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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.
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.
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.
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.
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
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.
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.
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.
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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.)
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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.
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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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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
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!"
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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.
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!
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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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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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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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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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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.
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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.
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.
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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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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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.
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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.
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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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