The selection and placement of microphones can have a major influence on the sound of an acoustic recording. For many, the best approach is to capture a performance by a skilled musician, using a quality instrument with proper microphone techniques – little or no modification to the signal required. This simple approach can often sound better than an instrument that has been reshaped by a multitude of signal processing gear.

In this guide, Shure Application Engineers describe particular microphone techniques and placement: techniques to pick up a natural tonal balance, techniques to help reject unwanted sounds, and even techniques to create special effects.

Following this, some fundamentals of microphones, instruments, and acoustics are presented.

Microphone Techniques

Before getting into specific instruments and sound sources, let's quickly look at is a very basic, general procedure to keep in mind when miking something that makes sound.


  1. Use a microphone with a frequency response that is suited to the frequency range of the sound, if possible, or filter out frequencies above and/or below the highest and lowest frequencies of the sound.
  2. Place the microphone at various distances and positions until you find a spot where you hear from the studio monitors the desired tonal balance and the desired amount of room acoustics. If you don't like it, try another position, try another microphone, try isolating the instrument further, or change the sound of the instrument itself. For example, replacing worn out strings will change the sound of a guitar.
  3. Often you will encounter poor room acoustics, or pickup of unwanted sounds. In these cases, place the microphone very close to the loudest part of the instrument or isolate the instrument. Again, experiment with microphone choice, placement and isolation, to minimize the undesirable and accentuate the desirable direct and ambient acoustics.

Microphone technique is largely a matter of personal taste. Whatever method sounds right for the particular sound, instrument, musician, and song is right. There is no one ideal way to place a microphone. There is also no one ideal microphone to use on any particular instrument. Choose and place the microphone to get the sound you want. We recommend experimenting with all sorts of microphones and positions until you create your desired sound. However, the desired sound can often be achieved more quickly by understanding basic microphone characteristics, sound-radiation properties of musical instruments, and basic room acoustics.

Vocal Microphone Techniques

Individual Vocals


Directional Microphones

The standard vocal recording environment usually captures the voice only. This typically requires isolation and the use of a unidirectional mic. Isolation can be achieved with baffles surrounding the vocalist like a "shell" or some other method of reducing reflected sound from the room. Remember even a music stand can cause some reflections back to the mic. Though the effect on the resulting audio is likely minimal, when recording something as important as the vocal track, all elements should be considered.

Microphone Positioning

The axis of the microphone should usually be pointed somewhere between the nose and mouth to pick up the complete sound of the voice. Though the mic is usually directly in front of the singer's mouth, a slightly off-axis placement may help to avoid explosive sounds from breath blasts or certain consonant sounds such as "p", "b", "d", or "t". Placing the mic even further off-axis, or the use of an accessory pop filter, may be necessary to fully eliminate this problem.

While many vocals are recorded professionally in an isolation booth with a cardioid condenser microphone, other methods of vocal recording are practiced. For instance, a rock band's singers may be uncomfortable in the isolated environment described earlier. They may be used to singing in a loud environment with a monitor loudspeaker as the reference. This is a typical performance situation and forces them to sing louder and push their voices in order to hear themselves. This is a difficult situation to recreate with headphones.

Recording in the Live Room

A technique that has been used successfully in this situation is to bring the singers into the control room to perform. This would be especially convenient for project studios that exist in only one room. Once in that environment, a supercardioid dynamic microphone could be used in conjunction with the studio monitors. The singer faces the monitors to hear a mix of music and voice together. The supercardioid mic rejects a large amount of the sound projected from the speakers if the rear axis of the microphone is aimed between the speakers and the speakers are aimed at the null angle of the mic (about 65 degrees on either side of its rear axis). Just as in live sound, you are using the polar pattern of the mic to improve gain-before-feedback and create an environment that is familiar and encouraging to the vocalists. Now the vocalist can scream into the late hours of the night until that vocal track is right.

Omnidirectional Microphones

Microphones with various polar patterns can be used in vocal recording techniques. Consider recording a choral group or vocal ensemble. Having the vocalists circle around an omnidirectional mic allows well trained singers to perform as they would live: creating a blend of voices by changing their individual singing levels and timbres. Two cardioid mics, positioned back to back could be used for this same application.

Multi-Pattern Microphone

An omnidirectional mic may be used for a single vocalist as well. If the singer is in a room with ambience and reverb that add to the desired effect, the omnidirectional mic will capture the room sound as well as the singer's direct voice. By changing the distance of the vocalist to the microphone, you can adjust the balance of the direct voice to the ambience. The closer the vocalist is to the mic, the more direct sound is picked up relative to the ambience.

Ensemble Vocals

Application of ensemble, or group, vocals falls into the category known as "area" coverage. Rather than one microphone per sound source, the object is to pick up multiple sound sources (or a "large" sound source) with one (or more) microphone(s). Obviously, this introduces the possibility of interference effects unless certain basic principles (such as the "3-to-1 rule") are followed, as discussed below.

For one microphone picking up a typical choir, the suggested placement is a few feet in front of, and a few feet above, the heads of the first row. It should be centered in front of the choir and aimed at the last row. In this configuration, a cardioid microphone can "cover" up to 15-20 voices, arranged in a rectangular or wedge-shaped section.

Mic Positioning for Vocal Groups

For larger or unusually shaped choirs, it may be necessary to use more than one microphone. Since the pickup angle of a microphone is a function of its directionality (approximately 130 degrees for a cardioid), broader coverage requires more distant placement.

Multiple Microphones for Large Groups

3-to-1 Rule

In order to determine the placement of multiple microphones for choir pickup, remember the following rules: observe the 3-to-1 rule (see glossary); avoid picking up the same sound source with more than one microphone; and finally, use the minimum number of microphones.

3-to-1 Rule Provides Best Coverage

For multiple microphones, the objective is to divide the choir into sections that can each be covered by a single microphone. If the choir has any existing physical divisions (aisles or boxes), use these to define basic sections. If the choir is grouped according to vocal range (soprano, alto, tenor, bass), these may serve as sections.

If the choir is a single, large entity, and it becomes necessary to choose sections based solely on the coverage of the individual microphones, use the following spacing: one microphone for each lateral section of approximately6 to 9 feet. If the choir is unusually deep (more than 6 or 8 rows), it may be divided into two vertical sections of several rows each, with aiming angles adjusted accordingly. In any case, it is better to use too few microphones than too many. In a good-sounding space, a pair of microphones in a stereo configuration can provide realistic reproduction.

Podcasting and Content Creation

Podcasting and content creation have become popular methods for people to create and distribute content for a wide audience online and over the airwaves. A podcast is essentially spoken word, so the same tips that apply to speech will apply to a podcast application. Watch the following video to get the best performance out of one of the most popular podcast microphones, the SM7B.

The SM7B is a popular choice, however there is nothing magic about achieving great sound. For any microphone, follow the same tips as shown in the video:

1. Keep the microphone 6 - 12" from your mouth.

Generally, keep the microphone as close as possible to your mouth to avoid picking up unwanted room reflections and reverberation. The point where the mic sounds best is often called the sweet spot. This is usually around 6 - 12" (15 - 30 cm) from your mouth.

Do not get too close either. Proximity effect, which is an increase in low frequency response that occurs as you get closer to a directional microphone, can cause your voice to sound "muddy" or overly bassy.

Aim for the Sweet Spot

2. Aim the microphone toward your mouth from below or above.

This placement minimizes "popping" caused by plosive consonants (e.g. "p" or "t").

Positioned Slightly from Below

3. Use an external pop filter.

Though most microphones have some sort of built-in windscreen, an additional filter will provide extra insurance against "p" pops. The pop filter can also serve as a reference to help you maintain a consistent distance from the microphone.

External Pop Filter

4. Consider your Surroundings.

Reflections are caused by hard surfaces, such as walls, ceilings, even tabletops or music stands. They can adversely affect the sound quality captured by the microphone.

Use an Optimized Room When Possible

5. Speak directly into the microphone.

High frequencies are very directional, and if you turn your head away from the microphone, the sound captured by the microphone will get noticeably duller.

Speak Directly Into the Mic

Acoustic Stringed Instruments

Acoustic string instruments, such as the acoustic guitar, upright bass, violin, and more are complex sound sources. They greatly benefit from ample experimentation with mic placement to achieve accurate and pleasing sound reproduction.

It is also an opportunity for exploring sound manipulation, giving the studio engineer many paths to the final mix. Whether you are involved in a music studio, a commercial studio, or a project studio, you should continue to explore different methods of achieving the desired results. The possibilities are limited only by time and curiosity.

One of Many Ways to Mic an Acoustic Guitar

As a starting point for acoustic instruments, try placing one mic three - six in (7 - 15 cm) away, directly in front of the sound hole. Then put another microphone, of the same type, four feet (1.2 m) away. This will allow you to hear the instrument and an element of room ambience. Record both mics dry and flat (no effects or EQ), each to its own track. These two tracks will sound vastly different. Combining them may provide an open sound with the addition of the distant mic. Giving the effect of two completely different instruments or one in a stereo hallway may be achieved by enhancing each signal with EQ and effects unique to the sound you want to hear.

8 in. (20 cm) From Sound Hole

Options for Acoustic Guitar

After getting a sense of the instrument and the room characteristics, you can start to experiment with other techniques to obtain specific results. Below are some additional guidelines for a variety of tones and qualities.

Placement Options

Microphone Placement Tonal Balance Comments
1. 8 in. (20 cm) from sound hole (see image below) Bassy Good starting placement when leakage is a problem. Roll off bass for a more natural sound (more for a uni than an omni).
2. 3 in (7 cm) from sound hole Very bassy, boomy, muddy, full Very good isolation. Bass roll-off needed for a natural sound.
3. 4 - 8 in (10 - 20 cm) from bridge Woody, warm, mellow. Mid-bassy, lacks detail Reduces pick and string noise.
4. 6 in (15 cm) above the side, over the bridge, and even with the front soundboard Natural, well-balanced, slightly bright Less pickup of ambiance and leakage than 3 feet from sound hole.
Miniature microphone clipped outside of sound hole Natural, well-balanced Good isolation. Allows freedom of movement.
Miniature microphone clipped inside sound hole Bassy, less string noise Reduces leakage. Test positions to find each guitar's sweet spot.

Acoustic Bass (Upright Bass, String Bass, Bass Violin)

Upright Acoustic Bass

Microphone Placement Tonal Balance Comments
6-12 in (15-30 cm) out front, just above bridge Well-defined Natural sound.
2-6 in. (5-15 cm) from f-hole Full Roll off bass if sound is too boomy.
Wrap microphone in foam padding (except for grille) and put behind bridge or between tailpiece and body Full, "tight" Minimizes feedback and leakage.


Microphone Placement Tonal Balance Comments
3 in (8 cm) from center of head Bassy, thumpy Limits leakage. Roll off bass for natural sound.
3 in (8 cm) from edge of head Bright Limits leakage.

Miniature microphone clipped to tailpiece aiming at bridge

Natural Limits leakage. Allows freedom of movement.

Additional String Instruments

You can try the previously mentioned mic technique on any acoustic instrument. Attempt to position the mic in different areas over the instruments, listening for changes in timbre. You will find different areas offer different tonal characteristics. Soon you should develop "an ear" for finding instruments' sweet spots. In addition, the artist and style of music should blend with your experiences and knowledge to generate the desired effect.

Microphone Placement Tonal Balance Comments
Violin (Fiddle):
2-6 in. (5-15 cm) from side Natural Well-balanced sound.
1 ft (30 cm) from bridge Well-defined Well-balanced sound, but little isolation.
Aiming toward player at part of soundboard, about 2 ft (0.6 m) away Natural See "Stereo Microphone Techniques" section for other possibilities.
Tape miniature microphone to soundboard Somewhat constricted Minimizes feedback and leakage.
All String Instruments:
Miniature microphone attached to strings between bridge and tailpiece Bright Minimizes feedback and leakage. Allows freedom of movement.


Grand Piano

A Common Technique for Grand Piano

The grand piano is one of the largest and most iconic instruments in music recording. From the deep low end to the highest keys, the piano soundboard produces a huge range of sounds that can make any recording sound rich and full. In addition, the adjustable piano lid can greatly affect the sound in the room, affecting microphone choice and placement.

Grand Piano Placement Options

Microphone Placement Tonal Balance Comments
1. 12 in. (30 cm) above middle strings, 8 in. (20 cm) horizontally from hammers with lid off or at full stick Natural, well-balanced Less pickup of ambience and leakage than 3 feet out front. Move microphone(s) farther from hammers to reduce attack and mechanical noises. Good coincident-stereo placement. See "Stereo Microphone Techniques" section.
2. 8 in. (20 cm) above treble strings, as above Natural, well-balanced, slightly bright Place one microphone over bass strings and one over treble strings for stereo. Phase cancellations may occur if the recording is heard in mono.
3. Aiming into sound holes Thin, dull, hard, constricted Very good isolation. Sometimes sounds good for rock music. Boost mid-bass and treble for more natural sound.
4. 6 in. (15 cm) over middle strings, 8 in. (20 cm) from hammers, with lid on short stick Muddy, boomy, dull, lacks attack Improves isolation. Bass roll-off and some treble boost required for more natural sound.
5. Next to the underside of raised lid, centered on lid Bassy, full Unobtrusive placement.
6. Underneath the piano, aiming up at the soundboard Bassy, dull, full Unobtrusive placement.
7. Surface-mount microphone mounted on underside of lid over lower treble strings, horizontally, close to hammers for brighter sound, further from hammers for more mellow sound Bright, well-balanced Excellent isolation. Experiment with lid height and microphone placement on piano lid for desired sounds.
8. Two surface-mount microphones positioned on the closed lid, under the edge at its keyboard edge, approximately 2/3 of the distance from middle A to each end of the keyboard Bright, well-balanced, strong attack Excellent isolation. Moving "low" mic away from keyboard six inches provides truer reproduction of the bass strings while reducing damper noise. By splaying these two mics outward slightly, the overlap in the middle registers can be minimized.
9. Surface-mount microphone placed vertically on the inside of the frame, or rim, of the piano, at or near the apex of the piano's curved wall Full, natural Excellent isolation. Minimizes hammer and damper noise. Best if used in conjunction with two surface-mount microphones mounted to closed lid, as above.

Upright Piano

Upright Piano with Lid Removed

The upright piano is a more compact design than the grand piano, and yet it produces sounds that are just as complex, and with results that are just as satisfying, for a recording. Experiment with placement in the lid, with the lid completely removed, or even from the bottom or rear of the piano for more body and tonal differences.

Upright Piano Placement Options

Microphone Placement Tonal Balance Comments
1. Just over open top, above treble strings Natural (but lacks deep bass), picks up hammer attack Good placement when only one microphone is used.
2. Just over open top, above bass strings Slightly full or tubby, picks up hammer attack Mike bass and treble strings for stereo.
3. Inside top near the bass and

treble stings

Natural, picks up hammer attack Minimizes feedback and leakage. Use two microphones for stereo.
4. 8 in. (20 cm) from bass side of soundboard Full, slightly tubby, no hammer attack Use this placement with the following placement for stereo.
5. 8 in. (20 cm) from treble side of soundboard Thin, constricted, no hammer attack Use this placement with the preceding placement for stereo.
6. Aiming at hammers from front, several inches away (remove front panel) Bright, picks up hammer attack Mike bass and treble strings for stereo.
12 in. (30 cm) from center of soundboard on hard floor or one-foot-square plate on carpeted floor, aiming at piano (soundboard should face into room) Natural, good presence Minimize pickup of floor vibrations by mounting microphone in low-profile shock-mounted microphone stand.



With the saxophone, the sound is fairly well distributed between the finger holes and the bell. Miking close to the finger holes will result in key noise.

The soprano sax must be considered separately because its bell does not curve upward. This means that, unlike all other saxophones, placing a microphone toward the middle of the instrument will not pick-up the sound from the key holes and the bell simultaneously. The saxophone has sound characteristics similar to the human voice. Thus, a shaped response microphone designed for voice works well.

Saxophone Placement Options

Microphone Placement Tonal Balance Comments
2-6 in. (5-15 cm) from and aiming into bell Bright Minimizes feedback and leakage.
2-6 in. (5-15 cm) from sound holes Warm, full Picks up fingering noise.
2-6 in. (5-15 cm) above bell and aiming at sound holes Natural Good recording technique.
Miniature microphone mounted on bell Bright, punchy Maximum isolation, up-front sound.


The sound energy from a flute is projected both by the embouchure and by the first open fingerhole. For good pickup, place the mic as close as possible to the instrument. However, if the mic is too close to the mouth, breath noise will be apparent. Use a windscreen on the mic to overcome this difficulty.

Common Flute Placement

Microphone Placement Tonal Balance Comments
2-6 in. (5-15 cm) from area between mouthpiece and first set of finger holes Natural, breathy Pop filter or windscreen may be required on microphone.
2-6 in. (5-15 cm) behind player's head, aiming at finger holes Natural Reduces breath noise.

Oboe, Bassoon, etc

Microphone Placement Tonal Balance Comments
About 12 in. (30 cm) from sound holes Natural Provides well-balanced sound.
2-6 in. (5-15 cm) from bell Bright Minimizes feedback and leakage.


Popular Harmonica Mic

Microphone Placement Tonal Balance Comments
Very close to instrument Full, bright Minimizes feedback and leakage Microphone may be cupped in hands.


Tonal balance can be dramatically altered by adjusting the mic's position relative to the accordion. A good place to start is positioning the microphone about one to two feet from the instrument. At that distance, the sounds radiating from the accordion's surfaces combine into a pleasing composite. In contrast, a mic placed very close to the accordion tends to emphasize the surface nearest the microphone. The sound from a closely placed mic won't accurately capture the sound of the whole instrument.

Place the Mic About 1-2 ft (30-60 cm) Away

Microphone Placement Tonal Balance Comments
1-2 ft. (30-60 cm) in front of instrument, centered Full range, natural sound Use two microphones for stereo or to pick up bass and treble sides separately.
Miniature microphone mounted internally Emphasizes mid-range Minimizes leakage. Allows freedom of movement.


The complete family of brass instruments as we know them today, including the cornet, trumpet, trombone, tuba, and euphonium, date from about 1850. A key characteristic of brass instruments (apparent to almost everyone) is that they are all very loud. However, their loudness varies with the pitch because it requires much more energy to force the tube to resonate at the higher harmonics.

Miking too close to the instruments distorts the tonal balance, exaggerating some elements and understating others. Brass instruments require some distance from the mic to sound natural, as the sound needs some space to develop.

Trumpet, Cornet Trombone, Tuba

In general, the sound from brass instruments is very directional. Placing the mic off-axis with the instrument's bell will result in less pickup of high frequencies – leading to the sound being more diffused, with fewer upper harmonics and not as much bite.

Trombone and Trumpet Placement

Microphone Placement Tonal Balance Comments
1-2 ft. (30-60 cm) from bell (a couple of instruments can play into one microphone) On-axis to bell sounds bright; to one side sounds natural or mellow Close miking sounds "tight" and minimizes feedback and leakage. More distant placement gives fuller, more dramatic sound.
Miniature microphone mounted on bell Bright Maximum isolation.

French Horn

Microphone Placement Tonal Balance Comments
Microphone aiming toward bell Natural Watch out for extreme fluctuations on VU meter.

Amplified Instruments

Guitar and Bass Guitar Amplifiers Ready for Recording

Another "instrument" with a wide range of characteristics is the loudspeaker. Anytime you are recording a guitar or bass cabinet, you are confronted with the acoustic nature of loudspeakers (more commonly referred to in this context as the amplifier, or simply, the amp). A single loudspeaker is directional and displays different frequency characteristics at different angles and distances. On-axis at the center of a speaker tends to produce the most "bite", while off-axis or edge placement of the microphone produces a more "mellow" sound. A cabinet with multiple loudspeakers has an even more complex output, especially if it has different speakers for bass and treble.

A common approach is to close-mic an individual speaker. This is a habit people develop from viewing or doing live sound. In the live sound environment, most audio sources are close-miked to achieve the highest direct to ambient pickup ratios. Using unidirectional mics for close miking maximizes off-axis sound rejection as well. These elements lead to reduction of potential feedback opportunities.

Options Close Against the Speaker

But in the recording environment, the loudspeaker cabinet can be isolated and distant-mic techniques can be used to capture a more representative sound. As with most acoustic instruments, the desired sound develops at some distance away from the speaker. Often, by using both a close and a distant (more than a few feet) mic placement at the same time, it is possible to record a sound which has a controllable balance between "presence" and "ambience".

Placement of the loudspeaker cabinet itself can also have a significant effect on their sound. Putting cabinets on carpets can reduce brightness, while raising them off the floor can reduce low end. Open-back cabinets can be miked from behind as well as from the front. The distance from the cabinet to walls or other objects can also vary the sound. Again, move the instrument and the mic(s) around until you achieve something that you like!

Microphone Techniques for Amplifiers

Microphone Placement Tonal Balance Comments
1. 4 in. (10 cm) from grille cloth at center of speaker cone Natural, well-balanced Small microphone desk stand may be used if loudspeaker is close to floor.
2. 1 in. (2.5 cm) from grille cloth at center of speaker cone Bassy Minimizes feedback and leakage.
3. 4 in. (10 cm) off-center with respect to speaker cone Dull or mellow Microphone closer to edge of speaker cone results in duller sound. Reduces amplifier hiss noise.
4. 1-2 ft. (30-60 cm) Natural, open Use condenser microphone for position 4 -
3.and 4. combined for good 2-mi technique Options for blending Adjust spacing to minimize phase issues.
Miniature microphone draped over amp in front of speaker Emphasized midrange Easy setup, minimizes leakage.
Microphone placed behind open back cabinet Depends on position Can be combined with mic in front of cabinet, but be careful of phase cancellation.

Mic Placed Off-Center on a Bass Amp

Drums and Percussion

Miking a Drum Kit

The drum kit is one of the most complicated sound sources to record. Although there are many different methods, some common techniques and principles should be understood. Since the different parts of the drum kit have widely varying sound they should be considered as individual instruments, or at least a small group of instrument types: Kick, Snare, Toms, Cymbals, and Percussion. Certain mic characteristics are extremely critical for drum usage - let's cover a few of them here:

Dynamic Range A drum can produce very high Sound Pressure Levels (SPLs). The microphone must be able to handle these levels. A dynamic microphone will usually handle high SPLs better than a condenser. Check the Maximum SPL in condenser microphone specifications. It should be at least 130 dB for closeup drum use.
Directionality Because we want to consider each part of the kit an individual instrument; each drum may have its own mic. Interference effects may occur due to the close proximity of the mics to each other and to the various drums. Choosing mics that can reject sound at certain angles and placing them properly can be pivotal in achieving an overall drum mix with minimal phase problems.
Proximity Effect Unidirectional mics may have excessive low frequency response when placed very close to the drums. A low frequency roll-off either on the microphone or at the mixer will help cure a "muddied" sound. However, proximity effect may also enhance low frequency response if desired. It can also be used to effectively reduce pickup of distant low frequency sources by the amount of low roll-off used to control the closeup source. Typically, drums are isolated in their own room to prevent bleed through to microphones on other instruments. In professional studios it is common for the drums to be raised above the floor. This helps reduce low frequency transmission through the floor

Here is a basic individual drum miking technique:

  1. Bass (Kick) Drums - This drum's purpose in most music is to provide transient, low-frequency energy bursts that help establish the primary rhythmic pattern of a song. The kick drum's energy is primarily focused in two areas: very low-end timbre and "attack". Although this varies by individual drum, the attack tends to be in the 2.5-5 kHz range. A microphone for this use should have good low frequency response and possibly a boost in the attack range, although this can be done easily with EQ. The mic should be placed in the drum, in close proximity 1 - 6 in. (2.5 - 15 cm), facing the beater head. Or for less "slap" just inside the hole.
    Mic Placed Inside the Sound Hole
    Mic Close to the Resonant Head
    Mic Offset to Kick Drum
  2. Snare Drum - This is the most piercing drum in the kit and almost always establishes tempo. In modern music it usually indicates when to clap your hands! This is an extremely transient drum with little or no sustain to it. Its attack energy is focused in the 4 - 6kHz range.

    Typically, the drum is miked on the top head at the edge of the drum with a cardioid or supercardioid microphone.

    Dynamic Microphone
    Condenser Microphone
    Top and Bottom Technique
  3. Hats - These cymbals are primarily short, high frequency bursts used for time keeping, although the cymbals can be opened for a more loose sound. Many times the overhead mics will provide enough response to the high hat to eliminate the need for a separate hi-hat microphone. If necessary, a mic placed away from the puff of air that happens when hi-hats close and within four inches to the cymbals should be a good starting point.
    Straight Down on the Hi-Hats
    Angled Slightly Away from the Kit
  4. Tom Toms - While the kick and snare establish the low and high rhythmic functions, the toms are multiple drums that will be tuned from high to low between the snare and kick. They are primarily used for fills, but may also be consistent parts of the rhythmic structure. The attack range is similar to the snare drum, but often with more sustain.

    An individual directional mic on the top head near the edge can be used on each drum and panned to create some spatial imaging. A simpler setup is to place one mic slightly above and directly between two toms.

    Small Clip-on Microphone for Tight Spaces
  5. Overheads - The cymbals perform a variety of sonic duties from sibilant transient exclamation points to high frequency time keeping. In any case, the energy is mostly of a high-frequency content. Flat frequency response condenser microphones will give accurate reproduction of these sounds. Having microphones with low frequency roll-off will help to reject some of the sound of the rest of the kit which may otherwise cause phase problems when the drum channels are being mixed. The common approach to capturing the array of cymbals that a drummer may use is an overhead stereo pair of microphones. (positions A and B)
    Large Diaphragm Overheads
    Small Diaphragm Overheads

Simpler methods of drum miking are used for jazz and any application where open, natural kit sounds are desired. Using fewer mics over sections of the drums is common. Also, one high quality mic placed at a distance facing the whole kit may capture the sounds of kit and room acoustics in an enjoyable balance. Additional mics may be added to reinforce certain parts of the kit that are used more frequently.

Microphone Techniques for Drum Kit

When there are limited microphones available to record a drum kit use the following guidelines:

Number of microphones Positioning Alternative (Positioning reference)
One Use as "overhead" (A/ B)
Two Kick drum and overhead (A/B and D)
Three Kick drum, snare, and overhead or kick drum (A/B, C, D)
Four Kick drum, snare, high hat, and overhead (A/B, C, D, G)
Five Kick drum, snare, high hat, tom-toms, and overhead (A/B, C, D, E, G)

Additional Percussion

Microphone Placement Tonal Balance Comments
Timbales, Congas, Bongos:
One microphone aiming down between pair of drums, just above top heads Natural Provides full sound with good attack.
One microphone placed 6-12 in. (15-30 cm) from instrument Natural Experiment with distance and angles if sound is too bright.
Steel Drums:
Tenor Pan, Second Pan, Guitar Pan
One microphone placed 6 in. (10 cm) above each pan Bright, with plenty of attack Allow clearance for movement of pan.
Microphone placed underneath pan Decent if used for tenor or second pans. Too boomy with lower voiced pans.
Cello Pan, Bass Pan
One microphone placed 4-6 in. (10-15 cm) above each pan Natural Can double up pans to a single microphone.
Xylophone, Marimba, Vibraphone:
Two microphones aiming down toward instrument about 18 in (45 cm) above it; spaced 2 ft. (60 cm) apart, or angled 135 degrees apart with grilles touching Natural Pan two microphones to left and right for stereo. See "Stereo Microphone Techniques" section.
One microphone placed 4-6 in. (10-15 cm) above bars Bright, with lots of attack For less attack, use rubber mallets instead of metal mallets. Plastic mallets will give a medium attack.


One of the most popular specialized microphone techniques is stereo miking. This use of two or more microphones to create a stereo image will often give depth and spatial placement to an instrument or overall recording. There are a number of different methods for stereo. Three of the most popular:

  1. Spaced pair (A/B)
  2. Coincident or near-coincident pair (X-Y configuration)
  3. Mid-Side (M-S) technique

Spaced Pair (A/B)

The spaced pair (A/B) technique uses two cardioid or omni directional microphones spaced 3 - 10 feet apart from each other panned in left/right configuration to capture the stereo image of an ensemble or instrument. Effective stereo separation is very wide. The distance between the two microphones is dependent on the physical size of the sound source. For instance, if two mics are placed ten feet apart to record an acoustic guitar; the guitar will appear in the center of the stereo image. This is probably too much spacing for such a small sound source. A closer, narrower mic placement should be used in this situation.

The drawback to A/B stereo is the potential for undesirable phase cancellation of the signals from the microphones. Due to the relatively large distance between the microphones and the resulting difference of sound arrival times at the microphones, phase cancellations and summing may be occurring. A mono reference source can be used to check for phase problems. When the program is switched to mono and frequencies jump out or fall out of the sound, you can assume that there is phase problem. This may be a serious problem if your recording is going to be heard in mono as is typical in broadcast or soundtrack playback.

X-Y Coincident

The X-Y technique uses two cardioid microphones of the same type and manufacture with the two mic capsules placed either as close as possible (coincident) or within 12 inches of each other (near-coincident) and facing each other at an angle ranging from 90 - 135 degrees, depending on the size of the sound source and the particular sound desired. The pair is placed with the center of the two mics facing directly at the sound source and panned left and right.

Due to the small distance between the microphones, sound arrives at the mics at nearly the same time, reducing (near coincident) or eliminating (coincident) the possible phase problems of the A/B techniques. The stereo separation of this technique is good but may be limited if the sound source is extremely wide. Mono compatibility is fair (near-coincident) to excellent (coincident).

Mid-Side (M-S) Stereo

The M-S or Mid-Side stereo technique involves a cardioid mic element and a bi-directional mic element, usually housed in a single case, mounted in a coincident arrangement. The ① cardioid (mid) faces directly at the source and picks up primarily on-axis sound while the ② bi-directional (side) faces left and right and picks up off-axis sound. The two signals are combined via the M-S matrix to give a variable controlled stereo image. By adjusting the level of mid versus side signals, a narrower or wider image can be created without moving the microphone. This technique is completely mono-compatible and is widely used in broadcast and film applications.

Fundamentals of Microphones, Instruments, and Acoustics

The world of studio recording is much different from that of live sound reinforcement, but the fundamental characteristics of the microphones and sound are the same. It is the ability to isolate individual instruments that gives a greater element of control and freedom for creativity in the studio. Since there are no live loudspeakers, feedback is not an issue. The natural sound of the instrument may be the desired effect, or the sound source can be manipulated into a sound never heard in the natural acoustic world.

In order to achieve the desired result for your recording, it is useful to understand some of the important characteristics of microphones, musical instruments, and acoustics.

Microphone Characteristics

Shure Microphones

The microphone is the first link in the audio chain and is critical to the overall performance of the sound system. Proper selection of microphones depends on an understanding of basic microphone characteristics and on a knowledge of the intended application.

There are four areas of microphone characteristics that should be considered when selecting a microphone for a particular application. They are:

Operating Principle: How does the microphone change sound into an electrical signal?

The operating principle describes the type of transducer inside the microphone. A transducer is a device that changes energy from one form into another, in this case, acoustic energy into electrical energy. It is the part of the microphone that actually picks up sound and converts it into an electrical signal. The operating principle determines some of the basic capabilities of the microphone. The two most common types are dynamic and condenser.

Dynamic Microphones

Dynamic microphones employ a diaphragm/voice coil/magnet assembly which forms a miniature sound driven electrical generator. The motion of the voice coil in the magnetic field generates an electrical signal that corresponds to the sound.

Dynamic Microphone

This design is extremely rugged, has good sensitivity and can handle the loudest possible sound pressure levels without distortion. The dynamic has some limitations at extreme high and low frequencies. To compensate, small resonant chambers are often used to extend the frequency range of dynamic microphones.

Dynamics Excel on Loud Sources

Loud sound sources, such as drums and electric guitar amps, won't distort a dynamic microphone. The Beta57A is shown here positioned on the snare drum.

Ribbon microphone elements, a variation of the dynamic microphone operating principle, consist of a thin piece of metal, typically corrugated aluminum, suspended between two magnetic pole pieces. As with moving-coil dynamics, no additional circuitry or powering is necessary for operation, however, the output of ribbon microphones tends to be quite low. Depending on the gain of the mixer or recording device to which the microphone is connected, additional pre-amplification may be necessary. Note that ribbon microphones are not as rugged as moving-coil dynamic microphones. The ribbon element itself is typically no more than a few microns thick, and can be deformed by a strong blast of air, or by blowing into the microphone. Ribbon microphones are highly regarded in studio recording for their "warmth" and good low frequency response.

Phantom power applied to the ribbon microphone could be harmful.

Condenser Microphones

Condenser microphones are based on an electrically-charged diaphragm/back plate assembly which forms a sound-sensitive capacitor. Sound waves cause the diaphragm to move, which varies the spacing between the diaphragm and back plate, which produces an electrical signal corresponding to the sound.

Condenser Microphone

All condenser microphones contain additional circuitry that requires power either from batteries or from “phantom” power that is supplied through the microphone cable. Note that the electronics produce a small amount of noise (hiss), and that there is a limit to the maximum signal level that the electronics can handle without causing distortion. Well-designed condenser microphones, however, have very low noise levels and can also tolerate high signal levels.

Condenser microphones are more sensitive than dynamics, and have extended high frequency response that adds detail to voices and instruments. They can also be made very small, making them ideal for unique form factors or instrument-mounts where tight space is beneficial.

Condensers Capture Transients and Ambiance

Due to their sensitivity, condensers are often used to capture vocals, cymbals, and acoustic instruments in their most natural state.

If you hear distortion when using a condenser microphone close to a very loud sound source, first make sure that the mixer input itself is not being overloaded. If not, switch in the attenuator in the mic (if equipped), move the mic farther away, or use a mic that can handle a higher level. In any case, the microphone will not be damaged by excess level.

Most modern condenser microphones use solid state components for the internal circuitry, but older designs employed vacuum tubes (also known as "valves") for this purpose. The subjective qualities imparted by vacuum tube electronics, often described as "warmth" or "smoothness," have led to a resurgence in the popularity of vacuum tube-based condenser microphones. These sonic advantages come at the expense of higher self-noise and fragility. Vacuum tubes typically have a limited life span, and eventually need to be replaced. Most vacuum tube microphones require an external power supply, as standard 48V phantom power is not sufficient. Some power supplies offer the ability to switch polar patterns remotely on microphones that feature dual-diaphragms (see Directionality for a discussion of microphone polar patterns).

Frequency Response: How does the microphone sound?

The frequency response of a microphone is defined by the range of sound (from lowest to highest frequency) that it can reproduce, and by its variation in output within that range. It is the frequency response that determines the characteristic “sound” of the microphone.

The two general types of frequency response are flat and shaped. These terms refer to the general shape of the graphical frequency response curve.

Flat and Shaped Frequency Response

A flat response on the left compared to a shaped response on the right

A microphone that provides the same output at every audible frequency has a frequency response graph that is a generally flat line, and is said to have a flat response. This means that the microphone reproduces sound with little or no variation from the original source. Flat response microphones typically have an extended frequency range and can reproduce very high and/or very low frequencies as well. Flat response microphones tend to be used to reproduce sound sources without coloring the original source. This is usually desired in reproducing instruments such as acoustic guitars or pianos. It is also common for stereo miking techniques and distant miking techniques.

When a microphone is more sensitive to certain frequency ranges than others, it has a shaped response. A shaped response mic is more sensitive to important or desirable frequency ranges and less sensitive to undesirable ones. The frequency graph appears as a varying line with specific peaks and dips.

This response is designed to enhance a frequency range that is specific to a given sound source. For instance, a microphone may have a peak in the 2-10Khz range to enhance the intelligibility or presence of vocals. This shape is said to have a "presence peak". A microphone's response may also be reduced at other frequencies. One example of this is a low frequency roll-off to reduce unwanted "boominess".

Normally, microphones with flat, wide-range response are recommended for pickup of acoustic instruments, vocal ensembles, and orchestras, especially when they must be placed at some distance from the sound source. Mics with shaped response are especially ideal for vocal use. The frequency response includes enhanced sensitivity to the upper mid-range (often called a "presence rise") for speech intelligibility and vocal clarity, and reduced sensitivity to low frequencies to reduce pickup of room noise and mechanical vibration.

The frequency response of some microphones is adjustable to tailor the microphone to different applications. Most common are low-frequency roll off controls, which can help prevent “rumble”, and presence rise switches to enhance intelligibility.

Directionality: How does the microphone respond to sound from different directions?

The directionality of a microphone is defined as the variation of its output when it is oriented at different angles to the direction of the sound. It determines how best to place the microphone relative to the sound source(s) in order to enhance pickup of desired sound and to minimize pickup of undesired sound. The polar pattern of a microphone is the graphical representation of its directionality. The two most common directional types are omnidirectional and unidirectional.

A microphone that exhibits the same output regardless of its orientation to the sound source has a polar graph that is a smooth circle and is said to have an omnidirectional pattern. This indicates that the microphone is equally sensitive to sound coming from all directions. An omnidirectional microphone can therefore pick up sound from a wide area, but cannot be “aimed” to favor one sound over another.

Omnidirectional Microphone

Omnidirectional Microphone

A unidirectional microphone, on the other hand, is most sensitive to sound coming from only one direction. On a polar graph, this will appear as a rounded but non-circular figure. The most common type of unidirectional microphone is called a cardioid, because of its heart-shaped polar pattern.

Cardioid (Unidirectional) Microphone

Cardioid Microphone

A cardioid type is most sensitive to sound coming from in front of the microphone (the bottom of the “heart”). On the polar graph this is at 0 degrees, or “on axis”. It is less sensitive to sound from the sides (“off-axis”), and the direction of least sensitivity is toward the rear (the notch at the top of the “heart”). For any microphone, the direction of least sensitivity (minimum output) is called the null angle. For a cardioid pattern, this is at 180 degrees or directly behind the microphone.

Thus, a unidirectional microphone may be aimed at a desired, direct sound by orienting its axis toward the sound. It may also be aimed away from an undesired, direct sound by orienting its null angle toward the sound. In addition, a unidirectional microphone picks up less ambient sound than an omnidirectional, due to its overall lower sensitivity at the sides and rear. For example, a cardioid picks up only one-third as much ambient sound as an omnidirectional type.

For example, the use of a cardioid microphone for a guitar amplifier, which is in the same room as the drum set, is one way to reduce the bleed-through of drums on to the recorded guitar track. The mic is aimed toward the amplifier and away from the drums. If the undesired sound source is extremely loud (as drums often are), other isolation techniques may be necessary.

Unlike the cardioid, other unidirectional microphones have some pickup directly behind the microphone. This is indicated in their polar patterns by a rounded projection, called a lobe, toward the rear of the microphone. The direction of least sensitivity (null angle) for these types is about 125 degrees for the supercardioid and 110 degrees for the hypercardioid. In general, any directional pattern that has a narrower front coverage angle than a cardioid will have some rear pickup and a different null angle.

Supercardioid Microphone

Supercardioid Microphone

The significance of these two polar patterns is their greater rejection of ambient sound in favor of on-axis sound: the supercardioid has the maximum ratio of on-axis pickup to ambient pickup, while the hypercardioid has the least overall pickup of ambient sound (only one quarter as much as an omni - 115 degrees for the supercardioid and 105 degrees for the hypercardioid). These can be useful types for certain situations, such as more distant pickup or in higher ambient noise levels, but they must be placed more carefully than a cardioid to get best performance.

Both patterns offer narrower front pickup angles than the cardioid and also greater rejection of ambient sound. While the cardioid is least sensitive at the rear (180 degrees off-axis), the least sensitive direction is at 125 degrees for the supercardioid and 110 degrees for the hypercardioid. When placed properly they can provide more "focused" pickup and less room ambience than the cardioid pattern, but they have less rejection at the rear: -12 dB for the supercardioid and only -6 dB for the hypercardioid.

The bidirectional microphone has full response at both 0 degrees (front) and at 180 degrees (back). It has its least response at the sides. The coverage or pickup angle is only about 90 degrees at the front (or the rear). It has the same amount of ambient pickup as the cardioid. This mic could be used for picking up two sound sources such as two vocalists facing each other. It is also used in certain stereo techniques.

Directional Characteristics

Comparison Chart - Polar Patterns and Directionality

Other directional-related microphone characteristics:

Ambient sound sensitivity Since unidirectional microphones are less sensitive to off-axis sound than omnidirectional types, they pick up less overall ambient or room sound. Unidirectional mics should be used to control ambient noise pickup to get a "cleaner" recording.
Distance factor Since directional microphones have more rejection of off-axis sound than omnidirectional types, they may be used at greater distances from a sound source and still achieve the same balance between the direct sound and background or ambient sound. An omnidirectional microphone will pick up more room (ambient) sound than a unidirectional microphone at the same distance. An omni should be placed closer to the sound source than a "uni"- about half the distance - to pick up the same balance between direct sound and room sound.
Off-axis coloration A microphone's frequency response may not be uniform at all angles. Typically, high frequencies are most affected, which may result in an unnatural sound for off-axis instruments or room ambience.
Proximity effect For most unidirectional types, bass response increases as the microphone is moved closer to the sound source. When miking close with unidirectional microphones (less than 1 foot), be aware of proximity effect: it may help to roll off the bass until you obtain a more natural sound. You can (1) roll off low frequencies at the mixer, (2) use a microphone designed to minimize proximity effect, (3) use a microphone with a bass roll-off switch, or (4) use an omnidirectional microphone (which does not exhibit proximity effect).

Understanding and choosing the frequency response and directionality of microphones are selective factors which can improve pickup of desired sound and reduce pickup of unwanted sound. This can greatly assist in achieving both natural sounding recordings and unique sounds for special applications.

Electrical Output: How does the microphone output match the sound system input?

The electrical output of a microphone is characterized by its sensitivity, its impedance, and by its configuration. The same characteristics are used to describe microphone inputs in sound systems. This determines the proper electrical match of a microphone to a given sound system.

The sensitivity of a microphone is defined as its electrical output level for a certain input sound level. Impedance is essentially the electrical resistance of the microphone's output signal: 150-600 ohms for low impedance (low Z), 10,000 ohms or more for high impedance (high Z).

The output configuration can be either balanced or unbalanced. A balanced output carries the signal on two conductors that are wrapped with a metallic shield. The signals on each conductor are the same level but opposite polarity (one signal is positive when the other is negative). An unbalanced output signal is carried on a single conductor (plus shield) and is common for instruments such as electric/acoustic guitars, keyboards and others.

Unbalanced Cables and Connectors

Unbalanced Inputs

Balanced, low impedance microphone connections (typically with XLR connectors) are the standard for all professional microphones because they allow for long cable runs (over 100 feet) with no loss of quality while reducing pickup of electrical noise. Audio mixers commonly include balanced mic inputs (usually with phantom power for condenser microphones).

Balanced Inputs

Balanced Inputs

Phantom Power

Phantom power is a DC voltage (usually 12-48 volts) used to power the electronics of a condenser microphone. For some (non-electret) condensers it may also be used to provide the polarizing voltage for the element itself. This voltage is supplied through the microphone cable by a mixer equipped with phantom power or by some type of in-line external source. The voltage is equal on Pin 2 and Pin 3 of a typical balanced, XLR-type connector. For a 48 volt phantom source, for example, Pin 2 is 48 VDC and Pin 3 is 48 VDC, both with respect to Pin 1 which is ground (shield).

The Beta91A Condenser Microphone

The Beta91A is a condenser microphone that excels for achieving thump and attack on the kick drum. It will not pass audio unless phantom power is supplied.

Because the voltage is exactly the same on Pin 2 and Pin 3, phantom power will have no effect on balanced dynamic microphones: no current will flow since there is no voltage difference across the output. In fact, phantom power supplies have current limiting which will prevent damage to a dynamic microphone even if it is shorted or miswired. In general, balanced dynamic microphones can be connected to phantom powered mixer inputs with no problem.

Use caution as phantom power can cause damage to ribbon microphones in some circumstances. Ensure phantom power is not applied to the channel when connecting the ribbon microphone, unless that microphone specifically requires it (check the user manual first).

Acoustic Characteristics

Since room acoustics have been mentioned repeatedly, here is a brief introduction to some basic factors involved in acoustics.

Sound Waves

Sound waves consist of pressure variations traveling through the air. When the sound wave travels, it compresses air molecules together at one point. This is called the high pressure zone or positive component(+). After the compression, an expansion of molecules occurs. This is the low pressure zone or negative component(-). This process continues along the path of the sound wave until its energy becomes too weak to hear. If you could view the sound wave of a pure tone traveling through air, it would appear as a smooth, regular variation of pressure that could be drawn as a sine wave. The diagram shows the relationship of the air molecules and a sine wave.

Pressure Variations Produce Sound Waves

A Closer Look at the Sound Wave

A sound wave can be described by its frequency, amplitude, and wavelength. The frequency of a sound wave is the rate at which the pressure changes occur, and determines the "pitch" of the sound. It is measured in Hertz (Hz), where 1 Hz is equal to 1 cycle-per second. The range of frequencies audible to the human ear extends from a low of about 20 Hz to a high of about 20,000 Hz.

The amplitude of a sound wave refers to the magnitude of the pressure changes and determines the “loudness” of the sound. Amplitude is measured in decibels (dB) of sound pressure level (SPL) and ranges from 0 dB SPL (the threshold of hearing), to above 120 dB SPL (the threshold of pain). The level of conversational speech is about 70 dB SPL. A change of 1 dB is about the smallest SPL difference that the human ear can detect, while 3 dB is a generally noticeable step, and an increase of 10 dB is perceived as a “doubling” of loudness. (See The Decibel)

Schematic of a Sound Wave

Schematic of a Sound Wave


The wavelength of a sound wave is the physical distance from the start of one cycle to the start of the next cycle, as the wave moves through the air. The higher the frequency of sound, the shorter the wavelength, and the lower the frequency, the longer the wavelength.


One Full Cycle Equals the Wavelength


The fluctuation of air pressure created by sound is a change above and below normal atmospheric pressure. This is what the human ear responds to. The varying amount of pressure of the air molecules compressing and expanding is related to the apparent loudness at the human ear. The greater the pressure change, the louder the sound.

Under ideal conditions the human ear can sense a pressure change as small as .0002 microbar. One microbar is equal to one millionth of atmospheric pressure. The threshold of pain is about 200 microbar. Obviously, the human ear responds to a wide range of amplitude of sound.

This amplitude range is more commonly referred to in decibels. Sound Pressure Level (dB SPL), relative to .0002 microbar (0dB SPL). 0 dB SPL is the threshold of hearing and 120 dB SPL is the threshold of pain. 1 dB is about the smallest change in SPL that can be heard. A 3 dB change is generally noticeable, while a 6 dB change is very noticeable. A 10 dB SPL increase is perceived to be twice as loud!

Loudness of Common Sounds

Sound Transmission

In a recording studio, it is possible to separate or isolate the sounds being recorded. The best way to do this is to put the different sound sources in different rooms. This provides almost complete isolation and control of the sound from the voice or instrument. Unfortunately, multiple rooms are not always an option in studios, and even one sound source in a room by itself is subject to the effects of the walls, floor, ceiling and various isolation barriers. All of these effects can alter the sound before it actually arrives at the microphone.

In the study of acoustics there are three basic ways in which sound is altered by its environment:

Reflection A sound wave can be reflected by a surface or other object if the object is physically as large or larger than the wavelength of the sound. Because low-frequency sounds have long wavelengths, they can only be reflected by large objects. Higher frequencies can be reflected by smaller objects and surfaces. The reflected sound will have a different frequency characteristic than the direct sound if all sounds are not reflected equally. Reflection is also the source of echo, reverb, and standing waves:
  • Echo occurs when an indirect sound is delayed long enough (by a distant reflective surface) to be heard by the listener as a distinct repetition of the direct sound.
  • Reverberation consists of many reflections of a sound, maintaining the sound in a room for a time even after the direct sound has stopped.
  • Standing waves in a room occur for certain frequencies related to the distance between parallel walls. The original sound and the reflected sound will begin to reinforce each other when the wavelength is equal to the distance between two walls. Typically, this happens at low frequencies due to their longer wavelengths and the difficulty of absorbing them.
Refraction The bending of a sound wave as it passes through some change in the density of the transmission environment. This change may be due to physical objects, such as blankets hung for isolation or thin gobos, or it may be due to atmospheric effects such as wind or temperature gradients. These effects are not noticeable in a studio environment.
Diffraction A sound wave will typically bend around obstacles in its path which are smaller than its wavelength. Because a low frequency sound wave is much longer than a high frequency wave, low frequencies will bend around objects that high frequencies cannot. The effect is that high frequencies are more easily blocked or absorbed while low frequencies are essentially omnidirectional. When isolating two instruments in one room with a gobo as an acoustic barrier, it is possible to notice the individual instruments are "muddy" in the low end response. This may be due to diffraction of low frequencies around the acoustic barrier.


The phase of a single frequency sound wave is always described relative to the starting point of the wave or 0 degrees. The pressure change is also zero at this point. The peak of the high pressure zone is at 90 degrees, and the pressure change falls to zero again at 180 degrees. The peak of the low pressure zone is at 270 degrees, and the pressure change rises to zero at 360 degrees for the start of the next cycle.

Two identical sound waves starting at the same point in time are called "in-phase" and will sum together creating a single wave with double the amplitude but otherwise identical to the original waves. Two identical sound waves with one wave's starting point occurring at the 180-degree point of the other wave are said to be "out of phase", and the two waves will cancel each other completely. When two sound waves of the same single frequency but different starting points are combined, the resulting wave as said to have "phase shift" or an apparent starting point somewhere between the original starting points. This new wave will have the same frequency as the original waves but will have increased or decreased amplitude depending on the degree of phase difference. Phase shift, in this case, indicates that the 0 degree points of two identical waves are not the same.


When identical multiple-frequency soundwaves combine, there are three possibilities for the resulting wave: a doubling of amplitude at all frequencies if the waves are "in phase", a complete cancellation at all frequencies if the waves are 180 degrees "out of phase", or partial cancellation and partial reinforcement at various frequencies if the waves have intermediate phase relationship.

Most soundwaves are not a single frequency but are made up of many frequencies. Partial cancellation, therefore, is more common, and the audible result is a degraded frequency response called "comb filtering." The pattern of peaks and dips resembles the teeth of a comb and the depth and location of these notches depend on the degree of phase shift.

Phase Inversion at the Mixer or Preamp

Phase reversal at the microphone is a form of intentional phase alteration. If there is cancellation occurring, a 180 degree phase flip will create phase summing of the same frequencies. A common approach to the snare drum is to place one mic on the top head and one on the bottom head. Because the mics are picking up relatively similar sound sources at different points in the sound wave, you are probably experiencing some phase cancellations. Inverting the phase of one mic will sum any frequencies being canceled. This may sometimes achieve a "fatter" snare drum sound. This effect will change dependent on mic locations.

The phase inversion can be done with an in-line phase reverse adapter or by a phase invert switch found on many mixer inputs and preamps.

Instrument Characteristics

Musical instruments have overall frequency ranges as found in the chart below. The dark section of each line indicates the range of fundamental frequencies and the shaded section represents the range of the highest harmonics or overtones of the instrument. The fundamental frequency establishes the basic pitch of a note played by an instrument while the harmonics produce the timbre or characteristic tone.

Frequencies for each Instrument

Also, an instrument radiates different frequencies at different levels in every direction, and each part of an instrument produces a different timbre. This is the directional output of an instrument. You can partly control the recorded tonal balance of an instrument by adjusting the microphone position relative to it. The fact that low frequencies tend to be omnidirectional while higher frequencies tend to be more directional is a basic audio principle to keep in mind.

Most acoustic instruments are designed to sound best at a distance (say, two or more feet away). The sounds of the various parts of the instrument combine into a complete audio picture at some distance from the instrument. So, a microphone placed at that distance will pick up a "natural" or well-balanced tone quality. On the other hand, a microphone placed close to the instrument emphasizes the part of the instrument that the microphone is near. The sound picked up very close may or may not be the sound you wish to capture in the recording.

Applications Tip: Room Treatment

When building a project studio or small commercial studio, it is usually necessary to do some work on the room to improve the sound in the room. This can include:

Sound Treatment Add sound treatment to the walls and ceiling for a more even response from reflections.
Gobos for Isolation Buy or build gobos that help to isolate instruments and microphones from other sounds in the room.
Strategic Placement Position instruments strategically to get the best use of your space for a recording session.

The goal here is to create an awareness of the sources of these potential influences on recorded sound and to provide insight into controlling them. When an effect of this sort is heard, and is undesirable, it is usually possible to move the sound source, use a microphone with a different directional characteristic, or physically isolate the sound source further to improve the situation.

What is Absorption?

Absorption is the changing of sound energy into heat as it tries to pass through some material. Different materials have different absorption effects at multiple frequencies. Each material is measured with an absorption coefficient ranging between 0-1 (sabins). This can be thought of as the percentage of sound that will be absorbed.

For instance: a material may have an absorption coefficient of .67 sabins at 1,000 Hz. This would mean the material absorbs 67% of the 1,000 Hz frequencies applied to it.

Here is a chart showing the advantages of acoustic foam over bare walls or carpeting.

Absorption Efficiency Per Material

  1. Brick (unglazed)
  2. Heavy Carpet (on concrete)
  3. Concrete Block (unpainted)
  4. Foam (Sonex 2")
  5. Acoustile

Many small studios assume they can save money and achieve the desired absorption effect by using inexpensive carpet. This is not necessarily true as the graphs illustrate.

Direct vs Ambient Sound

A very important property of direct sound is that it becomes weaker as it travels away from the sound source, at a rate governed by the inverse-square law.

Inverse Square Law

Inverse Square Law

As the microphone moves way from the sound source, the signal spreads out over space and looses energy in any given direction, resulting in the predicable rate.

For example, when the distance from the source to the microphone increases by a factor of two (doubles), the sound level at the microphone decreases by a factor of four (the square of two). This results in a drop of 6 dB in sound pressure level (SPL), a substantial decrease. Likewise, when the distance from the mic to the source is divided by two (cut in half), the sound level increases by 6 dB.

On the other hand, the ambient sound in a room is at nearly the same level throughout the room. This is because the ambient sound has been reflected many times within the room until it is essentially non-directional. Reverberation is an example of non-directional sound. This is why the ambient sound of the room will become increasingly apparent as a microphone is placed further away from the direct sound source. The amount of direct sound relative to ambient sound can be controlled by the distance of the microphone to the sound source and to a lesser degree by the polar pattern of the mic. This is called the "critical distance" and becomes shorter as the ambient noise and reverberation increase, forcing closer placement of the microphone to the source.

Critical Distance

Shure Microphone Selection Guide

As discussed previously in this guide, you will get the best results by experimenting with placement and listening critically. Microphone selection is another important consideration for getting the right sound. Below are some suggestions of Shure microphones for certain instruments and sound sources. As always, we encourage you to get creative and listen closely; over time you will better understand each microphone's characteristics and your preferences in your recordings.


Solo Vocal KSM42 SM7B SM27 KSM44A KSM353/ED SM58 PGA27
Ensemble/Choir KSM32 KSM137 SM27 KSM44A BETA 181 PGA81 PGA181
Spoken Word (Broadcast/Podcasting/Streaming) KSM42 SM7B SM27 KSM44A KSM8 SM58 PGA27


Electric Guitar (Amplifier) KSM32 SM57 BETA 27 KSM313/NE KSM353/ED PGA181 PGA57
Acoustic Guitar KSM32 SM81 KSM141 KSM44A BETA 181 PGA81 SM57
Electric Bass (Amplifier) KSM32 BETA 27 BETA 52A KSM353/ED SM7B SM57 PGA52
Acoustic Bass KSM32 KSM137 BETA 27 KSM44A KSM313/NE SM137 SM27
Piano KSM44A SM81 KSM137 VP88 KSM32 PGA81 PGA27
Orchestra/Ensemble KSM141 KSM137 KSM32 KSM353/ED KSM44A SM27 PGA81
Strings KSM32 SM81 KSM137 KSM44A TL45 MX150 SM11
Woodwinds KSM32 KSM137 BETA 98H/C KSM44A BETA 181 SM27 PGA27
Brass/Saxophone KSM32 BETA 56A BETA 98H/C BETA 181 KSM44A PGA27 PGA81
Leslie Cabinet (top) KSM32 BETA 57A SM57 BETA 181 SM81 PGA57 PGA181
Leslie Cabinet (Bottom) BETA 52A SM7B KSM8 SM81 PGA52 PGA81
Harmonica 520DX SM58 BETA 58A PGA58


Kick Drum KSM353/ED BETA 52A SM7B KSM313/NE BETA 91A PGA52 PGA57
Snare (Top) BETA 57A SM57 BETA 98AMP BETA 181 KSM137 PGA57 PGA27
Snare (Bottom) KSM137 SM7B BETA 181 KSM141 SM81 PGA81 PGA57
Rack/Floor Toms BETA 56A/57A SM57 BETA 98AMP BETA 181 SM7B PGA56 PGA57
Overheads KSM32 SM81 KSM137 KSM313/NE KSM353/ED PGA81 PGA181
Congas BETA 27 BETA 56A/57A SM57 BETA 181 KSM9 PGA56 PGA57
Mallets KSM32 SM27 KSM137 KSM137 BETA 181 SM137 PGA27
Auxilary Percussion KSM32 SM27 SM57 BETA 181 KSM137 PGA27 PGA81

Stereo Techniques

X-Y KSM137 SM81 BETA 181 KSM32 SM27 SM137 PGA81
M-S VP88 BETA 181 (pair) KSM44A (pair) MV88+
Spaced Pair KSM32 SM81 KSM137 KSM44A KSM141 PGA27 PGA81

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The Decibel

The decibel (dB) is an expression often used in electrical and acoustic measurements. The decibel is a number that represents a ratio of two values of a quantity such as voltage. It is actually a logarithmic ratio whose main purpose is to scale a large measurement range down to a much smaller and more useable range.

Audio equipment signal levels are generally broken into 3 main categories: Mic, Line, or Speaker Level. Aux level resides within the lower half of line level. The chart also shows at what voltages these categories exist.

Conversion Chart

Conversion Chart

One reason that the decibel is so useful in certain audio measurements is that this scaling function closely approximates the behavior of human hearing sensitivity. For example, a change of 1 dB SPL is about the smallest difference in loudness that can be perceived while a 3dB SPL change is generally noticeable. A 6 dB SPL change is quite noticeable and finally, a 10 dB SPL change is perceived as twice as loud.

Decibel Equation

Since the decibel is a ratio of two values, there must be an explicit or implicit reference value for any measurement given in dB. This is usually indicated by a suffix on the dB. Some devices are measured in dBV (reference to 1 Volt = 0 dBV), while others may be specified in dBu or dBm (reference to .775 V = 0dBu/dBm).

The form of the decibel relationship for voltage is:

dB = 20 x log(V1/V2)

where 20 is a constant, V1 is one voltage, V2 is a reference voltage, and log is logarithm base 10.


What is the relationship in decibels between 100 volts and 1 volt?

(dbV) dB = 20 x log(100/1)dB = 20 x log(100)dB = 20 x 2 (the log of 100 is 2)dB = 40

That is, 100 volts is 40dB greater than 1 volt.

What is the relationship in decibels between .0001 volt and 1 volt? (dbV)

dB = 20 x log(.001/1)dB = 20 x log(.001)dB = 20 x (-3) (the log of .001 is -3)dB = -60

That is, .001 volt is 60 dB less than 1 volt.

Similarly: If one voltage is equal to the other, they are 0 dB different.

If one voltage is twice the other, they are 6 dB different.

If one voltage is ten times the other, they are 20 dB different.

Transient Response

The ability of a microphone to respond to a rapidly changing sound wave.

A good way to understand why dynamic and condenser mics sound different is to understand the differences in their transient response.

In order for a microphone to convert sound energy into electrical energy, the sound wave must physically move the diaphragm of the microphone. The speed of this movement depends on the weight or mass of the diaphragm. For instance, the diaphragm and voice coil assembly of a dynamic microphone may have up to 1000 times the mass of the diaphragm of a condenser microphone. The lightweight condenser diaphragm starts moving much more quickly than the dynamic's diaphragm. It also takes longer for the dynamic's diaphragm to stop moving in comparison to the condenser's diaphragm. Thus, the dynamic's transient response is not as good as the condenser's transient response. This is similar to two vehicles in traffic: a truck and a sports car. They may have engines of equal power, but the truck weighs much more than the car. As traffic flow changes, the sports car can accelerate and brake very quickly, while the semi accelerates and brakes very slowly due to its greater weight. Both vehicles follow the overall traffic flow but the sports car responds better to sudden changes.

The picture below is of two studio microphones responding to the sound impulse produced by an electric spark: condenser mic on top, dynamic mic on bottom. It is evident that it takes almost twice as long for the dynamic microphone to respond to the sound. It also takes longer for the dynamic to stop moving after the impulse has passed (notice the ripple on the second half of the graph). Since condenser microphones generally have better transient response then dynamics, they are better suited for instruments that have very sharp attacks or extended high frequency output such as cymbals. It is this transient response difference that causes condenser mics to have a more crisp, detailed sound and dynamic mics to have a more mellow, rounded sound.


3-to-1 Rule - When using multiple microphones, the distance between microphones should be at least 3 times the distance from each microphone to its intended sound source.

Absorption - The dissipation of sound energy by losses due to sound absorbent materials.

Active Circuitry - Electrical circuitry which requires power to operate, such as transistors and vacuum tubes.

Ambience - Room acoustics or natural reverberation.

Amplitude - The strength or level of sound pressure or voltage.

Audio Chain - The series of interconnected audio equipment used for recording or PA.

Backplate - The solid conductive disk that forms the fixed half of a condenser element.

Balanced - A circuit that carries information by means of two equal but opposite polarity signals, on two conductors.

Bidirectional Microphone - A microphone that picks up equally from two opposite directions. The angle of best rejection is 90 degrees from the front (or rear) of the microphone, that is, directly at the sides.

Boundary/Surface Microphone - A microphone designed to be mounted on an acoustically reflective surface.

Cardioid Microphone - A unidirectional microphone with moderately wide front pickup (131 degrees). Angle of best rejection is 180 degrees from the front of the microphone, that is, directly at the rear.

Cartridge (Transducer) - The element in a microphone that converts acoustical energy (sound) into electrical energy (the signal).

Clipping Level - The maximum electrical output signal level (dBV or dBu) that the microphone can produce before the output becomes distorted.

Close Pickup - Microphone placement within 2 feet of a sound source.

Comb Filtering - An interference effect in which the frequency response exhibits regular deep notches.

Condenser Microphone - A microphone that generates an electrical signal when sound waves vary the spacing between two charged surfaces: the diaphragm and the backplate.

Critical Distance - In acoustics, the distance from a sound source in a room at which the direct sound level is equal to the reverberant sound level.

Current - Charge flowing in an electrical circuit. Analogous to the amount of a fluid flowing in a pipe.

Decibel (dB) - A number used to express relative output sensitivity. It is a logarithmic ratio.

Diaphragm - The thin membrane in a microphone which moves in response to sound waves.

Diffraction - The bending of sound waves around an object which is physically smaller than the wavelength of the sound.

Direct Sound - Sound which travels by a straight path from a sound source to a microphone or listener.

Distance Factor - The equivalent operating distance of a directional microphone compared to an omnidirectional microphone to achieve the same ratio of direct to reverberant sound.

Distant Pickup - Microphone placement farther than 2 feet from the sound source.

Dynamic Microphone - A microphone that generates an electrical signal when sound waves cause a conductor to vibrate in a magnetic field. In a moving-coil microphone, the conductor is a coil of wire attached to the diaphragm. In a ribbon microphone, the diaphragm is the conductor.

Dynamic Range - The range of amplitude of a sound source. Also, the range of sound level that a microphone can successfully pick up.

Echo - Reflection of sound that is delayed long enough (more than about 50 msec.) to be heard as a distinct repetition of the original sound.

Electret - A material (such as Teflon) that can retain a permanent electric charge.

EQ - Equalization or tone control to shape frequency response in some desired way.

Feedback - In a PA system consisting of a microphone, amplifier, and loudspeaker, feedback is the ringing or howling sound caused by amplified sound from the loudspeaker entering the microphone and being re-amplified.

Flat Response - A frequency response that is uniform and equal at all frequencies.

Frequency - The rate of repetition of a cyclic phenomenon such as a sound wave.

Frequency Response Tailoring Switch - A switch on a microphone that affects the tone quality reproduced by the microphone by means of an equalization circuit. (Similar to a bass or treble control on a hi-fi receiver.)

Frequency Response - A graph showing how a microphone responds to various sound frequencies. It is a plot of electrical output (in decibels) vs. frequency (in Hertz).

Fundamental - The lowest frequency component of a complex waveform such as musical note. It establishes the basic pitch of the note.

Gain - Amplification of sound level or voltage.

Gain-Before-Feedback - The amount of gain that can be achieved in a sound system before feedback or ringing occurs.

Gobos - Movable panels used to reduce reflected sound in the recording environment.

Harmonic - Frequency components above the fundamental of a complex waveform. They are generally multiples of the fundamental which establish the timbre or tone of the note.

Hypercardioid - A unidirectional microphone with tighter front pickup (105 degrees) than a supercardioid, but with more rear pickup. Angle of best rejection is about 110 degrees from the front of the microphone.

Impedance - In an electrical circuit, opposition to the flow of alternating current, measured in ohms. A high-impedance microphone has an impedance of 10,000 ohms or more. A low-impedance microphone has an impedance of 50 to 600 ohms.

Interference - Destructive combining of sound waves or electrical signals due to phase differences.

Inverse Square Law - States that direct sound levels increase (or decrease) by an amount proportional to the square of the change in distance.

Isolation - Freedom from leakage; the ability to reject unwanted sounds.

Leakage - Pickup of an instrument by a microphone intended to pick up another instrument. Creative leakage is artistically favorable leakage that adds a "loose" or "live" feel to a recording.

Maximum Sound Pressure Level - The maximum acoustic input signal level (dB SPL) that the microphone can accept before clipping occurs.

Microphone Sensitivity - A rating given in dBV to express how "hot" the microphone is by exposing the microphone to a specified sound field level (typically either 94 dB SPL or 74 dB SPL). This specification can be confusing because manufacturers designate the sound level different ways. Here is an easy reference guide: 94 dB SPL = 1 Pascal = 10 microbars. To compare a microphone that has been measured at 74 dB SPL with one that has been measured at 94 dB SPL, simply add 20 to the dBV rating.

NAG - Needed Acoustic Gain is the amount of gain that a sound system must provide for a distant listener to hear as if he or she was close to the unamplified sound source.

Noise - Unwanted electrical or acoustic interference.

Noise Cancelling - A microphone that rejects ambient or distant sound.

NOM - Number of open microphones in a sound system. Decreases gain-before-feedback by 3dB everytime NOM doubles.

Omnidirectional Microphone - A microphone that picks up sound equally well from all directions.

Output Noise (Self-Noise) - The amount of residual noise (dB SPL) generated by the electronics of a condenser microphone.

Overload - Exceeding the signal level capability of a microphone or electrical circuit.

PAG - Potential Acoustic Gain is the calculated gain that a sound system can achieve at or just below the point of feedback.

Phantom Power - A method of providing power to the electronics of a condenser microphone through the microphone cable.

Phase - The "time" relationship between cycles of different waves.

Pickup Angle/Coverage Angle - The effective arc of coverage of a microphone, usually taken to be within the 3dB down points in its directional response.

Pitch - The fundamental or basic frequency of a musical note.

Polar Pattern (Directional Pattern, Polar Response) - A graph showing how the sensitivity of a microphone varies with the angle of the sound source, at a particular frequency. Examples of polar patterns are unidirectional and omnidirectional.

Polarization - The charge or voltage on a condenser microphone element.

Pop Filter - An acoustically transparent shield around a microphone cartridge that reduces popping sounds. Often a ball-shaped grille, foam cover or fabric barrier.

Pop - A thump of explosive breath sound produced when a puff of air from the mouth strikes the microphone diaphragm. Occurs most often with "p", "t", and "b" sounds.

Presence Peak - An increase in microphone output in the "presence" frequency range of 2,000 Hz to 10,000 Hz. A presence peak increases clarity, articulation, apparent closeness, and "punch."

Proximity Effect - The increase in bass occurring with most unidirectional microphones when they are placed close to an instrument or vocalist (within 1 foot). Does not occur with omnidirectional microphones.

Rear Lobe - A region of pickup at the rear of a supercardioid or hypercardioid microphone polar pattern. A bidirectional microphone has a rear lobe equal to its front pickup.

Reflection - The bouncing of sound waves back from an object or surface which is physically larger than the wavelength of the sound.

Refraction - The bending of sound waves by a change in the density of the transmission medium, such as temperature gradients in air due to wind.

Resistance - The opposition to the flow of current in an electrical circuit. It is analogous to the friction of fluid flowing in a pipe.

Reverberation - The reflection of a sound a sufficient number of times that it becomes non-directional and persists for some time after the source has stopped. The amount of reverberation depends on the relative amount of sound reflection and absorption in the room.

Rolloff - A gradual decrease in response below or above some specified frequency.

Sensitivity - The electrical output that a microphone produces for a given sound pressure level.

Shaped Response - A frequency response that exhibits significant variation from flat within its range. It is usually designed to enhance the sound for a particular application.

Signal to Noise Ratio - The amount of signal (dBV) above the noise floor when a specified sound pressure level is applied to the microphone (usually 94 dB SPL).

Sound Chain - The series of interconnected audio equipment used for recording or PA.

Sound Reinforcement - Amplification of live sound sources.

Speed of Sound - The speed of sound waves, about 1130 feet per second in air.

SPL - Sound Pressure Level is the loudness of sound relative to a reference level of 0.0002 microbars.

Standing Wave - A stationary sound wave that is reinforced by reflection between two parallel surfaces that are spaced a wavelength apart.

Supercardioid Microphone - A unidirectional microphone with tighter front pickup angle (115 degrees) than a cardioid, but with some rear pickup. Angle of best rejection is 126 degrees from the front of the microphone, that is, 54 degrees from the rear.

3-to-1 Rule - (See top of page 34.)

Timbre - The characteristic tone of a voice or instrument; a function of harmonics.

Transducer - A device that converts one form of energy to another. A microphone transducer (cartridge) converts acoustical energy (sound) into electrical energy (the audio signal).

Transient Response - The ability of a device to respond to a rapidly changing input.

Unbalanced - A circuit that carries information by means of one signal on a single conductor.

Unidirectional Microphone - A microphone that is most sensitive to sound coming from a single direction-in front of the microphone. Cardioid, supercardioid, and hypercardioid microphones are examples of unidirectional microphones.

Vacuum Tube (valve) - An electric device generally used to amplify a signal by controlling the movement of electrons in a vacuum. Vacuum tubes were widely used in the early part of the 20th century, but have largely been replaced by transistors.

Voice Coil - Small coil of wire attached to the diaphragm of a dynamic microphone.

Voltage - The potential difference in an electric circuit. Analogous to the pressure on fluid flowing in a pipe.

Wavelength - The physical distance between the start and end of one cycle of a soundwave.