Shure's Microphone Techniques for Studio Recording Section Two: Microphone Characteristics
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About Shure's Microphone Techniques for Studio Recording
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Section Two: Microphone Characteristics
Introduction
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 it is useful to understand some of the important characteristics of microphones, musical instruments, and acoustics.
Microphone Characteristics
There are three main considerations when choosing a microphone for recording applications: operating principle, frequency response, and directionality.
Operating Principle
A microphone is an example of a “transducer, ”a device which changes energy from one form into another, in this case from acoustic into electrical. The type of transducer is defined by the operating principle. In the current era of recording, the two primary operating principles used in microphone design are the dynamic and the condenser.
Dynamic Microphone Elements
Condenser Microphone Elements
The main limitations of a condenser microphone relate to its electronics. These circuits can handle a specified maximum signal level from the condenser element, so a condenser mic has a maximum sound level before its output starts to be distorted. Some condensers have switchable pads or attenuators between the element and the electronics to allow them to handle higher sound levels. 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. A second side effect of the condenser/electronics design is that it generates a certain amount of electrical noise (self-noise) which may be heard as “hiss” when recording very quiet sources at high gain settings. Higher quality condenser mics have very low self-noise, a desirable characteristic for this type of recording application.
Frequency Response
Directionality
The sensitivity to sound relative to the direction or angle of arrival at the microphone. Directionality is usually plotted on a graph referred to as a polar pattern. The polar pattern shows the variation in sensitivity 360 degrees around the microphone, assuming that the microphone is in the center and 0 degrees represents the front or on-axis direction of the microphone. There are a number of different directional patterns designed into microphones. The three basic patterns are omnidirectional, unidirectional, and bidirectional.
Omnidirectional
Unidirectional
Bidirectional
The bidirectionalmicrophone 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.
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:
- roll off low frequencies at the mixer.
- use a microphone designed to minimize proximity effect,
- use a microphone with a bass roll-off switch, or
- 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.
Instrument Characteristics
First, let’s present a bit of background information about how instruments radiate sound. The sound from a musical instrument has a frequency output which is the range of frequencies produced and their relative amplitudes. The fundamental frequencies establish the basic pitch, while the harmonic frequencies produce the timbre or characteristic tone of the instrument. Here are frequency ranges for some commonly known instruments:
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.
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.
Frequency, Wavelength, and the Speed of Sound
The frequency of a sound wave indicates the rate of pressure variations or cycles. One cycle is a change from high pressure to low pressure and back to high pressure.
The number of cycles per second is called Hertz, abbreviated “Hz.” So, a 1, 000Hz tone has 1, 000 cycles per second.
The wavelength of a sound is the physical distance from the start of one cycle to the start of the next cycle. Wavelength is related to frequency by the speed of sound. The speed of sound in air is 1130 feet per second or 344 meters/second. The speed of sound is constant no matter what the frequency. You can determine the wavelength of a sound wave of any frequency if you understand these relationships:
Loudness
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!
Sound Transmission
It is important to remember that sound transmission does not normally happen in a completely controlled environment. In a recording studio, though, 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
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.
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 controlled by the inverse-square law. When the distance from a sound source doubles, the sound level decreases by 6dB. This is a noticeable audible decrease. For example, if the sound from a guitar amplifier is 100 dB SPL at 1 ft. from the cabinet it will be 94 dB at 2 ft., 88 dB at 4 ft., 82 dB at 8 ft., etc. When the distance is cut in half the sound level increases by 6dB: It will be 106 dB at 6 inches and 112 dB at 3 inches. 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. However, if the microphone is placed beyond a certain distance from the sound source, the ambient sound will begin to dominate the recording and the desired balance may not be possible to achieve, no matter what type of mic is used. 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.
Phase Relationships and Interference Effects
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. Most soundwaves are not a single frequency but are made up of many frequencies. 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.
The last case is the most likely, 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.
With microphones this effect can occur in two ways. The first is when two (or more) mics pick up the same sound source at different distances. Because it takes longer for the sound to arrive at the more distant microphone, there is effectively a phase difference between the signals from the mics when they are combined (electrically) in the mixer. The resulting comb filtering depends on the sound arrival time difference between the microphones:a large time difference (long distance) causes comb filtering to begin at low frequencies, while a small time difference (short distance) moves the comb filtering to higher frequencies.
When these sounds combine (acoustically) at the microphone, comb filtering results. This time the effect of the comb filtering depends on the distance between the microphone and the source of the reflection or the distance between the multiple sources. 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.
About the Authors
John Boudreau
John, a lifelong Chicago native, has had extensive experience as a musician, a recording engineer, and a composer. His desire to better combine the artistic and technical aspects of music led him to a career in the audio field. Having received a BS degree in Music Business from Elmhurst College, John performed and composed for both a Jazz and a Rock band prior to joining Shure in 1994 as an associate in the Applications Engineering group. While at Shure, John led many audio product training seminars and clinics, with an eye to helping musicians and others affiliated with the field use technology to better fulfill their artistic interpretations. No longer a Shure associate, John continues to pursue his interests as a live and recorded sound engineer for local bands and venues, as well as writing and recording for his own band. RICKFRANKOver his career, Rick has been involved in a wide variety of music and recording activities including composing, teaching, performing, and producing popular music, jazz and commercial jingles. He has spent his life in Illinois where he received his BS in English and his MBA from the University of Illinois, Urbana-Champaign. While in downstate Illinois he also operated a successful retail musical instrument business and teaching program that coincided with working as a professional guitarist and electric bassist. Rick’s work at Shure began as Marketing Specialist for Music and Sound Reinforcement where he was responsible for researching and analyzing new product concepts and introducing new wired and wireless products to the global market. He has presented training seminars to audiences and written instructional materials on a variety of audio subjects including Understanding Sound Reinforcement using the Potential Acoustic Gain Equation. He is currently Marketing Director for Wired Microphones at Shure where he is responsible for Music industry products including microphones and other products for recording and he continues to perform music professionally.
Tim Vear
Tim is a native of Chicago who has come to the audio field as a way of combining a lifelong interest in both entertainment and science. He has worked as an engineer in live sound, recording and broadcast, has operated his own recording studio and sound company, and has played music professionally since high school.
At the University of Illinois, Urbana-Champaign, Tim earned a BS in Aeronautical and Astronautical Engineering with a minor in Electrical Engineering. During this time he also worked as chief technician for both the Speech and Hearing Science and Linguistics departments. In his tenure at Shure, Tim has served in a technical support role for the sales and marketing departments, providing product and applications training for Shure customers, dealers, installers, and company staff. He has presented seminars for a variety of domestic and international audiences, including the National Systems Contractors Association, the Audio Engineering Society and the Society of Broadcast Engineers. Tim has authored several publications for Shure and his articles have appeared in several trade publications.
Rick Waller
Now residing in the Chicago area, Rick grew up near Peoria, Illinois. An interest in the technical and musical aspects of audio has led him to pursue a career as both engineer and musician. He received a BS degree in Electrical Engineering from the University of Illinois at Urbana/Champaign, where he specialized in acoustics, audio synthesis and radio frequency theory. Rick is an avid keyboardist, drummer and home theater hobbyist and has also worked as a sound engineer and disc jockey. Currently he is an associate in the Applications Engineering Group at Shure. In this capacity Rick provides technical support to customers, writing and conducting seminars on wired and wireless microphones, mixers and other audio topics.
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