Index


Speakers page

    General information

    Loudspeakers are avery important part of audio system. Speakers are the most visible and most crucial components of any audio system.Speakers are so vital to the sound of an audio system because they are the actual reproducers of the sound. More than any other component, the speakers impact the sound, character, feel and power of reproduced sound. The loudspeakers are almost always the limiting element on the fidelity of a reproduced sound in either home or theater. The other stages in sound reproduction are mostly electronic, and the electronic components are highly developed. The loudspeaker involves electromechanical processes where the amplified audio signal must move a cone or other mechanical device to produce sound like the original sound wave. Since Chester W Rice and Edward W Kellogg in the mid-1920s invented the moving-coil-drive unit at General Electric Co, conventional loudspeakers have operated as pistons. They operate in this way no matter what their method of transduction is?electromagnetic, electrostatic, or piezoelectric. Like a paddle in water, they move air and so generate sound with a relatively rigid diaphragm. For this reason, most speaker-drive units? diaphragms are very often cones or, in the case of tweeters, domes to confer on them a much higher stiffness than you could easily achieve with a flat alternative.

    This process of converting the electrical signal to sound waves involves many difficulties, and usually the speaker performance is the most imperfect of the steps in sound reproduction. In practice, the diaphragm materials that cover bass and midrange frequencies are too limp, even in cone form, to avoid the flexure of the diaphragm. This flexure can lead to resonance that cause severe sound quality problems. Lots of speaker research and developmnt has focused on the measurement and suppression of resonance. Pistonic operation of speaker element has one significant drawback: As frequency rises toward and beyond the point at which the wavelength in air is comparable with the dimensions of the diaphragm,the acoustic output becomes increasingly directional, and the speaker begins to "beam." You can avoid this problem by making the diaphragm sufficiently small (the wavelength in air at 20 kHz is only 17 mm). To generate a given sound-pressure level, the diaphragm?s acceleration must be constant, meaning that its excursion must quadruple with every halving of frequency. This phenomenon explains why high-quality moving-coil loudspeakers usually have two or more drive units: a large, low-frequency driver,which can move enough air to reproduce bass sounds, and a smaller, high-frequency driver.

    So, why do speakers have a very large impact on the sound of a given audio system and why they are not ideal? Because they are the physical components that actually create sound waves in the air. It is much more difficult to efficiently and effectively move air with little distortion thereby reproducing an audio signal in the form of sound waves than it is to reproduce small electrical signals that mimic those sound waves.

    Speakers are magnetic devices (at least, the common kinds are) containing a largish magnet and an electromagnet. It's not unheard-of for speakers to mess with TVs. It is usually the permanent magnet that causes the problem, because it generates a a steady magnetic field that can interfere with TV CRT operation (it uses magentic beam deflection, so extra external magnetics can put the electric beam to wrong place on the screen giving picture geometry and color errors). This magentic field gets smalle much faster than inverse square, so a smal distance between speaker and the sensitive device helps to keep problems away. Depending on the device, enough distance from speaker could be half meter or one meter. In case you need to place spaker near to TV or computer monitor, you need to get a pair of magnetically shielded speakers (PC multimedia spakers are built in this way, also many magnetically shielded hifi spakers are available in home theater markets).

    There are also non-magnetic speaker element types. The most well known types of those are piezo speaker elements and electrostatic speakers. Piezo speakers can generally used for high frequency playback. Piezo elements are generally used to make all kinds of simple beepers / alarms and sometimes used to make some high frequency speaker elements. Piezo elements consist of special crystals that change their shape when voltage is applied to them.

    Electrostatics speaker elements use electrical field to generate mechanical movement. Electrostatic element has lightweight membrane that is capable of producing audio frequencies, and that membrane is moved with aid of an electric field. Th audio signal from amplifier is converted to high voltage that then moved that membrane. Electrostatics speaker (EL) is based on dipole operation. The sound gest out from the speaker both from front and back of the speaker. Because there is only one element there is no phase errors or delay. And because diaphragm is lightweight transient response is nearly ideal if mechanical and electrical realization is made well enough. The big reason why ESLs sound like ESLs, is the coherent nature of a single, full range driver and the lack of cabinet diffraction problems. Very few manufacturers have managed to get moving coil driver systems to deliver a coherent wave-front.

    Speaker wiring

    Amplifier to speaker connection is an important part of audio system chain. There are fe thigns you should keep in mind related to speaker wiring:

    • Speaker and receiver/amplifier connections are usually identified as positive and negative. Make connections carefully. Positive to positive and negative to negative. If a speaker's connectionsare reversed, speakers will operate out of phase, resulting in unsatisfactory audio performance.
    • Be careful to prevent positive and negativespeaker wire conductors touching duringoperation. This could result in equipmentdamage.
    • Ensure that speaker system impedance(electrical resistance) is not lower than that specified by the manufacturer ofyour receiver or amplifier.
    • Choose speaker wiring size (gauge) to match amplifier power and cabling distance. The ugly truth: Any length of speaker cable degrades performance and efficiency. A simple fact to remember on speaker cables: Current needs copper, voltage needs insulation. No signal can travel well on a thin and long wire.
    • For ideal performance the cables should be of similar length. If the two speaker cables are not close to each other in resistance, and inductance, the damping will be different. The capacitance will also be different. How much effect those have in normal life on short distances is a question, and usually small differences do not matter. The signal delay difference caused by different cable lenghts is no problem, because the signal on the speaker cables travels at almost the speed of light.
    • It is not recommended to coil excess speaker cable on the floor, because coiling the cable greatly increases the inductance, and other interaction effects that may cause more than just an extra amount of high frequency roll-off. A snaking "S" pattern will avoid the worst of any interaction problems.
    Copper is a very good conductor of electricity, but it isn't perfect. It has a certain amount of resistance, determined primarily on its cross-sectional area (but also by its purity and temperature). This wiring resistance is "seen" by the amplifier output as part of the load in series of the actual load. Since increasing the load impedance decreases current flow, decreasing power delivery, we have lost some of the amplifier's power capability merely by adding the series resistance of the cable to the load. There is a definite impact on the amplifier damping factor caused by cabling resistance/impedance. Damping, the ability of the amplifier to control the movement of the speaker, is especially noticeable in percussive low-frequency program material like kick drum, bass guitar and tympani. Boomy, mushy bass is the result of poor damping and clean "tight" bass is a sign of good damping at work. In short, you need thick enough wires and you are happy witht he losses caused by resitance and on the damping factor. Another rule of thumb is that 4-ohm load should require conductors with twice the copper of an 8-ohm load, assuming the length of the run to the speaker is the same and you want same performance. For most speaker applications 14 gauge lamp cord ok (2.5mm^2). If really long runs or very high power system, get 12 gauge (4mm^2).

    The extremely low impedance nature of speaker circuits makes cable capacitance a very minor factor in overall performance and the dielectric properties of the insulation used are nowhere critical on speaker cables compared to the effect of speaker wire resistance. Very high capacitances should be avoided, because highly capacitive cables can trigger HF oscillation in amplifiers which are only marginally stable. Once some amplifiers has this problem, but nowadays no properly designed modern amplfier should have a problem in this regard. The cable capacitance effect is the reason why shielded speaker cables are not generally recommended (shielded cable has higher capacitance than similar non-shielded). In general, a cable which exhibits high capacitance usually also exhibit low inductance. Low inductance is theoretically desirable in a speaker cable, as it reduces the cable reactance at high frequencies. In practice, even driving a 3-ohm load over thirty feet of cable with highest inductance cable commonly available will result in a treble droop of less than 1dB at 20kHz. Some of those low-inductance cables claim to have a 'matched' impedance, as the characteristic impedance of the cable is only 6-8 ohms, as opposed to the 50-100 ohms of most speaker cable. In reality, this is an insupportable argument, since the cable is not being driven from a 6-8 ohm source, speaker impedance varies wildly, and a matched impedance would in any case only be of importance when cable length exceeds about 1/10 of a wavelength. The wavelength of a 20kHz signal in such a cable is about six miles. This type of 'impedance matched' speaker wire offers no real advantage in normal applications (will sound identical to 12AWG zipcord), and would not be really cost effective.

    Some speaker cable cable vendors talk about loudspeaker cables characteristic impedance, with claims that it should match the speaker impedance for "optimum results". One thing the cable vendors have completely neglected to point out, is that the characteristic impedance is only important (and relevant) when the source impedance, cable impedance and load impedance are all matched. Having an extremely low impedance at one end (the amplifier) and a variable impedance at the other (the majority of all loudspeakers) makes true matching impossible. The cable characteristic impedance is relevant and really show out only when cable is length approach or is considerably greater than the wavelenght of the signal in the cable. When the cable is considerable shorter than this, the characteristic impedance details are not relevant on the real performacne of the cable, and in normal electronics design they are neglected when cable distance is less than tenth of the wavelenght. The wavelength of highest audio frequencies (20 kHz) on the cable is around 10 kilometers. When the speaker cables in real-life applications are only very smallfractuion of that, the charateristic impedance does not have any real relevance on speaker cables because cable electrically so short.

    If the spaker cable is going to be soldered to the connector the most important consideration for insulation for speaker cables is then probably heat resistance(so that the cable insulation van take the soldering temperature nicely).

    Speakers are typically floating loads. This means that the magnetic fieldsaround speaker cables due to signals flowing through the cables aresymmetric and cancel quite nicely in in 'far field' some distance from cable. However closer in, the fields are the superposition of those of the outward and return conductor bundles which normally do not cancel to zero. If you want to get as much of this cancellation effect as is reasonably possible, twist the speaker leads. Twisting will probably reduce any overall H-field couping to other nearby cables. Although with most domestic audio, the signal connection from the power amp to the speaker is not balanced. The vast majority of power amps (particularly those without outputtransformers) have one side of the output grounded. Hence the E-fields near thecables (assuming a form of twin-feed) are non-zero. Hence capacitivecoupling to nearby objects may well occur. However, in most cases the wiring to the speaker, and the speaker itself arecompletely "floating" (i.e. not connected to anything else). This providesas much of a "balanced" circuit as you need for intereference-resistant speaker wiring. Because amplifier outputs and speakers are very low impedance devices, it is difficult to induce interference on them. You can probably run your speaker cables along any kind of signal or power cable you're likely to have in your home and not be able to detect any ill effect.

    There are many different connector types used to connect speaker cables to speakers. Banana plugs (4mm diameter connector pin) are very traditional speaker connectors. Banana plug is good connector for speaker signals, both mechanically and electrically. You can see those connectors on many audio amplifiers and speakers. Banana plugs make reliable connection and those connectors are available in hifi industry for even very thic cables. In Europe the audio industry is moving away from banana connectors. European safety standard EN 60065 for consumer electronicseffectively bans the use of the 4mm 'banana' type plug commonly used forconnecting loudspeakers to amplifiers. The reasoning behind the prohibitionof cable-mounted 4mm (and smaller-sized) plugs is that they can be inserted into a European mains socket with possibly fatal consequences. Another thing with banana connectors is that most banana connectors used in hifi systems are non-insulated bare metal connectors that leave the voltage going to speaker very easy to touch (this voltage can tens of volts on high power amplifiers and can be thus potentially dangerous).

    Audio industry seems to be moving away from banana connectors and replace them with other alternative. There has been many ideas what this connector could be. Professional audio industry seems to prefer Speakon connector. The major professional loudspeaker manufacturers have been using Speakon? connectors for the input termination on their products for several years now. With Speakon? connectors, you can plug straight from the amp to the speaker, and start making those great sounds right away. The Speakon? connector meets all known safety regulations. Once wired correctly, the connector cannot be plugged in backwards, causing the type of inverted polarity situations that are common with banana hookups. It will provide a safe, secure and reliable method of interfacing your amplifier to the load. Speakon connectors are available in several versions, from 2 pins to 8 pins. The four pin Neutrik Speakon NL4FC Connector is the most commonly used type.

    For home HiFi systems there is no uniform wiring connection standard (other than insulated screws which take bare wire and spade plugs on terminal screws). If the amp/receiver has spring loaded terminals, there are not a whole lot of options. The gold plated pins that are available for such connectors are not an improvement over a good bare stripped wire connection. If you have spring loaded terminals, twist wire hard and tight, and insert it naked. Those speakers with gold-plated heavy duty 5-way binding posts, the choice is clear: gold plated spades, preferably properly crimped. If you can not make a proper crimp via a crimping tool, then a proper solder joint will be very close in quality to a good crimp. After spade lugs, the next best connection is a gold-plated expanding/locking banana plug. Standard non-expanding or non-locking banana plugs with a nickel plating are not very good at all.

    You should always use the recommended load for your amplifier. This means selecting suitable speaker type or wiring multimpe speakers in the right way. Most solid-state amplifiers would rather look at an open circuit (no speaker at all) than a load. Therefore, you can usually use a load that is higher than the recommended load without any problems or damage. Most powerful tube amplifiers need a load to avoid transformer or tube socket damage. If a mismatch can not be avoided on a tube amp, it is usually better to go towards a lower impedance rather than too high of a speaker impedance (this may stress the tubes more than normal operation).

    Sometimes you might have heard term bi-wiring. Bi-wiring is where the crossover inside the speaker has been separated into it's HF and LF sections, and separate pairs of connecting terminals provided to access those separate sections independently. Normally, the LF and HF crossover sections are in parallel, connected internally to the same single pair of binding posts. For single cable use, a set of jumpers is provided to bridge the terminal pairs, paralleling the separated crossover sections outside the cabinet instead of inside. Then, separate speaker cables are run from the same amp output to these separated pairs of terminals at the speaker. With the electrical separation, differing currents will flow within the two cables that make up a bi-wire set. Two things happen due to this: The losses in the cable due to "eye-squared-are" losses (current squared time the resistance equals voltage drop) are reduced for each frequency band, so that any tendency for the woofer to modulate the tweeter due to current draw is greatly reduced. Secondly the magnetic fields due to the HF and LF currents have also been separated out, and any tendency for them to inter modulate and cause sonic artifacts has been greatly reduced.

    Speaker specifications information

      How to read speaker specs

      Typical specifications on speakers you can see is impedance, power rating, recommended amplifier power, frequency response, distortion and effiency. You can often see also other specifications. Impedance is the AC equivalent of resistance in a DC circuit. Speaker impedance is usually largely dependent on frequency. Remeber are not resistors, then impedance is a lot variable withfrequency. 4 Ohms or 8 Ohms is the impedance at a certain frequency, but notin the entire frequency spectrum. Real impedance of a nominal 8 Ohms speakercan vary for example from 6 Ohms to 20 Ohms (variation can be much higher on some speakers). The impedance rating told on speaker specifications is the "average impedance" or "nominal impedance" (the impedance stays in the operation usually at higher or lower than this, but generally stays above 75% of nominal impedance). Loudspeaker impedance has aspecific and well-understood meaning that is codified by severalwidely agreed-upon internationally-recognized standardsdocuments, IEC 268-5/60268-5 being one example:" The rated impedance of a loudspeaker or loudspeaker system is that value of a pure resistance which is to be substituted to the loudspaekr system when defining the available electric power of the source. This is to be specified by the manufacturer.""The rated impedance specified by the manufacturer normally represents the lowest value of the modulus of the impedance in that part of the frequency range, where the maximum power is to be expected, and is normally not more than 20% higher than the lowest value of the modulus of impedance at any frequency within the rated frequency range."Because the impedance varies with the requency, some peaker also show impedance curve. The impedance curve is a description, usually presented in the form of a graph,of the modulus of impedance as a function of frequency,measured under normal working conditions. This impedance can be measured using constant voltage or constant current source, but the method chosen shall be stated.Generally when you are buying or connecting a speaker to an amplifier, check that the speaker impedance is equal or higher than that is the lowest allowed impedance on your amplifier intended to drive those spakers (some aplifiers are limited to 8 ohms, some other can go to 4 ohms or even lower). Minimum and maximum power are pretty meaningliess specifications without more information. Minimum power specification is meaningless unless they specify minimum power for what. And maximum power specification does not help much either, unless the manufacturer specifies at what conditions the speaker can take this power. Often used speaker power handling capacity test standard is IEC 268-5, which specify filtered pink noise test signal.The nominal power (RMS power) for speakers is defined by supplying continous power is measured by pink noise rather than a sinousoidal signal and it is applied for 24 hours. Maximum power rating is a value which means almost nothing, but is used nonetheless by manufacturers to entice the unsuspecting into purchasing their product based solely on the big number. Technically, it is the maximum wattage that an audio component can deliver/handle as a brief burst during a musical peak. Most reputable manufacturers will provide both an RMS and Max power rating. Typically, the given value for the maximum power rating is twice to three times that of RMS. Frequency Response is the frequency range to which the speaker can respond. Full-range speaker is a speaker designed to reproduce all or most of the sound spectrum within human hearing (20Hz - 20KHz). Typical speakers are not able to play back this whole frequency range. Please note that in frequency response many companies publish only "Half Spec's". For example 50 to 20 kHz is meaningless with knowing what the deviation is from the reference level at a specified frequency (typically 1 kHz). For example 50 to 20 kHz + or - 3dB is decent and+ or - 20 dB is atrocious (quite useless product). So both pieces of information are necessary to be useful, be suspicious ofnew gear with low prices and great looking spec's. If the speaker just says frequency response 30-25,000 Hz and no other information like deviation and how it is measured, this "spec' tells you absolutely nothing. It could be interpreted to meaning nothing more than if you put any frequency in between 30 and 25,000 Hz, SOMETHING out. Frequency response, without some statement of the error band, the measurement conditions and such doesn't mean a thing. A good specification would be something like: 50-20,000 Hz, +- 3 dB measured at 2 meters on the principle axis. Harmonic Distortion means that harmonics are artificially added by the non-linearities of thespeaker, and are generally undesirable (but can not be completetely avoided). It is expressed as a percentage of the original signal. Magnetic shielding means that the speaker does not generate such large magnetic field as normal speakers around it. Magentic shielded speakers can be placed near sensitive equipment (like TVs and computer monitors) without causing them any problems (most "normal" speakers cause distortion to picture if they are too near to picture tube of TV or monitor). Efficiency tells how much incoming power the a speaker converts to sound. Typical closed and vented cabinets are less than 5% efficient, so you generate a lot more heat than sound volume. Because speaker efficiency is usually that low and does not tell much on audio itself, the speaker manufaturers usually use another measure on speaker "efficiency". Speaker manyfacturers generally list a spaker parameter that tells how much sound you can generate with the speaker with a given power fed to the speaker (reference power usually 1 watt). Typical "Hifi" spakers have efficiency of around 80-90 dB/m/W (decibels at one meter distance with one watt of power fed to the speaker). Typical PA speakers can have higher efficiencies, typically in 95-110 dB/m/W range.

    Speaker element information

    The loudspeaker element involves electromechanical processes where the amplified audio signal must move a cone or other mechanical device to produce sound like the original sound wave. This process involves many difficulties, and usually is the most imperfect of the steps in sound reproduction. An enormous amount of engineering work has gone into the design of today's dynamic loudspeaker.

    Most dynamic drivers come with a specification sheet with certain parameters that define the acoustical and electrical behavior of the speaker. These are often called "Thiele/Small" parameters, named after the two engineers who did much research trying to standardize and define driver/enclosure relationships.

    Speaker elements are usually characterized using followign parameter (all of above or usually only some of them):

    • F3: The roll-off frequency at which the speaker driver's response is down -3dB. F3 is determined by the frequency at which the output is 3dB lower than the level at 100Hz (or other frequncy if specified).
    • Fs: This is the resonance frequency of the speaker element (resonance in free air).
    • Impedance: The measure of the magnitude of an electrical load when using alternating currents, such as in audio. It describes the combined effect of resistance, capacitance and inductance. This is usually 4 ohms or 8 ohms depending on speaker element.
    • Le: The electrical inductance of a speaker?s voice coil. This is measured in millihenries (mH).
    • Power Handling: This the maximum continuous sine wave power that can be dissipated by the voice coil/magnet assembly without failure. This is measured in watts RMS.
    • Qes: The Q of a driver at its free air resonance considering only its electrical losses.
    • Qms: The Q of a driver at its free air resonance considering only its mechanical losses.
    • Qtc: The total Q of a woofer and sealed enclosure at the system?s resonance frequency, considering all resistive losses. Qtc: the . Q. of a sealed enclosure. A Qtc of .7 has smoothest response and the lowest F3. A Qtc of above 1.1 should only be used by those who like a . Boomy. response.
    • Qts: The total Q of a woofer at Fs, considering all driver resistances.
    • Re: This is the actual DC resistance of a speaker?s voice coil as measured with a standard volt/ohm meter. The reading will be lower than the speaker?s nominal impedance. A 4 ohm speaker will typically measure at 3.2 ohms, while an 8 ohm speaker will be about 6.4 ohms.
    • Sd: The active radiating area of a speaker cone, including the part of the surround which moves to produce acoustic output.
    • SPLo: The speaker?s reference efficiency measured with 1 watt input at a distance of 1 meter from the center of the cone. This is measured in decibels (dB). Typical values for hifi speaker elements for this is 85-95 dB/1w/1m.
    • Vas: Volume Acoustic Suspension. It is the volume of air having the same stiffness as the speaker?s suspension. This is measured in in cubic feet or liters.
    • Vd: The volume of air displaced by the speaker?s cone during an Xmax displacement
    • Xmax: This is the measure of a speaker cone?s maximum excursion in one direction while maintaining a linear behavior. This is meausred in millimeters.
    • Zs: This is the impedance of the speaker element at the resonance frequency.

    A louspeaker driver is a mechanical resonant system. Thecombination of the driver suspension's mechanical stiffness and the stiffness of the enclosed air together form the stuffness portion of the system, and the effective mass of the driver is,well, the mass. In any such mechanical resonant system, the constraint to motion, be it stiffness, mass or losses, depends upon whatfrequency you are operating at. Below resonance, it is thesystem stiffness that dominates. Below resonance, in the stiffness region, an applied force will only push the driver so far before the restoring force equals the applied force and the driver stops. Thus, excursion isconstant. Above resonance, it is the moving mass. Above resonance, an applied force will only accelratethe the mass of the driver so much, thus the driver is mass/acceleration limited and the excursion goes as the inversesquare of frequency (or, equivalently, the square of time,since x = 1/2 a t^2). At resonance, it is the combined mechanical losses of the system that dominate.

    In the simple example of a single driver, the driving force, below resonance, acts primarily against the mechanical stiffness of the driver's suspension, a reactance. Above resonance, it acts primarily against the mass of the speaker cone, again a reactiance. At resonance, it acts primarily against the frictional losses of the driver suspension. For speakers in enclosures, you can then add the enclosure stiff- ness, the port mass, leakage losses and the like. Across the entire range, a very SMALL part of this force acts against the driver's radiation impedance, which consists of both a resistive and a reactive part.

    Any fixed-size diaphragm will produce a sound level which isproportional to the area, proportional to the displacement, andproportional to the square of frequency. A constant excursion independentof frequency will result in a response that rise at 12 dB/octave.The issue of the resonance is important in many conventional speakersbecause the excursion "conveniently" decreases at -12 dB/octaveabove resonace, thus resulting in an output which is flat.

    Dust cup in spaker element is sometime talked about. The function of the dust cup is dust and other foreign matter out of the voice coil gap. Any small grit that gets in there could easily lodge in the gap and cause distortion and possible failure. Additionally, depending upon the design, it could seal the enclosure and it could be used to radiate higher frequencies on some speaker elements.

    The permanent magnet in the speaker element creates a suitible magnetic field that can be used together with voice coil generate force to move the diaphragm. There has been several different magent types used in speakers, but nowadays majority of the magnets are ceramic ferrite magnets. There are two popular magnetic materials: barium ferrite and strontium ferrite. They are practically equivalent performance and essentially cost wise and are used interchangeably. Mechanically, the ceramic magnets are quite hard and brittle and thus prone to cracking and shattering especially if mechanically shocked. The vast majority of ceramic-based magnet structures are assembled using anaerobic quick-set adhesives of the type typically known as cyanoacrylates. Trying to whack the magnet apart is almost always a guaranteed way of breaking it.

    The "strength" of the magnet, in term that are important to the operation of the speaker, is directly related to the flux density in which the wire of the voice coil is immersed. The relevant parameter is referred to as the "Bl product" (that's "bee-ell"). It is the product of the flux density, B, measured in Tesla and the length of the wire, in meters, that's immersed in that field. The Bl product is measured in units of Tesla-meters, and is equivalent to the amount of force per amount of current (actually 1 Tesla meter is equal to 1 Newton of force per ampere of current). The size or material of the hard magnet material is not what determines the important factor here, and that is the flux density in the voice coil cap. Generally for both Alnico- and ferrite ceramic-based magnet structures, the material used to actually focus and concentrate the magnetic field in the gap is almost always common, soft, low-carbon steels. Magnetic materials are capable of storing large amounts of magnet energy permanently, they have a limit: the flux density the can store is limited. The typical flux density at the surface of a ceramic magnet might be on the order of only 0.2 Tesla. When we concentrate that same total flux in a smaller area, the flux density would, obviously, be greater. However, there is a limit, and that is that the amount of flux you can direct through these auxiliary structures is limited. At best, soft, low-carbon steels "saturate" at flux densities of about 1.2 Tesla. Any attempt to put any more flux into them will not result in any increase in flux desnity, the extra flux will simple "spill out" of the structure into the air. In essence, the "strength" of the magnet field where it counts, that is in the voice coil gap, is limited usually not by the magnetics but the the soft iron that direct the magnetic field to the gap.

    Once you have chosen a good loudspeaker element from a reputable manufacturer and paid a good price for it, you might presume that you would get good sound reproduction from it. But you won't without a right enclosure. Right type of enclosure is an essential part of sound production.

    Some people believe that speakers need "break in" to sound good. That "break in" mans that the speakers should be played for some time with osund to make them sound right (for example few days with music or pink noise). There is some idea of breaking in a woofer to slightly "loosen" the suspension, slightly lowering resonant frequency, but that likely won't substantially change the sound of a speaker. If anything it would change the low end only slightly. Putting my hopes in speaker break in so that the speakers will change in a positive way if you "run them" for a while. I wouldn't expect the sound to change at all. If you hear considerable changes for better, most propably those changes are caused by the fact that you are used to the sound of the speakers and start to like it as it is.

      Information on speaker element parameters

    Speaker design basics

    Designing and building speaker systems can be fun and rewarding. Loudspeaker design looks, on the face of it, deceptively simple. Unfortunately, the number of variables are infinite, and the skill of the designer is one of balancing the variables to achieve an acceptable result within the design specification. There is no such thing as a perfect loudspeaker!

    When designing loudspeakers lots of things needs to be considered. First you need to consider the intended use for your spaker.The designer of a Public Address (PA) speaker would congratulate himself if he achieved a loud, highly focused sound that could be beamed to a section of the crowd at a football stadium; Quality of sound would be of much less importance. In PA spakers usually controlled directivity and good effiency (up to 100 dB/im/1w or more) are usually more important criterieas than flat frequency response.

    Typical PA speakers are designed to play bakc sound ar range of 60 Hz up to 15 KHz (some well designed ones can have better frequncy response). In home HIFI spakers the sound quality in listening position is the most important. The most important is that the sound quality is best at the intended listening position. In home spaker we are usuallyasking for good frequenyc response (as flat as possible from 20 Hz to 20 KHz is preferred), good transient response and good phase response.The speaker efficiency is not that important (85-90 dB/1m/1w is usuallyadequate), because usually few tens of watts of amplifier power is available and it will give usually more than edequate amound of soundto a typical room.

    There are an infinite number of ways of solving the speaker design maze. Designing a speaker ccmbines engineering, experimenting and some form of "art".You need to select suitable element, select right enclosure type, make right size enclosure and design right kind of crossover suitable for those elements. For selecting elements you need to know their technical parameters. The Thiele-Small parameters are needed to design the bass enclosure for best bass response. The Thiele-Small techniques are used world-wide for the rational design of low-frequency systems. Those techniques are used by both speaker manufactuers and amateurs alike. The more you get into speaker design, the more you realize that the nature of sound reproduction is very complex. Lots of papers are publicized on loudspeakers, there is a lot of good measuring equipment available. So this means that nearly anyone is able to make something competent.

    Closed enclosures are the simplest enclosures. They consists of speaker element(s) installed to a specified size enclosure. The suitable speaker enclosire size depends on the element parameters. The most typical closed enclosure system rtarget the total Q of a woofer and sealed enclosure at the system?s resonance frequecy (known as Qtc) to be around 0.7. The Qtc value of 0.707 has smoothest response and the lowest bass response (-3dB point). Sometimes differnet values (usually higher) are used. A Qtc of above 1.1 should only be used by those who like a boomy response Vented enclosures (also known as ported or bass-reflex enclosures) have been used since the early days of loudspeakers.A good vented design provides essentially flat response to just above a low-frequency 3dB-down point (f3), with system output rolling at some rate below that frquency (typically 24 or 36 dB per octave). When designing vented enclosures you need to know speaker element parameters, because certain speaker parameters work together with specific enclosure volume and tunign frequency produce a certain frequency response.

    Loudspeaker simulation programs can help the speaker design considerably.The loudspeaker simulation programs and measuring equipment often does not require that the user really understands the input he gives to the program. This may lead to simplified models.

    A very high proportion of all woofers in 2-way speakers have additional inductance placed in series with them. The inductance is part of the speaker's crossover, and the goal of the woofer crossover is to further limit the high frequency response of the speaker. The idea of fast woofers is a fallacy because almost all woofers are driven through low-pass filters that reduce the high frequency response and slow down the transient response of a woofer even more than the mechanical limits thusfar described. Another reason why electrical circuits are used to slow down woofers and reduce frequency response well below mechanical the mechanical limits of the driver relates to the smoothness of frequency response. Just because a speaker responds to some higher frequency doesn't mean that the response is smooth in that area. In many cases the voice coil inductance is relatively unimportant in woofers because it is common to put even larger external inductors in series with their voice coils.

    The mass of the cone has a lot to do with low frequency response, which is why you can't use a tweeter for a woofer. When you add mass to a speakers cone and retune the enclosure for the new massier woofer, you get greater bass extension at the cost of lower efficiency.

    Rememeber that frequency response isn't everything. Dispersion is also important. Dispersion is largely based on the diameter of the radiating surface. The larger the diameter of the speaker, the less dispersion at high frequencies. As a rule, people don't want beamy-sounding speakers, speakers that beam sound like tight laser beams, speakers that have a sweet spot the size of a dime.

    When designing speakers, remeber that for most thing there is no one "correct" answer what kind of best design would be. There "correct" answer for ported designs, there are a whole range ofpossibilities, depending on what your goals are. Some are better on one side, and some other on other aspects. Vented speaker system is very complex system. There does not exist and cannot exist anequation or system of equations to describe the behavior ofvented systems ("no closed form solution" in mathematical terms). Most speaker design equationa are simply least-squares fit.Start with a slightly different form and you can get verysimilar curves described by equations that look not verysimilar.For example when designing bass enclosures, many people look for "optimum enclosure" without specifying what they consider as the "optimumum performance". Optiumum bass responde could mean flatest response in the pass band, deepestresponse, highest efficiency, and so on. This depends from wo you ask. Many of these are somewhat mutually exclusive. So whoever you ask held from anybody you need to define what "optimum" means for you. There are no "correct"answers for closed box system either, but a range of possibilities (in fewer dimensions, to be sure), just likevented systems.

      Enclosure design

      So you just purchased that new set of drivers, and now your ready to build the enclosure. The bass response is the main focus of nearly every speaker system available today, and is the foundation for the rest of the system response. DIY amateur speaker designer must decide on the type of enclosure to build to house the woofer. The design of the speaker enclosure has most effect on the bass response of the speaker. The important parameter for bass response is the enclosure type, size of the box (volume) and how the element + enclosure system is tuned. Ported systems are all around good performers, and most commercial home speakers use some type of ported enclosure. The tuned port in these systems increases efficiency by nearly 3 dB in an optimum enclosure, and the roll-off frequency can be much lower, often by as much as 1/3 - 1 octave below a sealed enclosure. So, with nothing more than a properly designed optimum vented enclosure, you have very efficient bass reproduction with several advantages over an optimum sealed enclosure. The downside of the ported enclosure is that it must be carefully designed to give good performance. Ported boxes must be fairly precise in volume and tuning. If those are not right, the performance is not good. A ported system receives maximum damping at resonance, with minimal driver excursion at the encosure resonance. Below it ported system cut-off is a steep -24 dB/octave. Usually there are several possible alignments attainable by different tuning configurations and driver parameters when building a ported system. Usually there is an optimal alignment that gives a flat response to cutoff, a fair transient response, and good power handling. Ported enclosude is also sometimes called bass reflex enclosure. A sealed enclosure does not allow air to move from inside the enclosure to the outside and vice versa. It is basically just a closed box where the speaker element is mounted. A very clear advantage for a sealed enclosure is simplicity - you can get good performance with nearly any driver with an EBP of less than 90 in a simple sealed box. Enclosure volume is not critical with these designs, and a volume change of +/- 10 - 20% will not adversely affect the sound. Sealed boxes have very gentle roll-off characteristics after F3 at -12 dB/octave. On the downside, a sealed driver reaches maximum excursion at resonance, which adds considerably to the distortion produced at high output levels. Sealed enclosure damping is wholly dependent upon the air trapped behind it in the enclosure. A true optimum enclosure is a reality with a Qtc of 0.707, with superior transient response and a low F3. Sealed enclosure is also sometimes called acoustic suspension design.In some applications you can propably see a passive radiator enclosure. It is like a enclosude with two speaker elements, but only one of them has voice coil in them. The second element without voice coil acts like the enclosude tuning element, like the port in ported enclosure. Basically passive radiator spakers have similar characteristics as poerted enclosures. The main difference is just how the tuning is done.Sealed designs have several advantages for the car. One is enclosure size, as optimum ported systems generally must be larger. With all the new "small box" drivers available today for sealed enclosures, very good performance can be achieved with minimal space used in the vehicle. Building an optimum ported enclosure for the home is easy, as you usually have more than enough space for that big box.

      Crossover design

      Speaker drivers cannot easily recreate all the frequencies in the audible spectrum by themselves. Instead, specific drivers need to be used to recreate various chunks or bands of the frequency spectrum. In most cases, the frequency spectrum is divided up into two or three bands or groups of frequencies. Each band is then handed off to a specific driver well suited to reproduce it. A crossover is a device designed to divide audio information into smaller frequency ranges to comply with the requirements of different transducers in an audio reproduction system. This is accomplished by running the audio through a set of filters. There are four kind of filters that are used in this kind of applications:

      • a simple high pass filter can block the low frequency at designed range
      • a simple low pass filter can block the high frequency at designed range
      • a simple band pass filter can pass the frequency range at designed range
      • a simple band-stop filter block the sound at designed frequency range
      The real life speaker crossovers are combination of those filter types. Most often used filters are high pass and low pass filters. Crossovers can be passive or active designs. Crossovers used in home loudspeakers are almost exclusively of the passive variety meaning that they do not require electric power to operate. Passive crossovers are usually found inside speaker cabinets along with the speaker components. These often connect to the outside world via a single jack, but sometimes each speaker component also has its own jack in case one wants to bypass the built in passive crossover. Passive crossovers split up the frequency spectrum using a series of wire wrapped inductors, capacitors and resistors. Passive crossovers are geneally built once as iti is designed and it's perfomance can't be changed later in any way (changing it need chaging components in it). Active crossovers are placed before the power amp. In that application each frequency range is given its own power amp and its own drivers. Active crossover also needs some power to operate. The advantage of active crossovers is that they can be designed to have very well controllable frequency response and that they can be made adjustable (easy to adjust crossover frequencies and output signal levels). One reason for favouting active designs in high power audio systems is that crossovers can get bulky and expensive at high power levels (especially passive subwoofer filters). At high power systems Consider using a line-level crossover and multiple power amps. It could be simpler and cheaper ifyou're starting a new mid to high-end system from scratch. Well tuned active system sounds better too under some conditions, because the operation of active crossover can usually made more accurate. Commercially available passive crossover modules come with charts showing their frequency curves and recommended maximum load for different speaker impedances. You can alsobuy bare coils and capacitors to build your own. There are many online tools will help you calculate the raw components needed. The main design problem in using passive crossovers (both ready made modules and ones built from components) is that they are for the most part, designedfor 8-ohm (or whatever) RESISTIVE loads, and drivers presentsomething that's substantially different. The result is that thecorssover behavior and response work somewhat differently than youmight expect. This means that any of the off-the-shelf "standard" one-size-fits-all crossover are assured to have significantly compromised performance,simply because there cannot be any such things as a one-size-fits-all passive crossover. Once a passive crossover is set, it cannot be changed or adjusted (in other way than modifying the filte circuit or changing components in it). Active crossovers can be made more easily adjustable. The power handling of a crossover is largely depends upon the characteristics of the individual components in the crossover itself. The withstand voltage of the capacitors is one thing and the current handling capacity is another thing that limits the allowed power. Some crossiver designs have also resistors in them, and how they are connected and what is their power rating has effect on crossover power handling capacity.

      Magnetic shielding

      If you just want to build a speaker that you can position next to your TV,use self-shielded speaker driver(s).Magnetic shielding means that the speaker does not generate such large magnetic field as normal speakers around it. Magentic shielded speakers can be placed near sensitive equipment (like TVs and computer monitors) without causing them any problems (most "normal" speakers cause distortion to picture if they are too near to picture tube of TV or monitor). So called shielded speakers usualy have a thick guage mild steelcup over the magnet structure - specially made for the job. Majority of properly shielded low- and mid-frequency drivers use both the reverse magnet and the cupshield. If properly sized, the reverse magnet can eliminatealmost all of the stray field originating from the magnet andrear plate, while the cup shield attempts to deal with the fieldeminating from the front plate region.Some speaker element manufacturers even sell a range of cups and reverse magnets with differentdiameters for tweeters, midrangers and woofers.The idea of doing magnetic shielding is the following: Select a similar magnet as used in the speaker element. Put some glue on the magnet. Positionthe reverse magnet on the speaker magnet in the way the magnets wil repeleach other. Press them together and with a soft click they wil sticktogether. Wait until the glue is dry and glue the cup on the reverse magnet.That's all and you will have a more or less well shielded loudspeaker frame.Attaching external magnets may or may not result in low enough fields toleave the CRT undistorted.To make it still better, you can glue the shielding metal cup on thereverse magnet. To get this workign you might need to experiment with size of reverse magnet and the use of the shield cup.Getting the right sized bucking magnet can be difficult. Toosmall and the cancellation is insufficient. Too large and youhave the same problem only reversed in polarity and a morecomplex field distribution. If you have a toolbox of magnets youmight try various ones. The one with the LEAST attractive orreplusive force one is the one you want. Beware that the reversemagnet can and often does change the driver parameters.The cup method is usually used to redirect the residual fieldafter you already added the bucking magnet. The cup alone probably won't be enough in most cases. If you want to experiment with magnetic fields, go for what was described above. But if you justwant to build a speaker that is safe to use adjacent to a CRT, save yourselfa ton of money and frustration and just use self-shielded speaker driver(s).There are plenty of them out there in a variety of sizes and types.Sometimes mu-metal is mentioned as a good shielding material against magnetif fields. Unfortunately effective mu-metal shielding is not simple.Mu metal is very tricky stuff to apply, it isridiculously expensive, very difficult to work (cut, bend), and unlikely tostop any significant field when placed midpoint between the source (speakermagnet) and the object (CRT). Mu-metal is used in some devices for shielding, but it is not easy. For example the sort used on CRT tubeshave to be fashioned, spot welded and then annealed ( heat treated) to getthe magnetic properties back.And keep in mind that with many speakers the "enough distance" isanough to keep devices happy. A magnetic field's strength falls off as the fourth power of distance. Alittle more space is a safe solution. Before you build theenclosure, move the driver around near where you want it to be; atwhatever distance you see no effect on the picture, that's where to putit. Build the enclosure accordingly or select suitable shielding method if necessary.

    Speaker protection

    Speakers are expensive devices which can be dameged by excessive audio power or DC applied to them. Fot this reason some form of spaker protection is usually preferred. The type of speaker protection a designer chooses will depend on the application of the speaker. A fuse may be fine for home hi-fi applications for overload protection or no protection at all. If the overall sound system is correctly designed and operated within it?s limits then speaker protection will not usually be an issue. Most performers would rather have a speaker die a slow tortured death than to ever have a blown fuse shut down their performance before a live audience. There are three main reasons for speaker protection. First, to protect speakers from being overpowered during otherwise normal operation. Also, to protect from power surges and your mistakes such as knocked or dropped mics-faulty signal leads-squeal- ing, loud feedback - overly aggressive vocal performance. And, to protect speakers in event of an amplifier blowing up. You decide if you are at risk.A speaker breaks or blows out most often because it is overpowered or over driven. There are endless ways this can happen, but some amplifier distortion is nearly always involved (unless you use a very "over-powered" amplifier to blow up your speaker). In general, the problem of how to protect speakers is a difficult one with no universal solution. There are two types of protection systems for speakers: active and passive. Active types include compressor/ limiters, equalizer/filters, and purpose built speaker processors. Active devices are usually placed between the mixer and power amplifier in the signal chain. The most elaborate speaker processors monitor the power output of the power amp and adjust the input signal to keep the amp from overpowering the speaker. These are designed for specific speaker models (and used only in large PA systems). Far less expensive than active devices are different types of passive protection compo- nents usually mounted inside the speaker box. These include the simple fuse, thermal breaker (manual and auto resetting), polyswitch, capacitor, inductor, resistor, light bulb, zener diode, relay, and different combinations these. For example a well designed crossover provides the upper frequency drivers the protection they require from full range signals but cannot protect the drivers from excessive input power in their frequency range. Sometimes also low frequency driver protection is needed. If your speakers are experiencing mechanical failure because of too low frequencies fed to them, a good subsonic filter (at least 18 dB per octace) can help quite a bit by trimming out the most damaging frequencies.Naturally, one can simply rely on fuses. In line speaker fuses can be useful in non performance applications, protecting either complete speakers or individual elements. Fuses are cheap, speakers are usually not. Fuses have no (or minimal) impact on perceived audio quality. For woofer protection that offers flexibility and minimal cost is a simple fuse. Using a fuse requires you to be able to balance several factors at once. With trial and error you find the smallest fuse that will allow an uninterrupted performance in your situation, with your speaker, your amplifier, at your volume level. Some manufactures use a fuse or manual breaker in speaker. Over the years various manufacturers have employed various types of "light bulbs" wired in series with the speaker as a means of speaker protection. As the load current increases the bulbs filament heats up and the resistance of the light bulb increases thereby attenuating the power delivered to the speaker. The light bulb glows brighter and its resistance increases as the power to the driver is increased. Hence it sucks a higher percentage of the power than the driver.The bulb actually acts like a "compressor" by turning down the peaks. Selecting the correct bulb is tricky. For mid-/high- frequency horn driver protection, the series light bulb is in fashion these days. Manufacturers see this as an inexpensive way to protect drivers. Series buls can have some negative effects on the speaker performance. The speakers can be too bright at soft levels and too dull with fuzzy top end at loud levels. Usually a combinations of 12 and 24 volt automotive type bulbs of varying wattage are used depending on the size of the driver. Positive Temperature Coefficient (PTC) devices or thermistors can be used to protect speakers. They behave somewhat like a cross between fuses and bulbs. The poly-switch is a non-linear resistor, having a low resistance at normal temperatures and a much higher resistance at some designated temperature. Relays can be employed in conjunction with some signal rectifier circuitry to form a protection system. Many hi-fi amplifiers and professional power amps (and loudspeaker systems) provide some of protection, either to protect the speakers from an amp fault, and/or vice versa. The basic requirement of a speaker protector in requires that any potentially dangerous DC flow to the speakers should be interrupted as quickly as possible (this is typically done in amplifier end).

    Speaker measurements

    Speaker measurements are a wide and hard topic. Measuring loudspeakers correctly is a ?tough business: most people who alledgedly know what they're doing (or at least claim so) don't get it even remotely right. It requires very careful setup under well controlled conditions and consistent, well defined procedures.

    The simplest technique is simple SPL meter and signal source (test CD). A simple SPL meter and a test CD with different frequency test signals will probably give you numbers, but they will probably not be of any use as the measurement of speaker itself. In most situations, room effects actually dominate the frequency response performance of a sound system. Commercial speaker manufacturers typically measure their speakers in an anechoic chamber. The reason for this is to get rid of the room effect. Another technique used by commercial organizations are time gated measurements, which allow an user to make quite good speaker measurements (middle and high frequencies) on a normal room.

    When measuring speaker performance you should measuring one speaker at one time. If you are testing two speakers (left and right) at the same time, they will interfere with each other, which will give you wavful measurement results. The spikces and dips you get with two speakers are the direct result of comb filtering due to the fact that you have two real sources which are interfering constructively and destructively with one another. If you have two speakers as sound source, basically your measuring as much your improper setup as you are your speakers. Normal acoustical testing involves one speaker and one microphone.

    Measuring speakers inside "pretty live" room is also problematic, because in such case the room reflections easily dominate the sound. For this reason professionals test speakers in an anechoic chamber. If you don't have access to an anechoic chamber, try testing one speaker at a time, outdoors. An outdoor environment away from large structures is the cheapest useable alternative to an anechoic chamber. If possible, hoist the speaker and microphone way up in the air (for example set the speaker on the top rung of a ladder) to lower the frequency of the inevitable ground bounce. Forget about usign the swept sine wave which in generic frequency response measurements, stick with pink noise or band filtered noise.

    If you are using a decibel meter as your measurement instrument, be sure to use the "flat" frequency response. Be sure that normal A and C weightings are far from flat! A weighting is compleely useless for speaker measurements. And the frequency response nowhere near flat then if you were using C weighting, because the "C" setting is subject to a high frequency roll off above 8 kHz.

    If you want to test the acoustical response of the speakers, and eliminate the room reflections on a normal room, there is a method that uses impulse response testing where impulses are sent to the speaker and the early impulses are captured and the room reflections are not. The frequency response of the speaker alone can then be determined using math on the captured early impulses. Impulse systems like MLSSA are required to give you anything approaching accurate speaker response without a chamber. Some measurement methods use pure pulses, and some methods use pseudo-random noise (can be trated in mathematics as a series of pulses).

    The typical test signals used in speaker testing are:

    • A swept tone that has variable amplitude (decreasing amplitude at higher frequencies to protect the speaker tweeter)
    • A series of chirps at ascending frequencies (one octave or 1/3 octave chirps for example)
    • A series of band filtered noise signals at ascending frequencies (filtered to one octave or 1/3 octave wide bands for example)
    • Impulse signal (for impulse response)
    • A pseudo-random white or pink noise signal

    Nowadays when computing power is easily available, impulse response tests are generally done with help of pseudo-random noise signal. An impulse response derived directly from an actual impulse is the same as an impulse response derived by means of the cross-correlation of broadband signals. Only, the latter probably has a far better SNR.

    Getting meaningful results at low frequencies is very hard. It is very hard to make meaningful measurements in the sub 200 Hz region. It takes a very controlled environment and gated equipment, such as MLSSA. In any "ordinary" sized room, MLSSA is useless for measuring below200Hz as well, because the long time window required to get any kind ofresolution at low frequencies is more than long enough to let all thereflections in as well. (And thus show the room response, rather than theraw speaker response). When it comes to measuring bass, unfortunately there really is no substitutefor a BIG anechoic chamber, or the wide outdoors.

    Nowadays there are affordable PC based measurement systems that allow you to measure the speaker performance. In most systems the time-domain and frequency response, both amplitude and phase, of drivers and systems can be measured. Some also measures room reverberation time, SPL sound levels, and sundry other stuff.

      Speaker element parameters

      • Measuring Loudspeaker Parameters - There are several different ways to measure the Thiele/Small parameters of a loudspeaker driver. The method described here provides a way for the beginner and DIY enthusiast to measure the parameters without any expensive or specialised equipment.    Rate this link
      • Measuring speaker parameters - it's possible to get most of the Thiele-Small parameters from a loudspeaker by just accurately measuring the impedance versus frequency    Rate this link

    Headphones

    Headphones are categorized first by their type of transducer (earspeaker) technology and then by the style of wear. Dynamic (moving coil) headphones are the most common and are available in every form from lightweight foldables to heavy-duty studio monitors. Dynamic headphones have transducers that are basically miniature loudspeakers. The transducers in isodynamic headphones are miniature versions of magnetic planar loudspeakers. Electrostatic headphones have thin and lightweight diaphragms that vibrate inside in an electrostatic field. Electret headphones (also called fixed electrostatic) have permanently polarized diaphragms (or backing material), so a biasing power supply is not necessary. Both electrostatic and electret headphones operate at high signal voltages which is generally supplied by the signal converter supplied by the headphone manufacturer.

    Geneally dynamic headphone transducers are resistance-controlled, not mass-controlled like loudspeaker drivers above the main resonance. In any case 'damping factor' is largely nonsense. The rated load indicates the range of impedances that the amplifier can drive to full power (eg, 100mW at 30 ohms).

    International standard on audio interfaces, IEC 61938 (formerly IEC268- 15) calls for an intermediate-impedance source of 120 ohms and a source voltage of 5 V rms, which provide reasonably satisfactory performance (sound level, frequency response, distortion) with headphones of any impedance over the range of 8 to 600 ohms at least. The IEC 61938 international standard specifies that headphone manufacturers should assume that the amplifier has an output impedance of 120 ohms (the value of a typical output current-limiting resistor which protects the amp against shorts).

    The headphone output of home audio components is perfectly adequate to drive modern headphones to ear-splitting volume, but since headphones are often regarded as toys, little care may be gone into designing the circuitry. A simple headphone output might be nothing more than a couple of series resistors off the power amplifier's speaker terminals. Since power amps are supposed to drive loudspeakers, this configuration has the potential disadvantage of higher noise and output distortion than small signal amps, in addition to high output impedance, which could affect the sound of headphones.

    Other headphone outputs have their own driver circuits, but the question remains as to their quality. A highly regarded preamplifier may, nevertheless, be using a poorly implemented headphone driver that either audibly distorts the signal or has inadequate drive capability. Portable stereos are notorious for their weak headphone outputs, which are deliberately underpowered to conserve battery life.

    Modern dynamic headphones can reach maximum volume with only a few milliwatts of power. Therefore, unless the headphones are very inefficient, headphone amplifiers that are rated at over 500 milliwatts per channel are probably, extravagent - not to mention dangerous.

    A headphone distribution amplifier takes a single headphone input and splits it to drive multiple headphones. It is standard equipment in recording studios, and at its most basic, could be any power amplifier with a fan of resistors at the speaker terminals - one resistor for each headphone - to limit the current to the headphones and to help maintain a reasonable load across the amplifier. These are called "mass-feed systems." A distributed-feed amplifier will have separate gain blocks and level controls for each output.

    There are two basic types of adapters you might need with a headphone: converting between 1/4" and 1/8" (3.5mm) plugs and converting between stereo and monaural. The headphone jack that comes on most professional and home audio equipment is meant to mate with a 1/4" stereo plug. Portable stereos use the 1/8" mini-stereo plug.

    Wireless and cordless headphones operate without a cord. A transmitter (base) plugs into the sound source (the stereo), and the headphones (usually dynamic-type) have a built-in receiver and amplifier. "Wireless" sometimes refers to infrared-based systems and "cordless" to radio frequency (RF) transmission systems. Both infrared and RF systems are subject to background hiss (unless you have an expensive digital wireless headphone).

    Noise cancellation headphones have signal processing electronics that sample ambient sounds with miniature microphones and then generate an inverse signal inside the headphones to cancel the noise (up to 70% or 10dB). The active technology is most effective with low frequency noise, such as that from airplane engines, and when the headphones are closed-ear types to provide passive attenuation of ambient high frequencies. Passive noise reduction headphones are closed-ear types that are specially constructed to maximize noise filtering properties.

    Special surround sound headphone systems are also available. There are 4-channel headphonesa and personal surround systems. In addition there are special sound processors and normal headphoens togenerates a 3-D acoustic image by engaging the transfer function of the listener's head.

    In addition to normal stereo headphones there are mono headphones. Monaural headphones are used mostly with communications equipment (such as speech recognition systems, shortwave radios, scanners and cellular phone hands free units) and often come with attached microphones or voice tubes. They are commonly referred to as "headsets," although that term could also apply to stereophones.


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