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By far the most crucial aspect of recording acoustic guitars well are the microphones used to record them. For instance a dynamic vocal mic will neither be sensitive enough to handle the instrument's relatively small volume, nor able to cope with the fretboard's entire frequency range. Go for a good quality capacitor (condenser) microphone every time.

Tip: record in an environment that sounds good to start with. Ever tried the bathroom? You might get a great result from those super sound-reflecting tiles. Also using a room with a wooden floor and walls can sound very nice when recording acoustic guitars, electric guitars are different although most instruments sound better in wooden rooms.

The next most important thing is to position your mics correctly. I will list two ways that are used quite often in recording, nothing beats experimenting different positions with headphones on. A 30 foot extension cable for headphones come sin handy when doing this.

Position your mic about 24 inches (0.6m) from the neck to body join. The idea behind placing the head of the mic a fair distance from the instrument is to be able to capture some of that resonant air emanating from the instrument. Its a common rule in recording any instrument to record at least the distance of the widest part of the instrument. ie on a guitar to body that vibrates to create the sound is about 2 feet hence the 24 inches. If you mic up the guitar inside of these 24 inches then the mic position is very critical and the smallest movement in the mic position can create a hugely different sound.

Tip: If you want your acoustic to sound right up front in your face, record in stereo. That's to say, get two good quality capacitor mics, point them across each other at roughly 90 degrees and keep their heads real close. Ben Harper, one of my favorite musicians used this trick on his WILL TO LIVE album.... You can hear his hand sliding across the frett board in stereo ! You can hear the movement of his slide in the frett noise. An article on stereo miking techniques is found here.

Another method to try is place a mic slightly off axis to the sound hole between 1" and 6" above or below the sound hole. This technique is mostly used in live sound since it will give less of the room sound and less chance of feedback in a live environment. Rolling off some bass around the 100 Hz region is nearly always needed with miking at the sound hole.

You'll know when you have succeeded when you merely have to apply minimal corrections in EQ to achieve a strong and acoustically pure result.

Most people know about stereo how you can pan left and right to create effects what many people leave out is depth. Try this out, grab a mic and record your voice at 1" from the microphone and then again at 3 foot, give them a quick EQ to take off the extra bass on the 1" track. Play them back and you'll see that if you place a mic close you get a sound that's in your face at the front of the speakers. Then listen to the 3 foot mic and hear that's its back and behind the speakers. You can create depth by using near and far miking techniques. Experiment to give your music an added dimension that other home recordings don't have. Adding reverb to a track can give it the impression it was recorded away from the mic so can certain EQ techniques but you still cant give correct depth without moving the microphone away.  Mix

To make your recordings sound ultra professional, record flat with no effects and instead find the right microphone for the singer. In the mix, roll off everything below 100 Hz and above 12,000 Hz. Add 2-4dB at 160Hz for male vocals or 320Hz for female voice for warmth. Notch out the mid-range, 500-800Hz, by a few dB. Sometimes a little sparkle in the 7-8kHz area is good, if there's no sibilance there. Finally, a little compression after the EQ can smooth the vocals out nicely.

Effects

Effects should be used minimally. If a stereo effect is so great that you can no longer pinpoint the instrument, you used too much. Another tip here is doubling and de-tuning. You can make any instrument dramatically wide, yet centered, by putting the same instrument far right (127) and far left (0) and slightly detuning them by about 5-7 cents. This is a great technique for 'wall of sound' like

 mixes that have strings that appear to float on the mix. Use it sparingly though, as hard panned doubles can easily take up sonic space where other instruments need to go.

To pan your MIDI orchestra, 0 should be far left and 127 is far right. You rarely want to set any instrument to an extreme value. For example, Harp, might be set to 20, French horn to 40, Flute to 60, Oboe to 70 and double basses to 110. The Front strings might be at 40 and the Cellos at 89. Don't read these numbers as absolutes; they are just an estimate. Every piece of gear sounds a little different. While all synths have 128 theoretical pan values, many of these values do not do anything to the sound. Some only change the actual sonic position every 3, 7, 15 values, some even 31 values. So experiment, move things around a little and the sounds will fall into their place.

There is no absolute way to create a sonic image of an orchestra, but it does make sense to follow a classic seating chart. Be aware of the frequency ranges of the instruments (i.e., how bassy, mid range or treble-like the instruments are) in order to avoid conflict. The Bass Drum is far from the double basses. The Cellos and Violas can play one part distinctly on the right while the violins play a different part of the left. Note that the woodwinds, perhaps the most melodic of the orchestra, are centered. As you go to the right, the sound goes from soft to hard, from sweetness to brassy trumpets and tubas. As you go left, it gets more delicate, with soft horns, piano or harp. In the back, you have your short and louds, like Piatti (cymbal) Snare, Bass Drum and Timpani. In the front, you have the long and softs, the strings.

There's a number of ways to achieve that 3 dimensional sound in your mixes, the sound that seems to jump right out at you. The most dramatic is reversing the phase on one channel of a stereo mix. Sound just leaps out, but there is a problem. Sum to mono and the whole image disappears, what we know as phase cancellation. Another way to do this is with a combination of a delay and pan controls. You hard pan the mix left and right and add a tiny, infinitesimal delay to one channel. Keep the amount of delay really small or the mix will get lopsided. Our ears can appreciate subtle shifts in the direction a sound comes from. As you add the delay, listen for the sound to "open up".

Put a delay before your reverb and set it to a 100% short delay with no feedback. Send a vocal line to the delay and then on to the reverb. In the mix, you'll first hear the dry vocal. The delay line then creates a gap before the reverb begins. This makes the room seem bigger, without needing a long (muddy) reverb time. Adjust the delay time to fit your music. On choppy vocals it's cool.

Dry sound /silence/reverb splash. Vocal reverb sounding muddy? Don't send so much bass to the reverb. Use EQ before the reverb and take out everything below 3,000 Hz. This gives a nice, bright splash on the plosives and hard consonant sounds. This can make the words more intelligible in a busy mix, too.

Compressor and Gates

The compressor, when properly used, automatically lowers the volume when the input exceeds a certain threshold. This allows a vocalist to get louder without going into the red. Try to have the input to the compressor boosted so that all the "soft" words come through with a strong level. As soon as the vocalist gets louder, the clamping down begins and if they scream, it clamps down hard. The ideal is to have more consistent loudness no matter what they are doing.

Gates have two parameters: 1) The noise floor threshold, and the rate. The noise floor threshold eliminates the entire signal when it dips below the threshold, which is set from 50db to -10db. If the gate is set too high, then the attack of the vocalist's words may be cut off or come in too abruptly. The rate parameter "fades out" the audio signal as the gate come on. This is effective to prevent the gate from chopping off the tails of the words. Usually a rate of 1-1.5 sec is enough.

The Threshold is the all-important level at which the compressor kicks in. If you set the threshold to -10, it will leave all of the signal under -10 alone. When the signal exceeds -10 then it starts compressing at the ratio. -10 is an excellent place to start. Don't confuse this with the fact that your gear is outputting -10 or +4 impedance wise. Though the threshold seems like it is a volume control, it is not. It is merely telling the compressor at what level compression takes over the signal.

When it comes to setting the Ratio, 2:1 is probably the most common setting for a compressor recording or playing back nearly anything. This is a great starting point. What this means, simply, is that it takes 2 decibels of sound energy to raise the output meter by 1dB. You can read the 1st number as the dB in and the second as the dB out. Again, 2dB in equals 1 dB out.

Compressors do add noise to a signal, and they do destroy dynamic range. Noise is taken care of by gating the signal. When it dips below a certain threshold, the audio signal is muted. This is effective for getting rid of low level noise you do not want in the file, such as bleed from headphones, or the vocalist moving, turning pages on lyric sheets, etc.

The louder the vocalist, the less sensitive the mic needs to be. Some condenser mics will distort badly if the vocalist is too close when they scream and it is an awful sound. There is nothing you can do to fix that audio either. Because the distortion happened before the signal hits the compressor, all the compression in the world cannot help. If there is a -10 or -20 pad on the mic, use it with untrained vocalists. Otherwise, use a dynamic mic, which is less susceptible to break up under high sound pressure levels (SPL). Or you can have them take a step back before you record them.

Vocals

The most common mistake is recording vocals too loud or too soft. It's not an easy thing to set levels with a good, dynamic vocalist. As soon as you think you have the level pegged, the vocalist may move a few inches and you find out they are louder than you thought and meters are now in the red. So you lower the level and find out that the meters are barely moving at all. The human voice is extremely dynamic, from soft whispers to piercing screams. If the level is too low, you will be bringing in noise and hum if you amplify it later. However, if you record too loud, there will be times when the vocal clips. This is damage that cannot be corrected later. The solution is to use a compressor in the chain after the preamp.

The second process of getting a good vocal sound is 'post-processing'. This is done after your vocal tracks are recorded. Here's where you can fix things like off key notes (with pitch processors), surgically remove short clicks and pops (with a pencil tool in an audio editor), replace words or phrases if necessary to make a great sounding composite vocal track. This is also where you apply effects and other processors to make the vocal tracks sound more professional.

One of the most challenging things in home recording is to record the human voice. To get the right performance, you not only have to be a competent engineer, but sometimes a cheerleader and psychologist. First let's lay out what it takes to get a good vocal sound. The room, mic, cables, pre-amp, compressor and recorder will determine the sound you get. My recommendation is to use the best you can afford in all categories.

If you don't own the best in one gear area, borrow or rent something if you have an important session. My favorite choices for female vocal is an AKG C12, for male vocals it's a Neumann U87. Both of these are tube mics and sound great, but they're also expensive. If you have a few to choose from, don't be afraid to try them all. Time spent up front experimenting is never wasted.

The limiter/compressor you use is just as important. Some styles of music call for a compressor with a big footprint. This means that you can hear it working. Other styles, like Jazz, call for a more transparent sound. Train your ear to hear what compression sounds like in a track. Chances are great that there is more compression on 99% of the vocal tracks you've ever heard. The bottom line is always use the best gear you can.

Exactly how close the mic is to the mouth when using the frontal close placement technique should be set for each speaking person. Listen for the balance of attributes versus disadvantages and move the mic accordingly. The starting point can be about 4" from the mouth directly on axis. You may even angle it by up to 90 to help smooth out the offending sound component. The microphone design must be very clear in it's off axis response for this to work. Have your assistant move the mic in and out from the voice, about 2" to 6" while the speaker is rehearsing and listen for the proximity warmth boost as it balances to consonant brightness. The pop filter should always be employed when close micing the mouth.

The most common method of recording a talking head for any purpose is the direct on axis close proximity placement technique, or the 'frontal close placement technique.' Producers commonly call this "in your face". It has a number of advantages, which are intimate sound, good articulation of consonants, up front or leading sound, and proximity warmth. It also has a number of disadvantages, which are; the need to de-ess, plosive popping, the lack of depth, and lack of natural room tone.

Each room has its good spots and we hope you can find your own in your room. Listen for a short (0.9 ms or less) reverberant quality. A good starting point for this 'loose placement' technique is around 12" from the mouth. By working with your assistant move the mic in and out while the speaker is rehearsing and listen to the voice to room tonal balance. Be careful not to get too roomy because it is hard to reduce later. By getting a little room tone in with the voice a more natural and comparatively rare sound will emerge.

A less common vocal micing technique is the 'loose placement' in front of the mouth. Most microphones cannot be used this way due to the unevenness of the off axis response. However a microphone with smooth off axis response can excel at this placement technique if the recording room is of a certain acoustic merit. The first advantage, if you can learn to use it in your recording, is a natural room tone. The balance between voice and room is set by the distance to the mouth and the placement of the speaker in the room.

The 'loose placement' technique gives natural depth of narration and applies to scenes where the talking head is not "in your face". You should notice that the need to de-ess and filter at low frequencies is reduced. The proximity warmth is replaced by natural room warmth, which is again more rare in today's recording. The recorded natural depth has the power to draw the listener into the recording rather than blow them back in their seats. This adds another trick to your engineering collection.

Sometimes the direct on axis techniques of close and loose do not produce the desired effects. This is when creative recording teams start to discover new ground. First, it requires a microphone that's off axis response is extremely clear.

Unstick yourself from the belief that there is only one way to record a speaking voice, and that is with the microphone pointed straight at the mouth. If the microphone is of a certain quality level with respect to off axis response a whole new vista can be dissevered and used in your daily technique. Sometimes the mic can wind up over the head pointed down and away from the speaking voice.

Still another technique that can be learned is the diaphragmatic, or 'lower chest placement' placement. This style of placement cannot be used easily if there is a standard music stand involved. The reason is; the pointing the microphone at the chest cavity.

While out of phase signals can sometimes be a good thing and useful, but in the studio out of phase signals are something that are not desirable. Not to say it can't be used creatively, but in general it's something to avoid. As an engineer, you should know, by ear, what an out of phase signal sounds like. In mono, the effect of two stereo signals being out of phase is drastic and undeniable. Whatever signals are shared by both speakers in a two channel system, like your home stereo, will disappear. Sometimes completely and sometimes not but it will sound "wrong". In stereo the effect is not as drastic but with a few repetitions you can hear the difference. You will hear the following things in a stereo signal that is out of phase. 1) Absence of center image. 2) Absence of low frequency.

Using phase adjustment to get rid of unwanted ambient sounds is a bit trickier than simply placing another mic in the recording room and flipping the phase switch. There are some other things added to the equation to make this work. Powerful real-time adaptive filters are needed to constantly track the interference then account for the difference in the interference picked up by the cockpit mic and the interference picked up by the wanted signal mic.

Because phase relationships are always at play in the real world, they can be incorporated into products used to get rid of unwanted sound. Think back to the last time you heard a helicopter traffic report. What was missing? The sound of the helicopter.

To clarify, by center image I mean the effect of what's called the "phantom image". When you sit between the speakers, whatever is shared by both speakers, is heard in the center (that is on a system that's in the proper polarity alignment). When that same system is out of phase, that center image is gone and the sound seems to come from around the side of your head. Take "absence of low frequency" to mean the things in the mix that occupy the lower end, like Kick drum, bass guitar etc. When low frequency is absent, the signal sounds very "thin".

Finding out if your stereo signal is in good shape is an easy matter. A signal out of phase will have an absence of low end, sound very thin or even sound like it's coming from around the side of your head. The best way to troubleshoot this in your studio is to put the console output into mono by either pushing the mono button or simply panning the two channels up the center. Bring the volume of the two mics up at equal levels and then flip one or other of the mics out of phase using the phase button on the console. You should hear a marked change in the sound (for the worse) as you flip the polarity. If your home studio does not have a console with a phase reverse switch, you can wire a cable out of phase and put it somewhere in line with one of the mics. Although a bit cumbersome it is the same thing as pushing a phase button. In reality, the mics can be at any degree of "out-of-phaseness".

A compressor is like an invisible hand on a volume control. This allows a vocalist to get louder without going into the red. One of the coolest settings is to have the input to the compressor boosted so that all the "soft" words come through with a strong level. As soon as the vocalist gets louder, the clamping down begins and if they scream, it clamps down hard. The ideal is to have more consistent loudness no matter what they are doing.

The more dynamic (louder) the vocalist, the less sensitive the mic needs to be. Some condenser mics will distort like madness if the vocalist is too close when they scream and it is an awful sound, especially if you are wearing cans (headphones). There is nothing you can do to fix that audio in the mix. Because the distortion happened before the signal hits the compressor, all the compression in the world cannot help. If there is a -10 or -20 pad on the mic, use it with untrained wild vocalists. Otherwise, use a dynamic mic that is less susceptible to break up under high sound pressure levels (SPL). Or you can have them take a step back before they commit their below from their personal living hell.

It's a good practice to mix your vocal track early. This way you can build around the vocals. Remember that they vocals are the focus of the song. The majority of the vocal range is in the mids, with a bit of warmth below 500 Hz. Cutting the rest of the mix a little bit in this area will help the vocal track sit well. Be careful though, so the track sits in the mix, but isn't way out front. It's also best to fit the instruments around the singer, rather than trying to cut a hole in the instruments to squeeze the vocals in.

If you'd like to add warmth to the voice, try adding some at the 300Hz to 500Hz ranges. This is mostly below the vocal range, but adds nice resonances. This effect can sometimes be achieved by simply moving the mic closer to the singer.

Using a delay and a pitch shifter can thicken vocals a little bit. Try adding a quick delay, between 5 and 25 milliseconds, than pitch shifting the delayed track a few cents up or down. This just smears the vocal a little, making it sound a little fatter. This will mess with the phase of the signal, however, and may make it so the song is not mono-compatible, so be careful, and check the results in mono. If your song is not going to be released, this is not necessary.

Guitars (Acoustic)

Getting a faithful reproduction of an acoustic guitar or bass takes a good deal of patience to master. Generally speaking, don't put the mic directly in front of the sound hole, as this is an invitation to mud. Try placing the mic either ahead of or behind the sound hole, and aimed towards it. An SM-57 will give you a good tone with enough experimenting, although typically a more expensive and transparent condenser mic is preferred. Acoustic guitars also seem to respond very nicely with a room mic to pick up the ambience, or just use a single mic far enough away to pick up both the guitar and the room's reflections. If you'll be distant micing, it's better to use a large diaphragm condenser in omni mode. This will give you the most natural sound of the guitar's interaction with the room. For obvious reasons, you'll need a very quiet area to record this way.

For some stereo imaging on the acoustic guitar track, try an X-Y mic setup and pan the resulting channels a little left and right of center. This provides a full sound without being too wide. An X-Y pattern is using 2 of the same mics with the capsules about 1" apart. The body's of the mics will point away from each other at an angle between 75

Guitars (Electric)

The SM-57 is probably the most popular mic for a guitar cabinet, although many mics work great. In a multiple speaker cabinet (2 x 10, 2 x 12, 4 x 12, etc) check all the speakers because one may sound better than the others may. This takes a little patience until you know the general sweet spot. Keep the mic angled at a small angle to the grille cloth (not perpendicular), and about 1" out. Point the mic at a portion of the cone, not near the center or edge. Crank the amp as loud as possible to get the sweet tone. Make sure you turn your gain down on your mic pre, as this tends to be a hot signal with a loud amp.

The placement of the mic on the guitar amp offers you a great deal of sonic flexibility. Try monitoring through headphones while someone else plays while moving the mic to slightly different positions on the cone, at different distances and different angles. You should be able to get a good sound without using EQ if you're patient enough. This is the best way to get a natural sound as EQ colors your recording too much if used heavily.

Try using a little compression when tracking. Set the compressor to the point where you see just a little bit of flicker in the gain reduction LED's (1 or 2 dB) with a 2:1 or 4:1 ratio. Leave the attack and release times in the middle of their adjustment ranges (20ms-80ms attack, 350ms-650ms release). Besides the small amount of compression and the appropriate distortion, record it dry, and add effects at mixdown.

If you decide that you want to try recording an electric guitar direct into the board, one method is to use a preamp output from the back of your amp. It is important in this scenario to utilize a DI to match impedances and eliminate hums. Note that you absolutely must leave a speaker load attached to your amp. This is especially important with a tube amp. You'll notice that this type of signal generally sounds harsh and edgy (even more with solid state amps). This happens because a lot of the characteristic sound of a guitar amp comes from the power amp and speakers. This is why most engineers prefer to mic a guitar amp as opposed to running the guitar direct.

Bass

Micing a bass amp is very similar to micing a guitar amp, but there are a few other things to consider. One of the important things is speaker size; a 4x10 sounds considerably different than a 1x15. The smaller speakers have better defined low end. You can hear the dynamics of the player much better with a cabinet with the smaller speakers, but there tends to be a lot of midrange that some players will object to. This midrange hump also takes up valuable space in the mix, taking away from the guitars and other instruments. The 1 x 15" will produce deep bass much more power than the 10" speaker will give. A 15" has a higher possibility of getting muddy, especially with more articulate playing, because the large cone reacts more slowly than a 10".

When recording a bass guitar amp, typically you'd use a dynamic cardioid mic, but a slightly more distant condenser mic can give good results too. It helps to raise the bass cab off the floor a bit with a cushion to stop the room resonances you sometimes get. Also, make sure you don't turn the amp up too loud, as the low frequencies overload mics pretty easily. You might have problems with 'standing waves' creating a boomy sound. These are frequencies that have wavelengths that are multiples of your room's dimensions. You need some form of bass trap to absorb these low frequencies. The best positions for these are corners. Blankets, drapes and normal foam won't work well because they aren't dense enough to absorb low frequencies. Also note that foam and cardboard egg crate is very flammable and should never be used.

Recording bass direct is also a very popular method. There are several ways of going direct, but basses are generally not plugged directly into the mixer input. Doing so usually creates an impedance mismatch that loads down the pickups and affects the tone. You can use a passive direct box, which is essentially an impedance matching transformer. There are also active direct boxes. These are basically a simple preamp. Some preamps have solid state electronics, while some have tubes. Some preamps offer nothing more than adjustable gain stages and a small EQ, while others have compression, delays, reverbs, flanges, and speaker simulators.

Mixing a mic signal and a direct input can give really nice results. If you have enough tracks, record each one to its own track so you have more possibilities when you mix. You can combine the two to a single track if you have to. Use a stereo compressor for these tracks so the gain reduction is consistent on each channel. Another important thing to accomplish when recording is trying to get a good sound without using much EQ on the console. If you overuse EQ it doesn't sound natural. If you spend the time to dial in your equipment you'll only need to adjust the EQ's a little to help it sit in the mix. A bass roll off around 100 or 150 Hz is important to keep the low bass defined, and a slight boost around 1.5 kHz will add some finger squeak (which sounds good in moderation).