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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 the track, but avoid distorting the signal. This will allow you to maximize the signal-to-noise ratio of the signal. You want the signal to be loud enough to mask any noise in the system but you don'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 Digital Distortion Bad.

2. Don't immediately reach for the EQ knob and don't over do it with the Reverb. These are two of the biggest mistakes. Rather than fiddling with EQ (equalization) if you don't like the way something sounds, try changing the source. If you are miking a guitar for example, try moving the microphone around to alternate positions relative to the acoustic guitar (or amp, if it 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 will allow 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 close field monitors, your living room stereo, your car stereo, a cheap boom box, etc.). This will allow 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 simply
disappear due to cancellation.

Drum Machines
A Good Trick
Buy, beg, borrow, steal or if you have too, rent a Hi Hat and a Snare. Mic them up and play along 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 in getting a more real sound.

Studio Monitors
A Little History Lesson
In 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 monitoring system really is impressive to listen to, something not overlooked by studio owners who wanted to impress that high paying client. Eventually engineers and producers learned that this was not always the best way to accurately record and mix records because the average listener was listening to music through their inexpensive home stereos, radios and cassette decks. Also, large monitor systems and the cost for the required control room acoustic treatments were going through the roof. This is particularly 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-field monitoring.

Near and Mid-Field monitors

By their definition these speakers are intended for close in monitoring. The idea was to improve 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 (no pun intended). With near-field monitoring, the surrounding acoustic environment becomes a much less significant. The same holds true for mid-field monitoring except there is a little more distance placed between the speakers and the listing position. This result can be a larger sound field along with something closer to that large monitor sound. A good set of monitors properly located in a reasonably non-reverberating room and powered by a 100 to 200 watt amplifier will yield surprisingly accurate results at budget cost. Even the biggest studios use smaller speakers to augment their large monitoring systems. Today near and mid-field monitors have become proven tools in the recording business, and thats what it is, a business.

Monitor Placement
While near and mid-field monitors are more forgiving of the surrounding room acoustics it is always prudent to optimize the listening environment when ever possible. Be aware of the effect that the size of the listening room can have on low frequency response. The general guide line is that smaller rooms will have more bottom end, but still be aware of placement of your monitors in a large room. If you find that your monitors are to bass heavy or the reverse to 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 reflective walls and windows.

Monitor Spacing
Consideration should also be given to the physical spacing between the monitors and the listing position. The general rule is: The distance between the monitors is equal to the distance to the listener. In other words, the listener and the two monitors are at the three corners of a triangle having equal sides.

Monitor Placement Tip

Trying 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). Sit exactly 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 each monitor 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 becomes distinguishable separately, which is not what you want.

Microphones Explained
BASIC TYPES The most commonly microphones used in audio, "Dynamic" and "Condenser. DYNAMIC MICROPHONES
Dynamic microphones are commonly found in PA applications due to its general ruggedness and simplicity of use (no need for phantom power or batteries). It works rather like a speaker in that there is a diaphragm attached to a coil of hair-thin insulated wire flexibly suspended in a magnetic field. Sound waves set the diaphragm and coil in motion vibrating back and forth which causes the coil to cut lines of magnetic force, thus a small amount of voltage is induced in the coil. The voltage varies in polarity with the frequency of the sound waves and in strength with the amplitude 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). This voltage 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 the amplified signal by vibrating back and forth to create sound. In fact, dynamic microphones and speakers are almost interchangeable. Believe it or not, you can connect a raw speaker, a woofer 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 may promptly fry the mic, but this backwards PA will actually function (briefly). Dynamic mics are best for close-up use whether for vocals, instruments or instrument amplifiers. Certain models are also preferred for bass drum and others for brass instruments.

CONDENSER MICROPHONES
Condenser microphones offer high sensitivity and smooth frequency response. They operate on a small amount of DC voltage either from a built-in battery or a "phantom" power supply unit, or from the mixer if it has phantom power built in. This is deposited as positive and negative charges on two thin metal plates with a small airspace or other resistive material between them. This forms the diaphragm cartridge. Sound waves cause the top plate to vibrate which alternately compresses and de-compresses the resistance. It acts as a dielectric and a signal voltage is produced that varies in polarity and amplitude with the frequency and amplitude of the sound waves. This travels down the cable 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 of bass 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, especially guitar, banjo, mandolin, violin, upright bass, piano or anything with strings. They are also preferred for overhead coverage of drum sets. At one time it was thought that condenser mics were too fragile for PA applications, however they have greatly improved over the years in that regard with many models now designed for this kind of work which virtually equal dynamic mics for road-worthy-ness.

PICKUP PATTERNS Directionality or Polar Response

Most microphones are capable of picking up sounds over a wide area however they don't pick all 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 levels those 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 depending on the microphone's design. These shapes are what you see listed as "polar patterns" in mic literature. Although the polar plot diagram is flat-looking, in reality mics pick up sounds coming from above and below as well as the front and sides and even the back

Cardioid Polar Pattern

Cardioid The 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 from behind the capsule as well as sounds that come from the side. This is a good pattern for vocal microphones. 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-up patterns that those frequencies make. If you listen closely, not only will you hear the amplitude (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 the amplitude 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 and a tiny lobe of rear sensitivity.

Omni Polar Pattern

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

Half-omnidirectional or Hemispherical

Half-omnidirectional or hemispherical: Picks up equally over a 180 spherical angle. This is the pickup pattern of PZM (pressure zone microphone)

Bi-directional

Bi-directional It is not very difficult to produce a pickup pattern that accepts sound striking the front or rear of the diaphragm, but does not respond to sound from the sides. This is the way any diaphragm will behave if sound can strike the front and back equally. The rejection of undesired 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 above an instrument. Frequency response is just as good as an omni, at least for sounds that are not too close to the microphone.

Microphones Applications
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 least understood. One reason is compressions sonic results are often subtle and thus hard to hear especially for budding engineers. Another hurdle is presented by the various and differing compressor control parameters; those, too, are typically subtle in their individual sonic effects, and they work together interactively, further complicating the stew. Then theres the confusion that lies in the bewildering array of product types and models the engineer must choose from before even reaching for a control knob. For example, for a given application, should you select a VCA-based compressor or one controlled by an opto-electrical element? Solid-state or tube designed compressor or a hybrid of the two. Then there are the analog and digital compressors. 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 mystery for many users. Yet, if you want to master the arts of recording and mixing, learning compressions intricacies is imperative. After all, the production processes for most of todays popular music formswith the notable exceptions of classical and some jazzrely heavily on compression. Simply put, if youre not compressing properly, youre not getting the best sounds possible. This article will try to help you guide through the maze of compressor options and explain practical compression applications in plain English. We will start with the basics of compression, 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 theyre important.

Improvement Plan
Compression falls under the broader category of dynamics processing. The term dynamics refers 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 processors purpose is simply to increase or decrease a signals dynamic range, which alters how the levels fluctuate 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 rangethat is, it reduces the difference in volume, or level, between the loudest and softest parts of a performance. The process of reducing volume is called gain reduction. Properly applied, gain reduction makes a performance sound more consistent from beginning to end. For that reason, compression is a great remedy for a performance in which the levels fluctuate too widely.

8 By reducing dynamic range, a compressor also allows for the processed signals overall level to be raisedthat is, become hotterresulting in increased loudness without pushing the signals loudest parts into distortion. Bringing up the overall level has the additional benefit of making lower-level sounds louder than 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, and easier to hear. Of course, you may not want to make burps, string buzzes, and other incidental performance sounds more audible. Therefore, apply compression only when musically appropriatewhen the 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 it is desirable to use compression during the recording process. That approach has several potential benefits. For one, a compressor makes it easier to capture usable tracks when recording an instrument with a wide dynamic range. Moreover, solving level-fluctuation problems 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 creative aspects. For those recording to any digital medium, using a compressor during tracking ensures that sounds 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 on extra loud notes. For those recording to analog tape, compressing during tracking allows the signal level to be raised higher above the noise floor, which results in an improved signal-tonoise ratio.

Tricks of the Trade

In addition to problem solvingsmoothing out rough performances, improving digital resolution and signal-to-noise ratio, avoiding digital clipping, and the likeyou can also employ compressors in numerous creative applications. For example, a compressor can dramatically change the envelope of a sound in much the same way an envelope generator works in a synthesizer. That and other compression tricks can give a vicious attack to a lackluster snare drum, add crunchy edge and sustain to a mild-mannered electric guitar, make a lead vocal sound so urgent that listeners will dial 911, or pump up an entire mix until the band sounds like its exploding out of the speakers. In simplest terms, think of a compressor as an automatic volume controller. Indeed, before compressors were invented, engineers typically had to ride gain on a channel to maintain consistent volume levels. (Then again, many engineers still ride gain, even when using compressors.) 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 with lightning-fast reflexes. The compressors control settings determine when and how much that fader moves. Depending on how its controls are set, a compressor reduces either transient peaksthe short-lived, attack portions of a soundor the average-level portions of the sound, and sometimes both. Examples of transient peaks include the stick strike on a drum head and guitar-string plucks.

9 A sounds average-level portions include a snare drum shells ringing and the sustaining of a guitar note after it is plucked. Certain instrumentsa wood block, for instanceproduce mostly transients and very little sustain. Others, such as vocals and organs, typically produce mild 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 least five controls: threshold, ratio, attack time, release time, and output level. Full-featured VCA models may offer more than twice that many controls, whereas some expensive opto-electrical compressors may provide only two control knobs. Note: Units with fewer controls are not necessarily less capable; rather, they typically provide automatic control of parameters such as attack and release time, or they gang two parameters (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, all signals 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 above the average level of the signal. That way, peaks that exceed the threshold get attenuated while the average levels pass unaffected through the unit. Clearly, a proper threshold setting is critical to a compressors performance: if the threshold is set too high, the unit will not process 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 input and 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 it exceeds the threshold. For example, a 2:1 ratio will let a signal increase in level only 1 dB for every 2 dB it exceeds the threshold. Likewise, if the signal exceeds the threshold by 6 dB at a 2:1 ratio, the compressor attenuates the signal by 3 dB, a net gain increase of only 3 dB. In that case, the compressors gain-reduction meter (if it has one) will show 3 dB of gain reduction. Typically, different instruments and performances call for different compression ratios. For example, to compress a ballads near-perfect vocal track, a mild 2:1 ratio would probably suffice; at that ratio, and with the appropriate threshold dialed in, the compressor tightens up the performance enough to ensure quiet phrases are not lost in the mix and higher levels are not overbearing. At the other extreme, a bass guitar track that alternates between mellow finger-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 in order. 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 signal level subject to compression. The relationship between the two controls affords flexibility and sonic variation. There are, two different-sounding ways to get the same amount of gain reduction out of a compressorlow threshold and low ratio or high threshold and high ratio.

10 Attack time is how long it takesmeasured in milliseconds (ms) or microseconds ()for the compressor to kick in once the signal exceeds the threshold. A slow attack time lets inherently fast transient signals pass threshold before compressing the rest of the signal; a fast attack catches 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 is exceeded, 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 unruly peaks are tamed or increase it until average levels are lowered and desirable peaks get through unscathed. If youre having trouble hearing your settings effect, watching a downstream peak-level meter (that is, one that monitors the levels after the processthe compressors output-level meter, for example) will let you visually confirm what portion of the sound is attenuated. Release time is how longmeasured in seconds or hundredths of a secondit takes for the compressor to return the signal to unity gain (its unprocessed state) after the signal falls back below threshold. That is, once the release time passes, the compressor lets the signal pass through 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 and get out of the way quicklyfor instance, when you want to put a lid on transient guitar plucks but allow the ringing notes to pass through unaffected. Conversely, a moderate attack time coupled with a long release is perfect for those Santanatype guitar solos in which you want notes to sustain forever. At two seconds or longer, the extended release time causes the compressor to slowly restore compressed levels to their original (higher) gain, just as the sustained notes start to naturally die off, which counteracts the decay and makes the tails of the notes louder. Output Level a compressors last control stage. That control is also known as make-up gain because it is used to make up for the gain reduction caused by the compressor. The usual approach is to increase the processed signals output level so it matches the unprocessed signals level. That creates unity gain between the two signals, which makes it easier to compare them using the bypass switch and ensures appropriate levels when recording or mixing.

Path Not Taken

A compressor can degrade or ruin an audio signal as well as enhance it; therefore, one of the most useful features on any compressor is the bypass switch, which lets you compare processed and unprocessed signals. After using the output-level control to balance the levels of the processed and unprocessed signalsa critical step because louder signals sound brighter and fulleryou can judge whether your control settings actually improve the sound by switching the compressor in and out of the circuit. Fortunately, most compressors provide a bypass. Typically, this is a switch that disables the compressor circuitry; ideally, it also disables the input- and output-level controls. A hardwire bypass is usually the best design because it routes the input directly to the output and bypasses all compressor circuitry, such as input and output amplifiers and gain-control devices.

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You Dont Need a Input

Many compressors also provide an input-level control, but those are often superfluousif not undesirable. For one thing, a compressor with a wide-ranging threshold control can handle almost any input level. So an input-level control is necessary only if the threshold range is too high 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 at the 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, the compressor may not kick in when fed low levels. In that case, an input-level control is necessary to boost the input to an appropriate level. The reason an input-level control can be thought of as undesirable is it adds yet another amplification/attenuation stage to the circuitry, thus degrading signal quality.

Knee-jerk Reaction
In addition to the controls and parameters already discussed, several more-subtle parameters and design features often figure prominently into a compressors performance or sound. One such parameter is the knee, which is related to the compressors threshold control. The knee determines how quickly and smoothly the compressor will transition from no action to the full ratio of gain reduction set on the unit once the signal passes threshold. Generally, a compressors knee is hard or soft, though some units provide switch able hard- and soft-knee compression. Hard-knee Compression The unit processes the audio signal at the selected ratio once the input signal passes the threshold. Although useful for applications such as peak limiting and de-essing, a hard knee can sound abrupt, especially with higher ratios. Soft-knee Compressor (sometimes called overeasy) A compressor set to soft-knee compression, begins to compress as the signal approaches the threshold level and gradually increases the ratio until the signal attains threshold, at which point it equals the selected ratio value. The gentler, logarithmic increase of soft-knee processing tends to sound more transparent (less noticeable) than hard-knee compression, and thus is usually preferable for most vocals and instruments. In addition to manual controls for attack and release times, some compressors offer an automatic mode, called auto mode that does some of the tweaking for you. That is often referred to as program-dependent or adaptive processing. In auto mode, the compressors detector 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 on performances in which the dynamics change radically. It also lets you set up quickly yet still get good results when the pressure is on. The downside is you lose some control over the sound. For example, you may like those peaks when the guitarist picks harderin which case you 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 adaptive processing.

12 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 release timesmanually set or not change based on program content. That is due to the inherent nature of opto-electrical compressors, which in general are slower and less exacting than VCA-based designs. Because the attack and release controls on optical compressors provide only approximate response times, many manufacturers simply put fast and slow on either side of the knob, rather than hash marks indicating exact times.

Double Duty
Most dual-channel compressors offer stereo linking, a feature that lets you run two channels for example, stereo acoustic guitar or even an entire mixthrough the compressor and have each channel be attenuated the same amount. That keeps one sides level from dipping more than the other, which would throw the stereo image out of whack. True stereo linking works by having the channel that exhibits the most gain reduction determine the gain reduction for the other channel. Another form of linking establishes a master/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 is not the entire story. Actually, the detector circuits in compressors and true limiters differ by design. A compressors detector circuit is usually designed to detect RMS, or average, levels rather than transient peaks. Therefore, transient peaks almost always overshoot a compressors 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 peak energy levels and thus reacts faster. Whereas all true compressors use RMS-sensing detector circuits, detectors for different models can differ substantially in their reaction times. That means two different compressors set to the same attack, release, threshold, and ratio values may nevertheless respond quite differently to the same signal. (That is one of the many reasons it is difficult to recommend specific control settings for compressing various instruments.)

Chain, Chain, Chain

Every compressor has a side-chain detector circuit that sees when the threshold has been exceeded 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 to attenuate the signal. The circuits for threshold, ratio, attack, and release are also found in the side-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. Sidechain inserts therefore let you process the compressors input signal before it reaches the detector. That permits de-essing and other frequency-conscious applications.

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De-essing
To de-ess a vocal, first patch the send and receive from the compressors insert into an equalizers input and output, respectively. Next, boost the equalizers high frequencies and cut its lows and mids. That causes the compressors detector to hear the vocal as having excessive highs. 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 the audio path. With attack time set to around 50 and release time between 50 and 60 ms, the compressor can be made to quickly attenuate the sibilance and get out so the rest of the vocal is left unchanged. Of course, the compressors threshold must also be set properlyabove the vocals average levelsfor that to work. You can also use a side-chain insert to make the detector react to a signal entirely unrelated to the audio-input signal. The classic example here is ducking: a side-chain application in which an announcers voice is set to trigger a music beds attenuation. To set up this type of ducker, play stereo music tracks through a dual-channel compressor and patch the voice-over track (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, the detector hears the voice and instructs the compressor to lower the music bed

Freq Show
The misconception that split-band compression is the same as frequency-conscious compression is common. A split-band compressor splits the audio signal into two or more frequency 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 frequencies differently from the highs. A compressor that offersor is set up to providefrequency-conscious compression is still a full-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, userselected frequencies.

Ones and Zeros

One advantage of digital compressor is most of them offer the look ahead circuitry. Because the compression algorithm is in software, the compressor can analyze what it is about to process and place the attack time right at the onset ofor even beforethe sound, resulting in a zero attack time. However, while a super-quick (or zero) attack time is great for catching transients, it doesnt always sound the best. Therefore, use such power judiciously; the crack of 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 benefit of working with digital compressors is the ability to stay in the digital domain. If youre working with a digital audio workstation or digital mixer, there are strong arguments for not re-entering the analog circuits. Most importantly, by staying in the digital domain, you avoid the 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 userinstallable EPROM, CD-ROM, or some other user-friendly method. In addition, you should insist on a box with a word clock input. Without word clock inputs, you will be limited to using only one digital compressor at a time.

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Fresh Squeezed
Clearly, its important to choose the right compressor for the job at hand. With compressors it is not so much the design but the execution of the design that makes a compressor good or bad 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 the sound that counts, not the propaganda.

Starting from Scratch These general rules will get you started and prevent most processing
mistakes. Once you have some experience, you can tweak settings for more extreme processing. 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 a mono 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 its highest 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 markstypically labeled 0 dBscreened around the input and output control knobs. If your unit provides those reference marks, set both knobs at 0 dB for unity gain. If no marks are provided, youll either need to call the manufacturer to find the unity gain for each knob or use a tone generator in conjunction with the units input and output meters to determine unity settings. If the compressor 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 tone to the compressors input and set the input-control knob so the compressors input meter reads 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. Its not 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. Youre now ready to make ballpark settings for processing the signal. Set the attack and release time controls to an average value, usually close to the twelve oclock position, and the ratio to roughly 2:1 or 3:1. Those mild settings reduce the risk that you will over compress the signal. Switch the compressors meters to show gain reduction and lower the threshold until approximately 4 to 6 dB of gain reduction is attained on peaks. It is most important here that the lowest signal levels do not exceed the threshold and trigger the compressor. In other words, 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 and begin listening to the results. If you want more compression, increase the ratio; if you want less, reduce it. Use fast attack and release times for compressing only the peaks. Use slow attack and release times to make a signal sound denser. Most importantly, let your ears be the guide. After finding settings that provide the results you want, adjust the output control to make up the gain that was lost to gain reduction. Of course, you can add more or less than that amount if you wishjust make sure youre paying attention to proper gain staging with regard to any downstream gear. That is, dont boost the compressors output if doing so requires you to lower the input on the next device below its unity gain setting.

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Dialing in Hot Sounds

Ask any experienced engineer for suggestions on compressor settings for various instruments and youre likely to be met with a blank stare. He or she is not trying to be evasive. The best settings depend on a host of variables: the type of compressor used, the units detectorcircuitry 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, the outboard gears noise floor, and on and on. Use your ears is a tired phrase, but ears are still the 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 the compressors mentioned in this article. As always, use your ears to make additional adjustments if the initial sound is not to your liking.

The Exciting Compressor

Note: Motown developed a mixing method that allowed presence, bite and intensity on lead vocals. 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 Mix

In the 1950's and early 1960's records were generally mixed with the vocal far louder than the music. The vocal had a very natural sound to it except the there was a lot of reverb applied to the vocals. The artist that really had this sound was Frank Sinatra. Back then, when listening to 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 Nelson had the vocals way out front.

The Motown Mix

Berry Gordie had a "better" idea. Motown was selling "excitement." The thinking was that the rhythm of the music is what made the record exciting and what the kids danced to. There actually was a lot of melody and important lyrics in these old records - but rhythm was the key. 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 on the vocal. There were three custom built reverberation chambers at Motown - all used during a 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 could be added into the mix and would add "excitement" to the sound. A lot of people were impressed with this device (and clone devices that followed) especially to make the vocal sound brighter. Here is how to use compressor for an exciter.

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The Motown 1960's Exciting Compressor

With the Motown mix approach there were problems. If you wanted the lyrics to be heard you had to use a lot of compression on the vocal so that the softer words could still be heard over the higher-level music. In addition you boosted the "presence range" (around 5 kHz) with an equalizer. 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 vocal and split the signal so that it went to 2 different channels. Before the vocal signal went to the second channel, it went through a compressor. Now he had two channels of the vocal - one compressed and one uncompressed. On the uncompressed vocal he added very little with the equalizer and he added the reverb. On the compressed channel, he squeezed the S*#t out of it 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 and equalized channel, he brought this up just enough to add excitement and presence to the vocal sound. The result was nothing less than amazing. In the mix the vocal sounded very natural and bright. None of the music ever "stepped on" the vocal and you could hear each and every syllable 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 of their product and the older Motown technique seen basically the same. As you try this technique out you will find it works for other instruments as well. Often the frequency of EQ needs to be changed for the instrument. The vocal works well with tons of 5kHz to 8 kHz added to the "exciting compressor;" guitars work better with 3 kHz - 5 kHz and bass guitars work better with 800 hZ to 1.5 kHz. For analog recording or working with an analog console, splitting the vocal into two console channels is easily done with a Y-chord or similar function at the patch bay. For digital consoles, it's a little harder; usually the best results are obtained by actually having two vocal tracks recorded on the tape.

COMPRESSOR FX & TIPS

Fairly common one this, you probably know it, but if you don't it's pretty simple. Getting a snappy emphasis in this way works with most sounds, but obviously has greater effect the more 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 fast attack of the snare to bust through before the compressor kicks in to crush the sound...This little technique also works great on bass synth sounds with a fast attack...

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COMPRESSOR SETTINGS Sound Attack Release Vocal Loud vocal Acoustic G Electric G K & Snare Bass Mix General fast fast 5-10 ms 2-5 ms 1-3 ms 1-10 ms fast fast 0.5 0.3 0.5 0.5 0.2 0.5 0.4 0.5 sec sec sec sec sec sec sec sec

Ratio 2:1 - 8:1 4:1-10:1 5:1-10:1 8:1-10:1 5:1-10:1 4:1-12:1 2:1 - 6:1 5:1

Hard/Soft soft hard soft/hard hard hard hard soft soft

Gain reduction -3-8 db 5-15 db 5-15 db 5-15 db 5-15 db 5-15 db 2-10 db 2-10 db

AUDIO LIMITING Controlling Peaks In digital recording there are extreme peaks that can cause the overall average level to be low. If you are mixing down to analog tape, many of these peaks have been "rounded off" by the tape. You can control these peaks with the LIMITING function of most compressors. This is 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 are often not noticeable to the ear. This is quite common in mastering, since it allows mixes to be recorded much louder on digital media, such as CDs without going into digital clipping. In other words, -6dB of peak gain reduction will allow a song to be recorded twice as loud to your 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. Use smaller 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
Microphone

Miking Speakers

One 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 he preferred or recording electric guitar. He suggested trying to find an older Shure SM57s the ones with Unidyne III wrapped around the head.

Single Microphone Placement

Use a Shure SM 57 microphone; put it up in front of the speaker. Just kidding there is a little more to it.

General Placement
The most commonly used placements are; One, mic the center of the speaker, two, the sweet spot (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 microphone nearer and farther from the speaker. If the cabinet has no grille, you might start with the mic as close to the speaker as you can get, which will give you a nice proximity boost in the low end then try pulling back a bit to let the sound develop.

Axis Placement
Theres (at least) one more variable you can experiment with: the orientation of the microphone on- or off-axis. Many engineers will immediately go for on-axis (with the microphone aimed 90 degrees, or straight into the speaker) to get the best overall frequency response. However, theres an infinite range of off-axis positions, with the microphone turned at 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 just pointing the microphone straight into the cabinet.) Turning the microphone slightly or a great deal off-axis does two things. First, the frequency response of the mic will vary depending on the on-/off-axis positioning check out the polar pattern chart for any directional mic to see what I mean. Youll see that the highfrequency 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. If you rotate off-axis toward the center of the speaker, the front of the mic will see more of that as the main source (and again, take into account the slope of the cone of the speaker, and where it is hitting the mic to determine just how far off-axis you really are).

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EXPERIMENT
Try re-amplifying the guitar signal into the amp. That is, record a dry guitar signal using a direct box into my recorder. Then use an interface box there is a variety to chose from to send that direct recorded track into an amp/speaker. This provides you with the exact same guitar 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 the sonic 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 change anything else. Record another take. Angle the mic a bit more, record another take. Continue through a good range of mic rotation. When youre finished, youll have a session where you can A/B among the various mic angles. Make sure you take good notes, so when you listen back you know what youre listening to. Now put your ears to work. Which position sounds best to you? Keep in mind that what sounds best in isolation may not be what works best in the context of a song. And maybe the tone you like best is with the mic straight on if thats the case, wonderful!

Add a Room Microphone

If 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 feet depending on your needs. In this case the microphone can be pointed in the general direction of the speaker. Again start dialing in a sound by moving the microphone back and forth to achieve that sound you are looking for.

THATS STILL NOT ALL

One final tip: The idea of turning a mic off-axis isnt limited to guitar speakers. Any time youre positioning 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 expanded color 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 Blaster Flanger with its super wide stereo spread. Ah, but the way they get that stereo spread is by flipping 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 what a track sounds like in mono before you sign off. 2. Stringing along with dead strings. Yes, change your strings before that important recording session and no, adding compression to increase sustain is not a suitable substitute. With new strings, your axe will sound brighter, notes will sustain longer, and tuning will be more consistent. 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 preset in 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 straight signal. Although ADT is a valid effect in its own right if you want a sort of more focused version 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 stereo image 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: Sound travels 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 arrives at the speed of light. If the mic is one foot away from your speaker, zoom in on the tracks and shift 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 you need 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, and the like. The good news is that the pickup is directional, and changing the guitars position can make it less prone to picking up garbage. Dont use your ears; look at the meters, because the levels 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 be able 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 certain point to get a good tone. But dont go past that point widely known as 11. Why? Aside from the possibility of overloading your mic, things in the room will have more of a tendency to rattle, and poor room acoustics may be overemphasized. As Johnny Cochran once said, Once you get your tone, leave it alone.

21 8. Forgetting to bring a spare set of tubes. Tubes fail, tubes go soft, and they sometimes do it at in opportune moments. . nuff said. Remember, if one tube of a matched set fails, you need to replace them both. Its a good idea not to trust the tubes you buy, but to try them out immediately 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 all know thats a good idea. But have you adjusted bridge intonation lately? Just changing strings can be enough to throw the intonation out of whack. You may not notice that theres any problem until you start recording, and everyones listening to your guitar under the audio equivalent of a microscope. In my experience, few things can destroy a session faster than having to adjust intonation on a guitar with dead strings (mistake #2), because it will be next to impossible to get it in tune. Tempers will fray, harsh words may be exchanged. And while youre at it, leave a tuner in-line at all times, or use the tuner in a piece of software (e.g., Native Instruments Guitar Rig and Cakewalk 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 with a stompbox that can use batteries or AC, try both and see which sounds better. With some old stomp 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 of the story: When you show up at the session, bring both the AC adapter and a fresh set of batteries. Of course, there are plenty of other mistakes that guitar players make in the studio, from snorting 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 evolve according to the engineers working style. Your job is to play a great part; the engineers is to record. 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 to do two jobs at once. If youre really concerned that the recording isnt right, then record a dry part 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 remains one of the hardest to get a great sound on in the studio. Here are three basic rules to recording acoustic guitar. Rule 1 - Use condenser microphones. They will always sound better than dynamic microphones for acoustic guitars. There are several condenser microphones that are currently on 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 sound Bigger. 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 out a solid column of boomy low end. Avoid placing a mic in that column, or even aiming one directly at the sound hole, and youll get better results.

STRUMMED
How 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-driving strumming in a rhythm section. Many engineers start with the standard position: a large-diaphragm cardioid condenser mic placed 1224" away, at guitar neck level, slightly to the players left (assuming a right-handed player), and aimed at the point where the neck meets the guitars body. If the strummed guitar 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 slightly in 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 at the neck/body joint, but also experiment using dynamic microphones. Try using the standard Shure 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 fidelity is less important than punch. In general, one mic is fine for this style; you want the guitar to sit in the track and drive it. The subtleties of room or fancy stereo miking will be lost or clutter up the track.

FINGER PICKED
For a finger picked guitar, especially one that will be used solo, you want to capture all the detail 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 be had using small-diaphragm condensers. A finger picked guitar can be quiet and delicate; look for 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, especially if the guitar will be in a mix with other instruments. Try augmenting that mic with a stereo pair pulled back to get some room sound, and give sonic depth and space.

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CLASSICAL GUITAR
To record a true classical guitar performance (as opposed to a nylon-string guitar played in another 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 that are 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 the instrument/player than recordings for other styles. This means that the room will play a big part 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 six feet away, pretty much directly in front of the instrument. The idea is to capture what the audience would hear. The classical guitar is a low-volume, delicate instrument. Place your microphones too far back, and youll have too much room. You may also be forced to use so much 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 feet back this will add control over the stereo width of the final tracks, and are set up for mid/side decoding. NOTE: Use a stereo miking technique if the guitar will be solo; if its a duet with another instrument, try using one mic for a tighter sound.

That Sound
Country/Pop For that Eagles "Lyin' Eyes strummed sound. Place the microphone about 6 to 8 inches from
the guitar's sound hole, but angle the microphone toward the area where the fret board and the sound hole meet. If you point the microphone directly into the sound hole, it will be very full probably much too full, and very boomy. Use a compressor/limiter to knock down any peaks (3:1 ratio), and set the threshold a little lower to give it a slightly "squashed" or tighter sound. Set the threshold higher to just limit the peaks and give a more open sound. You may need to EQ out some bottom end Boom. If so, try rolling off some bottom (100Hz), or cutting a 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 the pick will give more attack-less sweetness.

Eric Clapton
For his classical/gut-string guitar sound. Use a condenser microphone and place it about ten inches away from the guitar, about 3 to 4 inches up the neck, but aim it at the players picking fingers. This angle will reduce boominess by virtue of the microphones cardioid polar pattern producing a natural roll off when it's aimed off-axis, while simultaneously delivering the attack of the fingers. Try and say that three times in a row! The added distance will pick up some of the guitar body's resonance. A compressor is a must for this case because of unexpected peaks. A 4:1 ratio is a good place to start, but set the threshold fairly high so that the most of the 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 need deep bottom end. Place one mic in front of the guitar, a bit further back than the standard position. Position a second 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 microphones position; try it at knee level, looking up toward the guitar body behind the bridge, or at ear level looking down at the guitar body behind the bridge. The meat of the sound will come from the front mic, but placed correctly (move it around, youll know when you hit the right spot) the right-hand mic will fill out the bottom end with tight, full, round lows. Use this approach with steel- and nylon-string guitars; for nylon, pull the right-hand mic back, or turn it down 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 the microphones 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, two spaced small diaphragm condensers a few feet back and a few feet apart. Two room microphones, pulled back enough to capture a balance of guitar and room. For added versatility, track a pickup on the guitar as well. Use the large-diaphragm with some of the pickup as the main sound, use the spaced small diaphragms to give the sound width and richness, and then mix in the room microphones for spaciousness and depth. This will also work well for surround applications. Warning: Getting all those tracks in-phase may take some work.

Positioning the Microphone

Try 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 the microphones around while you listen on the studio monitors.

Nashville Acoustic Tuning

The Low E A D G strings are replaced with thinner strings tuned one octave higher than normal, 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 electric guitar 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 will overpower the acoustic and the most effective way to even them out is to compress the acoustic a little bit more than what you may have already done going to tape so you can bring the 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 that counter each other rhythmically (giving them each their own space), and have them each play in a different octave. That will give you a full sounding track that remains open and airy at the same time. You can also make an acoustic guitar sound bigger or more rock-like by panning the 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 your delay to "de-tune" one of the guitars just a pinch (one cent is a good place to start). The delay will provide the brain with the psycho acoustic information it needs to perceive the guitar as bigger, while the pitch change will make it appear "fatter."

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Recording Bass Guitar

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

BASS-IC INSTINCTS
The central message is this: your goal when recording bass should be to get as clean and fullfrequency 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 mess around with the track to your heart's desire. That way, you have the option of going back to your original sound if you want to.

Keeping The Bass Guitar

In its first octave, the bass guitar generates fundamental frequencies between 40 Hz and 80 Hz. Thus you could say that the fundamental notes that the bass puts out are between 40 and 100 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 octave of 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 you do this you will find the instrument sounds like this: 1. It has adequate lows and body at loud listening levels. If you cannot say this, 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, often making 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 1
You will need a (DI) Direct Box to change the high impedance signal of the bass guitar into a low impedance signal that your console's microphone preamp needs to see. The DI will also have 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 when recording bass. Adjust EQ as necessary for the sound and style of the music. Generally the warmth area of the EQ spectrum is down in the 100-200 Hz area. Watch your recording levels and be careful not to overload.

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Method 2
Start 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 that you're just overloading your recorder. That way, when the bassist plays normal lines, you will have 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 without overloading 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 down anything 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 to start. Keep the compressor attack slow enough to let you hear the attack of each note. Keep the release fast enough so that each note is not affected by the note before it. Be careful: if the release is too fast, the compressor will chatter and distort as long notes sustain. (If you'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 more at mix time. You will probably have to increase the output gain of the compressor slightly to compensate for gain reductions.

Then there is the Bass POD and other direct recording devices. If you have access, give
them 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 vocal recording. Meaning: Not At The Same Time As Everything Else. However, recording a scratch vocal 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 can have the vocalist in the control room or in another adjoining room listening on headphones and singing into any available microphone. Pick an isolated room or corner away from any noise source. (Furnace ducts, etc.) Put down a carpet and cover your music stand with a towel or rug to avoid reflections. If the area is open and 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 the area is well lit. Check the headphone mix yourself on the same headphones the singer(s) will be using. Don't take for granted that the mix sounds the same in the studio as it does in the Control 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 pop screen will be needed in addition to precise positioning. Plosives are the Ps and Ts that ruffle the diaphragm of the microphone, causing unwanted low end information to get onto your recording. Some repositioning by the singer to the microphone may be necessary to achieve the 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 finished singing into. Its probably the most common because its the easiest. But its not always the best option.

LAYERING
This is another variation on the one-voice/one-mic easy method outlined above. Change up the recording path. Instead of just using the lead vocal mic and signal path, use something else. Set up a different mic, (through a different preamp if you can) so theres not so much layering 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 if youre using a cardioid mic, the outside singers may lose definition since theyre off-center on the mic and there is typically less presence as one moves off-axis. Try using the mic in a wider, hypo-cardioid (as opposed to hyper-cardioid) pattern or better yet, try switching the mic to the omni directional pattern. This delivers excellent results because most microphones have flatter frequency response in omni and sound much more real.

29 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 create comb 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 for reflections 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, the roominess may overwhelm the direct vocal sound.

TWO OF A KIND
When using two singers, you can have them sing into one mic, or give each vocalist their own mic. Two singers into a single mic is most common, but if the mic is directional, youre compromising 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 them simultaneously sing into opposite sides of a figure-8 patterned mic. Each singer can get as close 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 the singers. Make adjustments in the volume by having the singers move closer or farther from the mic. Make sure to put any reflective surfaces to the sides of the mic, which have the greatest rejection. Ribbon micphones are quite good for this, especially if you have a lessthan-wonderful sounding recording space, because theyll pick up less room than a condenser.

THREES A CROWD
When recording a background trio, its fairly common to have them gather around and sing into 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 moving from on-axis to off-axis you may be surprised at how much sonic difference being off-axis makes.) For stacking background vocals with a group of three, try this: Get two variable-pattern matching microphones and set them up as an M/S pair. Put the mid (M) mic where you would normally 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 left in 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 that you dont end up with mono left, center, right. You end up with a stereo background vocal group that surrounds the lead vocalist without occupying the same space. Put the singer at 11:30 whose part is farthest away from the lead singers melody, as their part will crowd the lead the least. By moving the singers around the microphones, you can fill the space from left to 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's out of phase. Be systematic in tracking down the culprit. If you follow this prescription closely and 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. Individual drummers 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 potential you have for phase problems.

Equalization for Drums

12.5 kHz - Air (transients) Region Above 6.3-8 kHz - Snare Crack Peak 3.1 to 2.5 kHz Midrange Harshness (strongest) 800 to 1.2 Hz Adds Papery Tone 500 Hz Prominent Head Ring 315 to 400 Hz Adds Woody Character to Drum 125 & 250 Hz / 200 Hz strongest - The Meat (fundamental) 80 Hz Lowest Useful Overtone

Snare Drum
For the snare drum, the always a safe and highly effective choice is the venerable Shure SM57 microphone. Set the mic up at a 45 to 60 degree angle with the capsule about an inch or two above the head and about two inches from the side, again pointing at approximately a 45degree angle into the middle. Hear is an interesting tip. Try pointing the microphone at the drummer's crotch, not that it's a particularly good sounding part of most drummers anatomy, but because it's away from the hi-hat and any potential leakage problems. You could also place a second microphone below the bottom head. This will really add to the sound. If you do this, you try reversing the phase on the bottom microphone. Fig.1 is a good representation of a snare drum microphone placement. About 2-3 inches off the head and pointed at the drummers, well you know. Then there is the standard in from the side position. This technique does help keep the microphone out of the drummers way and vice versa.

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Kick (Bass) Drum

The microphone should be placed about half way in to the drum itself and pointing at the beater. Here is a tip. Bring the mic in from the right side of the drum and angle it at the beater you will be avoiding leakage from the snare drum. You can experiment with the depth of the mic, 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 the bottom, and touching the front head. Another method is to place the mic inside the bass drum, 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, natural sounding ones. C) If there isn't enough attack; you can place a second mic on the drummer's side pointing at the 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 a 300 to 400 Hz dip using the equalizer. Use the amount of dip that makes the drum sound the best - usual amounts vary between 3 dB and 9 dB. When you reduce this frequency on the foot drum, you also tend to get better distinction between the foot drum and bass guitar. You can 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.

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

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Miking Cymbals In General

Cymbal microphones usually don't need too much in the way of EQ, but you may want to roll off the bottom end and add just a pinch of top end (around 8 - 10K). Keep the input levels of the cymbals fairly low as they have transients that can fool meters and blow tweeters faster than you can say, "#*-! :(

Hi Hats
In most cases, you don't really need to mic them. You'll get enough hi-ht bleeding in to the other microphones. If you have the luxury of plenty of inputs and tracks, mic the hi-hat, but chances are you won't need to. There are a number of different techniques for miking hi-hats. The object is to keep the other drums out of the high hat microphone as much as possible. Try to point the microphone away from the drummer and down at the outer edge of the hat from the top. You have to watch that 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 the hats close and that can ruffle your diaphragm and make for nasty sounds.

Cymbals
In this application a small diaphragm condenser is preferred. Place the microphones about 16 inches over the cymbals' centers and towed out at about 45 degrees. This will give better separation, and it will also reduce the amount of low end bleed from the toms that are picked up 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 an air of mystery surrounding mixing. It's not rocket science. It's really just the practical application 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 with nobody else around to bother you. Shut your eyes. Take a blank piece of paper and diagram what you hear. Draw a head in the middle of the page (bird's eye view). Listen for the kick drum. Where is it? Dead center? Great - then draw a little box near the center top of the page and 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 lay out 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 perspective of the player. Try mixing from the player's perspective. In other words, drums are panned with the high tom on the left and the floor tom on the right.

Mixing an Imaginary Pop/Rock Song

The kick drum, snare drum, and bass should all be down the middle, and should be the most predominant elements in the mix with the exception of the lead vocal. The bass and drums form the song's feel or groove. If they're mixed correctly, you are already half way home to a great 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 stereo output) up to -3db VU. Add a little 2.5 K for attack if you need to. Roll off a little 300HZ if the kick 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 level that rivals, but doesn't exceed the level of the kick and bass. Add a little plate or room reverb to 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 left side of the kit are panned to the same side of the mix as the high tom. The mid tom should appear in the middle, and the floor tom and cymbals from the right side of the kit all appear on the right. If your toms sound like cardboard boxes, try adding a little bottom, rolling off some 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 care of itself, and remarkably, it does! Add the guitars next. first, the electric. Pan it almost full left. Take a short delay from the guitar and pan it almost full right, but a slightly lower volume. Your brain will tell you that you hear a big, wide guitar that appears mostly on the left side. Pan the acoustic guitar to the right maybe add a little harmonizer to it. Detune it one cent. Pan it to the left. Result? a big acoustic guitar sound that cuts through the mix and doesn't require more volume to do it. Now add the keyboards now. Usually pan it as if you were sitting at it, but if the guitar on the left is playing in a lower register, then I don't pan the low end of the piano there as well. They'd compete for space with each other. In this case, let's assume it's okay to pan the piano to nine o'clock for the low end and three o'clock for the high end.

34 By using a stereo compressor set to a fast attack and slow release, you'll make the piano "tinkle" a little more on the top end, and "growl" a little more on the low end. Hence, you'll be adding another instrument, but once again, it won't fight for space. Adding background vocals: Let's make the assumption that we have two tracks of group vocals three voices in each stack. Let's make them sound like the Eagles. Pan one group far left, and the other hard right. Suck out some lower mid-range to make them sound airy and angelic. See? Just like the Eagles. Yeah Right -- better add some stereo reverb. A nice plate reverb with approximately 1.5 second decay ought to do it. There you go. Eagles. Lead vocal. Slam it right down the middle. Make it loud. It's important. Treat it as such. These days, the pros seem to like their lead vocals dry -- so you can eschew the reverb if you'd like. If not, try a little plate or chamber on it. Again, keep it short for most types of tunes. You can also try a little delay on the lead vocal. It will make it more apparent without adding volume. One of the real tricks to mixing, making instruments easy to find in the mix without using volume to do it. Eq can be a huge help in that department, but it takes time to understand what eq does to individual instruments, and how it affects a whole mix when the instruments are all added together.

The Motown Mix

Motown had a "better" idea. Motown was selling "excitement." The thinking was that the rhythm of the music is what made the record exciting and what the kids danced to. There actually was a lot of melody and important lyrics in these old records - but rhythm was the key. Actually Motown started a revolution in mixing and most modern rock (and even pop) releases are mixed in this style, even today. Another Motown innovation was to have more reverb on the music than on the vocal. There were three custom built reverberation chambers at Motown - all used during a 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).

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The Mix Before sending to a Mastering Engineer

Noise Is Your Enemy. A noisy microphone preamp, hiss on a guitar track or a whine or RF (radio interference) from a bad connection All of this will be easier to hear after mastering. Keep YOUR TRACKS QUIET. You have probably heard dont sweat the small stuff or the standard Fix it in the Mix. Do yourself a favor SWEAT THE SMALL STUFF. Do take advantage of Mute Groups and Noise Gates. This is as especially true for the
beginning and the end of a song. Effect processors, even good ones, can add noise that you might not want at the head of your song. Effects and other tracks not in use on the very first note of a song should be muted (not be in the mix). Example: If your drummer stars the song out with a hi-hat or stick click (intro count) you will probably not want this on your finished song. If you do, mute the effects just up to the first note. At the end of the song, whether it fades out or has a cold stop make sure that the extra noise is kept to a minimum.

Dont waste hours of time in the studio trying to get a mix to sound huge. Most recordings
especially those made on a limited budget get that huge sound during the mastering sessions. This is not to say settle for less and do not walk away from a recording you are not happy with. Instead, start by trying to make your mix sound will balanced and well rounded. Nothing should Jump Out of it. The instruments should all sound related to each other and they should not step on each other. There also should not be any irritating frequencies that spike up in your face. A mix that sounds a bit small or dull is a lot better than one that sounds irritating. It will also probably have more potential during mastering.

Do compare your mix to some of your favorites. Keep in mind that the overall volume comes
LATER, during mastering. Just listen to the overall tone and feel of the mix.

Do be realistic. Dont overdo the effects. For the most part, a mastering engineer will make compression and volume adjustments on the recording. Reverb, which is generally in the background, may seem to come up in level. Unless you are using it as a feature effect, leave it at a reasonable level. A general rule of thumb is that once you actually notice it, you may have used too much. Again it is a matter on taste. It is actually fairly common for a mastering engineer to lay a small amount of reverb on a series of mixes to bring them together a bit. Do take advantage of STEREO. That being said, be cautious of phase cancellation. Hit that
Mono button frequently and see if your mix collapses. Stereo doubled guitar tracks are famous for this. They can sound huge and feel like they are coming from everywhere until you hit the mono button when they disappear completely. Stereo imaging and density is a Massive Mastering specialty, and an art that is rarely practiced in project studio mastering. It gives a normal mix a solid anchor with mono compatibility, while expanding the air image in the highs for a rich, wide, yet realistic and dimensional soundstage. It also a chief source of Loud on a CD. There is no magic box or mastering program that does this on its own. This is truly about technique.

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Dont waste huge amounts of time trying to mix every song at the same volume. This is one
of the first things that your mastering engineer will deal with. You can spend hours bouncing back and forth trying to make sure that every mix rides at the same level. Find a reasonable level and mix your tune. However.

Do try to mix to a good level to your two-track. All you want is a good signal to noise ratio.
You do not want to smash it. Mix so the bulk of the mix rides at -18 or 16db is just fine. Leave some room for some peaks (-6db peaks are ideal for 24-bit mix) and leave some room for the mastering engineer to work.

Do Not Do Not mix with some mastering processor across the stereo bus. Once it is there
it is there forever. Same with compression, a little bit (1db or 2db) of compression can really help bring a mix together. Look for individual tracks that could use compression or try putting one across the drum or vocal bus. Be Careful. It is very easy to over due it, and impossible to correct it later.

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Studio Construction Ideas

Important thing to remember are:
Stereo room symmetry around your speakers. Glass windows or doors for communication. Low-mid frequency absorption from 150 -550Hz. High frequency absorption. Absorption across the rear of the control room wall. Whatever low frequency absorption you can fit in the space.

Things to avoid are:

- Having to go through the studio to get to the control room!! (because you will always get interruptions as people move in and out of the studio) - Creating studios with no visual communication. There is nothing worse than recording someone you can't see. - Big studios with a small pokey control room and visa versa.

THE BIG FACILITY

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THE CORNER CONTROL ROOM

THE GARAGE STUDIO 1

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THE GARAGE STUDIO 2

THE CONTROL ROOM