Академический Документы
Профессиональный Документы
Культура Документы
Transducers
The text and diagrams that are to follow will show how a voltage can be generated by moving a conductor through a magnetic field. The generator which does this might be thought of as a transducer, which is a device which converts one form of energy into another. A microphone is a transducer, in that it converts acoustic energy (variations in air pressure) first into mechanical energy (movement of diaphragm), and then into electrical energy. Of the many methods that have been tried over the years to converts acoustical events in time and space into electricity, today, most practical studio quality microphones may be classified under one of the two major categories, the dynamic microphone and the condenser (or capacitor) microphone, both of which are described below. Because microphones are concerned with electrical power and sound waves, they are appropriately called electro-acoustic transducers. Acoustics is the science of how the size, shape and contents of an enclosure affect the quality or characteristics of sound from a source within the enclosure. Before continuing, you should ensure that you have understood the content in the earlier subjects on logarithms, power ratios, the decibel, the mechanics of sound and acoustics. The microphone, like the loudspeaker, must perform in two different worlds; in the physical world of sound waves in air and in the electrical world of the audio systems, which amplifies processes, and distributes audio programs. The selection and placement of microphones can have a major influence on the sound of an acoustic recording. It is a common view in the recording industry that the music played by a skilled musician with a quality instrument properly miked can be sent directly to the recorder with little or no modification. This simple approach can often sound better than an instrument that has been reshaped by a multitude of signal processing gear. The content of this subject 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.
Condenser Microphones
Condenser microphones operate on an electrostatic principle rather than the electromagnetic principle used in dynamic systems. The head, or capsule, of the condenser mic consists of two very thin plates one movable and one fixed. These two plates form a capacitor. Microphones using the variable capacitor principle are universally referred to in the industry as condenser microphones, the term condenser being a holdover from past engineering terminology. A capacitor is an electrical device that is capable of storing electrical charges (refer to Basic Electronics). The amount of charge that a capacitor can store is determined by its value of capacitance and the applied voltage across its plates according to the formula ; Q=CV Where Q is the charge in Coulombs C is the capacitance in Farads V is the voltage in Volts The capacitance of the capsule is determined by the composition of the surface area of the plates (which are fixed values), the dielectric or substance between the plates (which in this case is air and is fixed), and the distance between the plates (which is a variable which varies with sound pressure impinging in the moving plate/diaphragm). Therefore, the plates of a condenser mic capsule form a sound pressure sensitive capacitor, one whose capacitance varies with the magnitude and frequency of the sound pressure hitting it. In the design used by most manufacturers, the plates are connected to opposite sides of a DC power supply, which provides a polarizing voltage for the capacitor. Electrons are drawn from the plate connected to the positive (+) side of the power supply and forced through a high value resistor onto the plate connected to the negative (-) side of the supply. This process continues until the charge on the capsule (that is the number of electrons on the positive and negative plates) is equal to the capacitance of the capsule multiplied by the polarizing voltage. When this equilibrium is reached, no further appreciable current flows through the resistor. This is the capsules condition when the air pressure around it is in its steady state (meaning there is no variation in the air pressure).
By introducing sound waves around the microphone, the variations in air pressure cause the diaphragm (front plate) to move back and forth in response. As the diaphragm (moving plate) vibrates as a result of variation in air pressure, the distance between it and the other plate of the capacitor varies and so does the devices capacitance. When the distance between the plates decreases, the capacitance increases; when the
Page 4 of 32 School of Audio Engineering Chennai Ver 1.0
distance increases, the capacitance decreases. The diagram below shows a illustration of this principle.
To put things in a nutshell, the capacitance of the microphone capsule is inversely proportional to the separation of the plates. According to the equation below, Q,C, and V are interrelated. So as the diaphragm moves in response to sound pressure variation, the capacitance C, changes in sympathy with it. Since charge (Q) is constant, Q=CV can be written as :-
Which shows that the voltage across the plates varies inversely with the capacitance, which in turn varies with the same rate as the sound pressure level hitting the front plate (diaphragm)? By keeping the charge across the plates constant, the voltage across the resistor can be made to vary as a function of the change in capacitance. The resulting AC signal component appears at the output terminals analogous to the sound pressure impinging upon the front plate/diaphragm.
Phantom Power In order to function, the capacitor/diaphragm must be biased with a polarizing DC voltage across its plates. This is supplied either by internally fitted batteries or through the use of phantom powering from a console or power supply unit. And, unlike dynamic moving-coil microphones, the circuit output voltage and impedance are both high. Consequently, a special pre-amplifier stage must be incorporated within the condenser microphone to lower the impedance and step up the voltage output to a useable levelIf internal batteries are to be avoided then the only solution is to supply the power via the microphone signal cable. Phantom Power Types P48, T, A-B, 12V, 24V, 48V P48, T, A-B, 12V, 24V, 48V are the types of phantom power usually encountered. There is often a lot of confusion over the differences and indeed similarities of the various types. DIN 45 596 defines that phantom powering may be achieved using either of three standard operating voltages; 12V,24V or 48V. The way that these voltages are presented to the microphone may vary depending upon the type of powering used. T & A-B These are the same thing, A-B being the old term for what is now referred to as T powering.Due to the potential difference between the A and B conductors, a current will flow through a dynamic (moving coil) microphone if it is connected to this sort of phantom power. This is not good and will probably cause distortion to the sound and perhaps longer term damage to the microphone. Only T-powered microphones should be used with T-type phantom power circuits. The T-powered microphones behave like capacitors and hence block D.C. current flow.
You may wonder why T power is used. The advantage is that the shield of the microphone cable need not be connected at both ends, thus this allows the common practice of disconnecting one end of the shield to a microphone in order to prevent hum (earth loops). This is not the case in P48 type powering below where all three conductors are required throughout. It would seem that 'T' power is mainly now only used by location recordist and specialists for specific applications, usually over long microphone cable lengths. P48 In this type of powering the same voltage is put onto both A and B conductors. The result of this is that the microphone sees no potential difference, hence no current flows through the coil. This type of powering is now the most common due to it being safe if a dynamic (moving coil) microphone is accidently or purposefully plugged into a powered microphone channel. The actual voltage used varies between the standard voltage of 48V, 24V and 12V 12V, 24V, 48V These are the common voltages used in all types of phantom powering. The voltage does not usually definitively indicate the way that the power is delivered to the microphone, although 48V is almost certainly P48 powering when it is encountered. Creating a clean 48V DC supply is difficult and expensive when on location with only 9V battery is available, partly because of this it should be noted that most modern microphones will work with voltages anywhere in the range 9-54V.
Page 6 of 32 School of Audio Engineering Chennai Ver 1.0
The permanent magnet and iron yoke The permanent magnet assembly (as shown in the diagram below) serves to produce a strong magnetic field inside the gap in which the voice coil will sit. When the diaphragm vibrates back and forth between its steady state conditions in response to an impinging sound wave, the coil moves back and forth between the magnetic field in which it is suspended. The motion of the coil in the fixed magnetic field in the gap causes the wires of the voice coil to cut across the flux in the gap. This in turn produces an induced current across the coil that is analogous to the pressure acting on the diaphragm, as shown in Figure 1. The output would be an electrical equivalent of the acoustic sound wave hitting the diaphragm. Here, simple alternating sound waves are shown impinging on the diaphragm (A), producing the output voltage shown at B. This is where the name moving-coil comes from. The signal from the dynamic element can be used directly, without the need for additional circuitry. This design is extremely rugged, has good sensitivity when close to the sound source and can handle the loudest possible sound pressure levels without distortion hence a preferred choice in the areas of live sound reinforcement and notwithstanding equal usage in studio environments as well. 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.
Ribbon Microphones
In contrast to the moving-coil design, some dynamic microphones use a metallic membrane, which functions as both the diaphragm and as the moving conductor. In a typical design, a thin, corrugated electrically conductive foil is suspended in a strong magnetic field supplied by permanent magnetic structures as can be seen in the diagram to your right. Wires attached to each end of the foil bring in the induced voltage to a transformer mounted in the microphone housing. The transformer serves two purposes:
i. ii.
to step up the voltage to raise the microphones output impedance (0.2) to approximate that of a moving coil microphone (150 ~ 600 range)
Because the diaphragm is usually formed in the shape of a thin ribbon of metallic foil (corrugated lengthwise to provide additional physical strength and durability to the thin foil), this type of microphone is usually referred to as a ribbon microphone. The earliest ribbon microphones were rather bulky devices with delicate ribbon structure. Such microphones do not stand up well against the demands of contemporary recording practices ; a misplaced puff of air is all it takes to put one out of commission. As a result, the older ribbon microphones could not compete with the more robust moving coil design and generally fell out of favour as a studio tool. However, the extremely low mass of a ribbon diaphragm does offer a potentially excellent transient response.
Inside a Ribbon Microphone Capsule
The Boundary Layer Microphones The surface at which sound is reflected is commonly referred to as a boundary and the boundary layer is the all paths from the source combine at the diaphragm with negligible phase cancellation within the audio frequency range of interest (in this case 20Hz-20kHz). The example below shows a lady singing into a microphone placed at differing heights.
(i)
the output of a microphone placed in the vicinity of a reflective surface is subject to response fluctuations (comb filtering) due to interaction between direct and reflected sound waves reaching its diaphragm. The severity of the comb filtering effect increases as the microphone is brought closer to the boundary surface. However, when the microphone is extremely close to the surface, the direct and reflected-path signals reinforce each other over most of the audio frequency spectrum, resulting in a 6dB level increase.
(ii)
(iii)
The advantages of placing a microphone close to a reflective surface have been well known and the incorporation of this principle brought about the development of the PZM microphone by Crown.
Pressure Operation
About the simplest microphone that can be deigned requires little more than a diaphragm stretched over a sealed cavity, as illustrated in the diagram to your right. A capillary tube (a small opening) drilled through one of the cavity walls functions as a pressure equalization vent; that is, it allows air to slowly leak into or out of- the cavity which this remains at the same static atmospheric pressure as the air outside the cavity. However the vent is insensitive to quick acoustic pressure changes, which are by comparison too rapid to be equalized via the capillary opening. You can compare this structure to the Eustachian tube which permits the atmospheric pressure in the inner ear and out ear to be balanced. When a varying acoustic pressure acts on the exposed surface of the diaphragm, the diaphragm vibrates accordingly and an output voltage is produced. This signal would be strictly a function of the varying acoustic pressure around the microphones diaphragm, for there is no way for the microphone to determine the angular direction from which this pressure wave arrived. Given the manner in which the microphone responds to sound, it is often referred to as pressure operated. The directional pattern of a pressure operated microphone is referred to as omni directional, in which omni means all. Hence in simple terms, a microphone which responds to sound pressure arriving from all directions.
Omni directional Mics The omni directional microphone has equal response at all angles. Its coverage or pickup angle is a full 360 degrees. This type of microphone can be used if more room ambience is desired. For example, when using an omni, the balance of direct and ambient sound depends on the distance of the microphone from the instrument, and can be adjusted to the desired effect. In the pressure operated type microphone, only one side of the diaphragm is allowed to interface with the outer atmospheric pressure whereas the faces the empty cavity. Now consider what happens when the diaphragm is exposed to the surrounding sound field on both of its sides. As illustrated in the diagram on your rights, there is an additional path to travel before the sound arriving on one side of the diagram reached the other side of the diaphragm. By contrast there is zero path length difference for a signal arriving at either side of the microphone. Both these conditions are illustrated in the diagrams below.
If we were to compare the pressure gradient system with the pressure operated microphone, the following observations can be made :iii. the pressure operated microphone compares sound pressure variations in the environment around the microphone with the fixed internal reference, as such iv. a pressure operated system is inherently capable of equal sensitivity to sounds arriving from all directions and across the entire frequency range.
The same is not true however of a pressure gradient microphone. The first problem is one of sensitivity to low frequencies. Sound especially low frequency sound, can find a way around a pressure gradient capsule by way of diffraction (ability of low frequencies to bend around a barrier remember Fundamentals of Sound). After the pressure wave arrives at the front of the diaphragm (causing the diaphragm to move), it passes around the microphone and impinges on the rear, creating a pressure gradient. High frequencies on the other hand cannot diffract around the capsule and will only hit the front of the diaphragm. Consequently the pressure gradient for low frequencies will be relatively higher than high frequencies, neutralizing somewhat the pressure applied by lower frequencies on the diaphragm due to phase cancellation. If we were to plot the sensitivity of the microphone across the entire audible frequency range, we would obtain a 6dB/octave rise in the frequency response with very little output at all at low frequencies. As shown in the graph below.
The resultant output rises at a rate of 6dB per octave until the transition frequency is reached. This is the point at which the wavelength of the signal is half the path length. At progressively higher frequencies, the output falls off rapidly, reaching zero output when the wavelength is equal to the path length. The two graphs on your right illustrate the rising frequency response. Clearly, this would not be a popular microphone as it cannot reproduce the frequency range required in audio engineering application. To compensate for this effect, the microphone housing and the electronics of the pressure gradient mic is usually designed with a complementary 6dB per octave roll-off characteristic, thus providing a flat response at frequencies below the transition frequency. As shown in the diagram following the first one. In each case, the upper curve (b) illustrate pressure gradient operation. The other two curves in each part of the figure (c & d) will be discussed later when we come to the subject of unidirectional microphones.
Bidirectional Microphones
A maximum pressure gradient will be created by a signal arriving from directly in front of, or directly behind, the diaphragm. In either case, the front-to-rear (or vice versa) path length difference is at its maximum. But a signal arriving at right angles to the diaphragm reaches both sides via the same or equal path length, Hence there is neither pressure difference nor phase shift from one side of the diaphragm to the other. With a pressure gradient of zero, there is no output at all for signals arriving from 90 angles to the side of the microphone. 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.
are out of phase by 180, the sum between them creates a cancellation of the sound waves arriving directly from the rear.
The unidirectional microphone is most sensitive to sound arriving from one particular direction and is less sensitive at other directions. The most common type is a cardioid (heart-shaped) response. This has full sensitivity at 0 degrees (on-axis) and is least sensitive at 180 degrees (off-axis). Unidirectional microphones are used to isolate the desired on-axis sound from unwanted off-axis sound. In addition, the cardioid mic picks up only about one-third as much ambient sound as an omni. Application Example 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.
Extensions of Cardioid patterns Unidirectional microphones are available with several variations of the cardioid pattern. Two of these are the supercardioid and hypercardioid. Both patterns offer narrower front pickup angles than the cardioid (115 degrees for the supercardioid and 105 degrees for the hypercardioid) 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 chart below shows the various polar patterns
Highly directional microphone pick up patterns can be achieved in two ways i. The use of a parabolic reflector The use of a parabolic reflector with the microphone at the focus can create very narrow lobes. But of course this will only be for frequencies above that with the wavelength equal to the diameter of the parabolic dish. Below this frequency the system will be more or less omni-directional. In practice, the highly directional pickup pattern will only start after about 2kHz.
The use of interference tubes (Gun or Rifle microphones) A long slotted tube is placed after the diaphragm. Sounds that arrive off axis of the tube enter the slots and will arrive at the diaphragm at different times depending on which part o the tube they enter. There would therefore be a phase cancellations. The most cancellation occurs for frequencies with wavelengths less than the acoustic length of the tube. Most tubes are about 50cm long. This microphone design is used mainly for news gathering for radio and TV work.
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: 1. 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. 2. 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. 3. 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 omni directional. 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
Page 21 of 32 School of Audio Engineering Chennai Ver 1.0
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.
Mono Microphone Techniques The basic idea of mono miking is the collection of various mono sources of sound for combination in a mix for a simulated stereo effect. Often mono and stereo sources are mixed together. There 4 Monomiking styles of mic placement directly related to the distance of a mic from its sound source. Distant Microphone Placement The positioning of one or more mics at 3 feet or more from the sound source such as the distance picks up a tonally balanced sound from the instrument or ensemble and also picks up the acoustic environment ie reflected sound. Using this style provides an open, live feeling to the sound. Distant miking is often used on large ensembles such as choirs or orchestras. Mic placement depends on size of sound source and the reverberent characteristics of the room. One must try to strike an overall balance between the ensemble and the overall acoustics. A problem with distant miking is that reflections from the floor which reach the mic out of phase with the direct sound will cause frequency cancellations. Moving the mic closer to the floor reduces the pathlength of reflected sound and raises the frequency of cancellation. A height of 1/8 to 1/16 inces will keep the lowest cancellation above 10KHz. Close Microphone Placement The mic is placed 1" to 3' from the source. Only direct, on-axis sound is captured. Creates a tight present sound quality which effectively excludes the acoustic environment. Very common technique in studio and live sound reinforcement applications where lots of unwanted sound (leakage) needs to be excluded. Multitrack recording often requires that individual instruments be as "clean" as possible when tracked to tape. Miking too close may colour the recorded tone quality of a source. Small variations in distance can drastically alter the way an instrument sounds through the mic. A common technique when close miking is to search for the instrument's "sweet spot" by making
Page 22 of 32 School of Audio Engineering Chennai Ver 1.0
small adjustments to mic placement near the surface of the instrument. The sweet spot is where the instrument sounds fullest and richest. Accent Miking A not too close miking technique used to highlight an instrument in an ensemble which is being picked by distant mikes. The accent mike will add more volume and presence to the highlighted instrument when mixed together with the main mic. Ambient Miking An ambient mike is placed at such a distance that the reverberent or room sound is more prominent than the direct signals. The ambeint mic is used to enhance the total recorded sound in a number of ways: restore natural reverb to a live recording used to pick up audience reaction in a live concert. in a studio, used to add the studio rooms acoustic back in to a close miked recording Stereo Miking Techniques The use of two identical microphones to obtain a stereo image in which sound sources can be identified by location, direction and distance. Stereo miking methods rely on principles similar to those utilized by the ear/brain to localise a sound source. These methods may be used in close or distant miking setups to record ensembles, orchestras or individual sound sources live or in the studio. AB or Spaced Pair The two mics (Omni or cardioid) are placed quite far from each other to preserve a L/R spread or soundstage. The AB method works on the arrival time differences between the two mics to obtain the stereo image. This is similar to the ear utilizing Interaural Arrival Time differences to perceive direction. In placing AB mics use the 3:1 Rule : the distance between the to mics should be at least 3 times the distance between the mics and the source. This help maintain phase integrity between the mics ie less chance of ou-of-phase cancellations occurring. The AB stereo method can give an exaggerated stereo spread and can suffer from a perceived hole in the center effect. The sound can be warm and ambient but off centre sources can seem diffuse ie not properly located.
XY, and Mid-side (MS) or Coincident Pair In both these techniques the mic capsules sit on top of each other. The stereo image is obtained by intensity differences produced by the sound source on each mic. This is similar to the Interaural Intensity Difference utilized by the ear. The images are usually sharp and accurate but the stereo spread can seem narrow. XY Pair are two cardioid mics (Top angled L ,bottom R) set at an angle of between 90 and 135 degrees. The angle increases the intensity differences and widens the stereo image. 2 omni mics can be used for more ambiances.
MS Technique
One first order cardioid microphone and one bi-directional microphone in the same point and angled 90 creating a stereo image through a so called MSmatrix. MS-Stereo uses a cardioid microphone capsule as center channel and a bi-directional microphone (figure-of-eight-microphone) at the same point, angled
Page 24 of 32 School of Audio Engineering Chennai Ver 1.0
at 90 as the so-called surround channel. The MS-signal can not be monitored directly on a left-right system. The MS-matrix uses the phase cues between the center and the surround microphone to produce a left-right signal suitable for a normal stereo system. Due to the presence of the center microphone, this technique is well suited for stereo recordings where a good mono-compatibility is needed, and is extremely popular in broadcasting.
SPLIT
Inverse
The M-S Matrix The output of the center cardiod (MID) is sent to a channel and panned to the CENTRE. The output of the figure 8 microphone is sent split into two identical signals. One of these splits will be phase reversed, while the other is not. Both these signals are routed to channels 2 and 3 respectively, panned LEFT and RIGHT and ganged. Channel 1 gives control over the MID content of the signal while Channels 2 and 3 will be simultaneously increased or decreased to the desired SIDE amount. This system gives the user a remote way of controlling how much width he/she wants in a recording. L= M+(+S) R = M+ (-S) In the case of the stereo signal being collapsed into Mono, such as in radio broadcast, the integrity of the MID signal is maintained. This can be proved by the equation below : If L = (M+S) , and R = (M-S) then L+R = (M+S) + (M-S) = 2M + 0S
The Blumlein Pair uses 2 bidirectional mics set at right angles to each other. The Blumlein stereo set-up is a coincidence stereo technique, which uses two bi-directional microphones in the same point and angled at 90 to each other. This stereo technique will normally give the best results when used at shorter distances to the sound source, as bi-directional microphones are using the pressure gradient transducer technology and therefore is under influence of the proximity effect. At larger distances these microphones therefore will loose the low frequencies. The Blumlein stereo is purely producing intensity related stereo information. It has a higher channel separation than the XY stereo, but has the disadvantage, that sound sources located behind the stereo pair also will be picked up and even be reproduced with inverted phase.
Baffled Stereo Spaced microphone stereo techniques using an acoustic absorbent baffle. Baffled stereo is a generic term for a lot of different stereo techniques using an acoustic baffle to enhance the channel separation of the stereo signals. When placed between the two microphones in a spaced stereo set-up like A-B stereo, ORTF stereo, DIN stereo or NOS stereo, the shadow effect from the baffle will have a positive influence on the attenuation of off-axis sound sources and thereby enhancing the channel separation. Baffles should be made from an acoustic absorbent and non-reflective material to prevent any reflections on the surface of the baffle to cause coloring of the audio. One of the more well known baffled stereo principles is the so called Jecklin Disc (or OSS for Optimal Stereo Signal) developed by the Swiss sound engineer Jrg Jecklin. This techniques uses two omni directional microphones spaced 36 cm and a special acoustic treated disc with a diameter of 35 cm placed between the microphones.
Near Coincident or OSS (Optimal Stereo Sound) Several versions of this method including the ORTF (Office for Radio and TV of France), NOS and Faulkner. Microphone pair is separated by a distance similar to that between the 2 ears. Therefore, both level and time differences are used to obtain the stero image. Uses the best features of AB and XY to produce a soundstage with sharply focused images and an accurate stero spread.
The Binaural Mic or Dummy Head is a development of the near coincident approach which mounts 2 mics in ear cavities of a model head. This technique produces realistic 3D stereo effects which can be heard through headphones. Two omni directional microphones placed in the ears of a dummy head creating the stereo image. The Binaural recording technique uses two omni directional microphones placed in the ears of a dummy head and torso. This two-channel system emulates the human perception of sound, and will provide the recording with important aural information about the distance and the direction of the sound-sources. When replayed on headphones, the listener will experience a spherical sound image, where all the sound-sources are reproduced with correct spherical direction. Binaural recordings are often used as ambience sound or in virtual reality applications. (also known as Kunstkopf in German) NOS stereo Two first order cardioid microphones spaced 30 cm and angled 90 creating the stereo image. The NOS Stereo Technique uses two cardioid microphones spaced 30 cm apart and angled at 90 to create a stereo image, which means a combination of difference-in-level stereo and difference-in-time stereo. If used at larger distances to the sound source the NOS stereo technique will loose the low frequencies due to the use of pressure gradient microphones and the influence of the proximity on these type of microphones. The NOS stereo technique is more useful at shorter distances, for example on piano, small ensembles or used for creating stereo on a instrument section in a classical orchestra. ORTF stereo Two first order cardioid microphones spaced 17 cm and angled 110 creating the stereo image. The ORTF stereo technique uses two first order cardioid microphones with a spacing of 17 cm between the microphone diaphragms, and with an 110 angle between the capsules. This technique is well suited for reproducing stereo cues that are similar to those that are used by the
Page 27 of 32 School of Audio Engineering Chennai Ver 1.0
human ear to perceive directional information in the horizontal plane. The spacing of the microphones emulates the distance between the human ears, and the angle between the two directional microphones emulates the shadow effect of the human head. The ORTF stereo technique provides the recording with a wider stereo image than XY stereo and still preserves a reasonable amount of mono-information. Care must be taken when using this technique at larger distances, as the directional microphones exhibit proximity effect and will result in low frequency loss. You could add low frequency with an equalizer to desired taste.
1. Frequency response The variation in output level or sensitivity of a microphone for a given input over its useable range, from lowest to highest frequency.
Virtually all microphone manufacturers will list the frequency response of their microphones as a range, for example 20Hz- 20,000Hz. This is usually illustrated with a graph that indicates relative amplitude at each frequency. The graph has the frequency in Hz on the x-axis and relative response in decibels on the y-axis. A microphone whose response is equal at all frequencies is said to have a flat frequency response. These microphones typically have a wide frequency range. 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. Although dynamic microphones and condenser microphones may have similar published frequency response specifications their sound qualities can be quite different.
2. 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. A graph paper drawn in a polar coordinate format is used for plotting the polar response patterns that will be described shortly in this subject. The concentric circles indicate dB of attenuation. In some polar charts, the dB attenuation is sometimes expressed as sensitivity with a maximum value of 1.0 and a minimum of 0.25.
3. Ambient sound sensitivity Since unidirectional microphones are less sensitive to off-axis sound than omni directional 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. 4. Distance factor Since directional microphones have more rejection of off-axis sound than omni directional types, they may be used at greater distances from a sound source and still achieve the same balance between the direct sound gand background or ambient sound. An omni directional 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. The diagram above shows the comparison of various directional pickup patterns relative to and omni directional mploar response.
5. Off-axis coloration A microphones 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. However, off-axis colouration can be used as a passive equalization system as turning the microphone slightly off axis reduces the amount of high frequency content in the signal that it is capturing. For example, placing the microphone directly in front of a guitar amp cabinet may sometimes result in an overly bright guitar sound. By using off-axis colouration to reduce the amount of high frequencies, we can mellow down the guitar tone by playing around with the distance from the source and directionality offsets. 6. Low frequency response characteristics At low frequencies, rumble (high level vibration in the 3Hz-50Hz region) may be transmitted in a studio or hall, or along the surface of a large unsupported floor space. You can eliminate this adverse effect in any of the following three ways;1. Use a shock mount to isolate the mic from the vibrating surface and floor stand 2. Choose a microphone that displays a restricted low frequency response.
3. Restrict the response of the wide range microphone through the use of a low frequency roll-off filter. Another low-frequency phenomenon inherent to most directional microphones is known as proximity effect. 7. 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 i. ii. iii. iv. 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).
Proximity effect also causes the exaggeration of plosive consonants such as p,t,b,d, by translating the sudden gust of wind emitted from the mouth with a popping effect. One way to eliminate this kind of breath pops is by replacing it with a omni directional microphone if needed in a close miking situation. On a more positive note, this increase in bass response has long been appreciated by vocalists for giving a full, larger that life quality to a thin voice. In many cases, the directional mic has become an important component of the artists sound. 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. 8. 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. A transient is a short duration, high level peak, such as a hand-clap or snare drum hit. Transients occur at the initial attack of a sound source. How a microphone reacts to a transient will directly effect its frequency response and how much SPL (sound pressure level) it can take. 9. Microphone Levels and Impedance Impedance is how much a device resists the flow of an AC signal, such as audio. Impedance is similar to resistance which is how much a device resists the flow of a DC signal. Both impedance and resistance are measured in ohms. When referring to microphones, low impedance is less than 600 medium impedance is 600 to 10,000 , high impedance is greater than 10,000 .
Microphone outputs are of necessity very weak signals, generally around -60dBm. (The specification is the power produced by a sound pressure of 10 Bar) The output impedance will depend on whether the mic has a transformer balanced output . If it does not, the microphone will be labeled "high impedance" or "Hi Z" and must be connected to an appropriate input. The cable used must be kept short, less than 10 feet or so,
trouble, and must be carefully designed and constructed of premium parts. Noise also includes unwanted pickup of mechanical vibration (also commonly referred to as micro phony) through the body of the microphone. Very sensitive designs require elastic shock mountings, and mics intended to be held in the hand need to have such mountings built inside the shell. The most common source of noise associated with microphones is the wire connecting the mic to the console or tape deck. A mic preamp is very similar to a radio receiver, so the cable must be prevented from becoming an antenna. The basic technique is to surround the wires that carry the current to and from the mic with a flexible metallic shield, which deflects most radio energy.