between the foot drum and bass guitar. Youcan also boost 50 Hz to give the drum proper fullness but be careful not to over-boost this.

To make the foot drum more prominent in the mix for low-level listening boost the "beater"

frequencies as follows:

3 kHz - boost to give a hard felt beater sound.

5 kHz - boost to give a hard wood beater sound.

7 kHz - boost for a metallic beater sound.

TomsPlace all tom microphones at a 45-degree (or there about) angle to the drumhead with theend of the mic (the capsule end) pointing at an imaginary spot about 2" past the rim nearestyou as you place the microphone. The floor tom microphone can be placed a little close to thecenter of the head, but not too close. The distance of the microphone from the actual headshould range between one inch and six inches depending on how "roomy" you like your drumsto sound. Once again, the further the microphones are from the drums, the roomier thesound, but you'll have to pay more attention to possible phase cancellation problems.


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Miking Cymbals In GeneralCymbal microphones usually don't need too much in the way of EQ, but you may want to rolloff the bottom end and add just a pinch of top end (around 8 - 10K). Keep the input levels ofthe cymbals fairly low as they have transients that can fool meters and blow tweeters faster

than you can say, "#*-! :(

Hi HatsIn most cases, you don't really need to mic them. You'll get enough hi-ht bleeding in to theother microphones. If you have the luxury of plenty of inputs and tracks, mic the hi-hat, butchances are you won't need to.There are a number of different techniques for miking hi-hats. The object is to keep the otherdrums out of the high hat microphone as much as possible. Try to point the microphone awayfrom the drummer and down at the outer edge of the hat from the top. You have to watchthat the microphone isn't pointed at the bell because it tends to sound very pingy and thin.

Also, don't get too close to the closing edge because a puff of air comes out every time thehats close and that can ruffle your diaphragm and make for nasty sounds.

CymbalsIn this application a small diaphragm condenser is preferred. Place the microphones about 16inches over the cymbals' centers and towed out at about 45 degrees. This will give betterseparation, and it will also reduce the amount of low end bleed from the toms that are pickedup in the cymbal microphones.

Drum Overheads (Stereo Tracking)These should be placed about 5-6 feet above the kit to start with. If they are too loud or quiet,

adjust them accordingly. The best mics to use overhead are small diaphragm condensers.They have a sharp, bright, clean sound to them.


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Mixing Made Easy(sounds like cooking show)

Most people think mixing is complicated. It's really not. Most pros like the fact that there's anair of mystery surrounding mixing. It's not rocket science. It's really just the practicalapplication of basic physics, a little bit of psychoacoustics, and a pinch of good taste.

The best way to learn anything is to copy the masters. Listen with headphones. Listen withnobody else around to bother you. Shut your eyes. Take a blank piece of paper and diagramwhat you hear. Draw a head in the middle of the page (bird's eye view). Listen for the kickdrum. Where is it? Dead center? Great - then draw a little box near the center top of the pageand write, "kick drum," in it. Snare Drum? Same deal. Bass guitar? Also down the middle.Piano? Low notes in the left ear. High notes in the right. Isn't that remarkable? The pianos layout just as if you were sitting at the keyboard. You're starting to get the idea.Most mixers will mix their instruments from the perspective of the listener or the perspectiveof the player. Try mixing from the player's perspective. In other words, drums are panned withthe high tom on the left and the floor tom on the right.

Mixing an Imaginary Pop/Rock SongThe kick drum, snare drum, and bass should all be down the middle, and should be the mostpredominant elements in the mix with the exception of the lead vocal. The bass and drumsform the song's feel or groove. If they're mixed correctly, you are already half way home to agreat mix.Start with the kick drum. Adjust your mix monitor level to where you normally like to listen.Bring the kick fader up to a point where it kicks the mix bus meters (the console's stereooutput) up to -3db VU. Add a little 2.5 K for attack if you need to. Roll off a little 300HZ if thekick is a little tubby in the lower mids.Bring up the bass guitar fader until the bass becomes a cohesive unit with the kick, and the

two of them seem to hit you in the chest. Now add the snare to the mix. Bring it up to a levelthat rivals, but doesn't exceed the level of the kick and bass. Add a little plate or room reverbto the snare. Try a 1 second decay time for starters. Adjust to taste.

Bring up the toms and overhead tracks. Keep them panned so that the cymbals on the leftside of the kit are panned to the same side of the mix as the high tom. The mid tom shouldappear in the middle, and the floor tom and cymbals from the right side of the kit all appearon the right. If your toms sound like cardboard boxes, try adding a little bottom, rolling offsome 300-500HZ in the lower mids, and adding a little top end to give them some crack.Don't bother with a hi-hat most of the time. Some one once told me that it usually takes careof itself, and remarkably, it does!

Add the guitars next. first, the electric. Pan it almost full left. Take a short delay from theguitar and pan it almost full right, but a slightly lower volume. Your brain will tell you that youhear a big, wide guitar that appears mostly on the left side. Pan the acoustic guitar to the rightmaybe add a

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Basic Recording Tips