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Becoming a Studio Engineer
If becoming a Studio Engineer is your problem, kindly go to this page to learn more on Audio Engineering.
What is Audio Engineering
Audio engineering is a part of audio science dealing with the recording and reproduction of sound through mechanical and electronic means.
The field draws on many disciplines, including electrical engineering, acoustics, psychoacoustics and music . Unlike acoustic engineering, audio engineering generally does not deal with noise control or acoustical design. However, an audio engineer is often closer to the creative and technical aspects of audio rather than formal engineering. An audio engineer must be proficient with different types of recording media, such as analog tape, digital multitrack recorders and workstations, and computer knowledge. With the advent of the digital age, it is becoming more and more important for the audio engineer to be versed in the understanding of software and hardware integration from synchronization to analog to digital transfers.RECORDING APPLICATION GUIDE
Hi-hat
Omnis can also be used for this application, but they need to be placed a bit closer.
The drumkit can produce very high peak sound levels. Levels in excess of 120 dB at a distance ofone meter and at a few cm from a drum or cymbal head 140 dB or more is not unusual. It's obvious that the microphones must be able to handle these levels without clipping, which is not always the case in many recording situations.
It's difficult to understand why we see large diaphragm microphones used so often for this application. It's much more logical to choose a microphone that can handle these high levels comfortably, i.e. with enough headroom. Unless you do not wish to capture the natural sound of the instrument.
When a microphone is driven to its limit, the sound quality is affected and one can clearly hear, when the mic "hits the ceiling". At this point, this "smear" sound is what most engineers and producers think a trumpet or trombone sounds like. Most of this is driven by the player claiming that their sound needs to be "warmer". However you may try to mic the trumpet from the close proximity back side of the bell. This will eliminate the strong directivity and "warm" up the sound without"smear". Also in an overdub; if the recording room is large enough try to mic the trumpet or trombone from 3 meters away, this also smoothes out the strong directivity and creates "fatness".
Tom-toms
A three microphone technique (could be a Decca Tree configuration), often used in Jazz recordings, gives a very natural sound and also puts the skills of the drummer up front. The stereo pair should be panned left/right and the third mic in the middle. Avoid placing the overheads too close, because it will make the sound wander from the left to the right speaker. An XY, ORTF, DIN, or NOS pair of cardioids is sometimes used too, but it gives generally a less open sound than with spaced Omnis. The spot mic does exactly what it sounds like it does, intensify a spot or one drum. After placing one spot mic, if you still feel the need for more intensification, you should move to the multiple miking mentioned earlier. At this point you have qualified the drummers lack of ability to balance or tune drums. Multi miking is perfect for this situation. The engineer should now be in charge of the drum sound, just don't let the drummer know.
. The miniature mics have a slight high frequency boost in the 12 kHz range. Advanced users can remove the protection grid which produces the high frequency boost and a linear and smooth sound is produced, but you must be extremely careful about not touching the exposed diaphragm. Because of the perfect omnidirectivity of the Miniatures the pointing direction is not critical.
The soprano saxophone (usually a straight model, the curved variety is far less common) could be miked at 10-15 cm distance, pointing at the position of the right hand. If fingering noise is too prominent, the distance could be increased.
The big challenge with soprano saxes is that the instrument carries 2 very different sound elements that are both indispensable. At a distance these elements blend together nicely, but when close miking you have to be aware to cover them both or preferably mic them both. You cannot achieve the character from the opposite element by EQing on one of them! Higher frequencies (above 8kHz) are only directed along the axis of the bell, whereas mids and below are prominent at the tone holes. The 2 mics should not be equally loud. Start with the one at the tone holes and mix in the bell mic. Pan them just a tiny bit apart from each other.
The most common method of recording a human voice for any purpose is the direct on axis close proximity placement technique. Producers/engineers commonly call this "in your face". It has a number of advantages; intimate sound, good articulation of consonants, up front or leading sound and proximity warmth. It can also have a number of disadvantages; the need to de-ess, plosive popping, the lack of depth and lack of natural room tone.
A less common technique is the loose placement in front of the mouth. Most microphones cannot be used this way due to the unevenness of the off axis response. However a microphone with smooth off axis response can excel at this placement technique if the recording room is of a certain acoustic merit. The first advantage, if you can learn to use it in your recording, is a natural room tone. The balance between voice and room is set by the distance to the mouth and the placement of the performer in the room. Each room has its good spots and we hope you can find your own in your room. Listen for a short (0.9 ms or less) reverberant quality. A good starting point for this technique is around 12 inches from the mouth. By working with your assistant move the mic in and out while the performer is rehearsing and listen to the voice-to-room tonal balance. Be careful not to get too roomy because it is hard to reduce later. By getting a little room tone in with the voice, a more natural and comparatively rare sound will emerge. This technique gives natural depth of narration and applies to scenes where the vocal artist is not "in your face". You should notice that the need to de-ess and filter at low frequencies is reduced. The proximity warmth is replaced by natural room timbre, which is again more rare in today's recording. The recorded natural depth has the power to draw the listener into the recording rather than blow them back in their seats. This adds another trick to your engineering collection.
Sometimes the direct on axis techniques of close and loose do not produce the desired effects. This is when creative recording teams start to discover new ground. First, it requires a microphone with an extremely clear off axis response. By having the assistant move the microphone from the nasal area up, a different blend of "direct to room tone" can be discovered. If the performer is very breathy and / or pops a lot of plosives or has a very strong "S" this "high on the head" technique should be tried. Also if the performer is very diaphragmatic in his performing technique and produces too much body from proximity effect this should be tried. The human head is a nasal resonator; try pointing the direct axis of the microphone at the nasal cavity, just above the eyebrow. While the performer is rehearsing change the angle from direct on axis to 45° down and even 45° up. You may even point it over the head, you won't know what can work if you don't try.
Untie yourself from the belief that there is only one way to record a human voice, and that is with the microphone pointed straight at the mouth. If the microphone is of a certain quality level with respect to off axis response a whole new vista can be discovered and used in your daily technique. Sometimes the mic can wind up over the head pointed down and away from the speaking voice.
Still another technique that can be learned is the diaphragmatic placement. This style of placement cannot be used easily if there is a standard music stand involved. The reason is; the pointing the microphone at the chest cavity will be interfered with by the music stand itself and cause reflections from the music stand to smear the voice. High tech anti resonant or mesh music stands eliminate this problem. If the script is one or two pages or memorized this should be tried. What it yields is natural diaphragmatic warmth. Start with the mic about 12 inch from the mouth, low and at the centre of the chest cavity. Try angling it up and down 45° while the talking head is rehearsing. Listen for the balance of articulation in the consonants to the lower mid frequency roll that the human chest cavity produces. You usually can find a very full bodied sound that is again different from the "in your face" sound most engineers discover and produce. One good reason to use this technique is if the speaker has an extremely strong "S", really pops or is very nasal.
Alternatively, the broad selection of DPA Stereo Kits can all be used for choirs with great success. All kits are carefully matched and are being delivered in sturdy cases with a range of accessories. Depending on style and acoustics of the recording room, choose one of the following kits:
3521 or 3511 for ORTF
Excellent as a spot pair for supporting a choir section of an orchestra. The stereo imaging and spread is superb and the small recording angle offers a good separation. Tip: The CXO4000 microphone holder included in the 3521 kit can be hung down from the ceiling in the microphone cables.
3503 or 350 for AB stereo
In solo recordings of a choir that should blend in with the timbre of the hall, our omnidirectional microphones offer you the most natural reproduction.
3532-S, 3532-SP or 3532-T for low noise, high sensitivity AB stereo
As with the 3503 and 3506, the 3532 kits use AB technique offered by omni microphones to capture true room atmosphere. These exclusive kits use two large diaphragm microphones Type 4041. The larger the diaphragm, the lower the self-noise and the higher the sensitivity.
The ultimate microphone choice for an acoustic guitar is one or two 4041s - probably the application the 4041 is best suited for. It will give you the most beautiful, clean and crispy timbre with everything you wish for in details and transparency. When using two 4041s, a nice choice is, to place one outside the bridge and one towards the end of the fretboard and pan them after desired taste. An XY configuration could also be a good choice to open up the timbre from a specific position. Use either 4011s, 4015s or 4041s according to the desired recording angle and stereo spread.
Nice to know:
When the sweet-spot of the instrument is located, it can bring a beautiful depth to the recording if the microphone is pointed from the sweet-spot towards the fingerboard and the fingers
1: Adjust the height of the microphone stand to the mid of the guitar strings.
2: Microphone positions close to the front body of the guitar will probably give the best result.
3: The DPA first order cardioid microphones are neutral with regard to proximity effect at 30 cm distance on-axis. Closer microphone positions will add more low frequencies. Use the cardioid microphone off-axis (30°-45°) to adjust the clarity spot.
4: Listen carefully to the acoustic instrument in the microphone position and compare it with the sound impression in more natural listening positions
You can also try blending in the IMK4061 with the pick-up signal from the guitar. The true, open acoustic sound colour from the DPA condenser offers a realistic guitar sound you simply can not achieve by any pick-up system.
You can also have great success using the 4011 in the same position, taking advantage of the proximity effect. It all depends on the sound you wish to produce and the instrument/drummer.
As with brass instruments, a drumkit can produce very high peak sound levels. It is not unusual to see levels in excess of 120 dB at a distance of one meter and at a few cm from a drum or cymbal head, 140 dB or more. It's obvious that the microphones must be able to handle these levels without clipping, which is not always the case in many recording situations Direct miking of the clarinet is very similar to that of the soprano saxophone, bassoon and oboe, e.g. aiming at the fingering holes, 1/3 of the length up from the bell at a distance of 15-20 cm. Use an Omni such as 4006 or a Compact Omni when possible. If leakage is unwanted or you want less of the room to blend in, a Cardioid 4011 or Wide Cardioid 4015 could be used instead. Its compact counterparts (4026/27/28) take up less space, which can be convenient.
As with the soprano sax and oboe a good solution for very close miking is the 2 mic technique; one at the bell and one at the tone holes. Please refer to the saxophone section for further description. The wide cardioid microphones can give the advantage over the traditional first order pattern, that all tone holes are covered by the wider pick up pattern.
Too often, in our opinion, engineers use Cardioids where Omnis would simply provide a better overall sound. If you are in a multitrack orchestral situation and the orchestra has two flutes, two oboes, two bassoons, and two clarinets; an off-axis Omni microphone technique will work very well. Place the microphone between the two instruments, at about head height, and pointing straight down at the floor, which should not be carpeted. This technique will eliminate "keyboarding" and the bizarre polar patterns of these instruments will find their way into the microphone. When mixed into the rest of the orchestra it will not demonstrate that "too close" sound that should not be in this type of recording. Woodwinds are never in the front of an orchestra. With this technique you can place the woodwind section into the mix with pan pots and faders to the position that they physically occupy.
Classical orchestra with main stereo pair only
NB: If you are not familiar with the A-B Stereo techniques it is recommended that you read the A-B Stereo section on the Microphone University/Stereo Microphone Techniques.
Placing the A-B Stereo pair:
reproduce the instruments, the tonal balance of the orchestra, the directivity of instrument sections The position of the main stereo pair will also be the perspective of the future listener; therefore the sound engineer's goal is to create the illusion of natural perspective in placing the main stereo pair. In doing so, there is not necessarily any correlation between the actual placement of the microphones and the best seats in the hall, and consequently, placing the stereo pair correctly takes much effort. The distance to the orchestra and the height of the microphones above floor level should be adjusted for equal coverage of the different orchestra sections and for the amount of ambience required in the recording. Usually the optimum position is above or right behind the conductor's podium at a height of between three and four meters, so that no musicians obscure the instrument behind them. A good rule of thumb is: if you can see the sound source you can also hear it.
When using 4006s in large orchestral setups, the distance from the sound sources to the main stereo pair normally requires the use of the DD0297 Diffuse-field Grids. These grids give the microphones a linear diffuse-field response and help compensate for the air absorption losses of higher frequencies due to the distance.
NB: In rare setups the orchestra is arranged around the conductor in a semi circle. In these situations it is important that the main stereo pair isn't placed too close to the centre of the semi circle as the instruments around the edges will be recorded too far off axis, creating a heavy coloration (comb-filter) of sound due to the direct time delay occurring between the microphones.
Distance between the microphones:
Channel separation in the A-B Stereo technique is determined by the distance between the microphones. Compromises have to be made so that the orchestra sounds natural with a proper stereo width. Normally the spacing is adjusted from between 40 and 60 cm. Some producers favour greater spacings of between 1 to 2.5 meters and sometimes more, but in these cases a hole will start to appear in the middle of the stereo image. This can only be compensated by using a third microphone placed between the two others. DPA's UA0836 and UA0837 stereo booms allow the use of two microphones separated by distances of between 15-60 cm.
Experience will show that the optimum microphone spacing is dependent on both the size of the orchestra and the reverberation time of the concert hall. When increasing the number of instruments in the orchestra, the microphones need to be spaced further apart to give the listener the optimum stereo image. The microphone spacing should also be increased to improve channel separation in more reverberant rooms.
Microphone angle and acoustical attachments:
More ambience can be added to the recording without losing the fine position of the main stereo pair by fitting the UA0777 Nose Cone onto the microphones. This makes it possible to capture the full spectrum of the reflected sound from the walls in the concert hall, as the Nose Cones turns the microphones into perfect omnidirectionals across the whole frequency range.
The unique Acoustic Pressure Equalizers complete the range of possibilities. When fitted onto the standard grid of the 4006, the APEs function as both spatial and spectral equalizers. If it is necessary to add a slight colour to the recording, the APEs provide a soft on-axis boost defined by the size of the APE, without adding any noise to the recording.
The APEs also provide spatial properties, for example by focusing the 4006s on the orchestra in a more reverberant hall removing some of the unwanted ambience without compromising the position of the main pair. Even though the APEs are making the microphones more directional for higher frequencies, they do not affect their low-frequency qualities. In this frequency range the microphones still have the advantages of being omnidirectional. Therefore, they do not suffer from the proximity effect that makes normal directional microphones lose low end on larger distances.
NB: A-B Stereo is generally quite sensitive to any kind of phase difference between the two channels. Therefore it is quite difficult to add electronic EQ to an A-B Stereo recording without compromising the stereo image. However, the use of Acoustic Pressure Equalizers introduces exactly the same mechanical shape to both microphones, which again means no phase difference between the channels.
Advantages: The use of A-B Stereo techniques without support microphones can create an extremely convincing depth in the stereo image and capture a realistic room impression. The sound sources, ie musical instruments and room reflections, are picked up with the correct time alignment relative to the placement of the main stereo pair, which explains why this method is often regarded as the purist's choice.
1. Approx. 5-10 cm away from the instrument, aiming half way the mouth piece and the left hand. Breathing can be a problem in this position so an omni such as 4006 or Compact omni 4051 or 4052 may be an advantage.
2. Due to its polar character the flute can also be miked behind and slightly above the head of the player, pointing at the finger holes. In fact in an overdub situation a myriad of places around the head yield very good balance.
In this case a cardioid such as 4011 or Compact cardioid 4022 is recommended.
In another overdub situation you can try micing the flute from the front with a vertical elevation at head level and about a metre away, this really reduces the mechanical sounds.
For live use the headband 4066 is a great choice. This gives a fixed position, even when the musician moves around on stage.
There is a brilliance control that is available. It is the 20kHz on axis peak the UA0777 gives us; however I recommend that the brilliance peak be pointed straight through the kit and not at any part of it. Microphone number two should be placed in the vortex of the floor tom, ping, sizzler and left of the rack tom. Again balance control is easy, as your eyes and ears lead you to the spot.
You may have noticed there is no over-head. This is correct; the balancing of the two mics gives you all the cymbals with stunning realism. The other thing is the kick drum comes in with equal excitement because it is captured in true stereo. These two microphones can be panned in many ways to close, open, or shift the drum kit image to any part of the visualised stage. In a live jazz recording, I recommend some type of brick wall clipping. The Aphex Dominator, T.C: M-5000 MD-2, UREI 1170 all do the job nicely. They each have their own sound quality.
These two 4003/6 UA0777 mics can be supplemented with spot 4011-TL/12 on any part of the kit. Distant overheads, 4011-TL/12, will add a surreal ambiance to the entire kit while specialising their spread.
If the kick drum is to be miked; try a 4004/7 on the kicker side of the drum. With the microphone placed halfway up the drum, and on the edge of the diaphragm, near the bottom of the snare. This single placement technique, combined with the already placed 4003/6 UA0777 stereo pair, will yield a nice emphasis to the strike of the kick and the rattle of the snare bottom. It pans in to the left invisibly and brings up the definition.
Harp
In many cases, pedal harps and smaller (or Celtic) harps have been recorded with an unreflected point-and-shoot microphone position, which results in the subjective urge to apply compression, equalization, and plenty of reverb to get to an agreeable sound. While this may be applicable to various music styles, the harp can successfully be recorded without having to resort to electronic countermeasures. Let's simply take some time to experiment with microphone positions and the relation between instrument and recording room.
This article focuses on how to achieve a natural sound when recording the harp, like it is perceived in the front row when visiting a concert. Three basic applications for harp pickup will be described, namely solo recording, spot miking in an ensemble or orchestra, and stage sound.
It is generally challenging to find a room that fits both the instrument and the chosen music. Generally, a studio environment without a big enough room may not lead to a satisfactory tonal balance. The harp – just like the grand piano – needs a room in which it can unfold its sound.
Once a location has been chosen, the next step is to find a sweet spot for the harp in that room; the best approach is to try out extreme positions and to close in on the more amiable places in relation to the room. It is often found that the harp fits well between a corner and the centre of the room, not too close to either of them, while the microphones are placed closer to the centre of the room, and not too low to avoid floor reflections. A high ceiling with broken lines that provide some diffraction, such as in a church, is desirable.
When miking the harp, we follow the basic rule that the placement should be at least as far from the instrument as the instrument is big, so for a full sized pedal harp with its 190 cm height, a distance of about 200 cm is recommended as minimum distance. This way, the various components can mix into the complete harp sound while they travel in the air between instrument and microphones.
Move the microphones closer to, or further away from the instrument to find the position that represents the best mix of detailed instrument sound and room response.
A spot microphone closer to the harp can be added to the main stereo pair to provide more direct sound and detail. Be careful about two potential issues here, namely too much unwanted noise from the pedals or the player's fingers hitting and damping the strings, and secondly picking up only one component of the harp's sound while omitting another. See part 2) for further details on spot microphones.
Needless to say, our stereo array can also accommodate more instruments, such as a harp & flute duo. A baffle disc between the microphone pair will be of considerable help in achieving a natural sounding stereo image.
This placement will bring out the higher frequency aspects of the harp without sounding boomy. As a side effect, it will often transport the harp slightly to the foreground of the orchestral sound image. If this effect is not desired, a different position can be tried: slightly behind the harp, pointing downwards from the right – essentially opposite the player's head, just a little higher.
Using another spot microphone for the higher strings is helpful, but could potentially distract the player. Good results have been reported by using compact cardioid microphones such as the DPA 4026 et al. One from right behind the harp, facing the sound board from about 30 cm distance, the other looking at the sound board horizontally, from about the same distance. The two are then mixed with the top microphone twice as loud as the lower one. As a side note, the two microphones need not be panned to the left and right channel, as the harp has very little horizontal stereo image.
A common placement is to put one microphone – often with a cardioid pickup pattern – right in front of the harp, directly looking onto the sound board. Unfortunately, the resulting sound represents only a fraction of the complete instrument and often sounds boomy, with a peak in the bass or the low mids. A directional microphone will enhance the boomy low end if placed too closely, therefore great care should be taken to avoid this effect.
Alternatively, or additionally, a small microphone can be placed on top of the harp's pillar, similar to the spot microphone position described above. However, there will be more bleed from other sound sources than with the previous method. Theoretically, the microphone could be placed near the sound board, but firstly, the resulting sound would emphasise the neighbouring strings too much, and secondly, the microphone would inadvertently act as a boundary layer microphone, picking up too much unwanted sound – however pristine – from the stage environment.
Generally, the harp is very sensitive to microphone positioning, and we are trying to evade the disturbing aspects of the harp sound, such as pedal action noise or disharmonic ringing of the bass strings. Instead, we want to bring across the more pleasant ones, such as a bright attack, a well-pitched dosage of harmonic resonances, and an overall balance of the 47 strings which span almost seven octaves. It is essential to use microphones that can handle this task without distortion or colouring, and to take time for finding the sweet spots in a given recording situation.
There are some general statements such as "close miking for jazz, distant miking for classical music," which should always be taken with a grain of salt; a particular classical music piece may call for a straight response without sounding too spacious, and a piece written in a modern style may need exactly that spaciousness to sound right. The engineer should always feel some artistic freedom to make the overall sound work together with a style, a situation, or an album's theme that is reflected in different microphone positioning and balance.
The room acoustics
The room acoustics are such an important factor when recording a concert grand piano, that it is important to assess whether the room will do justice to the instrument or not. A concert grand piano is build for playing in concert halls where the reverberation times normally are acoustically tuned between 0,9 seconds to 1,3 seconds at 500 Hz and the room volume is minimum 10.000 cubic meters. Concert halls specially designed for playing Wagner can even have reverberation times around 2 seconds. With the lid open, the concert grand piano is capable of giving a full musical experience to the audience throughout the hall. This must be taken into account if the location is a recording studio. Finding a good location with a well tuned grand piano is the first thing to do - and the hardest.
The music
The choice of recording method also depends on the repertoire to be played. Classical piano music deserves a natural blend of ambience, but different composers all have their own characteristics which leave more or less space for the ambience to influence the music. Many a record producer and sound engineer has added his own interpretation of the grand masters' notes while immortalizing their compositions through a pair of microphones.
Rhythmic music and jazz traditionally are played in different types of locations. Therefore the listener's expectations of the timbre and the ambiance are somewhat different. In many cases, it was the composer's intention that his work be replayed on a hi-fi system in the home environment. The room acoustic of the location in which the jazz piano has been recorded is therefore judged by different and possibly less critical criteria, but, to give the grand piano its unique timbre, the recording still needs to convey a sense of location. This will furthermore give the listener a sensation that will bring him to the edge of his chair, so to speak.
Classical piano music and A-B stereo
When recording classical piano music, the finest results come out of two omnidirectional microphones in an A-B stereo setup in front of the grand piano. The A-B stereo pair is placed on the side of the piano to give the listener an illusion of being a part of the audience. By adjusting the distance to the piano and the exact placement next to the piano, the amount of ambience and the timbre of the instrument can be tuned. Placements around the mid of the piano 1 to 2 meters away are often preferred. The microphones are normally spaced between 40 to 60 cm and the stereo image is adjusted, so the pianist is to the left, of course. The height of the stereo boom is quite low, 1.2 m to 1.5 m above the boundary on which the piano is placed. By pointing the microphones to the open lid of the piano, the sound reflecting on the inner side of the lid will be brightened up and a beautiful depth will be added to the recording.
In rhythmic music and jazz, all the musical instruments are more or less used as rhythmic instruments, maybe even a sort of percussion. This also goes for the grand piano. It is therefore important to capture the attack of the player. The beautiful sound of the mechanics and the hands against the keys are often preferred to a correct stereo perspective. The "produced" sound is more accepted even for acoustic instruments. The pace and the rhythm does not leave much space for reverberation, which only will make the music sound muddy and unclear. The goal is therefore a tight and fast instrument which of course also sounds incredibly good. To achieve this goal, the microphones need to be placed close to the piano, maybe even inside the open lid over the strings or the hammers. Only here is it possible to capture the attack and the more rough side of the versatile queen of instruments. To distinguish the room reflections and the reverberation, the use of directional microphones is recommendable. Two extremely successful stereo set-ups using directional microphones inside a grand piano are mentioned following:
1: ORTF stereo set-up approximately 30 cm over the strings at the mid frame. The microphones are pointed 45° downwards and at the pianist.
2: Two parallel cardioid microphones spaced 60 cm and placed over the mid hammers pointing 45° downwards and at the pianist.
Please note that the sound pressure level inside a concert grand piano can exceed 130 dB SPL peak 20 cm over the strings. So be careful when choosing microphone types.
With e.g. the SMK4061 Stereo Microphone Kit a number of useful piano set-ups can be achieved:
This is a low cost stereo kit, equally suited for project studios and for hidden mic setup on stage, using two high quality 4061 omnidirectional miniature microphones and a variety of mounting accessories for placement in and on the piano, with either open or closed lid.
Try placing 4061 miniature microphones in foam windscreens with the DMM0011-B magnet holder in and around the sound holes of the piano. You can use 2 or 3 4061’s and create a balanced multidimensional sound with good front of house and monitor volume along with recordability. Place one mic in or near the high hole, another mic over the last octave of low strings (maybe the second to last octave). This method will give you high gain before feedback because of the close distance to the frame. It will also provide a thick full-bodied midrange sound that often fits well into rhythmic genres.
Boundary Layer Mounts (BLM6000) are also included in the SMK4061 kit. Mount two BLM6000s inside the open piano lid to get a natural sound for recording. The pressure zone technique will “draw in” and accumulate the piano’s timbre nicely and at the same time be very discrete. Placing them directly on the sounding board under the strings gives a sharp pop sound for on-stage use with a penetrating “honky-tonk” sound colour.
With the DMM0007 Universal Surface Mount you can fix the mics directly on the inside of the piano lid. This is also an almost invisible mounting technique. The microphone element is able to hang from its own integrated cable with the double-sided tape pads and the height/acoustic balance can hereby be adjusted. A well balanced, open-sounded position is 12 - 16 in (30-40 cm) over and in front of the hammers with approx. 24 in (60 cm) spacing.
About balanced and unbalanced lines
Signal balancing
Signal balancing can either be achieved by electronic balancing or by using a transformer. Both ends are hot, but one with reversed phase. At the same time the impedance from each wire to ground is exactly the same. If this is NOT the case, the line is not balanced. And with reference to microphones: Phantom power cannot be supplied.
The impedance balancing does not necessarily involve signal balancing. The explanation for this is that the impedance between pin 2 (in the 3-pin XLR) and ground is equivalent to the impedance between pin 3 and ground. But the signal is only present on pin 2. Pin 3 is the reference like in the unbalanced coupling. Hence the impedance is in balance, but not the signal.
The advantage of using an unbalanced signal in an impedance-balanced coupling is that the size (amplitude) of the signal will remain constant, independent of the input connection being balanced or unbalanced, with or without an input transformer.
The manufacturer can choose the amount of distortion he wants to specify the SPL at, as long as the THD appears in the printed literature. Normally, microphone manufacturers all over the world specify the equivalent sound pressure level at 0.5% THD, but as distortion normally increases with a certain linearity according to the input level, you will always be able to estimate the SPL at any amount of THD, because the THD is double with a 6dB increase of the input level.
The real trouble is not that the manufacturer sets the THD himself, or that he does not inform you about the clipping level of the microphone. The confusion is how the microphone manufacturer measures noise floor, equivalent SPL and clipping level. Does the user get the right information in the data sheet?
The illustration below shows the measured data and calculated dynamic range and headroom of the omnidirectional microphones Types 4003/4006 and Types 4004/4007.
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Furthermore,.Leakage is only a problem if it sounds bad. If the leakage from one sound source in another microphone sounds natural, it can purely be beneficial in the way that it adds natural room tone to the character of the sound source.Some microphones will not only sound good on axis but also off-axis, hereby offer the engineer the possibility to achieve more honest and natural pick ups.
Channel separation
If you choose an omnidirectional microphone, channel separation may be less precise than with a directional microphone, because the omni will pick up sound from all directions. Therefore, if channel separation is preferred, the ratio between direct and indirect sound can become more unfavourable with an omni.
The omni, however, can be moved closer to the source, without the penalty of the proximity effect that occurs with a directional microphone (pressure gradient transducers). The leakage that still occurs - in an acoustic setting it cannot entirely be avoided - is at least of neutral tonal quality and may add some beautiful "air" around the instrument.
As a general rule it can be said that if we place a cardioid at a distance of 17 cm to the source, then an omni placed at 10 cm gives the same ratio of direct and indirect sound as the cardioid.
What about high level sound sources and a condenser microphone then? And how do I cope with the previous advice about moving an omni even closer to the sound source? Well, a DPA microphone can handle that in most cases. One of our specialities is to handle extremely high sound pressure levels (SPL). We use a pre-polarized back plate that is charged with approximately 230 V, which allows us to move the diaphragm and the back plate further away from each other, without losing sensitivity. Hereby the diaphragm is able to have a bigger displacement without touching the back plate, which would make the signal clip
Using omnidirectional microphones in live sound reinforcement applications makes special demands on both the quality of the equipment and the experience of the sound engineer.
In general, the gain-to-feedback ratio will be reduced somewhat, which in itself can have a wide-ranging effect or almost none at all. Additionally, the characteristic of the feedback itself changes. Directional microphones tend to feedback at high frequencies while most omnis commonly feedback in the lower mid-range or bass. The feedback also builds up in a different way. Where directional microphones feedback suddenly and unexpectedly, omnis build up the feedback slowly, often starting as a low hum.Using omnis live on stage offers a special advantage, as it is possible to adjust the gain to a constant gain-to-feedback ratio across the stage. This makes them more effective as it. becomes possible for the artist to move all over the stage without suddenly stepping into a feedback zone.
A directional microphone (the cardioid is mostly used) has - as its name implies - a directional response, with a coverage angle of approx. 130°. Sounds from the rear are at its maximum attenuated by some 30 dB but this attenuation is dependent of frequency. In other words, the cardioid might have a nice flat frequency response on-axis, but off-axis this may not be the case. In fact, some directional microphones have a notably poor off-axis response. This means that sounds entering the microphone from the sides and the rear are more or less strongly coloured - the industry names this "the curtain effect".
An omnidirectional microphone will in principle pick up sound equally from all directions. The microphone will though become more and more directional the higher the frequency. The smaller the capsule, the more true omni the microphone is.
The omnidirectional microphone (a pressure type transducer) is in its working principle a more simple capsule construction than a directional microphone (pressure gradient transducer). Simplicity can result in a cleaner and more dynamic sound with a flatter frequency response. Due to the nature of a single diaphragm omnidirectional microphone (the sound only excites the diaphragm from the front) the construction can also be more rugged and therefore offers even better reliability and thermal stability.
Omnidirectional condenser microphones have in general a more extended low frequency response and lower distortion than directional microphones in a distance of over 30 cm. In listening tests, this is often described as a "fuller or warmer response in the bass". Again, equalization is needed on the directional microphone to compensate for the bass loss.
Wind and pop-noises and handling noise may also become a problem when using directional microphones. Directional microphones are far more sensitive to wind and pop noises than omnidirectional microphones, due to the use of more compliant diaphragms on directional microphones, which are more easily excited. The diaphragms made of stainless steel or nickel, these materials can be tightened very hard and are less compliant.
Distortion levels should also be considered when choosing between directional and omnidirectional microphones. Directional microphones tend to distort more than omnidirectional microphones, which is especially important when working with high SPL's in close miking situations. The difference in distortion between directional and omnidirectional microphones is evident, when comparing the THD specifications, If using a directional microphone, be sure to use one with low distortion and high headroom before clipping.
Conclusion
From the above it can be concluded that in close-miking situations, the choice of a DPA omni should be seriously considered. We strongly recommend to always make a habit of trying an omni first! It will often give a more natural sound, it can handle extremely high sound pressure levels, it does not suffer from proximity effect and is not that sensitive to wind, pop or handling noise!
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Speech
Common to all auditoria, churches, rooms for education, conference rooms, courtrooms, or theaters, is that one or more persons address an audience. From time to time the problem turns out that the speaker is not trained; he has not learned to speak up in a clear and well-articulated way. Moreover you cannot be sure that the speaker has learned to use a microphone in a proper way. The immediate solution is to gain the signal in order to obtain adequate amplification. This may cause excessive sensitivity to acoustical feedback.
A good solution to a problem like this is to use a microphone that is worn by the speaker himself. A microphone mounted on a headband used in connection with a wireless system provides a number of advantages. First of all the microphone is always close to the sound source. As a consequence, the microphone does not need to be gained as much. This is especially an advantage if several microphones are used simultaneously. Furthermore, it is a great advantage that the speaker has total freedom of movement. The speaker is not chained to the pulpit, the teacher can move freely in front of screens, or black-/whiteboards.
As the microphone can be sensitive to breath if placed incorrectly, it is important that the microphone is always placed according to the instructions given in the manual.
A headband microphone consists of a neck brace with two ear hangers and a microphone boom. Both the brace and boom can be adjusted for optimum fit and comfort.
Choose an omni
- when you need a microphone that is not sensitive to positioning- when the microphone is used by untrained personnel
- when the background noise is not a problem
- when feedback is not a problem
Choose a cardioid
- when background noise is a problem (especially low frequency contents)
- when feedback is a problem
How to read microphone specifications
When you read microphone specifications, it is extremely important that you understand how to interpret them. In most cases the specifications can be measured or calculated in many different ways. This article is designed to help evaluate specifications in a meaningful way.
While microphone specifications provide an indication of a microphone's electro-acoustic performance, they will not give you the total appreciation of how it will sound. Specifications can detail objective information but cannot convey the subjective sonic experience. For example, a frequency response curve can show you how faithfully the microphone will reproduce the incoming pure sinusoidal frequencies, but not how detailed, well dissolved or transparent the result will be.
The decibel (dB) scale
The basis for most microphone specifications is the decibel scale. The dB scale is logarithmic and is used because of its equivalence to the way the human ear perceives changes in sound pressure. Furthermore, the changes in dB are smoother and more understandable than the very large numbers that might occur in pressure scales (Pascal, Newton or Bar). The dB scale states a given pressure in proportion to a reference pressure., mostly 20 micro
Frequency Response
The frequency response curve illustrates the microphone's ability to transform acoustic energy into electric signals, and whether it will do so faithfully or will introduce colouration. Take care not to mistake frequency response for frequency range. The microphone's frequency range, will only give you a rough indication of which frequency area the microphone will be able to reproduce sound within a given tolerance. The frequency range is sometimes also referred to as "bandwidth".
On-axis: 20Hz - 20kHz ±2dB
Frequency Response:
Manufacturers of professional equipment will always provide more than one frequency response curve, as it is essential to see how the microphone will respond to sound coming from different directions and in different acoustic sound fields.
On-axis response
The on-axis response demonstrates the microphone's response to sound coming directly on-axis towards its diaphragm (0°). Be aware that the on-axis response may be measured from different distances, which may influence the response on directional microphones because of the proximity effect.
The off-axis responses will reveal the microphone's response to sound coming from different angles. This is particularly interesting when you want to discover how a directional (i.e. cardioid) microphone will eliminate sound coming from other angles than directly towards the diaphragm. Even though the off-axis responses are attenuated on directional microphones, it is of extreme importance that these curves also show a straight frequency response, as it will otherwise introduce an off-axis coloration (curtain effect).
A polar diagram is used to show how certain frequencies are reproduced when they enter the microphone from different angles. The polar diagram can provide an indication of how smooth (or uneven) the off-axis coloration will be.
The diffuse field response curve will illustrate how the microphone will respond in a highly reverberant sound field. This will be an acoustic environment where the sound has no specific direction but where all directions are equally probable. The reflections from walls, floor, ceiling etc. are as loud or louder than the direct sound and the sound pressure level is the same everywhere. This is especially interesting when considering omnidirectional microphones, because they are able to register the full frequency range in the lower frequencies. The diffuse field response will show a roll-off in the higher frequencies, partly due to the air's absorption of higher frequencies
Sensitivity expresses the microphone's ability to convert acoustic pressure to electric voltage. The sensitivity states what voltage a microphone will produce at a certain sound pressure level. A microphone with high sensitivity will give a high voltage output and will therefore not need as much amplification (gain) as a model with lower sensitivity. In applications with low sound pressure levels, a microphone with a high sensitivity is required in order to keep the amplification noise low.
According to the American tradition, which states the sensitivity in dB, relatively to 1 V/Pa, which will give a negative value. A serious microphone manufacturer will also state tolerances in sensitivity, according to production differences - such tolerances would normally be in the region of 2 dB.
SPL handling capability
In many recording situations it is essential to know the maximum Sound Pressure Level (SPL) the microphone can handle. Please note that in most music recording maximum peak SPL's easily supersede the RMS value by more than 20 dB. The RMS value indicates an average SPL and will not show the true SPL peaks.
It is important to know
1. The SPL where a certain Total Harmonic Distortion (THD) occurs.
2. The SPL where the signal from the microphone will clip, that is the waveforms will become squares. This is the term: Max. SPL and it refers to peak values in SPL.
Maximum sound pressure level:
168 dB SPL peak
Total harmonic distortion:
142 dB SPL peak (<0.5% THD)
148dB SPL peak (<1% THD)
It is important to bear in mind that the acoustic memory of the human being is only a few seconds, which leads to the so-called simultaneous A-B test
- or A-B-C test if more microphones are to be considered. The microphones need to be present simultaneously, picking up exactly the same sound source. You need to align the test and reference microphones bringing the diaphragms as close to the same point as possible. Note that the distance to the sound source needs to be at least four (4) times greater than the maximum distance between the centres of the microphone diaphragms. Some microphone housings and bodies are quite bulky. Use one microphone stand for each microphone in the set-up to find a position, which ensures a minimum influence on the acoustic field around the diaphragms from the more bulky microphone bodies. Do not hesitate to use a pop-filter if you intend to test the microphone with vocals, but use one pop-filter only.
Testing microphones with vocals
The most common tests of studio microphones are done with vocals, but do not hesitate to use more complex sound sources like guitar, piano, and wind or percussion instruments to spice up your evaluation. Most microphones at least have a decent on-axis response and you will only be able to evaluate the true quality of a microphone if you also test its off-axis qualities. Musical instruments are extremely qualified sound sources for testing both the on- and off-axis qualities of microphones simultaneously, but you can also get a good idea of the microphone's performance when using speech or singing using the following procedure. Make sure the headphone feed is from one microphone only, as it could otherwise influence the vocalist's performance.
Start here. This is more or less the normal distance to a studio microphone when used for vocals. Adjust the sensitivity on test and reference microphones to exactly the same level using voice or tone generation as the sound source, double-checking the levels with the peak meters in the console. Make sure that all equalizers are bypassed or in neutral position. Select the microphone you want to listen to by using the MUTE button in the console - not by using the faders.
In the reference position you will probably have some kind of preference of what an uncoloured voice should sound like. Here a directional microphone (i.e. a cardioid, hyper-cardioid or a figure-of-eight microphone) will normally not exhibit any or very little proximity effect. The weighting of the lower frequencies can therefore be expected to be neutral if you are testing a directional microphone. An omnidirectional microphone will not be influenced by the proximity effect, regardless of the distance, but you will use this distance as reference anyhow. The reference position will help you to discover any unwanted off-axis coloration when you move around the microphone later on. Return to the reference position as often as you like during your test to calibrate your ears.
It is extremely difficult to design studio microphones with no off-axis coloration characteristics, especially directional microphones. However, the off-axis qualities of a microphone are of the utmost importance if the aim is a clear and transparent recording. Off-axis sounds are allowed to be attenuated (if directional microphones). An increased attenuation of the higher frequencies can also be expected in cases of larger diaphragms, but an off-axis comb-effect (curtain effect) is definitely unwanted.
30 cm 45° off-axis up (Off-axis coloration test #2)
If the test microphone has a bulky design and is not rotationally symmetrical, this test will reveal any unsymmetrical coloration that might occur. "Up" means talking/singing into the microphone in an angle from the top of the protection grid provoking a sonic reflection from the base of the cartridge where the capsule is connected to the preamplifier housing - the bottom of the cartridge chamber. A microphone, which exhibits unsymmetrical off-axis coloration, has an extremely limited applicability and is not suitable for the more demanding recordings like ambience or suchlike.
If the test microphone is a directional microphone this close-up test will give you a picture of the microphone's sensitivity to pop noises even when using a pop-screen. In this position you can also expect an extreme enhancement of the lower frequencies due to the proximity effect of a directional microphone. In cases of some male voices or rock 'n roll bass drums this effect might be something you are looking for, but normally the proximity effect is an unwanted side effect - or at least something you try to use as discretely as possible. Omnidirectional microphones do not suffer from the proximity effect and you should not be able to hear any coloration of the lower frequencies when moving close to an omni. Furthermore, omnidirectional microphones are less sensitive to pop noises than their directional counterparts. Shouting into the microphone at close distance will reveal any possible limitation of the dynamic range of the microphone. Make sure that it is not your console or microphone amplifier that is the limiting factor in this test.
If the recording room allows it, it is now time for the ambience test, where you move as far away from the microphone as possible - preferably at least 3-4 m. Directional microphones will again reveal the unwanted proximity effect and will now sound thin with a severe bass roll-off. Omnidirectional microphones will be able to do the job better and keep an uncoloured response. The amount of sonic reflections from the walls in the recording room will now create a complex sound field at the diaphragm and the true directional quality of the microphone will reveal itself. Here it is important to cross-reference with the probe-like reference microphones.
30 cm 180° off-axis (Front-to-back attenuation and coloration test)
The reason for designing a directional microphone is, of course, to attenuate sounds from unwanted directions. To get a good front-to-back-attenuation on a cardioid microphone is quite difficult and to obtain a perfect polar pattern on an omnidirectional microphone is also quite an achievement. Talking/singing directly into the microphone from behind will help you to discover any possible unwanted back loops of the directional polar pattern or, if an omni, any unwanted coloration of the sound besides for the expected attenuation of higher frequencies.
Handling noise
Double-check the sensitivity adjustment on test and reference microphones to make sure that levels are identical. Tap and/or rub the microphones (including the reference microphones) on the preamplifier housing and/or on the microphone stand to get an idea of the microphone's sensitivity to handling noise. Generally you will find that a directional microphone is more sensitive to handling than an omni.
Microphone Stability
It is essential to the stability of a microphone that the materials in the cartridge work well together, i.e. expand and contract together while exposed to heat or cold. We have chosen a nickel foil for our diaphragm material in our pressure microphones. For the microphone housing we have chosen German silver, an alloy with a very high content of nickel. These two materials are so close to one another, that they respond extremely similarly towards temperature. This way we are able to manufacture microphone cartridges with a stable sensitivity regardless of the temperature. Furthermore nickel is extremely resistant to humidity When it comes to pressure gradient microphones, it gets more difficult, because this microphone principle needs a soft and very compliant diaphragm. The tight and hard metal foil diaphragms will not perform to their optimum when used in directional microphones. We have chosen a plastic material called PVDF with a vaporised layer of aluminum on. This material is one of the few plastic materials that is non-hydroscopic, i.e. it does not absorb water, and is truly resistant to the most aggressive kinds of humidity.
The distance between the diaphragm and the back plate in our pressure gradient microphones is about three times as large as in our pressure microphones. Therefore the pressure gradient microphone concept is less sensitive to temperature expansions and contractions.
The Pre-aging process
When materials are stressed, bent or worked on in any way, tensions will start to occur. Only time can release the tensions again. Microphone stability is of course dependent on the stability of the materials, so it is essential that the tensions in the housing and the diaphragm are released. Heating up the material will speed up the aging-process and therefore release the tension build-up inside it. Brüel & Kjær has named the process "pre-aging" and it has been used on measuring microphones for many years.
Tests are made during the whole process, where the sensitivity of the microphone capsules is measured while being heated up to 200° C. When the sensitivity is stable, the microphone capsules are put into a humidity room with 90% humidity and 40°C. The studio microphone capsules have now undergone more than 150 handlings and more than 300 for the 100 capsules for the 4040 Hybrid Microphones.
Electronic components and circuits
To get a low noise amplifier stable under all kinds of conditions takes hard work and a steady hand during the design process. Mainly it is the feedback loops in the low noise amplifier which are problematic, and, when not dimensioned accurately, the amplifier will start to oscillate. Also the components themselves in the feedback loops need to be of highest quality, if you want your amplifier to be stable under all thermal conditions. Running the amplifiers with low amplification will help this process, but then the output of the cartridge needs to have a decent level. We can obtain a high sensitivity on our capsules by having a high polarisation voltage on the back plate. This allows us to use unity gain amplifiers, running them as impedance converters and ending up with extremely stable microphones.
Microphones, High Wind and Rain
Sensitivity
In most microphone transformer designs the signal is transformed downwards. A transformer will reduce the sensitivity of the microphone with approxmately the same ratio as the signal is transformed down to, because the sensitivity specification concerns the microphone’s output signal in relation to the pressure on the diaphragm (mV/Pa).
Noise immunity on balanced audio lines is totally dependant on how well the in- and output impedances are balanced and the common mode rejection ratio of the audio input channel. The balanced audio transformer introduces a higher common mode rejection ratio than any other electric circuit and is perfect for balancing loads. Using in- and output transformers for balancing audio signal lines will provide the highest possible immunity against common mode induced noise on the audio lines. Transformerless microphone preamplifiers will from the mere tolerance of the electronic components introduce a slightly unbalanced load on the audio lines.
Cable drive capability
When a signal is transformed down, the signal voltage becomes lower according to the conversion ratio of the transformer, while the signal current becomes accordingly higher. The increased signal current will increase the microphone’s ability to drive long cables before noticeable signal deterioration occurs.
Low frequency handling
Transformers are audible and will introduce some low frequency distortion, as transformers are susceptible to saturation by the high energy in the low frequency signals. Not only will the frequency range grow for the transformerless variant, but more interesting is the tighter bass response in the very important region around 80 Hz.
Conclusion
Both transformerless as well as transformer output design have different advantages and disadvantages. This is illustrated in the following table. For this reason DPA offers both types of preamplifier design. However, the cleanest approach to authentic sound reproduction is choosing the transformerless type. Today, most microphone lines do not run more than 100 m.
Mic amps and AD converters are often placed pretty close to the mic which is why the primary advantage of the transformer microphone becomes weaker.
Background on two-channel stereo
Psychoacoustic research has quantified the time and level differences adequate for directional imaging to any position on the line between left and right loudspeaker in a standard loudspeaker setup (fig . 1) .
Additionally, a combination of time difference and level difference can act together. For instance the sound will be reproduced at 30° if the signal in one channel is delayed by 0.5 ms and the level is approximately 6 dB below the other channel (see dotted lines in fig. 2).
In fig. 2 the inter channel differences adequate for 10° and 20° reproduction respectively are also found.
First quantified, this information can be combined with the directional characteristics of the units in a two-microphone setup. Then it is possible to determine the optimum positioning of the microphones for a stereo recording
Microphone spacing
An important consideration when setting up for A-B stereo recordings is the distance between the two microphones. Since the acoustic character of the stereo recording is very much a question of taste, it is impossible to give fast rules for stereo microphone spacing, although it is a good idea to keep some important acoustic factors in mind.
Since the stereo width of a recording is frequency-dependent, the deeper the tonal qualities you wish to reproduce in stereo, the wider your microphone spacing should be. Using a recommended microphone spacing of a quarter of the wavelength of the deepest tone, and taking into account the human ear's reduced ability to localise frequencies below 150Hz, leads to an optimal microphone spacing of between 40 and 60 cm. Smaller microphone spacings are often used close to sound-sources to prevent the sound image of a particular musical instrument from becoming "too wide" and unnatural. Spacings down to 17 to 20 cm are detectable by the human ear, as this distance is equivalent to the distance between the two ears themselves.
It should also be noted, that an increase in microphone spacing will decrease the system's ability to reproduce the signals from sound-sources positioned directly between the microphones. This will also lead to a reduction in the quality of the stereo recording when it is played in mono.
Distance between microphones and sound-source
The ideal distance from the microphone pair to the sound-source not only depends on the type and size of the sound-source and on the surroundings in which the recording is to be made, but also on individual taste. The position from which the listener experiences the event - and hence the position from which the micro-phones record the event - should be chosen with care and feeling.
Critical musical recordings, such as a full orchestra in a concert hall, are good examples of the importance of correct stereo microphone positioning. Here the microphones would typically be placed above or behind the conductor. And although most instruments project their sound in an upwards direction, the microphones should be placed high enough so that the individual musicians do not shadow each other.
The mix of direct and diffuse sound in a recording is also of crucial importance, so much time can often be used establishing the optimum positioning of the microphones. It is here that the versatility of our A-B Stereo Kits comes into play. Using the different acoustical attachments for the microphones, the amount of ambience and the tonal colour of the recording can be adjusted without adding any noise. The choice of floor and ceiling mounting of the boom can give you added flexibility when positioning the microphones.
Omnidirectional microphones and A-B Stereo are often the preferred choice when the distance between microphone and the sound source is large. The reason is that omnidirectional microphones are able to capture the true low frequencies of the sound-source regardless of the distance, while directional microphones are influenced by the proximity effect. Directional microphones will therefore exhibit loss of low frequencies at larger distances. The DPA Wide Cardioid microphones are however designed with a richer bass response, making the bass loss at distances much less critical. Therefore these mics are a good alternative to omnis when a little directionality is preferred or needed. The sound colour is very similar to our omnis.
One of the more well known baffled stereo principles is the so called Jecklin Disc developed by the Swiss sound engineer Jürg Jecklin. This techniques uses two Type 4003 or 4006 omnidirectional microphones spaced 17.5 cm and a special acoustic treated disc with a diameter of Ø35 cm placed between the microphones.
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Binaural recordings are often used as ambience sound or in virtual reality applications.
Mono compatibility and dialogue
The MS signal can not be monitored directly on a left-right system. The MS matrix uses the phase cues between the Mid and the Side microphone to produce a left-right signal suitable for a normal stereo system. Due to the presence of the centre microphone, this technique is well suited for stereo recordings where a good mono-compatibility is needed, and is extremely popular in broadcasting.
When recording the two microphones seperately the stereo spread can be adjusted afterwards with the MS matrix by blending the M and S components in appropriate balance.
Right channel=M - S
The MS system will give directional information since the bi-directional microphone captures two directions and only puts out one signal with mutual opposite polarity. When sound approaches the MS arrray from the right, it enters the bi-directional Side mic in its phase inversed side. The matrix calculation says that Right output is M minus S. If you subtract a phase inversed signal from something it is like adding it. Mathematically speaking minus minus is plus. So, voila, sound approaching the MS array from the right side creates a signal in the right output of the matrix due to a simple subtraction.
For the left side, the sound meets the Side mic in its in-phase lobe and is simply added to the Mid signal.
NOS stereo
ORTF stereo
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 Equaliser to desired taste.
The stereo image is produced by the off-axis attenuation of the cardioid microphones. While A-B stereo is a difference-in-time-stereo, the XY stereo is a difference-in-level-stereo. But as the off-axis attenuation of a first-order cardioid microphone is only 6 dB in 90°, the channel separation is limited, and wide stereo images are not possible with this recording method. Therefore, XY stereo is often used where high mono-compatibility is needed - for example, in broadcasting situations where many listeners still receive the audio on mono equipment.
Opening angles of 120° to 135° or even up to 180° between the capsules are also seen, which will change the recording angle and stereo spread. Bidirectional (figure-of-eight) microphones and more directional cardioidtypes like hypercardioids can also be used for XY Stereo.