Do I need a new Stereo?

You should consider a new stereo if:

You have distorted audio at normal listening levels.
You want expandability. (Amplifier, Processors)
You are concerned about sound quality.
You can't seam to find 8-Tracks.

Remember, Your factory stereo was chosen by the auto manufacture, Because the stereo manufacture gave the lowest bid. So this does not mean that it is one of the best on the market.
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Do I need new Speakers?

You should consider new speakers if:

You have distorted audio at normal listening levels.
You are concerned about sound quality.
You don't want to blow another set.
You don't have any.

NOTE:Your factory speakers were chosen by the auto manufacture to keep your automobiles total sticker price down. You see, Speakers are not what the automakers are trying to sell you. We don't specialize in car sales and they don't specialize in speaker sales.
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Do I need an Amplifier?

You should consider a new Amplifier if:

You want to play music with less distortion.
You listen to Classical music, Jazz music.(Music with a wide range of instruments.)
You would like great dynamic range.
You are concerned about sound quality.
Your ears are yet to bleed.

Having an amplifier is not just for playing your music loud, but also to play with clarity. You should always have more power (watts) that what you need. This allows you to have reserve for peaks in music. This way you are not pushing your amplifier to its limits, and distorting during musical peaks.
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Multiple Amplifiers?

You should consider Multiple Amplifiers if:

You want to run multiple speakers.
You want to play a little louder without distortion.
You are concerned about sound quality.
You would like good staging and imaging.
Only one ear is bleeding so far.

Multiple amplifiers are good when you want to run separate speakers. For example a Tweeter, Midrange, Subwoofer. One amp would be good for the Tweeter/Midrange combo, And another amplifier for the Subwoofer.
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200 Watt Speakers Multiple Amplifiers?

Your 50W amp will not run your 200W speakers.

If your speakers are rated at 200-Watt continuous (RMS), then you should at least use a 150-watt amp. Some will argue that you should use at least half the 200w power. Well this is ok if you constantly play your speakers loud. I recommend a minimum of 3/4 of the 200w. It is possible to under power a speaker and also ruin it. Do not get me wrong, a lot of times speakers are way over rated, if you paid $45 for a pair of speakers, there is a good chance that they are over rated. I Bet you never thought you could ruin a speaker by under powering it!
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Do I need Subwoofers?

You should consider SMALL Subwoofers if:

You have no space in your automobile.
You listen to Jazz or music without a lot of bass.
You seem to be lacking something.
You just want to fill in the bottom portion of music.

You should consider LARGE Subwoofers if:

You listen to a lot of Classical music with drums, cannons.
You listen to Rap music.
You listen to a lot of heavy bass music.
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Getting Louder?

Does your radio stop getting louder?

Have you have noticed that when you turn your stereo up more than half way, It does not seem to play any louder? If so read on...

I do not have an Amplifier:

What is happening is one of two things:

1. Your speakers have reached their limits. This is very unlikely without an amplifier.
2. You stereo has reached is maximum pure output, any more will just add noise and distortion. You should look into getting an amplifier.

Yes, I do have an Amplifier:

What is happening is one of two things:

1. Your speakers have reached their limits. This is very unlikely if you have upgraded speakers or the proper amplifier.
2. More than likely you have reached the limits of your amplifier The gain setting is probably too high. Your gain settings are not volume controls, But tell the amp how soon to max out. If your amp is 100 watts, then no matter how low or high the gain setting is, it is still only a 100-watt amp! Say your volume control goes from 1 to 10, do you want to reach 100 watts at 4? No, you want a good span to control volume, so we set the gain to reach 100 watts when the volume is at 10. If it is set up at 4, then it will get loud quick, but will get no louder past 4! So, once your gain is set correctly, And you find that your stereo is not loud enough for you, then you need a bigger amp!
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Speaker Crackle Why do my Speakers Crackle?

If you noticed that your speakers are crackling, There is a good chance that your speakers could be blown. But do not rule out the fact that any other component could also be the source of this crackling. A good installation facility would be able to find the source of your problem, most at no charge. Now back to the speakers, If these are the original speakers that came with your car then yes, they are more than likely blown. Original speakers are usually made of just paper and will fall apart after some time.
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Whiny Noise Why do I have Whiny Noise?

Noise can be generated by several different things in your automobile.

Here is a quick list of the most common.

Alternator Noise.
Alternator noise is the most common noise found in our sound system. You will recognize it as a high pitched sound that seems to get louder and higher pitched when you step on the gas pedal. Or possibly louder when you turn on the headlights or the A.C. Something in your system is grounded better or worse than another and is causing a small voltage to exist. You should consult with our qualified personnel to solve this problem.

Popping Noises.
Popping noises are generally caused from spark plug wires, relays on the car and generally require a different remedy to fix than alternator noise. Sometimes popping will enter through your antenna.

Hissing Noise.
This type of noise is generally from improper set up of your audio components. Possibly your gain structure is not correctly set up. If this happens and the shop that installed your equipment can't solve your problem I strongly recommend a shop that has the proper equipment. Not to worry!. Now do not get me wrong, Some equipment is just noisy in itself, So to remove noise from these pieces of equipment may require additional parts. In summary, Do not settle for any noise in your new audio system.
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Need Bass!

So you want BASS! Because of new materials and technology, we are able to put bass in many automobiles without taking up your entire trunk! I want to integrate into my automobile. If you are looking to add bass to your automobile without giving up any space, you might want to check out our custom installation, and prefabricated, vehicle specific enclosures. If you can not find your car or you need more bass we have the facility to custom fabricate a bass system for you.
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What size wire should I run?

There is much debate over the benefit of certain wiring schemes (oxygen-free, multistranded, braided, twisted, air core, you name it). However, most people do agree that the most important factor in selecting power wire is to use the proper size. Wire is generally rated in size by American Wire Gauge, abbreviated AWG, or commonly just gauge. To determine the correct wire size for your application, you should first determine the maximum current flow through the cable (looking at the amplifier's fuse is a relatively simple and conservative way to do this). Then determine the length of the cable that your will use, and consult the following chart, taken from the IASCA handbook.

You should also consider the installation demands: will you need to run the wire around corners or through doors or into the engine compartment? These sorts of problems in the car audio application require some special care in cable selection. You will want to have cable that is flexible; it should have thick insulation as well, and not melt at low temperatures. You don't want to install wire that is rigid and prone to cracks and cuts, or else the results could literally be explosive.

Again, there is much debate over the benefit of the various schemes that are being used by different manufacturers. In general, however, you will probably want to upgrade your speaker wire from the factory ~20 gauge to something bigger when you upgrade your amplifiers and speakers. In most cases, 16 or 18 gauge should be sufficient, with the possible exception of high-power subwoofers. According to an example by Jerry Williamson, using 18 gauge instead of 12 gauge would only result in a power loss of 0.1dB, which is essentially undetectable by humans. Thus, other factors play more important roles in the selection of speaker wire. One issue is that different wires will have different line capacitances, which could cause the wire to act as a low pass filter. Generally, however, the capacitances involved are so small that this is not a significant problem. Be sure to heed the warnings above regarding cable flexibility and insulation, especially when running wire into doors and other areas with an abundance of sharp metal.

	Length of run (in feet)
Current	0-4	4-7	7-10	10-13	13-16	16-19	19-22	22-28
0-20A	14	12	12	10	10	8	8	8
20-35A	12	10	8	8	6	6	6	4
35-50A	10	8	8	6	6	4	4	4
50-65A	8	8	6	4	4	4	4	2
65-85A	6	6	4	4	2	2	2	0
85-105A	6	6	4	2	2	2	2	0
105-125A	4	4	4	2	2	0	0	0
125-150A	2	2	2	2	0	0	0	00

`A' is for amperes, which is a measurement of current equal to one coulomb of charge per second. You usually speak of positive current - current which flows from the more positive potential to the more negative potential, with respect to some reference point (usually ground, which is designated as zero potential). The electrons in a circuit flow in the opposite direction as the current itself. Ampere is commonly abbreviated as "amp", not to be confused with amplifiers, of course, which are also commonly abbreviated "amp". In computation, the abbreviation for amps is commonly "I".

`V' is for volts, which is a measurement of electric potential. Voltages doesn't "go" or "move", they simply exist as a measurement (like saying that there is one mile between you and some other point).

`DC' is for direct current, which is a type of circuit. In a DC circuit, all of the current always flows in one direction, and so it is important to understand which points are at a high potential and which points are at a low potential. For example, cars are typically 12VDC (twelve volts direct current) systems, and it is important to keep track of which wires in a circuit are attached to the +12V (positive twelve volts) lead of the battery, and which wires are attached to the ground (or "negative") lead of the battery. In reality, car batteries tend to have a potential difference of slightly higher than 12V, and the charging system can produce upwards of 14.5V when the engine is running.

`AC' is for alternating current, which is a type of circuit in which the voltage potential fluctuates so that current can flow in either direction through the circuit. In an AC circuit, it is typically not as important to keep track of which lead is which, which is why you can plug household appliances into an outlet the "wrong way" and still have a functioning device. The speaker portions of an audio system comprise an AC circuit. In certain situations, it is indeed important to understand which lead is "positive" and which lead is "negative" (although these are just reference terms and not technically correct). See below for examples. The voltage of an AC circuit is usually given as the RMS (root mean square) voltage, which, for sinusoidal waves, is simply the peak voltage divided by the square root of two.

`W' is for watts, a measurement of electrical power. One watt is equal to one volt times one amp, or one joule of energy per second. In a DC circuit, the power is calculated as the voltage times the current (P=V x I). In an AC circuit, the RMS power is calculated as the RMS voltage times the RMS current (Prms=Vrms x Irms).

`Hz' is for hertz, a measurement of frequency. One hertz is equal to one inverse second (1/s); that is, one cycle per second, where a cycle is the duration between similar portions of a wave (between two peaks, for instance). Frequency can describe both electrical circuits and sound waves, and sometimes both. For example, if an electrical signal in a speaker circuit is going through one thousand cycles per second (1000Hz, or 1kHz), the speaker will resonate at 1kHz, producing a 1kHz sound wave. The standard range of human hearing is "twenty to twenty", or 20Hz-20kHz, which is three decades (three tenfold changes in frequency) or a little under ten octaves (ten twofold changes in frequency).

`dB' is for decibel, and is a measurement for power ratios. To measure dB, you must always measure with respect to something else. The formula for determining these ratios is P=10^(dB/10), which can be rewritten as dB=10log(P). For example, to gain 3dB of output compared to your current output, you must change your current power by a factor of 10^(3/10) = 10^0.3 = 2.00 (that is, double your power). The other way around, if you triple your power (say, from 20W to 60W) and want to know the corresponding change in dB, it is dB=10log(60/20)=4.77 (that is, an increase of 4.77dB). If you know your logarithms, you know that a negative number simply inverts your answer, so that 3dB corresponding to double power is the same as -3dB corresponding to half power. There are several other dB formulas; for instance, the voltage measurement is dB=20log(V). For example, a doubling of voltage produces 20log2 = 6.0dB more output, which makes sense since power is proportional to the square of voltage, so a doubling in voltage produces a quadrupling in power.

`SPL' is for sound pressure level and is similar to dB. SPL measurements are also ratios, but are always measured relative to a constant. This constant is 0dB which is defined as the smallest level of sound pressure that the human ear can detect. 0dB is equal to 10^-12 (ten to the negative twelfth power) W/m^2 (watts per square meter). As such, when a speaker is rated to produce 92dB at 1m when given 1W (92dB/Wm), you know that they mean that it is 92dB louder than 10^-12W/m^2. You also know than if you double the power (from 1W to 2W), you add 3dB, so it will produce 95dB at 1m with 2W, 98dB at 1m with 4W, 101dB at 1m with 8W, etc.

`THD' is for total harmonic distortion, and is a measure of the how much a certain device may distort a signal. These figures are usually given as percentages. It is believed that THD figures below approximately 0.1% are inaudible. However, it should be realized that distortion adds, so that if a head unit, equalizer, signal processor, crossover, amplifier and speaker are all rated at "no greater than 0.1%THD", together, they could produce 0.6%THD, which could be noticeable in the output.

An Ohm is a measure of resistance and impedance, which tells you how much a device will resist the flow of current in a circuit. For example, if the same signal at the same voltage is sent into two speakers - one of which is nominally rated at 4 ohms of impedance, the other at 8 ohms impedance - twice as much current will flow through the 4 ohm speaker as the 8 ohm speaker, which requires twice as much power, since power is proportional to current.

The soundstage is the position (front/back and high/low) that the music appears to be coming from, as well as the depth of the stage. A car with speakers only in the front will likely have a forward soundstage, but may not have enough rear fill to make the music seem live. A car with both front and rear speakers may have anything from a forward to a rear soundstage, with an accompanying fill from the softer drivers depending on the relative power levels and the frequencies reproduced. The high/low position of the soundstage is generally only obvious in a car with a forward soundstage. The music may seem to be originating in the foot wells, the dash, or out on the hood, depending on how the drivers interact with the environment.

The stereo image is the width and definition of the soundstage. Instruments should appear to be coming from their correct positions, relative to the recording. The position of the instruments should be solid and easily identifiable, not changing with varying frequencies. A car can image perfectly with only a center-mounted mono speaker, but the stereo placement of the music will be absent.

The frequency response of a device is the range of frequencies over which that device can perform in some fashion. The action is specific to the device in question. For example, the frequency response of the human ear is around 20Hz-20kHz, which is the range of frequencies which can be resolved by the eardrum. The frequency response of an amplifier may be 50Hz-40kHz, and that of a certain speaker may be 120Hz-17kHz. In the car audio world, frequency responses should usually be given with a power ratio range as well, such as (in the case of the speaker) 120Hz-17kHz +/-3dB. What this means is that given an input signal anywhere from 120Hz to 17kHz, the output signal is guaranteed to be within an "envelope" that is 6dB tall. Typically the extreme ends of the frequency range are the hardest to reproduce, so in this example, the 120Hz and 17kHz points may be referred to as the "-3dB points" of the amplifier. When no dB range is given with a frequency response specification, it can sometimes be assumed to be +/-3dB.

• What does a voice coil do?
• What is a dual voice coil speaker?
• What is the advantage of dual voice coils?
• What happens when you run different signals into each voice coil of a dual voice coil speaker?
• Whats the point of a dual voice coil speaker if I have a mono-bridgeable amp?
• Does It Matter How The Voice Coils Are Wired To Each Other?
• How are dual voice coil speakers rated for power handling?
• Sub-30Hz Behavior
• 30-80Hz Behavior
• Transient Response
• Efficiency
• Midbass Transition
• There is No Free Lunch
• The Sealed Box
• The Ported Box
• The Bandpass Box
• Isobarik Loading
• Cone Area(Sd) and Linear Excursion Capability(Xmax)
• A Head to Head Comparison
• So How Do They Work
• The Sealed Chamber
• The Ported Chamber
• The Port
• "Universal" Bandpass Designs
• Port Basics
• Multiple Ports
• Enclosure Shapes
• Bracing and Strength
• A True Story
• The Jumbo Test
• Sealing the Box

What does a voice coil do?
The vast majority of speakers available on the market today are referred to as electrodynamic. All electrodynamic speakers share a fundamental aspect of operation: the reaction of a fixed magnetic field against a changing one. In most electrodynamic speakers, a voice coil, which is a single coiled length of wire wrapped around a cylinder called a former, produces the changing magnetic field when alternating current from the amplifier flows through it. This current is an electrical representation of the sound that was produced by the musicians in the recording studio and causes the voice coil (and therefore the cone or dome attached to it) to react against the fixed magnetic field produced by the speakers fixed magnet. A positive pulse should cause the cone to move outward and a negative pulse should cause the cone to move inward. When the cone moves as a result of being propelled by the voice coil, it produces the changes in the air pressure of the listening environment that we perceive as sound.

What is a dual voice coil speaker?
A dual voice coil speaker is simply one in which two separate lengths of wire are wound together around the same former and terminated independently. Except for some exotic exceptions, both voice coils have the same number of turns and length of wire, resulting in identical electrical characteristics. In most cases, one coil is wound onto the former first, and the second one is wound over the first one. Naturally, it is more expensive to wind and terminate dual voice coils and you will typically pay a small premium compared to a similar single voice coil speaker. So what do you get for the extra few bucks? Do dual voice coils offer a performance advantage? Not really. Do they offer any benefit over a conventional single voice coil design? Definitely.

What is the advantage of dual voice coils?
The primary advantage of the dual voice coil speaker is wiring flexibility. A single dual voice coil driver offers the user three hookup choices...parallel, series and independent. In a parallel hook-up the drivers impedance will be half that of each individual coil (a dual 4 ohm speaker would be a 2 ohm speaker in parallel.) A series hook-up results in twice the impedance of each single coil (a dual 4 ohm speaker results in 8 ohms if its coils are wired in series.) Finally, you can wire each voice coil to a separate channel of your amplifier, which can be useful if your amplifier is not mono-bridgeable or if you are bridging a four channel amplifier down to two channels to run your sub.The independent wiring application is the one that brought about the need for dual voice coil speakers in home audio. Unlike most good car amplifiers, home amplifiers and receivers are typically not mono-bridgeable. For this reason, dual voice coil woofers were developed so that a subwoofer or center speaker could be driven from the left and right channels of the average stereo home amp/receiver. Since sub-bass frequencies are hard to localize, the dual voice coil subwoofer allowed sub-bass reinforcement within one cabinet and one speaker. This cabinet could be placed inconspicuously in a corner or along a wall of the listening room, with the obvious benefits being space-efficiency and lower cost than two independent bass cabinets or a larger cabinet with two subs in it. Many popular home subwoofer / satellite speaker systems still use this basic configuration.

What happens when you run different signals into each voice coil of a dual voice coil speaker?
Essentially, if there is any difference between the signals driving each coil at any given point in time at a given frequency, the voice coils will either fight each other or help each other, depending on the phase relationship of the two signals at that frequency. This is not the same thing as bridging an amplifier and can create undesirable non-linearities and distortion because different input signals at each voice coil create shifts in the speakers electrical parameters.For this reason, it is advisable to mono-bridge the amplifier whenever possible and connect the voice coils of the dual voice coil speaker together in parallel or series. If a dual voice coil subwoofer must be wired to two independent channels, the inputs to both channels should ideally be the same (summed mono) and every effort should be made to match the gains of both channels as closely as possible.

Whats the point of a dual voice coil speaker if I have a mono-bridgeable amp?
Because most decent car audio amplifiers are mono-bridgeable, it is just as easy to run a single voice coil sub in mono as it is to run a dual voice coil sub in mono. Where a dual voice coil subwoofer has an advantage is in giving the user greater wiring flexibility while avoiding speaker-to-speaker series connections. As an example, a dual 4 ohm subwoofer with its coils in series behaves as an 8 ohm speaker. The same speaker, with its voice coils wired in parallel, behaves as a 2 ohm speaker. If your amplifier is designed to run at higher impedances, you would wire the coils in series. If your amplifier is a high-current design which produces optimum performance at lower impedances, the parallel connection makes more sense. The benefits of the dual voice coil design are even more apparent with multiple subwoofer installations. Lets say you are using a five-channel amplifier with a single mono subwoofer channel that produces optimum power at 4 Ohms mono. You want to run four subwoofers in your enclosure. You have a choice between using four 4 ohm single voice coil subs or four dual 8 ohm voice coil subs. With the single voice coil speakers, the only way to achieve a 4 ohm mono load is to wire them in a series-parallel arrangement. The problem with this, is that it is less desirable to wire subwoofers in series to each other (as opposed to parallel). Because of slight and unavoidable differences between speakers and because of the high likelihood of uneven loading between different speakers in a car, there will be slight differences in the mechanical behavior of the two speakers in series. These differences in movement result in induced voltage (called back EMF) being created by the speakers across the series connection. This effect causes a problem when two speakers which behave differently are connected in series because the speakers can modulate each other (cause each other to move), resulting in distortion. The problem becomes more serious as more speakers are connected in series. A good experiment to show the effect of back EMF is the following: connect four speakers in series and short the positive and negative input leads of the series circuit. Push down on one cone with your hand; you will notice that the three other speakers will move in the opposite direction of the one you are pushing. Now, reconnect the speakers in parallel, short the inputs and push down on one cone. The speakers will not modulate each other because each one is shorted directly. Now, lets get back to our original scenario of wanting to hook up four ten inch woofers to a 4 ohm mono load. The dual 8 ohm voice coil subwoofers allow us to do this without worrying about back EMF-induced non-linearities. Back EMF modulation is not a concern when the voice coils of a dual voice coil speaker are wired in series to each other because the coils are physically coupled on one moving mass. Therefore, they cannot possibly modulate each other because they cannot move differently. With this in mind, we can take our dual 8 ohm voice coil drivers and hook up the voice coils of each speaker to each other in series, making each one a 16 ohm driver. Then, we simply hook up the four 16 ohm speakers in parallel to the amp, giving us our 4 ohm mono load without a speaker-to-speaker series connection and any of the negative effects described above.

Does It Matter How The Voice Coils Are Wired To Each Other?
A dual voice coil speaker will behave exactly the same way whether it is wired with its coils in series or parallel. The only thing that changes is the impedance that the amplifier sees. This means that enclosure calculations are constant for dual voice coil woofers no matter how the coils are connected to each other, as long as both are connected.A common misconception with regard to dual voice coil speakers is the assumption that nothing changes if you power only one of the voice coils. With only one coil hooked up, a dual voice coil speaker will suffer a loss in reference efficiency of about 3dB (only half the coil windings are being energized) as well as a significant shift in its Thiele/Small parameters. This renders any enclosure calculations inaccurate unless you remeasure the speakers parameters with only one coil hooked up. Failure to account for the different parameters of a dual voice coil speaker with only one coil powered can result in very poor performance.

How are dual voice coil speakers rated for power handling?
A dual voice coil speakers power handling is typically specified by manufacturers for the whole speaker. This means that a 250 Watt dual voice coil driver is designed to handle a total of 250 Watts whether the coils are wired independently, in series or in parallel.

Sub-30Hz Behavior
Sealed box designs and single-reflex bandpasses are much better at controlling excursion at extremely low-frequencies (below 30Hz.) For this reason, they can usually handle more power in these frequency ranges than ported designs and dual-reflex bandpass designs which makes them less prone to low-frequency induced speaker damage. At frequencies below the tuning frequency of the port, a woofer in a ported box (or a dual-reflex bandpass) starts to de-couple. This means that the controlling function of the enclosure begins to disappear. The collapse is gradual rather than immediate, but at some point below the tuning of the port, the speaker behaves as if it were operating without an enclosure and suffers from potentially damaging over-excursion. (This is why it is a good practice to use a sub-sonic filter when running a ported enclosure or a dual-reflex bandpass. Some high-quality electronic crossovers like the AudioControl 4XS incorporate a programmable subsonic filter circuit.) Related to the loss of enclosure damping, ported and dual-reflex bandpass designs also exhibit higher distortion levels at very low frequencies than sealed or single-reflex bandpass designs. The importance of this is questionable, however, since little program material extends to below 30Hz. Sealed enclosures and single-reflex bandpass designs have a rather shallow low-frequency roll-off rate of around 12dB/octave, whereas ported enclosures and dual-reflex bandpasses typically exhibit 18- 24dB/octave roll-off. For this reason, sealed enclosures and single-reflex bandpass boxes can have much higher -3dB points (the frequency at which the output dips 3dB below the reference efficiency of the speaker) than ported designs while still producing very good ultra-low frequency output.

30-80Hz Behavior
This is the frequency range that is most important in that it encompasses the vast majority of low-frequency information present in music. Serious audiophiles assign much more importance to good performance in this range than in the extreme low-frequency range. At moderate power levels all of these enclosure types exhibit pretty decent manners. The ported box and the bandpass designs produce less distortion than the sealed box, but the difference is marginal. At higher power levels things change considerably. The dual-reflex bandpass, due to the fact that its ports control cone motion over a wider range of frequencies, produces the least distortion and exhibits the best power-handling characteristics. The ported enclosure and the single-reflex bandpass also do a very good job producing high-levels of undistorted bass output, again due to reduced cone motion in this frequency range. Bringing up the rear in this category is the sealed enclosure, which produces higher levels of distortion at high power levels. There is a common misconception that ported designs produce more distortion than sealed boxes. As you can see this is not entirely accurate; it depends on the frequency and the power level.

Transient Response
Transient response refers to the ability of the subwoofer system to reproduce quick changes (transients) in the program material accurately. This is often interpreted as "tightness" or "looseness" which is maybe a dangerous terminology since many people are more influenced by tonal characteristics when asked to qualify the "tightness" of the bass. Transient response is actually a function of accuracy in relation to time rather than frequency. In music, sounds like drum strikes and quick bass guitar pulses are good tests of a subwoofer system's transient performance. A system with good transient response will reproduce these sounds with clear, "tight" definition. A system with poor transient response tends to blur these sounds over time, due to the speaker's inability to stop and start quickly enough to react to the signal accurately. It is generally accepted that an optimized sealed enclosure exhibits the best transient response characteristics. The control provided by the air-spring in a good sealed system contribute to generally outstanding transient behavior (at very high power levels, the increased distortion can overshadow this advantage, however.) A ported enclosure can also achieve good transient behavior but never as good as an optimized sealed enclosure. It is possible, however, for a well-designed ported enclosure to have better transient response characteristics than sealed enclosures with higher Qtc's (above 1.0.) The specific alignment of the sealed and ported enclosures plays a huge role in determining the transient characteristics of each individual subwoofer system. Single-Reflex bandpass designs can also have good transient characteristics if their bandwidth is fairly narrow, but again, not as good as an optimized sealed enclosure. As the bandwidth becomes wider, their transient response can degrade considerably. Dual-Reflex designs generally exhibit inferior transient response characteristics when compared to the other designs. As with single-reflex designs, narrower bandwidths produce better transient performance than wider ones.

Efficiency
The term "efficiency" refers to the ability of a speaker system to convert electrical energy (power from the amplifier) into acoustic output. Consequently, it also serves as an indication of which system will produce the loudest possible output given the same size amplifier (assuming they can all handle the power.) For the purposes of this comparison, we are looking at efficiency in the 40-80Hz octave. Generally speaking, the most efficient enclosures are the two narrow-bandwidth bandpass designs with the dual-reflex version having a slight edge. Next in line, the wide-bandwidth dual-reflex and the ported enclosure exhibit very good efficiency as well. The sealed enclosure and the wide-bandwidth single-reflex bandpass are the least efficient designs.

Midbass Transition
For sub-bass to sound natural, the system must have good midbass capability as well. These two are interrelated because harmonic components of the sounds produced by instruments that play in the sub-bass range must be accurately reproduced in the mid-bass range for a system to sound accurate. In car audio, we normally don't have the luxury of using very large drivers to reproduce midbass. For this reason, the ability of a subwoofer system to smoothly transition to the mid-bass region becomes very important to achieving top-notch fidelity. The sealed and ported enclosures, because the speakers play directly into the listening environment usually produce the smoothest midbass transition. Wide bandwidth bandpass designs are a little more ragged, but still deliver good midbass reinforcement. The narrow bandwidth bandpass designs can create serious problems because their high-frequency roll-off can begin as low as 75-80 Hz and the amplitude of their response peaks is very high, which necessitates the use of larger, very capable midbass speakers in order to blend smoothly with the sub-bass.

There is No Free Lunch
As you can see by the comparison, no enclosure design is superior in all respects. They all have advantages and disadvantages. Analyzing the characteristics of each enclosure type will help you decide which enclosure type is right for your application. An informed decision involves an analysis of the following factors: the space that you want to make available in your car for the enclosure, your performance expectations (loudness, tonal qualities, etc,) the amount of amplifier power you will be using, and of course, your budget. Top-notch car audio specialists will weigh all the factors and consider all enclosure types before recommending a subwoofer system. Many will even show you specific data to support their suggestions. Remember that the information presented here assumes that each enclosure type has been properly designed and executed. This means that the speaker and the enclosure are carefully matched as a system. The skills of a competent designer, installer and cabinet builder are every bit as important to the end result as the design of the box or the type of woofers that you buy. Despite the very general scope of this piece, we hope it leaves you with a better understanding of subwoofer systems. At the very least, we hope that the next time you hear someone say "whatever you do, don't port the box" or "sealed boxes don't sound good," you will ask them to thoroughly explain their position. It could be amusing.

The Sealed Box
(aka: air-suspension, acoustic-suspension) enclosure is a classic box design. Patented in 1949 by Harry Olson and popularized in the 1950's by Acoustic Research, this design has stood the test of time and has been adopted by many home and car audio companies. In a sealed enclosure, the woofer is tightly controlled by a trapped volume of air in the enclosure which acts as a spring (hence the name "air-suspension.") The woofer must literally pull the air with it as it moves outward thus decreasing the air pressure inside the box and compress the air inside the box when it moves inward, which increases the air pressure inside the box. Since the air pressure inside the box seeks to equal the barometric pressure of the atmosphere, it acts as a controlling force over the motion of the speaker. The more the speaker moves inward or outward, the greater the pressure exerted by the air-spring of the sealed enclosure in the opposite direction. The relationship between the parameters of the speaker being used and the volume of air inside the enclosure dictates the performance of the sealed subwoofer system. By making the box larger, the air spring limits cone motion less and allows the system to play lower and with flatter overall response (lower Qtc) at the expense of power handling. If you go too large, however, you begin to lose efficiency in order to gain the additional low frequency extension. By making the box smaller, the air spring exerts more control and limits cone motion at low frequencies which increases power handling but does not let the system play as low and produces a more peaked response (higher Qtc.) For any speaker competently designed for sealed box applications there is a range of enclosure volumes that will produce good high-fidelity sound. Changing the enclosure volume within that range can fine-tune the response to suit the tastes of the listener and/or the acoustic properties of the vehicle.

The Ported Box
(aka: Bass-Reflex, Vented) have actually been around longer than sealed designs. The ported enclosure was patented in 1932 by A.C. Thuras. Further development since then has defined the behavior of ported systems much more precisely. A.N. Thiele and Richard Small are generally credited with having done the most definitive work in this area, which is why enclosure/speaker parameters are commonly referred to as Thiele-Small parameters. The coupling of a port or duct to the air inside the enclosure allows the subwoofer system to take advantage of the work being done by the rear of the woofer cone to reinforce the low-frequency response. The resonant characteristics of the column of air in a port, when installed in a given box, are adjusted by altering its resistance to motion, which is accomplished by changing the dimensions of the port. In some designs, instead of a port, a speaker cone with no motor assembly or a flat diaphragm is used to achieve the same effect. This is known as a passive radiator. The resonance of a passive radiator system can be adjusted by altering the radiator's surface area, mass and compliance (stiffness of suspension.) In a ported enclosure, there is a delicate relationship between the volume of air in the box, the resonant effect of the port, and the parameters of the speaker being used. When these three factors are correctly integrated, the rear output wave of the speaker is delayed just enough so that when it comes out of the port, it is in relative phase with the wave being produced by the front of the speaker. The result is constructive output from the port limited to a desired low- frequency range. This low-frequency reinforcement is one of the big advantages of a well-designed ported system. Using the work of the rear of the cone in a constructive manner means that a gain in efficiency of about 3dB over a broad band in the sub-bass range can be achieved as compared to a sealed enclosure using the same woofer. The other big advantage is that the interaction of the port, the enclosure and the speaker's resonant characteristics also reduces cone motion and, therefore, distortion at higher volume levels in the frequency range controlled by the port. The down side is that at frequencies below the tuning of the port, the speaker gradually begins to act as if it were not enclosed at all (more on this later.) The increased output combined with reduced distortion in the "meat" of the bass range (35-60Hz) is a big reason why many home speakers and high-power sound-reinforcement systems use ported enclosures for low-frequency reproduction. The vast majority of recording studios also use ported enclosures as monitors for the same reasons. The rules governing the behavior and proper design of ported speaker systems are considerably more complex than those for sealed enclosures. For this reason, it is a good practice to follow the advice of the speaker manufacturer or an experienced enclosure designer when it comes to designing a ported system. It is very easy to screw up a ported box if you just guess at the size and length of the port or the tuning frequency for the box. Not only will a poorly designed box sound bad, but it can easily damage the speaker if it is played hard.

The Bandpass Box
These enclosures seem to be the latest rage in the car audio world. It would probably surprise many people to know that these designs have been around for many years. The first patent for a bandpass enclosure was filed in 1934 by Andre d'Alton. In the last ten years, interest has been renewed in these enclosure designs and substantial strides have been made in defining their behavior. Many home sub/satellite speaker systems currently use bandpass designs for low-frequency reproduction. Designs from Bose, KEF, AR, and many others have become very popular in home audio circles. In a bandpass box design, the woofer no longer plays directly into the listening area. Instead, the entire output of the subwoofer system is produced through the port or ports. In a conventional sealed or ported subwoofer system the low-frequency extension is controlled by the interaction of the speaker and the enclosure design, but the high frequency response is a result of the speaker's natural frequency response capability (unless limited by a crossover.) In a bandpass enclosure, the front of the speaker fires into a chamber which is tuned by a port. This ported front chamber acts as a low-pass filter which acoustically limits the high- frequency response of the subwoofer system. The name "bandpass" is really pretty descriptive in that it refers to the fact that the enclosure will only allow a certain frequency "band" (range) to "pass" into the listening environment. So what? Couldn't the same thing be accomplished by placing a low pass crossover on the subwoofer system? Yes, it could, but a bandpass enclosure can produce significant performance benefits in terms of efficiency and/or deep bass extension that would not be possible in conventional designs of equal size. By adjusting the volumes of the front and rear chambers and the tuning of the port or ports, significant performance trade-offs can be created. When box parameters are adjusted for a narrower bandwidth, the efficiency of the subwoofer system within that bandwidth increases and can reach gains of up to 8dB (sometimes even higher.) As box parameters are adjusted for wider bandwidths, very impressive low-frequency extension can be produced from extremely compact enclosures at the expense of efficiency and good transient response. Intermediate bandwidths can also be designed which create a compromise between all these characteristics. As if that is not confusing enough, within each bandwidth range, the designer can also manipulate box parameters to shift the range of operation up or down the sub-bass range which also has an effect on efficiency. As you can see, bandpass enclosures can have very different sound characteristics based on the designer's choice of box parameters. As such, it is not always possible to make blanket statements as to the performance benefits and drawbacks of bandpass enclosures in general. One characteristic of bandpass enclosures which is universal is that they exert greater control over cone motion over a wider frequency band than conventional designs. Due to controlled, rapidly changing air pressure on either side of the woofer, the woofer is capable of producing high levels of acoustic output without physically moving very much. This means that the woofer is less likely to encounter excursion limits in the main part of the sub-bass range. However, just because the cone isn't moving as much doesn't mean that the speaker's motor assembly isn't still trying to drive the cone hard; it just means that the speaker cone is encountering resistance to motion. This resistance can be very hard on speakers, especially when crazy car audiophiles are at the controls. The conflict between the force generated by the motor assembly and the air pressure in the enclosure can impose extreme stress on the glue joints and suspensions of the woofers. You can literally tear a speaker apart in a bandpass enclosure if you apply too much power. Because the speaker is not moving as much and because noises are masked by the front chamber, it is also very difficult to hear when a woofer is in serious trouble. Many people have been known to crank bandpass enclosures up and blow the speaker to bits within a few minutes because they did not realize that the speaker was having a heart attack. Choosing the right amount of power and carefully setting amplifier gains is very important in order to ensure long- term reliability. Bandpass enclosures can be divided into two basic types: single- reflex and dual-reflex. In a single-reflex design, the rear chamber is sealed and the front chamber is ported. In a dual-reflex design, both front and rear chambers are ported into the listening area. A variation of the dual-reflex and single-reflex, known as "series-tuned," has a port which connects the rear and front chambers. The differences between single-reflex and dual-reflex bandpasses are similar to the differences between sealed and ported enclosures. A single-reflex typically exhibits a shallower low-frequency roll- off rate (approximately12dB/octave) and better transient response. A dual-reflex is more efficient and controls cone-motion over a wider range but typically has a sharper (18-24dB/octave) low- frequency roll-off. Because of the difference in low-frequency roll- off rates, a dual-reflex usually has to be larger in size to produce the same low-frequency extension as a single-reflex design. As compared to more conventional enclosure designs, bandpass enclosures are very complex to design and build. The rules governing the performance of bandpass enclosures leave no room for error. Slight volume miscalculations or sloppy construction can turn a good design into a poor-performing box. Integrating the proper size port or ports can be extremely challenging and often renders designs that looked great on paper completely impractical. The design of these boxes should definitely be left to people with extensive enclosure- building experience.

Isobarik Loading
Isobarik loading has become pretty popular for car audio use in the last few years. Again this is not a new concept, having been originally introduced by Harry Olson in the early 1950's. Technically, "isobarik" is not really an enclosure type; it is a loading method. This loading method involves the coupling of two woofers to work together as one unit. This is typically accomplished either by placing two woofers face to face or by coupling two woofers with a small chamber. The result of coupling the two speakers is that the coupled pair (iso-group) can now produce the same frequency response in half the box volume as a single speaker of the same type would require. For example, if a speaker is optimized for performance in a 1 cu.ft. sealed enclosure, one iso-group of the same speakers can achieve the same low frequency extension and overall response characteristics in a 0.5 cu.ft. sealed enclosure.

There is, of course, a penalty involved. Whenever you use isobarik loading, you are sacrificing 3dB of efficiency compared to a single driver in twice the air space. In practical terms, this is not usually a big deal since the powerhandling is doubled (two speakers instead of one) and the impedance is typically cut in half if we parallel the two speakers (twice the power, assuming the amplifier can deliver the necessary current.) The end result is about the same output as the single driver in the bigger box but at twice the amplifier power (and twice the speaker cost.)

Isobarik loading can be used within any enclosure type, including bandpass designs. The ported and bandpass isobarik designs can be difficult to design and build due to very small enclosures with large port requirements. Isobarik bandpass designs, in particular, can be literally impossible to build with certain speakers. There are some things to look out for with each type of isobarik design, such as mechanical noise and uneven heat dissipation which can present potential sound quality and reliability problems. All the methods which involve opposite cone motion require that the speakers be wired in reverse polarity relative to each other. These designs also provide a performance advantage in that their opposed cone motion averages out suspension non-linearities (differences in inward and outward suspension control,) which reduces distortion.

If you are strapped for space and can afford the extra speakers and more complex enclosure, the ability to have a compact subwoofer system with no real sacrifice in performance is well worth the extra effort and expense. On the other hand, if you have a lot of space and are looking to get the maximum amount of output without sacrificing sound quality, using multiple iso-groups can give you the best cone area/box volume ratio while still retaining good fidelity.

There are several different isobarik loading configurations to choose from, so which is the right one for you? This is a loaded question that quite often has no clear-cut answer. We do not recommend the use of isobarik loading to just anyone...we do not believe that the more subwoofers that you can shoe-horn into any given volume of air produces the best results. In fact, we're sure you've seen plenty of examples to support this line of thinking. So, we only recommend that you consider using isobarik loading if one or more of the following conditions holds true:

1.	You have more subwoofers than space to properly utilize them. The first condition is a common occurrence if you decide to change vehicles and keep the sound system or when you out-grow that desire to consume every cubic inch of your vehicle's storage space with subwoofers. You don't want to sell your equipment because you've fallen in love with it and so you begin to look for any way you can to keep it all (after all it *does* sound impressive to say "Hey, I've got 16 eight-inch woofers in the back of my Jetta"). Since iso-loading allows us to use the same number of pistons (a piston is one air-moving unit--think of it as you would the pistons in your engine) in half the space, an isobarik loading scheme might prove very attractive here.
2.	You have more power than you know what to do with. The second condition may hold true if you've suddenly acquired a 500W amplifier for your subs and your poor little 125W subwoofers just cannot handle that kind of power. In this case, you might switch from one 125W driver in a .875 cubic foot enclosure to 4 of the same drivers (2 isogroups) in the same sized box. This would bump your effective power handling up to 500W -- which, assuming the box was built according to the manufacturer's specifications, should handle the added power just fine.
3.	You have more money than you know what to do with The third condition is pretty self-explanatory...you've got money burning a hole in your pocket and rather than purchase a new set of tires to replace your balding Dunlops, you decide to splurge on stereo equipment and try to put as much stuff into your car as possible.
4.	You are a golden-eared tweak who can detect subtle non-linearities in your sub-bass. The fourth and final condition is a subtlety that most probably won't be familiar with. One of the nice little side-effects of using a face-to-face (or back-to-back) loading arrangements is the cancellation or driver non-linearities. This will be explored a little later though.

In short, if you're out to try isobarik loads for reasons such as "It looks cool" or "I heard that they 'hit harder' (or any one of a number of other colloquialisms) than anything else" then your money would more likely be better spent on something else a little less costly. Of course, we wholeheartedly support those who love to try new things just for the sake of trying out new things or to further their understanding of various subwoofer systems so don't take what you read here as discouragement...just fair warning of what to expect.

With all this said and done, let's explore some of the advantages and disadvantages of the "piggy-back" tunnel load, back-to-back tunnel load, the planar load and the "clamshell" isobarik configurations.

The piggy-back tunnel-loaded isobarik configuration is probably the second most popular isobarik arrangement in use today (the first being the face-to-face or "clamshell" configuration. It is cosmetically easier to integrate into the vehicle (as it does not have any potentially ugly subwoofer baskets protruding into the vehicle) but unfortunately this aesthetic benefit is offset by several important detractors:


1.The coupled air between the two drivers adds to the moving mass of the system and thus results in a less than optimal coupling between the drivers. Remember that the idea is to get these two subwoofers to act as one driver, and by adding a springy mass between them this ideal is somewhat compromised. Some might find that this leads to a beneficial lowering of the system Q (when the volume indicated in blue in the picture is sealed) but more often than not this effect is undesirable as it makes response predictions more difficult.
2.The coupling chamber negates one of the primary benefits of isoloading--small enclosure size. By the time we account for the displacement of this coupling tunnel in determining the gross volume of the blue chamber, the enclosure starts to approach the volume required by a single conventionally mounted driver.
3.Since the drivers are both firing in the same direction, there we do not reap the benefit of cancelled driver non-linearities as we would with a design implementing a push-pull configuration.
4.The driver whose magnet structure is housed in the coupling tunnel is in a highly unfavorable cooling environment and will be subject to power compression at lower levels. Basically, the drivers will be more or less equal performers at first, but as things start to heat up and the impedance of the front driver rises due to rising voice coil temperatures, the drivers start to fight each other to some degree rather than complement one another. This results in increasingly non-linear behavior with possible unpleasant audible side effects (e.g. sloppy transient behavior).

In essence, this configuration is more of a cosmetic "oh neat-o" design more than anything else, and we recommend that it not be used, especially for high-powered applications where the thermal power handling of the drivers would be called into question.

This design was thought up by someone who wanted to reap the advantages of canceling driver non-linearities without having to resort to the "clamshell" loading and it's inherent cosmetic problem (namely that of hiding an exposed subwoofer basket). This design, like its cousin the tunnel-loaded isobarik also has several detractions that make it an undesirable choice:
1.It shares the same problems with the added springy mass of air that couples the two drivers but with the back-to-back isoload, this problem is made even worse by the fact that the coupling chamber is now even larger, adding more moving mass and springiness over the tunnel-load and thus making frequency response predictions even more difficult.
2.The increased coupling chamber (pink volume) means that the blue volume and thus the entire enclosure must be even larger, even more closely approaching the volume of a conventionally loaded single subwoofer. In a home this might not be a problem, but in the vehicle where space is at a premium, this is a definite disadvantage!
3.Now that both magnet structures are in identical cooling environments, they will more closely track each other's performance but unfortunately, now we have two heat dissipating structures in the same tiny enclosure which will greatly reduce the thermal power handling of both drivers, not to mention the fact that as the air heats up, it expands thus pushing each of the subs outward and thus further limiting output by reducing each driver's potential excursion! While the original creator of this design should be given a pat on the back for creativity, it is definitely not an alignment that we recommend under any circumstances.

This alignment is similar in concept to a "clamshell" or face-to-face isobarik and will behave in a similar manner, but it also has some of the detractions of the tunnel-loaded isobarik that are associated with having a coupling chamber between the two drivers. It is somewhat space-inefficient in that you are giving up usable space behind the outside driver, but it produces a very interesting visual effect if you put a sheet of plexiglass in front of the speakers.

Construction tips
The loading chamber (indicated by the pink shaded region) should be between 0.75" and 1" deep. You should also do everything possible to minimize the surface area of the loading chamber since any trapped air in it essentially becomes part of the moving mass of the speaker system.

If you are more industrious, you can round off the ends of the loading chamber as seen in the diagram at right.

If you intend to utilize a ported design, port lengths can be rather large. This is common with single iso-group enclosures because of the small box volumes. For this reason, you may want to fire the port as shown in the diagram below. This should allow you to extend part of the port tube outside the enclosure without it being visible.

The face-to-face or "clamshell" configuration as it commonly called is the most compact and therefore the most practical isobaric loading method to use considering the tight confines of the average automobile. This configuration also provides the beneficial side effect of canceling driver non-linearities.

If there is one recurring theme in engineering it's that Mother Nature is lazy. She has made it a law that anything at rest wants to stay at rest and similarly anything in motion would much rather stay in motion in a straight line. Such is life in general and a speaker's dynamics are no exception. It's called the law of inertia and there is no escape.

When a subwoofer does its job, it is called upon to compress and rarefy the air in the listening environment many times per second and more often than not is required to do so over great distances. This places a great strain on the cone itself as it fights to retain its shape in the face of intense acceleration and deceleration. Ideally, a speaker's cone would be infinitely rigid and wouldn't deform under any circumstances, but obviously a perfect world this is not so we have to deal with the consequences of fighting Mother Nature.

As the cone pushes outward, it is somewhat flattened out as it attempts to kick-start the air in front of it into motion. Likewise, the cone is deformed the other way when the cone returns and attempts to compress the air in the subwoofer enclosure. The extent of this deformation is a function of the cone's geometry, construction and the amount of power with which the subwoofer is driven. A good engineer will design his cones such that this effect is minimized but there is only so much engineers can do if he's to make an affordable product.

Construction tips
It is important to note that when mounting the drivers to each other and then to the enclosure, a separator of some sorts must be used to space the drivers apart. If the drivers are not physically separated, their surrounds may rub against each other which will lead to premature failure of the driver. We recommend the use of a 5/8" thick ring of Medium Density Fiberboard (MDF) with appropriately spaced holes to pass the mounting bolts/screws through.

Lay the bottom driver in the box after wiring it up (this driver should have its positive leads wired to the positive terminal(s) of the amplifier and its negative terminals wired to the amp's negative terminal(s). Lay the MDF Iso-Ring atop this driver, invert the second driver over the first, line up the mounting holes, and screw the whole assembly to the enclosure.

Assuming your driver's gaskets are clean and unscathed, and the MDF ring is equally smooth on its contact surfaces, no other sealing agents need be used to assure a good air-tight seal at the driver/ring interfaces. If you decide to use silicon or some other sealant, be prepared to go through one hell of a fight if you need for some reason to disassemble the isogroup!

Some prefer to mount one driver inside the box and its partner atop the box thus using the enclosure wall itself as the spacing mechanism, but we have found that this makes driver servicing unnecessarily difficult--rather than just undo the eight bolts/screws using our suggested mounting method, one would have to have someone come in from a removable panel on the other side of the enclosure and hold up the other driver. In short, mounting everything from the outside makes much more sense and is infinitely easier to service.

When all is securely mounted, wire the outer subwoofers (the ones with their magnets exposed) such that the (+) on the speaker is wired to the (-) on the amplifier and visa versa. This will assure that both drivers are moving in the same direction when a voltage is applied. If you hook everything up and get no bass from your new isoload, chances are that either a lead fell off inside the enclosure or you've got a driver's polarity reversed....double check everything before powering up.

A lot of misinformation has been spread in the industry with regard to the issues that affect the SPL capability of a speaker system. The fact is that the factors which control SPL capability are very defined and simple:

Cone Area(Sd) and Linear Excursion Capability(Xmax)
The ability of the speaker to displace air in the listening environment is a function of the two factors above and is very similar to how the bore and stroke of a piston in an engine determine the displacement of the cylinder.

It is commonly understood that larger diameter woofers are louder than smaller diameter woofers (assuming equal excursion). In car audio, however, it is not often possible to fit large drivers into vehicles without a substantial sacrifice in usable space. For this reason, car audio subwoofer performance benefits greatly from maximizing displacement through increased excursion capability within a given frame size.

The specification which indicates linear excursion capability is "Xmax". This spec designates the amount of cone travel in one direction while maintaining linear motor behavior and is usually listed in inches or millimeters.

Linear motor behavior means that there is always a constant length of voice coil winding in the magnetic gap of the motor structure. If the voice coil is pushed beyond the linear limit, the output becomes more distorted and, if pushed too far, the speaker can suffer a failure of its suspension components or voice coil windings. Well-designed woofers can be played beyond their Xmax to some extent without audible low-frequency distortion or damage. The design of the suspension plays a large role in determining how acceptable the non-linear behavior will be.

Xmax does not indicate how far the cone can be physically moved. Just because a woofer cone can be moved by hand a great deal does not mean that its voice coil is capable of moving it that far. Just because you can go 100 mph on a bicycle being towed by a Porsche doesn't mean that you can achieve that speed using leg power! You should also be conscious of "peak to peak" Xmax specs which need to be divided by two to compare to one-way specs.

Long-excursion woofers require very rugged and precise suspension and motor design as well as sufficient thermal powerhandling to take advantage of their excursion potential.

A Head to Head Comparison
Let's compare two 10" speakers and determine their ultimate linear output capability. Speaker A is a 10" woofer with an Xmax of .468" (12 mm). Speaker B is a 10" woofer with an Xmax of 0.25" (6.5 mm), which at the time of this writing is pretty average in the industry.

Below you will see the maximum SPL that each speaker can produce at each frequency in a sealed enclosure with a Qtc of 0.7 (for maximally flat response). Next to the SPL figure in parentheses you will see the amount of power being handled to produce this maximum excursion. This figure is the effective mechanical powerhandling of each driver at each frequency. The numbers below do not indicate frequency response.

Maximum(Displacement Limited) Output and Powerhandling
	Speaker "A" (Xmax = 6.5mm)		Speaker "B" (Xmax = 12mm)
20 Hz	95.7 dB	189.2 W	90.2 dB	78.2 W
30 Hz	102.7 dB	244.3 W	97.3 dB	81.5 W
40 Hz	107.7 dB	392.6 W	102.3 dB	90.6 W
50 Hz	111.6 dB	705.5 W	106.2 dB	109.7 W
60 Hz	114.8 dB	1275 W	109.3 dB	144.6 W
80 Hz	119.8 dB	3649 W	114.3 dB	290.2 W
100 Hz	123.7 dB	8655 W	118.2 dB	597.5 W

The data show how direct the link is between Xmax and ultimate output capability when comparing speakers of equal size. As you can see, Speaker A outperforms Speaker B by 5.5 dB consistently up the scale. The difference in low-frequency output capability between these two drivers is staggering. You would need two Speaker B's to equal the output capability of one Speaker A. That makes sense when you consider that Speaker A is moving virtually twice as much air as one Speaker B.

If you refer to the plot to the right you will see a comparison to ultimate output with each speaker being driven by the amount of nominal broad-band power necessary to reach its linear excursion limits in that particular sealed box (again with Qtc = 0.7). You will see that Speaker A handles twice the power and is easily capable of outperforming Speaker B in this real-world situation. You will also notice that Speaker A does not begin to approach its excursion limits until the frequency drops below 25 Hz, whereas Speaker B approaches its limits starting at 45 Hz.

For every doubling of excursion capability (Xmax) you gain 6 dB of ultimate output capability. This may seem a bit counter-intuitive because we have all been taught that a doubling of acoustic power only produces a 3 dB increase. What we must keep in mind is that the acoustic power is proportional to the square of the pressure, just as electrical power is proportional to the square of voltage. A doubling of excursion requires 4x the input power and produces 4x the acoustic power, all other factors being equal. Here are the relationships in summary form:

1.26 x power (watts) = 1.12 x excursion = + 1 dB	1.59 x power (watts) = 1.26 x excursion = + 2 dB	2.00 x power (watts) = 1.41 x excursion = + 3 dB
2.52 x power (watts) = 1.59 x excursion = + 4 dB	3.18 x power (watts) = 1.78 x excursion = + 5 dB	4.00 x power (watts) = 2.00 x excursion = + 6 dB
5.04 x power (watts) = 2.24 x excursion = + 7 dB	6.35 x power (watts) = 2.52 x excursion = + 8 dB	8.0 x power (watts) = 2.83 x excursion = + 9 dB
10.0 x power (watts) = 3.16 x excursion = +10 dB

From these numbers you can quickly see that the change in power is always the square of the change in excursion. This is true both for input power and acoustic power as excursion is directly proportional to voltage, not power.

Going back to the comparison between he Speaker A and Speaker B, you can also see that low-frequency power handling is directly linked to Xmax. The Speaker A is capable of handling very high power levels in the heart of the sub-bass region range without it coils jumping like suicidal lemmings out of the gap. This means that it is in control and reproducing the signal faithfully. If you pump more than 90 watts into Speaker B at 40 Hz it will begin to distort and could potentially be damaged. The Speaker A handles almost 400 watts mechanically at 40 Hz.

The importance of mechanical power handling is undeniable when it comes to subwoofers. Especially when one considers the output capability of today's high performance car amplifiers. A speaker may be able to handle 1000 watts thermally but if it has a short voice coil and short excursion capability it will not handle power well, mechanically speaking.

So How Do They Work?
A bandpass enclosure is, by definition, simply a sealed enclosure with an acoustical filter in front of it that serves to limit the upper-end of the driver's frequency response. This natural limiting of the high-frequency response of the system makes the selection of mid-bass drivers critical. If your vehicle cannot fit larger midbass drivers (such as a 6 1/2" or larger) then a bandpass enclosure is probably not the best choice for you. Using a bandpass enclosure with insufficient mid-bass reinforcement will lead to sluggish, sloppy, muddy, impact-less low frequency response. In short--it will sound like a soggy pancake hitting a cardboard box.

Once adequate mid-bass reinforcement has been selected to complement the sub-system it will be necessary to add additional electronic filtering to further limit the upper frequency output of the enclosure. Contrary to popular belief, a bandpass enclosure (of any type--single reflex, dual reflex, series-tuned, etc.) does require the use of an electronic crossover to achieve optimum performance since the acoustical low-pass filter is not a very effective filter. What proponents of "crossover-less" bandpass enclosures neglect is that there is a considerable amount of high frequency output (called "out-of-band noise") that can get to be quite annoying. It is for this reason that we recommends that all bandpass enclosures be supplemented with an electronic crossover. If you would like to find out more about electronic crossovers, contact us.

In order to understand how a bandpass enclosure works, it helps to break the enclosure itself into three parts: the sealed (rear) chamber, the ported (front) chamber, and the port itself; but before we get started, we need to define some of the basic terminology used in this tutorial so we can make sure everyone is on the same page.

The Sealed Chamber
The sealed chamber's primary purpose is to serve as a high-pass acoustical (as opposed to electrical) filter and it's volume controls the lower -3dB point or FL. By changing the size of the sealed chamber , we can see a corresponding shift in FL that follows these simple guidelines:

The bigger the sealed chamber is, the lower the FL will be.

The smaller the sealed chamber is, the higher the FL will be.

Any changes made in the rear box volume require a corresponding change in the tuning of the front chamber(s) of the enclosure. Failure to retune the front chamber(s) will result in a mis-tuning and the box will more than likely sound really, really bad.

As is Mother Nature's style, we can't get something for nothing so as we adjust the volume of the rear chamber(s) it is important to keep the following in mind:

The bigger the sealed chamber is, the lower the driver's mechanical (also called "displacement-limited") power handling will be. If the rear chamber(s) is/are too big, the amount of power the system can handle is reduced.

The smaller the rear chamber(s) is/are, the higher the driver's displacement-limited power handling will be but of course, the low-frequency extension will suffer as a result.

The Ported Chamber
The ported chamber controls the bandwidth and efficiency of the system and behaves as follows:

As front chamber volume(s) increase, the system becomes more efficient. As front chamber volume(s) decrease, system efficiency decreases.

As always, Mother Nature has her dirty little hands in our enclosure design so we have to consider what she is doing:

As the size of the ported chamber increases, the bandwidth decreases. So, the more efficient we make the system, the smaller the passband will be. If the system is made too efficient (outrageously large front chamber), we'll wind up with "one-note" bass so typical of a myriad of mis-tuned bandpass enclosures on the market today. On the other hand, as the front chamber volume increases, the better the transient response of the system will be, and this is good.

As the size of the ported chamber decreases, the bandwidth increases. This may sound desirable (especially to those with smaller mid-bass drivers), but of course Mother has dictated that as the ported chamber shrinks the group delay becomes undesirably large and the system starts to sound very sluggish and muddy thus ruining our hopes of using a bandpass system with our factory-installed 4" speakers!

The Port
The port is probably the single most critical variable in the bandpass equation. The port MUST tune the front chamber to the exact center of the passband or the box will sound like total garbage. The center of the passband corresponds directly to the sealed-box resonant frequency or fc of the rear chamber. If for some reason the port is of the incorrect dimensions and tunes the front chamber too high, the frequency response will be skewed creating a really nasty response peak in the lower mid-bass range, whereas if it is tuned too low the response will be very peaky in the lower frequencies and the bass will sound unnatural and boomy.

"Universal" Bandpass Enclosures
The latest twist to the bandpass scene is the "universal" bandpass enclosure that promises to give all the benefits of a tailor-made bandpass design without the price or the time required to design one properly. The term "universal bandpass" itself is really an oxymoron (like "fresh-frozen", "jumbo-shrimp", "one size fits all", etc.) in that there truly is no one design that will work with all drivers! To imply that such a design is feasible is to totally ignore the very nature of bandpass enclosures: they are extremely driver sensitive and enclosure sensitive! If you have read all of this tutorial up until this point, it shouldn't take much persuasion to lead you to believe that "yes, bandpass designs are picky and easy to mess up". Not all 10" drivers are alike...not all 10" drivers from one manufacturer are alike, so why should they use the exact same box?

To illustrate what happens when a bandpass is designed for one driver and someone decides to place another driver in there, we offer the following scenario: You and a buddy got bored one day and decided, that you wanted to try something new. So, with the help of your friendly authorized car stereo dealer, you get the specs and start construction on a bandpass enclosure for the 10inch sub you had in a ported box at the time. When you finished and everything was in place, you turned on the system and were impressed with your handiwork. You were happy, your friend was happy, and with the 100W amp you had powering the 10inch suv in it's new home, the driver was happy.

Time goes on and one day you decide that it's not loud enough for you anymore. So for your birthday, your significant other gives you a brand new 300W amp and a better 10inch sub to replace your old 10inch. You call up your buddy and run out to your car and tag-team the installation. He installs the amp and you swap out the old 10 for the new 10. In a matter of minutes, the two of you are eagerly awaiting your first listen. As you reach for the power switch on your head unit, you tremble with excitement, eagerly anticipating being blown out of the car by the tremendous increase in bass output. . . then you hear it. The bass is absolutely horrendous and boomy. You begin to curse the gods of car audio and the next day you demand an explanation from your authorized car stereo dealer (your birthday was on a Sunday and the store wasn't open then).

Your dealer kindly explains that what you had was a well-shaped frequency response curve with a bandpass from 42-98 Hz with a pretty flat pass-band; but when you perterbed the gods of car stereo by using the wrong driver in the wrong box, what you wound up with was an acoustical nightmare with a frequency response curve that looks more like Mt. Everest than the plateau it should emulate. When the two plots are superimposed you can see just how dramatic the difference is. Remember, the only thing that has changed is the driver--the box and the ports were unchanged when these plots were drafted.

The moral of the story is that even though two drivers may be the same size, even though two drivers may be made by the same company, that does not make them the same. Likewise, a bandpass enclosure made for one driver will not necessarily work well with another driver.

Port Basics
When tuning a ported enclosure there are two widely used methods implemented. These two methods involve the use of a port, generally made from a simple piece of PVC pipe; or a duct (sometimes called a slot port), which is normally constructed out of the same material the box is made of (normally wood).

Before we can discuss how to make a port, it is important to know what factors affect the tuning frequency of the enclosure. It is a common misconception that the tuning frequency (fb) is a function of port volume when in fact, it is actually a function of the port's cross-sectional area and its length as given by the formula:

Where Av is the cross-sectional area of the port (in square inches),
Lv is the length of the port (in inches)
and Vb is the enclosure's net volume (in cubic inches).

It looks really hairy, and it is, but the thing to notice is that the volume of the port does not come into play. It is also interesting to note that contrary to what one might think, the bigger the diameter of port you use (bigger Av), the longer the port has to be (assuming box volume and tuning frequency are constant).

Round ports are really simple to execute since most loudspeaker manufacturers will specify a diameter and length of port for you to use in your particular design. Just remember that the port diameter that all manufacturers speak of is the port's internal diameter, not the outside diameter as seen in this picture.

The length specified is simply the length of the port from end to end, not just the length of the port inside the enclosure.

When using round or slot ports, it is important to use either a file or a piece of sandpaper and round off the inside edges of both ends of the port to minimize the likelyhood that your port would make whistling noises (caused by air moving rapidly over a sharp edge like that found on a whistle or a 1978 Cadillac doing 70mph).

Ducts are often used when a particular alignment calls for an outrageously long port to be squeezed in a very tiny enclosure due to a very low tuning. This scenario is commonly encountered when constructing ported enclosures for our W6 series drivers.

Designing and implementing a duct in your own project is really not as hard as it may seem at first, but there are a few guidelines you must follow if you are to experience any degree of success utilizing this porting technique.

Below is a perspective view of a typical duct port along with a few helpful tips on how to get the best results from your duct.

Av	The cross-sectional area (found by multiplying h and W) should be the same as that prescribed for a round port. For example, if our design calls for a 4" diameter port, our duct's cross-sectional area should be 12.57 square inches.
W / h	The ratio of W to h should not be any more than 9 to 1 to prevent tuning shifts introduced by excessive friction between the rapidly moving air and the port's surface. It should be noted that this is not a hard-and-fast rule, merely a rule of thumb to help you prevent a mis-tuning. In short, keep the port as square as aesthetically and physically possible.
Lv (phys)	The physical length of the port is measured down the dead center of the port from end to end. In the picture above, this would correspond to L1 + L2, but the physical length of the port isn't really what is important, it's the effective length of the port (with end correction) that is important.
Lv (eff)	The effective length of the port is found by adding an end correction factor. An end correction factor is necessary because more often than not, one wall of the port is also one wall of the enclosure and this wall extends beyond the end of the port thus effectively adding length to the port (remember, the driver can't "see" the length of the port, it can only go by what it "feels" is going on). Calculating end factor may sound like it would be more trouble than it's worth, but it's actually quite simple. To calculate end correction factor, simply add one-half of h to the physical length you calculated above (L1 + L2). Since pictures are worth a thousand words, we've included a diagram above to help illustrate the point. Click on the image to see it at full size.

Multiple Ports
There are two widely used methods for calculating multiple ports for a single chamber. Only one method is correct but unfortunately it is the least commonly used.
The first and incorrect method takes it's thinking from the original port formula and says basically that if we take two ports and sum their cross-sectional areas, we can just plug this total into the port formula for Av to get our port length. This would sound reasonable, but it can lead to serious mis-tunings in some cases as we'll see in an example below.

The second and correct way to figure out how long each port should be follows this simple three-step procedure:

• Divide the chamber volume by the number of ports you wish to use for that one chamber.

• Take the quotient and use that as your Vb (box volume) in the port formula

• Do the number crunching and figure out how long each port should be.

Example:
Let's take an arbitrary box volume of 2.5 cubic feet that we want to tune to 25 Hz with a 4" diameter port. If we plug and chug with that big hairy formula (or let our favorite software package churn out the numbers), we'll find that Lv = 18.844 inches.

Now let's decide that we don't want just a single port because it looks boring. Let's put a 2" port in each corner of the box for a total of 4 ports and see what the two methods give us:

Method 1:
Each 2" port has a cross-sectional area of 3.142 square inches so we multiply that by 4 to get 12.57 square inches. Plugging in 12.57 for Av in the port formula yields Lv = 18.844 inches for each port.

Method 2:
We want to use 4 ports so we divide 2.5 cubic feet by 4 and get .625 cubic feet. Vb now becomes .625 cubic feet. We are using 2 inch diameter ports so Av is 3.142 square inches. Plugging these numbers into the equation leads to Lv = 20.302 inches for each port.

Notice that Method 1 produces the same port length as did our single 4" diamater port as it should (after all, we have the same total port cross-sectional area which this school of thought proclaims is correct!). But the first method is incorrect because it neglects the frictional losses encountered by using many smaller ports--there is a higher port wall surface area to cross-sectional area ratio which raises the total amount of frictional losses in the ports and thus shifts the tuning!

Enclosure Shapes
While it is always a pretty good idea to stay away from perfect cubes, they don't necessarily have to be avoided like The Plague. Due to the very small dimensions of most mobile subwoofer enclosures, there is little chance of generating standing waves in the enclosure (standing waves cause nasty response fluctuations). For a standing wave to exist, the distance between parallel boundaries must be 1/2 the wavelength of the frequency at which the standing wave exists. Considering that sub-bass waves vary from 56.4 feet (20 Hz) to 11.28 feet (100 Hz), the generation of a standing wave is going to be impossible....after all, the enclosures we're speaking of have to fit in the average sedan or hatchback!

Any standing waves that might be generated by upper ordered harmonics (caused by distortion) in the enclosure can be readily absorbed with the addition of damping material such as polyfill (available at your local cloth store--it is used to stuff pillows and quilts) or Fiberglastm (the pink stuff) and/or they can be broken up with strategically placed bracing within the enclosure.

In short, don't worry too much about shape. Make the box to fit the space you can allot to the enclosure and forget about it--there are more important things to worry about...like bracing.

Bracing and Strength
Of all the things to worry about when constructing an enclosure, this is probably the most critical element. If an enclosure cannot adequately contain the tremendous amounts of pressure generated by today's high-powered subwoofer systems, the results will be marginal bass quality at best or total destruction of the enclosure at worst.

A flexing enclosure is a lossy enclosure. If the panels on your subwoofer enclosure vibrate, you lose output (SPL) and clarity. The solution is two-fold: use only 3/4" or 5/8" thick medium density fiberboard (MDF) and brace (reinforce) the life out of the box. If MDF (or the brand name Medite) is not available in your area (it can be quite hard to find, but most custom cabinet making shops should be able to supply you with what you need), the only other real solution (barring exotic materials like sheet PVC) is to use a super high-quality plywood like birch or some other marine-grade plyboard. Avoid using particle board at all costs as it is too flaky (literally), doesn't hold screws well and swells like a sponge when water hits it. In short, particle board comes from the Pit of Helltm. Avoid it at all costs.

After the proper materials have been chosen for box construction, the subject of bracing must be addressed. Bracing is very important!

A true story We once heard of a fellow using a triplet of 10inch subwoofers in a sealed box powered with a 1200 watt amplifier. The box was not braced at all, was one big chamber and was made of particle board. While demonstrating the system for a crowd of people, the box literally exploded in his trunk, coating his trunk with sawdust and enclosure shrapnel, leaving the subs to play in free-air (not cool). Not only is it frustrating (or really cool, depending upon your personality) to have a box explode in your car, but it can also be very dangerous....after all, you could poke an eye out!

If you remember just one thing about bracing, remember the following scenario:

The Jumbo Test Imagine, if you will, a rather bitter mid-sized African elephant named Jumbo (how original) that has been taking steroids and lifting religiously for the past 10 years. Jumbo's ex-girlfriend left him for a box (don't ask--because we don't know) and so every time Jumbo sees a box, he does his best to obliterate it. After building your enclosure, you should be able to place your box in front of Jumbo and let him tap dance on it all day long knowing that when Jumbo finally gives up (he never does) your enclosure will emerge unscathed. It sounds a bit extreme, but it's true: strength is key!

Sealing the Box
Whether you are planning to use a bandpass, ported or sealed box, sealing the edges is very important (isn't everything?). The first step to take in assuring a good tight seal at all joints is to use copious amounts of wood glue. Don't be shy with it--keep a wet rag handy to wipe up the excess. Like bracing, you can never use too much.

There have been some debates on rec.audio.car regarding the use of silicone caulk to seal enclosures since the caustic fumes (acetic acid) released during curing have an appetite for foam surrounds, but with a little understanding of what is going on, this problem can easily be avoided.

Fortunately, most good quality subwoofers have a specially treated surround that protects them from hungry acetic acid fumes which is cause #1 not to be overly concerned with using silicone to seal your box. Secondly, the fumes are only released during curing (the time when the caulk goes from a free-flowing gel to an amorphous solid) so all you have to do to prevent damage to the drivers is to wait until the silicone has cured (8-12 hours usually) before dropping the subs in. As one member of the rec.audio.car newsgroup (who shall remain anonymous) can attest, it is not a good idea to stick your head inside the box while the silicone is curing unless you are in search of the world's most obnoxious buzz (don't try this at home kids).

If time is of the essence and you are worried about your subwoofer, you might want to look into other sealants that are less caustic.

Power per channel:

Describes the output power in watts into a resistive (ohm) load


• Power- Measure of a power amplifier's ability (in watts) to deliver electrical voltage and current to a speaker

Total Power:
Describes the total output power of all channels added together into a resistive (ohm) load.

• RMS - Root Mean Square. The amount of continuous power (measured in watts) that an amplifier produces is called RMS power. The higher the RMS figure, the louder and cleaner your music sounds. When choosing an amplifier, the RMS rating is the power rating you should pay most attention to. Also, keep in mind that some manufacturers calculate the RMS power ratings of their amplifiers at different input voltages. For example, an amplifier rated at 100 watts RMS at 12 volts can produce considerably more power than an amp rated at 100 watts RMS at the more typical 14.4 volts.
• Peak - Stereo manufacturers often display peak power ratings on the face of their products. The peak power rating tells you the maximum wattage an amplifier can deliver as a brief burst during a musical peak, like a dramatic drum accent. The RMS figure is more significant.
• RMS Power at 2 ohms - This specification tells you how much more power your amp delivers when presented with a 2-ohm stereo load. You can achieve a 2-ohm load by using parallel wiring or by using 2-ohm speakers. Theoretically, amplifier output should exactly double as the impedance drops from the usual 4 ohms to 2 ohms. However, amp makers use different degrees of regulation on power supplies, which can restrict the actual increase in output. Less regulated power supplies come closer to doubling their output into 2-ohm loads. An amp with little regulation can achieve higher wattage into lower impedances. An amplifier with stiffer regulation maintains rated output as other electrical accessories demand voltage from the battery

Crossover:
Describes a filter that passes a specific range of frequencies, while blocking others.

• HP (high pass) - A filter that passes signals above a certain frequency (called cutoff frequency). A high-pass crossover allows only frequencies above the crossover point to be amplified. Useful for keeping low bass away from small speakers, so they can play more efficiently.
  
• LP (low pass) - A filter that passes signals below a certain frequency (called cutoff frequency). Often used to keep high frequencies from reaching a subwoofer, a low-pass crossover allows only frequencies below the crossover point to be amplified.
• BP (band pass) - A filter that passes a certain portion, or band of frequencies. Consisting of a HP and LP cutoff.
• AP (all pass) - Passes all frequencies.

Amplifier Class:
Describes the different classifications of amplifiers, depending on how the biasing of the amplifier circuitry is done.

• Class A - Class A operation is where both devices conduct continuously for the entire cycle of signal swing, or the bias current flows in the output devices at all times. The key ingredient of class an operation is that both devices are always on. There is no condition where one or the other is turned off. Because of this, class amplifiers are single-ended designs with only one-type polarity output devices. Class A is the most inefficient of all power amplifier designs, averaging only around 20%. Because of this, class an amplifiers are large, heavy and run very hot. All this is due to the amplifier constantly operating at full power. The positive effects of all this is that class designs are inherently the most linear, with the least amount of distortion. Devices operate during each cycle of signal swing. Also defined in terms of output bias current, (the amount of current flowing in the output devices with no signal).
• Class B operation is the opposite of class A. Both output devices are never allowed to be on at the same time, or the bias is set so that current flow in a specific output device is zero when not stimulated with an input signal, i.e., the current in a specific output flows for one half cycle. Thus each output device is on for exactly one half of a complete sinusoidal signal cycle. Due to this operation, class B designs show high efficiency but poor linearity around the crossover region. This is due to the time it takes to turn one device off and the other device on, which translates into extreme crossover distortion.
• Class AB operation allows both devices to be on at the same time (like in class A), but just barely. The output bias is set so that current flows in a specific output device appreciably more than a half cycle but less than the entire cycle. That is, only a small amount of current is allowed to flow through both devices, unlike the complete load current of class A designs, but enough to keep each device operating so they respond instantly to input voltage demands. Thus the inherent non-linearity of class B designs is eliminated, without the gross inefficiencies of the class a design. It is this combination of good efficiency (around 50%) with excellent linearity that makes class AB the most popular audio amplifier design.
• Class D operation is switching, hence the term switching power amplifier. Here the output devices are rapidly switched on and off at least twice for each cycle. Since the output devices are either completely on or completely off they do not theoretically dissipate any power. Consequently class D operation is theoretically 100% efficient, but this requires zero on-impedance switches with infinitely fast switching times -- a product we're still waiting for; meanwhile designs do exist with true efficiencies approaching 90%.
• Class G operation involves changing the power supply voltage from a lower level to a higher level when larger output swings are required. There have been several ways to do this. The simplest involves a single class AB output stage that is connected to two power supply rails by a diode, or a transistor switch. The design is such that for most musical program material, the output stage is connected to the lower supply voltage, and automatically switches to the higher rails for large signal peaks. Another approach uses two class AB output stages, each connected to a different power supply voltage, with the magnitude of the input signal determining the signal path.
• Class H operation takes the class G design one step further and actually modulates the higher power supply voltage by the input signal. This allows the power supply to track the audio input and provide just enough voltage for optimum operation of the output devices. The efficiency of class H is comparable to class G designs.
• Tube - Electron tube evacuated to such a degree that its electrical characteristics are essentially unaffected by the presence of residual gas or vapor. Amplifier using tube type electrical devices.

Output Topology
Type of devices used in an amplifiers output section. There are three basic types of output devices found on car audio amplifiers - integrated circuits, bipolar transistors, and MOSFETs. Bipolar and MOSFET transistors are considered discrete output devices. Usually there are two per channel, but some amps feature as many as four per channel.

• Bipolar - Transistor that contains two p or n junctions or diodes between two layers of opposite polarity material. Controlled by current rather than voltage. Found on the output stages of high-powered amplifiers. They are fast enough and can handle enough current to send wattage greater than 20 watts per channel to your speakers.
• MOSFET - Metal Oxide Semiconductor-Field Effect Transistor. A form of transistor controlled by voltage rather than current. MOSFETs have significantly higher switching speed than bipolar transistors. They generate almost no loss (little heat generation). MOSFETs are found on the output stages of high-powered amplifiers. They are fast enough and can handle enough voltage to send wattage greater than 20 watts per channel to your speakers.
• IC (integrated circuit) - An integrated circuit is found only on relatively low-wattage (20 watts RMS per channel or less) amplifiers called "bridged transformer-less" amps. An IC cannot pass enough current to work on a more powerful amp and is not considered a discrete output device

Power Supply Topology
Describes the type of device used in amplifiers power supply.

• Bipolar - Transistor that contains two p or n junctions or diodes between two layers of opposite polarity material. Controlled by current rather than voltage.
• MOSFET - Metal Oxide Semiconductor Field Effect Transistor. A form of transistor controlled by voltage rather than current. MOSFETs have significantly higher switching speed than bipolar transistors. They generate almost no loss (little heat generation), which lends the power supply fast response, excellent linearity, and high efficiency.
• PWM - Pulse Width Modulation. A signal generation method used for making all of the signal pulses the same amplitude but of varying durations or widths. This helps increase the supply voltage.
• Darlington configuration. A method of using two transistors to provide very high current gain.

Regulated power supply. Type of power supply that maintains supply voltage even as battery voltage fluctuates.

Ohm's Law
Ohm’s Law is a set of formulas used in electronics to calculate an unknown amount of current, voltage or resistance. It was named after the German physicist Georg Simon Ohm. Born 1787. Died 1854.

Knowledge of this Law is often under-estimated by beginners. I have talked to people that can design complex circuitry and microprocessor systems that have said, “Ohm’s Law? What’s that?”.

Unless you know this basic fundamental building block of electronics, you will never have a strong foundation to hold up the electronics towers you will be constructing in the future. Learn Ohm’s Law. Learn it inside and out!

In simpler terms, Ohm’s Law means: 1) A steady increase in voltage, in a circuit with constant resistance, produces a constant linear rise in current.

2) A steady increase in resistance, in a circuit with constant voltage, produces a progressively (not a straight-line if graphed) weaker current.

TECHNICAL DEFINITION ALERT!

Ohm's Law is a formulation of the relationship of voltage, current, and resistance, expressed as:

Where:

V

is the Voltage measured in volts

I

is the Current measured in amperes

R

is the resistance measured in Ohms


Therefore:

Volts = Amps times Resistance

Ohms Law is used to calculate a missing value in a circuit.

In this simple circuit there is a current of 12 amps (12A) and a resistive load of 1 Ohm (1W). Using the first formula from above we determine the Voltage:

V = 12 x 1 : V = 12 Volts (12V)

If we knew the battery was supplying 12 volt of pressure (voltage), and there was a resistive load of 1 Ohm placed in series, the current would be:

I = 12 / 1 : I = 12 Amps (12A)

If we knew the battery was supplying 12V and the current being generated was 12A, then the Resistance would be:

R = 12/12 : R = 1W

Note: Remember a battery is not measured in amperage as is commonly believed with beginners to electronics. The battery supplies the pressure that creates the flow (current) in a given circuit. The amperage rating on a battery is "How long the battery will last for one hour while driving a circuit of that amperage". It is measured in Amperage-Hours. So a 1000mAh would last for 1 hour in a one amp circuit. (1000mAh is 1A for one hour)

An easy way to remember the formulas is by using this diagram.

To determine a missing value, cover it with your finger. The horizontal line in the middle means to divide the two remaining values. The "X" in the bottom section of the circle means to multiply the remaining values.

• If you are calculating voltage, cover it and you have I X R left (V= I times R).
• If you are calculating amperage, cover it, and you have V divided by R left (I=V/R).
• If you are calculating resistance, cover it, and you have V divide by I left (R=V/I).

Note: The letter E is sometimes used instead of V for voltage