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Frequently Asked Questions

Here you will find a collection of frequently asked questions or frequent problems with microphones and headphones and the appropriate answers and solutions.
This area is still under construction, but new questions and answer will gradually be released.

If you should have questions which are not answered here, please contact our service staff.

 

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Headphone & Headset FAQ´sMicrophone FAQ´s
The beyerdynamic limiter concept

We all know the soft sound of a high-quality, electronic limiter as it is used in studios and for recording purposes. The sound of such an often VCA-based circuit corresponds to a remote-controlled volume increasing and decreasing and not a circuit which clips signals. 

Of course it is possible to integrate such an extensive electronic circuit directly into a headphone, but this makes the headphone useless for daily use, as the headphone always has to be equipped with a supply voltage for the operation of the limiter electronics. But this is not available with standard headphone outputs of players. 

In order to avoid this and to leave the headphone as passive listening tool, limiter circuits with passive components such as diodes are used. These circuits are known as diode limiter or brick wall limiter and can clip audio signals very fast and secure when the threshold is exceeded. In principle this is the same function as with an equalizer. 

At beyerdynamic we have invested a lot of work to achieve a similar sound with a pure passive circuit as with an extensive electronic limiter circuit. This includes a two-stage concept to reduce gently the audio signal by 6 dB when the limiter threshold is achieved (similar to a compressor with a quick attack and slow release time). Then there is a second hard limiter downstream, which limits the audio signal with a brick wall limiter when the input volume is further increased. 

The big advantage of this concept compared to others is the soft, compressor-similar range between the unlimited audio signal and the hard second limiter stage. This range still sounds very good and signals the user the achieved limiter threshold without producing immediately hard distortions. 

As mentioned before, the question arises which weighting filter should be used for the limiter circuit. An A-weighted limiter function is close to standards and measuring specifications, but does not quite reflect the properties of the human hearing at high volumes. At beyerdynamic we decided to use a limiter concept WITHOUT weighting filter. Why? 

What do we want to do with our limiter? We want to protect people against damaging their hearing. Due to the fact that the one and only weighting filter does not exist which reflects the properties of the hearing with all volume levels, we decided to assume the worst case, what means measuring the sound pressures unweightedly in dB (SPL). The procedure automatically meets all weighting filters which are provided for low sound pressures (A-, B-, C-weighting). Although passive limiters which operate with an A-weighting with normal, deep music material allow a higher sound pressure level before they respond, but the question arises if this really makes sense talking about limiters for high sound pressure levels. In our opinion a limiter should operate to protect the human hearing against damages and operate more conservatively than optimistically. This means the use of a limiter without a weighting filter. People who are looking for high sound pressure levels should prefer a headphone without limiter. 

More information: Spot on Technology - Passive Limiters in Headphones 

Difference between DT 880 PRO - DT 880 Edition - (PRO vs. Consumer)

The DT 880 PRO is designed for use in studio and on stage for professional use. It's headband has a more rugged construction (thicker material) and gives a more secure fit for the listener in comparison to the DT 880 Edition. This provides a better fit and causes also a more powerful compressed sound which is necessary for professional applications in studio, OB and FOH. The DT 880 Edition therefore is designed for easy listening. The headband provides a long wearing comfort for hours, causing the DT 880 Edition to have a more transparent and open sound in comparison to the DT 880 PRO.

The differences of both headphones are not huge in mechanical design, but only that small change causes a big difference in sound and application.

 

The following headphones DT 990 PRO - DT 990 Edition have a very similar difference.

 

PICTURES

 

Different impedances

The higher the impedance, the more power is needed to get a proper output volume from the headphone:

32 or 80 ohms = mobile use with laptop, MP3 Player, portable recorder etc..

250 ohms and higher = for permanent installations, headphone amplifiers etc.

Background:

Impedance ist the AC resistance of the headphones' voice coil, which is connected to the headphone amplifier. A impedance of 0 ohms would be a short-circuit of the headphone amplifier output; the headphone amplifier supplies an extreme current and after getting very hot, it either turns off automatically - or dies. The other side of the story is infinite resistance (broken cable); no current flows, but also no audio signals arrives - so, we have to be in between these two: 0 and infinite.

In general, headphones with low impedance are designed for use with mobile devices; mobile devices use low power from batteries and therefore also the output power is limited. A low impedance headphone can play (slightly) louder at a low power output. But why high impedance headphones?

The impedance is determined by the voice coil (dynamic headphones), which is a winded copper wire (coated to avoid a short-circuit). This copper wire is available in nearly every length, but not in every gauge (thickness) and a thicker wire has less resistance than a thin wire ("less fits through"). The magnetic field of the voice coil depends on the number of windings of the coil, causing a low impedance system to use a thicker (also heavier) wire and since the membrane foil can't be infinitely light-weight, the moving mass (voice coil and diaphragm) is relatively high. It's pretty clear that a higher mass can't move as easily (following an audio signal) as a lower mass. This low mass can easily be accomplished with thinner (lower weight) wire, but the thinner wire has a higher impedance. This means that the DT 770 PRO with 250 ohms sound more natural, but plays (depending on the used headphone amplifier) not as loud as the 80 ohms version.

The transducers of the 80 ohms versions are stronger and more powerful, a bit more low-mid accentuated and therefore this version is ideal for powerful reproducing of low-frequency material f.e. coming from a bass guitar. The 250 ohms version sounds more smooth and voluminous and can be used for mixing situations within the studio to analyse the whole mix.

What is "Impedance"?

Also known as nominal impedance. The impedance is the AC resistance of the coils of loudspeakers and headphones in ohms. Since impedance depends on the frequency, it is always specified at a frequency of one kilohertz. 

If you take a look at the offerings of dynamic headphones, you will find a very wide range of impedances. The spectrum ranges from 16 ohm headphones to 600 ohm headphones. Where does this wide range come from? And which headphones are suitable for what applications?

In order to get to the bottom of these questions, a power evaluation method must first be established. The basic task of the headphone is to convert the arriving electrical signal into sound pressure. The extent that this succeeds is described by the nominal sound pressure level of the headphones. This value (specified in the unit dB SPL) describes how high the generated sound pressure is when 1 mW of electrical power is supplied. If you take a look at the nominal sound pressure level of similar headphones with differing impedances, you will find that the value is roughly the same. That means that you need to consider the electrical power converted in the headphones in order to make a statement about the sound pressure attained.

For the following considerations, the impedance of the headphones is assumed to be real (described as resistance with the formula symbol R). This is not completely correct, but is sufficient for our purposes here.

You can calculate the electrical power converted in a resistor from the applied voltage:

P = V²/R

As an alternative, you can also use the current:

P = I² x R

Since we are talking about AC voltage and AC current, the RMS value must be used in both cases.

Low impedance (up to 100 ohms): connection to mobile devices (MP3 players, laptops, etc.). 

Medium impedance (between 100 and 300 ohms): connection to stationary installations (HiFi amplifiers etc.).

High impedance (over 300 ohms): connection to high-quality headphone amplifiers. Higher impedance = better sound = higher power requirements.

Difference between open and closed headphones

The differences between closed and open headphones are:

1) the strong ambient noise attenuation with closed headphones (and vice versa: the “outside world” cannot hear what is playing on the headphones) and 

2) the better spatial sound with open headphones. In principle, semi-open headphones are a mix of both and attempt to combine the respective advantages of each type. 

Open, semi-open or closed? What is the difference? 

As already explained in the in-ear headphones topic, the bass response is very good with in-ear headphones, since the space between the diaphragm and eardrum is “closed”, so to speak. In principle, these are closed systems. However, the conclusion that closed headphones have the best bass response is not completely correct, because the system works in a manner that is completely different from that of in-ear headphones. This is a topic that is not easy to explain and would make this article too long.  

The biggest differences between closed and open headphones are: 1) the strong ambient noise attenuation with closed headphones (and vice versa: the outside world cannot hear what is playing on the headphones) and 2) the better spatial sound with open headphones. In principle, semi-open headphones (such as the DT 880 PRO) are a mix of both and attempt to combine the respective advantages of each type. If we look at the issue from a mechanical standpoint, we recognize that open headphones have an advantage in comparison to those that are closed: the air volume that is closed off between the diaphragm and the headphone shell attenuates the vibration of the diaphragm. With open headphones, there is pressure compensation through the shell, which has a positive influence on the impulse fidelity, among other effects. The greater attenuation of the diaphragm also decreases the risk of uncontrolled vibrations.

The choice of headphones on the basis of how they sound depends on what we want to listen to, of course. For classical and jazz music, which have less of a bass component, but for which very high impulse fidelity is of particular importance, open headphones are the perfect choice. For pop and rock music, semi-open or closed headphones are the preferred choice. 

In the end, which operating principle is suitable depends on the application (where do we want to use the headphones?). If the headphones are to be used in a quiet environment (in a music studio for mixing or listening to music, for example) you can freely choose headphones according to type and personal taste. If the headphones are to be used by a musician for monitoring purposes during a recording, headphones should be selected that attenuate ambient noise very well and that in turn shield sound from escaping into the environment, so that the sound produced by the headphones is not picked up by the microphone.

Difference between DT 770 PRO and DT 770 M

Both DT 770 PRO and DT 770 M are basically one and the same headphone and also for similar applications. 

The difference between these two models though, is as follows:

1. The DT 770 PRO has so called "Bass-Relfex-Openings" at the side of each housing to make sure the diaphragm can "breath", causing a better reproduction of low- and high frequencies (since the diaphragm can move a lot easier). A small amount of isolation against ambient noise has to be sacrificed for the better sound. The DT 770 M doesn't have these openings, therefore sounds slightly "thinner" (slightly less low end), but offers up to 35 dB(A) of ambient noise isolation. That's why the DT 770 M is prefererd by drummers (like the German drummer Ralf Gustke) or percussionists (like the young percussionist Farouk Gomati).

2. The DT 770 PRO is equipped with silver velours ear pads for better wearing comfort and cooler ears. The DT 770 M is equipped with the so called "Soft-Skin" ear pads to provide maximum isolation of ambient noise.

Conclusion:

DT 770 PRO is for studio use where sound quality has a higher priority than the isolation of ambient noise.

DT 770 M is also for studio use, put for those to whom the isolation of ambient noise if most important.

The DT 770 M also includes the DT-Bag.

What does „Nominal Power Handling Capacity“ mean?

The nominal power handling capacity is the power that the headphone systems can be supplied with over a longer period of time without being damaged. While the power specified as the “maximum power handling capacity” may only be supplied for a short amount of time, the power specified in the “nominal power handling capacity” should be able to be handled on a long-term basis without any problems.

What is T.H.D. (Total Harmonic Distortion)?

We have already made brief mention of total harmonic distortion (THD) in another article. Without now making this too complicated, total harmonic distortion is the relationship between the original signal (fundamental) and the sum of the signals produced by the acoustic transducer itself or signals produced through housing parts (harmonics), signals that were not present in the original signal. Since the total harmonic distortion is always less than or equal to 1, it is usually specified as a percentage. The lower this value is, the fewer harmonics are generated by the acoustic transducer and/or housing parts, which is exactly what we want. This is because we have little to no control over the harmonics – if they exist – and they do not belong to the original signal.

What is the Frequency Response?

The frequency response specifies the lowest and highest frequency that the acoustic transducer can reproduce. These limits are always set at the point where there is a 3 dB drop (on the lower and upper frequency limit). Unfortunately, this is not standardized, so it is often the case that some manufacturers “cheat” a bit. Whether headphones really have a frequency response of 20 to 20,000 Hz, for example, remains an open question. The question of what the curve of the frequency response looks like also remains open. Is it linear, or does it go up and down? You would be able to see this on the frequency response curve, but it is rarely included, since it looks pretty “horrible” for most headphones. This matter will be explained in another article. Ideally, you should really rely on your own hearing and simply try out headphones you are considering in a side-by-side comparison. The best way to do this is to compare the headphones to headphones with which you are already familiar.

What does „Nominal SPL“ mean?

SPL is the pressure in Pascal (Pa) with which the air and the eardrum are being vibrated, also known as loudness.

SPL is also a degree of the loudness of a headphone at standardized conditions. Be warned though: high SPL does not guarantee that the Total Harmonic Distortion is low and the sound is good.

Difference between design (circumaural, supraaural, etc.)

Supraaural headphones

Once we know the differences between open, closed and semi-open headphones (the operating principles) we can return to supraaural headphones. Since supraaural headphones “only” rest on the ear concha (the outer, visible portion of the ear), it is not possible to completely close the space between the eardrum and diaphragm. With supraaural headphones, a differentiation is made only between variants with open or closed housings. As we have now learned, the use depends on the type of music and/or application area.

Ear pads are also involved here; they have a great influence on wearing comfort. In addition to different materials, there are also different forms: 1.) The “flat”, closed variant that is completely “closed” in the centre (familiar from the old Walkman headphones). This variant is becoming less and less common because such headphones are not particularly comfortable. 2.) The ring-shaped open variant (as with the beyerdynamic DT 231 PRO, for example), which is used on many supraaural headphones today. With the “flat”, closed variant, a great amount of heat builds up when worn for a long time, which in turn has a negative impact on wearing comfort. Moreover, they exert pressure on many parts of the ear concha (the outer, visible portion of the ear), which can also be unpleasant when they are worn for a long time. With the ring-shaped, open variant, this problem is nearly non-existent, so that the headphones are always comfortable even when worn for a long time. 

Circumaural headphones

The circumaural headphone is the largest design, used in studio headphones. The ear pads are large enough to be worn on the head around the ears. They “surround” the ear and do not touch it. This results in a clear differentiation between closed, open and semi-open operating principles because the space behind the diaphragm and between the eardrum and the diaphragm is either open or closed depending on the ear pad and housing.

The different ear pads (material, shape, thickness) also have an influence on the acoustics of the headphone. If the ear pad does not correctly connect to and seal on the head, for example, the space between eardrum and diaphragm is not closed, making the sound from the headphones less than ideal. However, the ear pads that do the best job of closing off this space are at the same time not always the most comfortable or the most hygienic. For example, imitation leather ear pads or the “softskin” ear pads that are now quite widely used are very easy to clean, but due to the fact that they do a very good job of closing off the space, they very quickly develop a great amount of heat, which in turn promotes sweating. The sweat is not absorbed by softskin ear pads and can be washed off with a normal wet rag. In contrast, velour and cloth ear pads do not close off the space very well and breathe a bit better as a result. Nevertheless, heat builds up when they are worn for a long time. The resulting sweat is absorbed by the ear pads. This has a negative effect on hygiene when the headphones are worn by many different people, of course. For this reason, velour and cloth ear pads should be washed on a regular basis. There are still more variants, such as gel ear pads, which consist of a flexible film that is filled with gel. Due to the fact that the gel distributes itself well over the entire surface when placed on the head, the ear pad fits the head perfectly, which results in ideal ambient noise attenuation. Real leather is also used for ear pads and has problems relating to heat build-up and hygiene that are similar to those of “softskin”.

In-ear headphones and intra-concha earphones

The smallest designs are in-ear headphones (also referred to as In-Ear-Monitors, abbreviated as IEM) and intra-concha earphones. The difference between the two designs is that in-ear headphones (such as DT 60 PRO from beyerdynamic, for example) are inserted into the ear canal like earplugs, while intra-concha earphones (such the MX series from Sennheiser) are only placed or wedged in front of the ear canal in the concha. This allows in-ear headphones to offer a much higher level of ambient noise attenuation, something that can be very important for a variety of reasons. 

We can start with the acoustics. If the in-ear headphone sits correctly in the ear canal so that it “seals” well, the space between the eardrum and the diaphragm is closed and very small. It functions like a sort of suspension system (or a “push-pull mechanism”); the diaphragm can easily move the eardrum with little amplitude and energy, something that results in very good bass response. As soon as the seal is lost in this system, it is immediately noticeable due to the loss of low frequencies (as is the case with intra-concha earphones). This is because the human ear is less sensitive to low frequencies (below approx. 150 Hz) than to high frequencies. So if we want to hear low frequencies better, a great amount of energy must be exerted in their amplification. When using loudspeakers, low frequencies can even be sensed by the body. This is not the case with headphones. Loudspeaker diaphragms are also larger and more robust (with thicker material), allowing them to move substantially more air than is the case with headphones. In order to optimally exploit the low level of energy developed by the headphone system, care must be taken so that the headphones or in-ear headphones close off the space as well as possible. 

The second reason why it is important that in-ear headphones seal well is that the volume does not need to be set as high. This protects our hearing. An in-ear headphone is therefore really “plugged” into the ear, allowing it to better seal out ambient noise and use the ear canal as a resonating body in order to produce better sound. The bass response in particular is substantially better than with intra-concha earphones. In order to do even more to improve ambient noise attenuation, sound and wearing comfort, many manufacturers offer the option of having a hearing aid audiologist make what are referred to as “ear moulds”, which are fitted exactly to your own ears and replace the included standard earplugs.

A few transducers...

Without exception, beyerdynamic manufactures dynamic transducer systems for headphones. These consist of a permanent magnet and a diaphragm capable of vibration, in the centre of which a coil is located, through which the current of the music signal flows, causing it to function as an electromagnet. Both magnets strongly attract and repel each other at varying levels, causing the diaphragm to vibrate and generate sound pressure waves (=tones). 

Electrostatic acoustic transducers

The transducer consists of two grid-like electrodes that are directly across from each other, between which the diaphragm (thickness: < 2 microns) is located. Depending on the audio signal, the electrodes are always charged and attract or repel the diaphragm, depending on their charge. Electrostatic acoustic transducers are extremely popular among audiophiles due to their extreme precision and low total harmonic distortion. Disadvantages include the high operating voltage required, the mechanical sensitivity and the relatively high purchase price, of course.

Orthodynamic acoustic transducers

Another transducer is one that many have almost forgotten: the orthodynamic acoustic transducer. The transducer consists of two grid-like ferrite magnets that are placed directly across from each other at a certain distance, with a curved diaphragm located between them. The diaphragm consists of two layers, between which the (flat) coil is attached. The diaphragm (coil) moves between both magnets (depending on the audio signal), setting the air in motion and generating sound. Orthodynamic transducers also have very precise sound and an extremely low total harmonic distortion. In the 80s, this type of transducer was often used, but it can hardly be found today. The biggest disadvantage was that due to the low forces of the magnets used, only a small amount of sound pressure could be achieved. 

Dynamic acoustic transducers

The third transducer is also the transducer that is most often used: the dynamic acoustic transducer, which is structured like a loudspeaker in principle: A ring coil (also referred to as a moving coil) is attached to the rear of the diaphragm. The coil moves in an air gap of a permanent ring magnet. This transducer offers high reproduction quality, is mechanically very robust, needs only a small amount of operating voltage and has a much lower purchase price in comparison with electrostatic transducers. For these reasons, dynamic acoustic transducers are the transducers most used today and are found in almost every set of studio headphones worldwide. 

(Electro)magnetic acoustic transducers

The fourth acoustic transducer is the magnetic acoustic transducer, also referred to as the electromagnetic acoustic transducer. These transducers are often used in high-quality/expensive in-ear headphones. The electromagnetic transducer is basically similar to the dynamic acoustic transducer, with the difference that the diaphragm is simultaneously the magnet and the coil is located in a fixed position under it (similar to the permanent ring magnet with dynamic transducers). The clear advantage is the sound pressure. These transducers produce more sound pressure than dynamic acoustic transducers, for example. The disadvantages are the higher purchase price, the somewhat worse total harmonic distortion and the resonance peak. The resonance peak is a certain frequency range (depending on the model) that is distinctly raised by a few dBs and that negatively influences the sound, since a certain range (e.g. the speech or bass range) is always boosted. This is also the reason why two-way and three-way systems are used in high-quality in-ear headphones. This makes it possible for the entire frequency response curve to be flatter. (With a three-way system, the resonance peaks of the three individual systems are adjusted to each other.)

What is diffuse-field equalisation?

What is diffuse-field equalisation?

Have you ever wondered why a frequency response curve is almost never included with headphones? I can let you in on the secret: they look terrible! Such an erratic frequency response graph would hardly encourage customers to make a purchase. What the customer wants in the end is something that is linear. Uncoloured. Solid.

But why do these frequency response curves look so horrible? And why do you not clearly hear these glaring leaps and drop-offs?

How we hear

From childhood on, humans are accustomed to perceiving acoustic events. We grow up with a variety of sound sources and get used to them. The baby rattle, the clatter of dishes from the kitchen, pedestrians on the street, music from loudspeakers, etc. – all of these sound sources have something in common: they are located relatively far from the ear.

Before the sound from these sources reaches our eardrum, it is coloured by the shape of our head and our ear. Depending on the angle, many frequencies are accentuated and others are attenuated. With time, we learn these frequency patterns and are able to do things such as recognise the direction in which the sound source is located. Therefore, we do not hear sound as it was produced at the source, but instead in coloured form.

Loudspeakers and headphones

When we listen to music over loudspeakers with a linear frequency response curve, we are actually hearing a spectrum that is influenced by the distinctive shape of our head. We perceive this as linear.

When listening with headphones, the headphones do not even try to generate any effects on the outer ear, since the sound source is so close to the ear. What comes out of the headphones arrives at the eardrum in relatively uncoloured form. In order for the headphones to still sound natural, the sound must be coloured so that it is as similar as possible to the colourations caused by the shape of the head and ear. In other words, the headphones must have the frequency response set so that it sounds like the sound is coming from a distant source.

Diffuse-field equalisation

In order to adjust headphones to our listening habits, we must first use technical means to measure the colourations caused by our head. For example, an artificial head with microphones in the ears is used. When this artificial head is exposed to sound, you can use the microphones to measure how the sound would be perceived by us instead of the artificial head.

So that the headphones do not have a sound that always seems to come from one direction, but instead can reproduce all sound directions equally, the artificial head must be exposed to sound from many directions and the result averaged. This does not perfectly reproduce any direction perfectly, but no direction is completely suppressed. 

At beyerdynamic, there is an echo chamber for this purpose. It is a small, five-sided room with acoustic sails on the ceiling that looks quite bare and empty. The fascinating thing about it is that, although it is the size of child’s room, it sounds like a cathedral! An octahedron loudspeaker that radiates sound in eight directions is in one corner. If you are far enough away from the loudspeaker, the strong echo causes you to no longer be in the direct field, but instead in the diffuse field of the loudspeaker, i.e. the area in which the sound reflected off the walls is louder than the sound that is coming directly from the loudspeaker.

If artificial head measurements are carried out in this chamber, many sound directions overlap due to the echo, allowing us to obtain the required averaging. This averaging (the measurement in the diffuse field) gives diffuse field equalisation its name.

In order to equalise the headphones, they are placed on the artificial head and the frequency response is adjusted so that the measured frequency behaviour corresponds to that of the diffuse field.

Discussion

Since the mechanical and electronic options for changing the frequency response of headphones are limited, the equalisation cannot be carried out perfectly. Different headphones are also adjusted to various tastes. It is by no means the case that all diffuse-field equalised headphones sound the same. In addition, the frequency patterns for directional hearing depend on the shape of the head and ears. For this reason, they are a little different for everyone. Hence, measuring with an artificial head is a pretty arbitrary choice.

Diffuse-field equalisation is therefore an important part of improving localisation with headphones and avoiding “in-head localisation”, but it is not guaranteed to work and is no replacement for extensive test listening.

120 ohms nominal source impedance for the DT 48 E

Where do those mysterious 120Ohm come from?

The basic headphone measurements are described in the german standard DIN-EN 60268-7 (based on the international standard IEC 268-7). According to this standard, measurements shall be performed using this source impedance of 120Ohm.

That’s why those 120Ohm are mentioned in the DT 48 E datasheet: to describe under which conditions the stated measurements where made.

Why does it make sense to use such a high output impedance?

The primary reason why this impedance is suggested is to avoid large differences in resulting SPL when using headphones of different impedances with the same headphone amplifier. 

Headphone impedance can vary from 16 to as high as 600 Ohms. Let’s assume a headphone amplifier with an output impedance of an ideal zero ohms that is adjusted to a constant output voltage of 1 Vrms (sine, 500 Hz). Connecting headphones of different impedances will yield the following resulting power at the headphone system:

600 Ohms – 1,7 mW

250 Ohms – 4 mW

80 Ohms – 12,5 mW

32 Ohms – 31 mW

16 Ohms – 62,5 mW

In general, equal power corresponds to equal SPL, independent of the chosen system impedance (at least this is the case with most of our headphones). That means, switching from a 600 Ohms headphone to a 16 Ohms headphone yields a change in SPL of 16 dB!

A series resistor of 120 Ohms reduces this difference considerably, the resulting power is:

600 Ohms – 1,2 mW

250 Ohms – 1,8 mW

80 Ohms – 2,0 mW

32 Ohms – 1,4 mW

16 Ohms – 0,9 mW

Do I have to use an amplifier with this high output impedance?

No, you don’t. 

In fact, most current low-power devices, like MP3-players, mobile phones etc. use headphone outputs with low output impedance. In conjunction with low impedance headphones, this allows delivering enough power to the headphone system even with low supply voltages. Connecting a headphone with high impedance to such an output results in low SPL, because the amplifier just cannot provide enough voltage to deliver power to the high impedance system.

Is there a difference in sound with respect to the amplifier’s output impedance?

To a certain degree, yes. But the effects are rather subtle.

There are two effects to be mentioned:

First, the impedance of the headphone system isn’t real. The complex part is rather small, but it exists. In conjunction with the series resistor, this results in a frequency dependent voltage divider. This will influence the frequency response, dependent on the output resistor. 

The other effect is damping. As stated above, the headphone system’s impedance has a complex part, and thus the membrane develops its own life when excited with voltage. The lower the output impedance of the amplifier, the closer the action of the membrane is to what it should be. So a low output impedance of the amplifier (compared to the system impedance) will deliver a better impulse response.

Conclusion

A series resistor at the amplifier’s output has it advantages (driving different headphone impedances, short-circuit protection). 

With proper circuit design, this technique yields a very versatile amplifier, even capable of driving very low impedances.

But a high output impedance isn’t actually necessary, especially if the amplifier is dedicated to low impedance headphones.

What is a pop shield?

A pop or wind shield protects microphones against air flows, which occur outside or when speaking mainly with breathed consonants such as “P”, “T”, “S” or “F”.

The materials used are foam or furry covers as  well as nylon-mesh screens in studios. 

In order to protect the diaphragm, many microphones provide a fixed basket made of metal and gauze, which also protects against wind to a certain extent. 

With studio microphones the pop shield has to protect the sensitive condenser diaphragm against the humidity and condensate which arise when speaking and singing.

What is signal to noise ratio, according to 1 Pa?

The difference between the selfnoise of microphone capsule and internal electronics (stated as acoustical level, which would result in the according electrical output level) and a reference level of 94 dB SPL (equivalent to 1 Pascal).

In case of identical measuring methods, either a weighted (adapted to human hearing) or CCIR, according selfnoise and signal to noise ratio add up to 94 dB.

Differences between beyerdynamic M 201 N, M 201 N (C) and the current M 201 TG

Basically they are the same microphones, but differ in age and connector: the M 201 N provides a three-pin DIN-connector and is the  original version of the 60’s of the last century (“N” means low-impedance, at that time this had to be mentioned).

Shortly  afterwards, the version M 201 N (C) was available, whereas “(C)” stands for “Cannon”, the inventor of the XLR connector. 

These two versions could be converted into one another. This is the reason why the respective connector was in a screw sleeve which poked out of the microphone shaft a little bit. 

Over the years the DIN version was discontinued and in the 80’s the so-called “Tour Group” microphone  series was introduced, where the microphone was integrated as M 201 TG. 

Now the XLR connector is flush with the microphone shaft. Apart from  that we manufacture the M 201 technically almost unchanged for many  years. 

Differences between beyerdynamic MCE 86 II, MCE 86 S II and MCE 86 S II CAM

The MCE 86 II can only be powered with phantom power. Your camera  provides this power; therefore you do not need a battery-powered microphone. 

The MCE 86 S II is the same microphone, but it can also be powered by a 1.5 V Mignon battery. 

The MCE 86 S II CAM is only a version of the MCE 86 S II for cameras (it can also be battery powered) with appropriate accessories such as the EA 86 elastic suspension, an adapter cable with mini jack plug and WS 716 wind shield. 

How do you have to position the capsule of the CK 930 T Set?

In the voice frequency response the CK 930 can be used as an acoustical boundary microphone and can be placed on the middle of a table.

Thus the direct sound and the sound reflected from the table top reach the microphone capsule almost equiphasely, a theoretical level gain of 6 dB (in practice it is approx. 5 dB) results compared to a set-up in the free field. 

The selected distance between microphone capsule and acoustical boundary of the CK 930 significantly improves the rear attenuation and reduces the risk of feedback, but has been optimised for the voice range. 

Therefore, the capsule should only point to the sound source in applications with acoustically effective boundary, when placed on a boundary it should be positioned  horizontally.

What ist meant by transducer type?

The vibrating unit in a dynamic moving coil microphone (i.e. the diaphragm and the oscillating coil attached to it) has a greater weight than a ribbon or the diaphragm of a condenser microphone. 

More energy from the sound signal is required to make this greater mass vibrate (which is necessary for sound conversion) than is needed to make the lighter ribbons and diaphragms vibrate.

A heavier diaphragm also follows a complex audio signal more lethargically than a lighter diaphragm. 

What initially sounds like characteristics that would eliminate moving coil microphones from competition with ribbon and condenser microphones can actually be extremely useful in many applications: 

a well designed moving coil microphone can often suppress interference noise on a stage (other instruments, monitor speakers, etc.) better than a corresponding condenser microphone. 

Dynamic microphones (with few exceptions) do not need any supply voltage at all and are often mechanically somewhat more robust. 

It is generally the case that a very good condenser microphone can be constructed so that it is more neutral in sound than a dynamic microphone.

What is meant by operating principle?

With “pressure” microphones, the back of the microphone diaphragm is isolated from the sound field, they always have an omni-directional polar pattern, i.e. they pick up sound from all directions at almost exactly the same level.

With “pressure gradient” microphones, the pressure difference between the front and back sides of the diaphragm determines the preferred pick-up direction; they are always unidirectional microphones.

What is the polar pattern?

Even though many users believe that only the long cylindrical microphones that are seen on television are unidirectional microphones, the reality is that most models used in video, studio and live applications are unidirectional microphones – no matter what they look like.

The name comes from the fact that these microphones transmit the sound only from one direction at the full level (for the most part), while the signals from all other directions are picked up as only attenuated (softer) sounds. The designation “cardioid polar pattern” results from the two-dimensional representation of the corresponding three-dimensional measurement: it has a cardioid-shaped area around the front of the microphone, in which all signals are picked up at approximately the same level (this shape looks something like a heart, which is why the pattern is called “cardioid”). It permits a relatively large range of movement in front of the microphone, while the attenuation of signals coming from behind the microphone (180°) is particularly good. With super- and hyper-cardioid polar patterns, the front pick-up range is increasingly narrowed and the range of strongest attenuation moves toward the diagonal rear (126° – 110° from the microphone axis).

With good designs, the reduced range of movement in front of the microphone (in comparison to the cardioid) is rewarded with low crosstalk with other signals (monitor speakers, instruments, etc.) and greater feedback rejection. 

With increasing directivity, i.e. starting with cardioid and continuing to super- and hyper-cardioid and on to lobar, unidirectional microphones must be directed more precisely toward the sound source; signals from the side are increasingly coloured in terms of sound quality and are transmitted at a lower level. For these reasons and others, microphones with a lobar polar pattern are not suitable for applications in front of large sound sources like choirs or orchestras.

This also explains why the question regarding the “range” of a microphone – e.g. in video applications – cannot be answered: Signals that strike the microphone from the side are transmitted more softly than those from the front; a voice from the other side of a busy street cannot be picked up in an intelligible manner even by a good microphone with a lobar polar pattern. Here it is physics that requires the user to either close the street to traffic or cross over to the other side.

What is the max. SPL at 1 kHz

Acoustic level at and above which the microphone produces a specified total harmonic distortion, usually 1% at 1 kHz.

This level defines the upper limit of the linear operating range of a microphone. This value is not specified for most dynamic microphones because it is so high that it can no longer be measured accurately.

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